From billreid at shaw.ca Wed Sep 1 11:09:14 2010 From: billreid at shaw.ca (Bill Reid) Date: Wed, 01 Sep 2010 11:09:14 -0500 Subject: [*] Tues Sept 7th meeting Message-ID: <4C7E7AAA.30004@shaw.ca> Hi All, Our next meeting will be at Les.Net. (Note:I am changing the time to 7:30) 7:30PM Tues, Sept 7th 130 Portage Avenue East. One of the nutty club buildings, 2 story building on the corner of Portage + Westbrook. I've painted the door bright Red, so it's easy to spot. There is a doorbell too. The executive entrance ladder on the side of the building has been retired. If you wouldn't mind giving me(mailto:sales at les.net) a rough head count, I'll make sure to have enough seating in a round-table configuration. Agenda: Adam Thompson will demo the ATCOM product line. Les will give an update on his CLEC roll out. Look forward to seeing you at the meeting, Bill From mbergen at obsglobal.com Wed Sep 1 13:09:52 2010 From: mbergen at obsglobal.com (Bergen, Mark) Date: Wed, 1 Sep 2010 13:09:52 -0500 Subject: [*] Long Distance PIN Message-ID: Hello all; We are using Trixbox 2.8 in one of our offices connected to a PRI (Allstream) and there are two things we are struggling with: 1. We use PIN's for all of our long distance calls, is there some way to have the PIN included when a user makes a long distance call by using DTMF maybe? It is for the VMX locator, some of our consultants work outside of Winnipeg and would like to give users the option of contacting them at the office they are working in or by cell phone, but when a user tries they hear MTS prompting them for the PIN. We have managed to create custom outbound routes for individual users but can't get the DTMF part to work. Here is some of what we have tried (5020 represents the users extension, so each outbound route can be unique): exten => _50201NXXNXXXXXX,1,Dial(dialout-trunk,1,,D(www1234${EXTEN:4})) and [outrt-005-VmXtoLONGDISTANCEcalls-custom] exten => _50201[12345678]XXNXXXXXX,1,Macro(user-callerid,SKIPTTL,) exten => _50201[12345678]XXNXXXXXX,n,Set(_NODEST=) exten => _50201[12345678]XXNXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,) exten => _50201[12345678]XXNXXXXXX,n,Macro(dialout-trunk,1,${EXTEN:4},,) exten => _50201[12345678]XXNXXXXXX,n,Macro(senddtmf) exten => _50201[12345678]XXNXXXXXX,n,Macro(outisbusy,) [macro-senddtmf] exten => s,1,SendDTMF (1234) 2. The second problem is when a someone calls in and wishes to be transferred to an office or user that is not local and does not have a toll free number. Blind transfer will not work as the caller does not know the PIN and trying a attended transfer did not work either, Asterisk gives us a beep but we don't receive one from the phone company, how can we transfer a long distance call and enter the long distance PIN before the transfer completes? Other than that the little Trixbox appliance is working fine, the Sangoma T1 card works great so far, and the FXO ports have not been a problem, faxing and analog phone are working. Any direction or help with the above would be greatly appreciated. Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100901/bc043e62/attachment.html From mbergen at obsglobal.com Thu Sep 2 10:00:23 2010 From: mbergen at obsglobal.com (Bergen, Mark) Date: Thu, 2 Sep 2010 10:00:23 -0500 Subject: [*] Long Distance PIN In-Reply-To: <1283394216.4635.3.camel@linux-k6vx.site> References: <1283394216.4635.3.camel@linux-k6vx.site> Message-ID: Thank you for the reply John, really appreciate it. I know you probably tried everything (yes, I'm reinventing the wheel, please be patient) but shouldn't something like this work: exten => _50201NXXNXXXXXXwwww1234,1,Dial(dialout-trunk,1,,(${EXTEN:4})) Then dial 5020 1 204 982 0218 1234 (1234 being the PIN) MTS still operates in the same manner regarding answering the line for the pin while the call is still in progress. I saw a fix for it (patch) but it was for Asterisk 1.4, not sure if it would work for Trixbox 2.8 (Asterisk 1.6). https://issues.asterisk.org/view.php?id=12123 Thanks again for your time, I really appreciate it. Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 mbergen at obsglobal.com |? Direct Line: 204.982-0218 Office: 204.982.0200? |? Fax: 204.982.0201 www.obsglobal.com -----Original Message----- From: John Lange [mailto:john at johnlange.ca] Sent: September-01-10 9:24 PM To: Bergen, Mark Subject: Re: [*] Long Distance PIN For a brief time we had an Allstream PRI with PINs and I never got this working with Asterisk. The problem is, the PRI does not send the "answer" signal when prompting for the PIN leaving asterisk to think that the call is still making progress. In other words, Asterisk will not move to the next step in the dialplan so you can't do any sending of DTMF etc. It was over 3 years ago that I tried it so things may be different now. -- John Lange http://www.johnlange.ca On Wed, 2010-09-01 at 13:09 -0500, Bergen, Mark wrote: > Hello all; > > We are using Trixbox 2.8 in one of our offices connected to a PRI > (Allstream) and there are two things we are struggling with: > > 1. We use PIN?s for all of our long distance calls, is there some > way to have the PIN included when a user makes a long distance call by > using DTMF maybe? It is for the VMX locator, some of our consultants > work outside of Winnipeg and would like to give users the option of > contacting them at the office they are working in or by cell phone, > but when a user tries they hear MTS prompting them for the PIN. We > have managed to create custom outbound routes for individual users but > can?t get the DTMF part to work. > > Here is some of what we have tried (5020 represents the users > extension, so each outbound route can be unique): > > exten => _50201NXXNXXXXXX,1,Dial(dialout-trunk,1,,D(www1234 > ${EXTEN:4})) > > and > > [outrt-005-VmXtoLONGDISTANCEcalls-custom] > > exten => _50201[12345678]XXNXXXXXX,1,Macro(user-callerid,SKIPTTL,) > > exten => _50201[12345678]XXNXXXXXX,n,Set(_NODEST=) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(record-enable, > ${AMPUSER},OUT,) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(dialout-trunk,1, > ${EXTEN:4},,) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(senddtmf) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(outisbusy,) > > [macro-senddtmf] > > exten => s,1,SendDTMF (1234) > > 2. The second problem is when a someone calls in and wishes to be > transferred to an office or user that is not local and does not have a > toll free number. Blind transfer will not work as the caller does not > know the PIN and trying a attended transfer did not work either, > Asterisk gives us a beep but we don?t receive one from the phone > company, how can we transfer a long distance call and enter the long > distance PIN before the transfer completes? > > Other than that the little Trixbox appliance is working fine, the > Sangoma T1 card works great so far, and the FXO ports have not been a > problem, faxing and analog phone are working. > > Any direction or help with the above would be greatly appreciated. > > Mark > > > > Mark Bergen > > Network Support Analyst > > ONLINE BUSINESS SYSTEMS > > Explore | Innovate | Lead > > > > 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 > > mbergen at obsglobal.com | Direct Line: 204.982-0218 > > Office: 204.982.0200 | Fax: 204.982.0201 > > www.obsglobal.com > > > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk From cfriesen at telenium.ca Thu Sep 2 10:48:14 2010 From: cfriesen at telenium.ca (Chris Friesen) Date: Thu, 2 Sep 2010 10:48:14 -0500 Subject: [*] Long Distance PIN In-Reply-To: References: <1283394216.4635.3.camel@linux-k6vx.site> Message-ID: <001901cb4ab6$41172410$c3456c30$@telenium.ca> Did you try adding something like this to the dial command? D([called][:calling]) Sends DTMF digits after the call is answered but before it is bridged. The called digits are transmitted to the called party, the calling digits to the calling party. One or both parameters may be set. I've seen and example like this ... ,1,D([1234]) And I wonder , if you didn't want them to hear the DTMF tones could you use m to play some music on hold while the call is connected or r to play ringing. w for wait ,1,rD(w[1234]) -----Original Message----- From: asterisk-bounces at muug.mb.ca [mailto:asterisk-bounces at muug.mb.ca] On Behalf Of Bergen, Mark Sent: September-02-10 10:00 AM To: John Lange; asterisk at muug.mb.ca Subject: Re: [*] Long Distance PIN Thank you for the reply John, really appreciate it. I know you probably tried everything (yes, I'm reinventing the wheel, please be patient) but shouldn't something like this work: exten => _50201NXXNXXXXXXwwww1234,1,Dial(dialout-trunk,1,,(${EXTEN:4})) Then dial 5020 1 204 982 0218 1234 (1234 being the PIN) MTS still operates in the same manner regarding answering the line for the pin while the call is still in progress. I saw a fix for it (patch) but it was for Asterisk 1.4, not sure if it would work for Trixbox 2.8 (Asterisk 1.6). https://issues.asterisk.org/view.php?id=12123 Thanks again for your time, I really appreciate it. Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com -----Original Message----- From: John Lange [mailto:john at johnlange.ca] Sent: September-01-10 9:24 PM To: Bergen, Mark Subject: Re: [*] Long Distance PIN For a brief time we had an Allstream PRI with PINs and I never got this working with Asterisk. The problem is, the PRI does not send the "answer" signal when prompting for the PIN leaving asterisk to think that the call is still making progress. In other words, Asterisk will not move to the next step in the dialplan so you can't do any sending of DTMF etc. It was over 3 years ago that I tried it so things may be different now. -- John Lange http://www.johnlange.ca On Wed, 2010-09-01 at 13:09 -0500, Bergen, Mark wrote: > Hello all; > > We are using Trixbox 2.8 in one of our offices connected to a PRI > (Allstream) and there are two things we are struggling with: > > 1. We use PIN?s for all of our long distance calls, is there some > way to have the PIN included when a user makes a long distance call by > using DTMF maybe? It is for the VMX locator, some of our consultants > work outside of Winnipeg and would like to give users the option of > contacting them at the office they are working in or by cell phone, > but when a user tries they hear MTS prompting them for the PIN. We > have managed to create custom outbound routes for individual users but > can?t get the DTMF part to work. > > Here is some of what we have tried (5020 represents the users > extension, so each outbound route can be unique): > > exten => _50201NXXNXXXXXX,1,Dial(dialout-trunk,1,,D(www1234 > ${EXTEN:4})) > > and > > [outrt-005-VmXtoLONGDISTANCEcalls-custom] > > exten => _50201[12345678]XXNXXXXXX,1,Macro(user-callerid,SKIPTTL,) > > exten => _50201[12345678]XXNXXXXXX,n,Set(_NODEST=) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(record-enable, > ${AMPUSER},OUT,) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(dialout-trunk,1, > ${EXTEN:4},,) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(senddtmf) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(outisbusy,) > > [macro-senddtmf] > > exten => s,1,SendDTMF (1234) > > 2. The second problem is when a someone calls in and wishes to be > transferred to an office or user that is not local and does not have a > toll free number. Blind transfer will not work as the caller does not > know the PIN and trying a attended transfer did not work either, > Asterisk gives us a beep but we don?t receive one from the phone > company, how can we transfer a long distance call and enter the long > distance PIN before the transfer completes? > > Other than that the little Trixbox appliance is working fine, the > Sangoma T1 card works great so far, and the FXO ports have not been a > problem, faxing and analog phone are working. > > Any direction or help with the above would be greatly appreciated. > > Mark > > > > Mark Bergen > > Network Support Analyst From sourceforge at glazer.ca Thu Sep 2 13:49:26 2010 From: sourceforge at glazer.ca (Martin Glazer) Date: Thu, 02 Sep 2010 12:49:26 -0600 Subject: [*] Asterisk Digest, Vol 69, Issue 2 In-Reply-To: References: Message-ID: <4C7FF1B6.2010802@glazer.ca> Sorry about the digest reply... Mark, why don't you look at this another way - why don't you have Trixbox handle the long distance PIN dialing and remove the "feature" from Allstream? That way you will have more control over when a PIN is required, on which trunks and using which extensions? No need to worry about sending additional digits, etc. A client of ours here in Calgary had something similar and we moved them to PINs with FreePBX - much easier now. Just a thought.... Martin On 09/02/2010 11:00 AM, asterisk-request at muug.mb.ca wrote: > Send Asterisk mailing list submissions to > asterisk at muug.mb.ca > > To subscribe or unsubscribe via the World Wide Web, visit > http://www.muug.mb.ca/mailman/listinfo/asterisk > or, via email, send a message with subject or body 'help' to > asterisk-request at muug.mb.ca > > You can reach the person managing the list at > asterisk-owner at muug.mb.ca > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk digest..." > > > Today's Topics: > > 1. Long Distance PIN (Bergen, Mark) > 2. Re: Long Distance PIN (Bergen, Mark) > 3. Re: Long Distance PIN (Chris Friesen) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 1 Sep 2010 13:09:52 -0500 > From: "Bergen, Mark" > Subject: [*] Long Distance PIN > To: "asterisk at muug.mb.ca" > Cc: "Bergen, Mark" > Message-ID: > > Content-Type: text/plain; charset="us-ascii" > > Hello all; > We are using Trixbox 2.8 in one of our offices connected to a PRI (Allstream) and there are two things we are struggling with: > > 1. We use PIN's for all of our long distance calls, is there some way to have the PIN included when a user makes a long distance call by using DTMF maybe? It is for the VMX locator, some of our consultants work outside of Winnipeg and would like to give users the option of contacting them at the office they are working in or by cell phone, but when a user tries they hear MTS prompting them for the PIN. We have managed to create custom outbound routes for individual users but can't get the DTMF part to work. > > Here is some of what we have tried (5020 represents the users extension, so each outbound route can be unique): > > exten => _50201NXXNXXXXXX,1,Dial(dialout-trunk,1,,D(www1234${EXTEN:4})) > > and > > [outrt-005-VmXtoLONGDISTANCEcalls-custom] > > exten => _50201[12345678]XXNXXXXXX,1,Macro(user-callerid,SKIPTTL,) > > exten => _50201[12345678]XXNXXXXXX,n,Set(_NODEST=) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(record-enable,${AMPUSER},OUT,) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(dialout-trunk,1,${EXTEN:4},,) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(senddtmf) > > exten => _50201[12345678]XXNXXXXXX,n,Macro(outisbusy,) > > [macro-senddtmf] > > exten => s,1,SendDTMF (1234) > > 2. The second problem is when a someone calls in and wishes to be transferred to an office or user that is not local and does not have a toll free number. Blind transfer will not work as the caller does not know the PIN and trying a attended transfer did not work either, Asterisk gives us a beep but we don't receive one from the phone company, how can we transfer a long distance call and enter the long distance PIN before the transfer completes? > Other than that the little Trixbox appliance is working fine, the Sangoma T1 card works great so far, and the FXO ports have not been a problem, faxing and analog phone are working. > Any direction or help with the above would be greatly appreciated. > Mark > > Mark Bergen > Network Support Analyst > ONLINE BUSINESS SYSTEMS > Explore | Innovate | Lead > > 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 > mbergen at obsglobal.com | Direct Line: 204.982-0218 > Office: 204.982.0200 | Fax: 204.982.0201 > www.obsglobal.com > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100901/bc043e62/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Thu, 2 Sep 2010 10:00:23 -0500 > From: "Bergen, Mark" > Subject: Re: [*] Long Distance PIN > To: John Lange, "asterisk at muug.mb.ca" > > Message-ID: > > Content-Type: text/plain; charset="utf-8" > > Thank you for the reply John, really appreciate it. > I know you probably tried everything (yes, I'm reinventing the wheel, please be patient) but shouldn't something like this work: > exten => _50201NXXNXXXXXXwwww1234,1,Dial(dialout-trunk,1,,(${EXTEN:4})) > Then dial 5020 1 204 982 0218 1234 (1234 being the PIN) > MTS still operates in the same manner regarding answering the line for the pin while the call is still in progress. I saw a fix for it (patch) but it was for Asterisk 1.4, not sure if it would work for Trixbox 2.8 (Asterisk 1.6). > https://issues.asterisk.org/view.php?id=12123 > Thanks again for your time, I really appreciate it. > Mark > > Mark Bergen > Network Support Analyst > ONLINE BUSINESS SYSTEMS > Explore | Innovate | Lead > > 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 > mbergen at obsglobal.com |? Direct Line: 204.982-0218 > Office: 204.982.0200? |? Fax: 204.982.0201 > www.obsglobal.com > > > -----Original Message----- > From: John Lange [mailto:john at johnlange.ca] > Sent: September-01-10 9:24 PM > To: Bergen, Mark > Subject: Re: [*] Long Distance PIN > > For a brief time we had an Allstream PRI with PINs and I never got this working with Asterisk. > > The problem is, the PRI does not send the "answer" signal when prompting for the PIN leaving asterisk to think that the call is still making progress. In other words, Asterisk will not move to the next step in the dialplan so you can't do any sending of DTMF etc. > > It was over 3 years ago that I tried it so things may be different now. > > -- > John Lange > http://www.johnlange.ca > > > On Wed, 2010-09-01 at 13:09 -0500, Bergen, Mark wrote: > >> Hello all; >> >> We are using Trixbox 2.8 in one of our offices connected to a PRI >> (Allstream) and there are two things we are struggling with: >> >> 1. We use PIN?s for all of our long distance calls, is there some >> way to have the PIN included when a user makes a long distance call by >> using DTMF maybe? It is for the VMX locator, some of our consultants >> work outside of Winnipeg and would like to give users the option of >> contacting them at the office they are working in or by cell phone, >> but when a user tries they hear MTS prompting them for the PIN. We >> have managed to create custom outbound routes for individual users but >> can?t get the DTMF part to work. >> >> Here is some of what we have tried (5020 represents the users >> extension, so each outbound route can be unique): >> >> exten => _50201NXXNXXXXXX,1,Dial(dialout-trunk,1,,D(www1234 >> ${EXTEN:4})) >> >> and >> >> [outrt-005-VmXtoLONGDISTANCEcalls-custom] >> >> exten => _50201[12345678]XXNXXXXXX,1,Macro(user-callerid,SKIPTTL,) >> >> exten => _50201[12345678]XXNXXXXXX,n,Set(_NODEST=) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(record-enable, >> ${AMPUSER},OUT,) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(dialout-trunk,1, >> ${EXTEN:4},,) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(senddtmf) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(outisbusy,) >> >> [macro-senddtmf] >> >> exten => s,1,SendDTMF (1234) >> >> 2. The second problem is when a someone calls in and wishes to be >> transferred to an office or user that is not local and does not have a >> toll free number. Blind transfer will not work as the caller does not >> know the PIN and trying a attended transfer did not work either, >> Asterisk gives us a beep but we don?t receive one from the phone >> company, how can we transfer a long distance call and enter the long >> distance PIN before the transfer completes? >> >> Other than that the little Trixbox appliance is working fine, the >> Sangoma T1 card works great so far, and the FXO ports have not been a >> problem, faxing and analog phone are working. >> >> Any direction or help with the above would be greatly appreciated. >> >> Mark >> >> >> >> Mark Bergen >> >> Network Support Analyst >> >> ONLINE BUSINESS SYSTEMS >> >> Explore | Innovate | Lead >> >> >> >> 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 >> >> mbergen at obsglobal.com | Direct Line: 204.982-0218 >> >> Office: 204.982.0200 | Fax: 204.982.0201 >> >> www.obsglobal.com >> >> >> >> >> _______________________________________________ >> Asterisk mailing list >> Asterisk at muug.mb.ca >> http://www.muug.mb.ca/mailman/listinfo/asterisk >> > > > > ------------------------------ > > Message: 3 > Date: Thu, 2 Sep 2010 10:48:14 -0500 > From: "Chris Friesen" > Subject: Re: [*] Long Distance PIN > To: "'Bergen, Mark'", > Message-ID:<001901cb4ab6$41172410$c3456c30$@telenium.ca> > Content-Type: text/plain; charset="utf-8" > > Did you try adding something like this to the dial command? > > D([called][:calling]) > Sends DTMF digits after the call is answered but before it is bridged. The called digits are transmitted to the called party, the calling digits to the calling party. One or both parameters may be set. > > > I've seen and example like this > ... ,1,D([1234]) > > And I wonder , if you didn't want them to hear the DTMF tones could you use m to play some music on hold while the call is connected or r to play ringing. w for wait > ,1,rD(w[1234]) > > > -----Original Message----- > From: asterisk-bounces at muug.mb.ca [mailto:asterisk-bounces at muug.mb.ca] On Behalf Of Bergen, Mark > Sent: September-02-10 10:00 AM > To: John Lange; asterisk at muug.mb.ca > Subject: Re: [*] Long Distance PIN > > Thank you for the reply John, really appreciate it. > I know you probably tried everything (yes, I'm reinventing the wheel, please be patient) but shouldn't something like this work: > exten => _50201NXXNXXXXXXwwww1234,1,Dial(dialout-trunk,1,,(${EXTEN:4})) > Then dial 5020 1 204 982 0218 1234 (1234 being the PIN) MTS still operates in the same manner regarding answering the line for the pin while the call is still in progress. I saw a fix for it (patch) but it was for Asterisk 1.4, not sure if it would work for Trixbox 2.8 (Asterisk 1.6). > https://issues.asterisk.org/view.php?id=12123 > Thanks again for your time, I really appreciate it. > Mark > > Mark Bergen > Network Support Analyst > ONLINE BUSINESS SYSTEMS > Explore | Innovate | Lead > > 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 > Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com > > > -----Original Message----- > From: John Lange [mailto:john at johnlange.ca] > Sent: September-01-10 9:24 PM > To: Bergen, Mark > Subject: Re: [*] Long Distance PIN > > For a brief time we had an Allstream PRI with PINs and I never got this working with Asterisk. > > The problem is, the PRI does not send the "answer" signal when prompting for the PIN leaving asterisk to think that the call is still making progress. In other words, Asterisk will not move to the next step in the dialplan so you can't do any sending of DTMF etc. > > It was over 3 years ago that I tried it so things may be different now. > > -- > John Lange > http://www.johnlange.ca > > > On Wed, 2010-09-01 at 13:09 -0500, Bergen, Mark wrote: > >> Hello all; >> >> We are using Trixbox 2.8 in one of our offices connected to a PRI >> (Allstream) and there are two things we are struggling with: >> >> 1. We use PIN?s for all of our long distance calls, is there some >> way to have the PIN included when a user makes a long distance call by >> using DTMF maybe? It is for the VMX locator, some of our consultants >> work outside of Winnipeg and would like to give users the option of >> contacting them at the office they are working in or by cell phone, >> but when a user tries they hear MTS prompting them for the PIN. We >> have managed to create custom outbound routes for individual users but >> can?t get the DTMF part to work. >> >> Here is some of what we have tried (5020 represents the users >> extension, so each outbound route can be unique): >> >> exten => _50201NXXNXXXXXX,1,Dial(dialout-trunk,1,,D(www1234 >> ${EXTEN:4})) >> >> and >> >> [outrt-005-VmXtoLONGDISTANCEcalls-custom] >> >> exten => _50201[12345678]XXNXXXXXX,1,Macro(user-callerid,SKIPTTL,) >> >> exten => _50201[12345678]XXNXXXXXX,n,Set(_NODEST=) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(record-enable, >> ${AMPUSER},OUT,) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(dialout-trunk,1, >> ${EXTEN:4},,) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(senddtmf) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(outisbusy,) >> >> [macro-senddtmf] >> >> exten => s,1,SendDTMF (1234) >> >> 2. The second problem is when a someone calls in and wishes to be >> transferred to an office or user that is not local and does not have a >> toll free number. Blind transfer will not work as the caller does not >> know the PIN and trying a attended transfer did not work either, >> Asterisk gives us a beep but we don?t receive one from the phone >> company, how can we transfer a long distance call and enter the long >> distance PIN before the transfer completes? >> >> Other than that the little Trixbox appliance is working fine, the >> Sangoma T1 card works great so far, and the FXO ports have not been a >> problem, faxing and analog phone are working. >> >> Any direction or help with the above would be greatly appreciated. >> >> Mark >> >> >> >> Mark Bergen >> >> Network Support Analyst >> > > > > ------------------------------ > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk > > > End of Asterisk Digest, Vol 69, Issue 2 > *************************************** > From mbergen at obsglobal.com Thu Sep 2 14:30:58 2010 From: mbergen at obsglobal.com (Bergen, Mark) Date: Thu, 2 Sep 2010 14:30:58 -0500 Subject: [*] Long Distance PIN In-Reply-To: References: <1283394216.4635.3.camel@linux-k6vx.site> Message-ID: Hi John; Thanks to you and Chris for your replies. Is it possible to use a :progress command when calling out? Something like D([called[:calling[:progress]]])? This is really bugging me now, seems like such a simple thing to do :-). I'd rather not have to drop our MTS PINS but its starting to look like we'll have to. Take care; Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 mbergen at obsglobal.com |? Direct Line: 204.982-0218 Office: 204.982.0200? |? Fax: 204.982.0201 www.obsglobal.com -----Original Message----- From: John Lange [mailto:john at johnlange.ca] Sent: September-02-10 11:22 AM To: Bergen, Mark Subject: Re: [*] Long Distance PIN > exten => > _50201NXXNXXXXXXwwww1234,1,Dial(dialout-trunk,1,,(${EXTEN:4})) I'm pretty sure this won't work. The 'w' in the 'exten' just literally matches the character 'w'. It's not a "wait" signal like I assume you are thinking it is. What your after is something like: exten => _50201NXXNXXXXXX,1,Dial(DAHDI/g1/${EXTEN}||D(1234)) D([called][:calling]) - Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. The 'called' DTMF string is sent to the called party, and the 'calling' DTMF string is sent to the calling party. Both parameters can be used alone. However, as I said it doesn't work because asteirsk does not interpret the Allstream PIN prompt as an answered call so it never sends the DTMF. All this is based on my 1.4.X knowledge since I've never run 1.6 so you might want to see if things have changed in 1.6 that can allow this. Regards, John Lange On Thu, Sep 2, 2010 at 10:00 AM, Bergen, Mark wrote: > Thank you for the reply John, really appreciate it. > I know you probably tried everything (yes, I'm reinventing the wheel, please be patient) but shouldn't something like this work: > exten => > _50201NXXNXXXXXXwwww1234,1,Dial(dialout-trunk,1,,(${EXTEN:4})) > Then dial 5020 1 204 982 0218 1234 (1234 being the PIN) MTS still > operates in the same manner regarding answering the line for the pin while the call is still in progress. I saw a fix for it (patch) but it was for Asterisk 1.4, not sure if it would work for Trixbox 2.8 (Asterisk 1.6). > https://issues.asterisk.org/view.php?id=12123 > Thanks again for your time, I really appreciate it. > Mark > > Mark Bergen > Network Support Analyst > ONLINE BUSINESS SYSTEMS > Explore | Innovate | Lead > > 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 mbergen at obsglobal.com |? > Direct Line: 204.982-0218 > Office: 204.982.0200? |? Fax: 204.982.0201 www.obsglobal.com > > > -----Original Message----- > From: John Lange [mailto:john at johnlange.ca] > Sent: September-01-10 9:24 PM > To: Bergen, Mark > Subject: Re: [*] Long Distance PIN > > For a brief time we had an Allstream PRI with PINs and I never got this working with Asterisk. > > The problem is, the PRI does not send the "answer" signal when prompting for the PIN leaving asterisk to think that the call is still making progress. In other words, Asterisk will not move to the next step in the dialplan so you can't do any sending of DTMF etc. > > It was over 3 years ago that I tried it so things may be different now. > > -- > John Lange > http://www.johnlange.ca > > > On Wed, 2010-09-01 at 13:09 -0500, Bergen, Mark wrote: >> Hello all; >> >> We are using Trixbox 2.8 in one of our offices connected to a PRI >> (Allstream) and there are two things we are struggling with: >> >> 1. ? ? ?We use PIN's for all of our long distance calls, is there >> some way to have the PIN included when a user makes a long distance >> call by using DTMF maybe? It is for the VMX locator, some of our >> consultants work outside of Winnipeg and would like to give users the >> option of contacting them at the office they are working in or by >> cell phone, but when a user tries they hear MTS prompting them for >> the PIN. We have managed to create custom outbound routes for >> individual users but can't get the DTMF part to work. >> >> Here is some of what we have tried (5020 represents the users >> extension, so each outbound route can be unique): >> >> exten => _50201NXXNXXXXXX,1,Dial(dialout-trunk,1,,D(www1234 >> ${EXTEN:4})) >> >> and >> >> [outrt-005-VmXtoLONGDISTANCEcalls-custom] >> >> exten => _50201[12345678]XXNXXXXXX,1,Macro(user-callerid,SKIPTTL,) >> >> exten => _50201[12345678]XXNXXXXXX,n,Set(_NODEST=) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(record-enable, >> ${AMPUSER},OUT,) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(dialout-trunk,1, >> ${EXTEN:4},,) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(senddtmf) >> >> exten => _50201[12345678]XXNXXXXXX,n,Macro(outisbusy,) >> >> [macro-senddtmf] >> >> exten => s,1,SendDTMF (1234) >> >> 2. ? ? ?The second problem is when a someone calls in and wishes to >> be transferred to an office or user that is not local and does not >> have a toll free number. Blind transfer will not work as the caller >> does not know the PIN and trying a attended transfer did not work >> either, Asterisk gives us a beep but we don't receive one from the >> phone company, how can we transfer a long distance call and enter the >> long distance PIN before the transfer completes? >> >> Other than that the little Trixbox appliance is working fine, the >> Sangoma T1 card works great so far, and the FXO ports have not been a >> problem, faxing and analog phone are working. >> >> Any direction or help with the above would be greatly appreciated. >> >> Mark >> >> >> >> Mark Bergen >> >> Network Support Analyst >> >> ONLINE BUSINESS SYSTEMS >> >> Explore | Innovate | Lead >> >> >> >> 200-115 Bannatyne Ave., Winnipeg MB ?R3B 0R3 >> >> mbergen at obsglobal.com | ?Direct Line: 204.982-0218 >> >> Office: 204.982.0200 ?| ?Fax: 204.982.0201 >> >> www.obsglobal.com >> >> >> >> >> _______________________________________________ >> Asterisk mailing list >> Asterisk at muug.mb.ca >> http://www.muug.mb.ca/mailman/listinfo/asterisk > > -- John Lange www.johnlange.ca From mbergen at obsglobal.com Fri Sep 3 12:06:24 2010 From: mbergen at obsglobal.com (Bergen, Mark) Date: Fri, 3 Sep 2010 12:06:24 -0500 Subject: [*] Asterisk Digest, Vol 69, Issue 3 In-Reply-To: References: Message-ID: Hi Martin; I'm pretty sure that is how we will end up doing it and I agree it would work much better for us and the users. The finance department likes having all of the long distance records together in one bundle for our Calgary and Winnipeg offices, so worse case it's in two bundles :-). Funny note, the trixbox setup is also in a Calgary office. Do you keep track of long distance usage? ------ >From Martin: Sorry about the digest reply... Mark, why don't you look at this another way - why don't you have Trixbox handle the long distance PIN dialing and remove the "feature" from Allstream? That way you will have more control over when a PIN is required, on which trunks and using which extensions? No need to worry about sending additional digits, etc. A client of ours here in Calgary had something similar and we moved them to PINs with FreePBX - much easier now. Just a thought.... Martin *********** From mbergen at obsglobal.com Mon Sep 13 20:26:45 2010 From: mbergen at obsglobal.com (Bergen, Mark) Date: Mon, 13 Sep 2010 20:26:45 -0500 Subject: [*] Garbled Outbound Audio Message-ID: We have noticed on occasion the outbound audio on our phone system is garbled. We have checked the network and it seems to be in good order. It happens randomly and usually only for a few seconds (just long enough to annoy users). It has also happened when there was only a light load on the network. We configured one of the snom 320 handsets to only use the G711u codec and no real change (outbound audio was still garbled a couple of times briefly over a 1.5hr phone call). Our system uses VoIP only inside our local network, we have an MTS PRI for outbound calls. Anyone have any ideas or pointers? Thanks; Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100913/c3446664/attachment.html From sales at les.net Mon Sep 13 22:54:44 2010 From: sales at les.net (LES.NET (1996) INC.) Date: Mon, 13 Sep 2010 22:54:44 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: References: Message-ID: <4C8EF204.50708@les.net> One thing to do, is run MTR or something that does 10 pings per second. Monitor the pings in real-time, and when the audio anomaly is occurring, see what MTR says for ping times. Or better yet, an SNMP octet counter for that interface that is your bottleneck (default gateway) My suspicion is that something is consuming your full bandwidth for a short period of time. Pinging on 1 second intervals would not reveal this.. Les On 13/09/2010 8:26 PM, Bergen, Mark wrote: > > We have noticed on occasion the outbound audio on our phone system is > garbled. We have checked the network and it seems to be in good order. > It happens randomly and usually only for a few seconds (just long > enough to annoy users). It has also happened when there was only a > light load on the network. > > We configured one of the snom 320 handsets to only use the G711u codec > and no real change (outbound audio was still garbled a couple of times > briefly over a 1.5hr phone call). Our system uses VoIP only inside our > local network, we have an MTS PRI for outbound calls. > > Anyone have any ideas or pointers? > > Thanks; > > Mark > > *Mark Bergen* > > *Network Support Analyst* > > *ONLINE BUSINESS SYSTEMS*** > > *Explore | Innovate | Lead* > > * * > > 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3** > > _mbergen at obsglobal.com _ | Direct Line: > 204.982-0218 > > Office: 204.982.0200 | Fax: 204.982.0201 > > www.obsglobal.com > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100913/df5de6d9/attachment.html From sean at ertw.com Tue Sep 14 07:30:43 2010 From: sean at ertw.com (Sean Walberg) Date: Tue, 14 Sep 2010 07:30:43 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: <4C8EF204.50708@les.net> References: <4C8EF204.50708@les.net> Message-ID: What does it mean to check the network and find it in good order? A few things to watch for: - FDX/HDX mismatches (which shows up as collisions and runts) - multicast/broadcast bursts - queue drops on switches - stp events If you're handy with wireshark you can also set it to watch a port and look for problems in the RTP. Garbled voice is either packet loss, latency, or phone hardware. Wireshark will show you if it's one of the first two. Sean On Mon, Sep 13, 2010 at 10:54 PM, LES.NET (1996) INC. wrote: > One thing to do, is run MTR or something that does 10 pings per second. > Monitor the pings in real-time, and when the audio anomaly is occurring, > see what MTR says for ping times. > Or better yet, an SNMP octet counter for that interface that is your > bottleneck (default gateway) > > My suspicion is that something is consuming your full bandwidth for a short > period of time. > > Pinging on 1 second intervals would not reveal this.. > > Les > > On 13/09/2010 8:26 PM, Bergen, Mark wrote: > > We have noticed on occasion the outbound audio on our phone system is > garbled. We have checked the network and it seems to be in good order. It > happens randomly and usually only for a few seconds (just long enough to > annoy users). It has also happened when there was only a light load on the > network. > > We configured one of the snom 320 handsets to only use the G711u codec and > no real change (outbound audio was still garbled a couple of times briefly > over a 1.5hr phone call). Our system uses VoIP only inside our local > network, we have an MTS PRI for outbound calls. > > Anyone have any ideas or pointers? > > Thanks; > > Mark > > > > *Mark Bergen* > > *Network Support Analyst* > > *ONLINE BUSINESS SYSTEMS*** > > *Explore | Innovate | Lead* > > * * > > 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3** > > *mbergen at obsglobal.com* | Direct Line: 204.982-0218 > > Office: 204.982.0200 | Fax: 204.982.0201 > > www.obsglobal.com > > > > > _______________________________________________ > Asterisk mailing listAsterisk at muug.mb.cahttp://www.muug.mb.ca/mailman/listinfo/asterisk > > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk > > -- Sean Walberg http://ertw.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100914/456b032b/attachment-0001.html From Roland.Gallinera at MCICoach.com Tue Sep 14 09:25:03 2010 From: Roland.Gallinera at MCICoach.com (Roland.Gallinera at MCICoach.com) Date: Tue, 14 Sep 2010 09:25:03 -0500 Subject: [*] Garbled Outbound Audio Message-ID: You might want to try a different router or check your physical MTS lines. In some cases, if your physical line has problems, the router chokes and needs a couple of milliseconds to compensate hence an send/receive error occurs and retries occur. Check your router error messages (clear it first or take note of the first occurrence of the send/receive retries, and then check the time when the garbled audio occurred) Regards, Roland Gallinera Senior Analyst Programmer Motor Coach Industries 1475 Clarence Avenue Winnipeg, MB R3T 1T5 Direct Line: (204) 287-GEEK (287-4335) roland.gallinera at mcicoach.com "Bergen, Mark" Sent by: asterisk-bounces at muug.mb.ca 09/13/2010 08:26 PM To "asterisk at muug.mb.ca" cc Subject [*] Garbled Outbound Audio We have noticed on occasion the outbound audio on our phone system is garbled. We have checked the network and it seems to be in good order. It happens randomly and usually only for a few seconds (just long enough to annoy users). It has also happened when there was only a light load on the network. We configured one of the snom 320 handsets to only use the G711u codec and no real change (outbound audio was still garbled a couple of times briefly over a 1.5hr phone call). Our system uses VoIP only inside our local network, we have an MTS PRI for outbound calls. Anyone have any ideas or pointers? Thanks; Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com _______________________________________________ Asterisk mailing list Asterisk at muug.mb.ca http://www.muug.mb.ca/mailman/listinfo/asterisk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100914/9d2265c7/attachment.html From cfriesen at telenium.ca Tue Sep 14 09:55:11 2010 From: cfriesen at telenium.ca (Chris Friesen) Date: Tue, 14 Sep 2010 09:55:11 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: References: Message-ID: <020201cb541c$d50652d0$7f12f870$@telenium.ca> What card and server are you using to run Trixbox? We had a similar issue using Switchvox on an HP DL360 and a Digium PCIe card. We were able to match the random static with error messages in the /var/log, it turned out. What we were able to determine is that there was an issue with the digium driver and asterisk regarding timing that was causing this. We changed cards (which changed drivers) and the random static stopped. chris From: asterisk-bounces at muug.mb.ca [mailto:asterisk-bounces at muug.mb.ca] On Behalf Of Bergen, Mark Sent: September-13-10 8:27 PM To: asterisk at muug.mb.ca Subject: [*] Garbled Outbound Audio We have noticed on occasion the outbound audio on our phone system is garbled. We have checked the network and it seems to be in good order. It happens randomly and usually only for a few seconds (just long enough to annoy users). It has also happened when there was only a light load on the network. We configured one of the snom 320 handsets to only use the G711u codec and no real change (outbound audio was still garbled a couple of times briefly over a 1.5hr phone call). Our system uses VoIP only inside our local network, we have an MTS PRI for outbound calls. Anyone have any ideas or pointers? Thanks; Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100914/cd3d76b8/attachment.html From mbergen at obsglobal.com Tue Sep 14 10:11:23 2010 From: mbergen at obsglobal.com (Bergen, Mark) Date: Tue, 14 Sep 2010 10:11:23 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: <020201cb541c$d50652d0$7f12f870$@telenium.ca> References: <020201cb541c$d50652d0$7f12f870$@telenium.ca> Message-ID: Thank you for the input Chris; We are using Sangoma A101DE T1 and Sangoma A20400 8 FXS analog card with 35 Snom 320 handsets. I've heard the jitter earlier on when talking to a user directly (extension to extension) but assumed it was a network issue, I'm in Winnipeg and the user was in Calgary with the phones connecting to each other through a VPN. We did have our trixbox on a different vlan than the handsets, putting them both on the same vlan allowed us to bypass the router completely and the problem has gotten much better but is still present. It is generally about 6 to 10 seconds of 'garbled' outbound audio after about 20min, just long enough to be annoying. A user described it as sounding like someone talking underwater. I'll contact Sangoma and see if maybe it is a driver or setting issue, no users have complained yet about this issue happening between extension, just with outside calls. Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com From: Chris Friesen [mailto:cfriesen at telenium.ca] Sent: September-14-10 9:55 AM To: Bergen, Mark; asterisk at muug.mb.ca Subject: RE: [*] Garbled Outbound Audio What card and server are you using to run Trixbox? We had a similar issue using Switchvox on an HP DL360 and a Digium PCIe card. We were able to match the random static with error messages in the /var/log, it turned out. What we were able to determine is that there was an issue with the digium driver and asterisk regarding timing that was causing this. We changed cards (which changed drivers) and the random static stopped. chris From: asterisk-bounces at muug.mb.ca [mailto:asterisk-bounces at muug.mb.ca] On Behalf Of Bergen, Mark Sent: September-13-10 8:27 PM To: asterisk at muug.mb.ca Subject: [*] Garbled Outbound Audio We have noticed on occasion the outbound audio on our phone system is garbled. We have checked the network and it seems to be in good order. It happens randomly and usually only for a few seconds (just long enough to annoy users). It has also happened when there was only a light load on the network. We configured one of the snom 320 handsets to only use the G711u codec and no real change (outbound audio was still garbled a couple of times briefly over a 1.5hr phone call). Our system uses VoIP only inside our local network, we have an MTS PRI for outbound calls. Anyone have any ideas or pointers? Thanks; Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100914/c777092e/attachment-0001.html From ve4drk at gmail.com Tue Sep 14 10:50:03 2010 From: ve4drk at gmail.com (Dan Keizer) Date: Tue, 14 Sep 2010 10:50:03 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: References: <020201cb541c$d50652d0$7f12f870$@telenium.ca> Message-ID: One of the more common issues we hear on our phones during conference calls is the persistent interference of cell phones (specifically blackberries) .. it is not uncommon to have people asked to move their BB's away from the units .. this has a warbling type of effect on an audio level. Dan. On Tue, Sep 14, 2010 at 10:11 AM, Bergen, Mark wrote: > Thank you for the input Chris; > > We are using Sangoma A101DE T1 and Sangoma A20400 8 FXS analog card with 35 > Snom 320 handsets. I?ve heard the jitter earlier on when talking to a user > directly (extension to extension) but assumed it was a network issue, I?m in > Winnipeg and the user was in Calgary with the phones connecting to each > other through a VPN. > > We did have our trixbox on a different vlan than the handsets, putting them > both on the same vlan allowed us to bypass the router completely and the > problem has gotten much better but is still present. It is generally about 6 > to 10 seconds of ?garbled? outbound audio after about 20min, just long > enough to be annoying.? A user described it as sounding like someone talking > underwater. > > I?ll contact Sangoma and see if maybe it is a driver or setting issue, no > users have complained yet about this issue happening between extension, just > with outside calls. > > Mark > > > > Mark Bergen > > Network Support Analyst > > ONLINE BUSINESS SYSTEMS > > Explore | Innovate | Lead > > > > 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 > > mbergen at obsglobal.com |? Direct Line: 204.982-0218 > > Office: 204.982.0200? |? Fax: 204.982.0201 > > www.obsglobal.com > > > > From: Chris Friesen [mailto:cfriesen at telenium.ca] > Sent: September-14-10 9:55 AM > To: Bergen, Mark; asterisk at muug.mb.ca > Subject: RE: [*] Garbled Outbound Audio > > > > What card and server are you using to run Trixbox? > > > > We had a similar issue using Switchvox on an HP DL360 and a Digium PCIe > card. We were able to match the random static with error messages in the > /var/log, it turned out. What we were able to determine is that there was an > issue with the digium driver and asterisk regarding timing that was causing > this. We changed cards (which changed drivers) and the random static > stopped. > > > > chris > > > > From: asterisk-bounces at muug.mb.ca [mailto:asterisk-bounces at muug.mb.ca] On > Behalf Of Bergen, Mark > Sent: September-13-10 8:27 PM > To: asterisk at muug.mb.ca > Subject: [*] Garbled Outbound Audio > > > > We have noticed on occasion the outbound audio on our phone system is > garbled. We have checked the network and it seems to be in good order. It > happens randomly and usually only for a few seconds (just long enough to > annoy users). It has also happened when there was only a light load on the > network. > > We configured one of the snom 320 handsets to only use the G711u codec and > no real change (outbound audio was still garbled a couple of times briefly > over a 1.5hr phone call). Our system uses VoIP only inside our local > network, we have an MTS PRI for outbound calls. > > Anyone have any ideas or pointers? > > Thanks; > > Mark > > > > Mark Bergen > > Network Support Analyst > > ONLINE BUSINESS SYSTEMS > > Explore | Innovate | Lead > > > > 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 > > mbergen at obsglobal.com |? Direct Line: 204.982-0218 > > Office: 204.982.0200? |? Fax: 204.982.0201 > > www.obsglobal.com > > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk > > From mbergen at obsglobal.com Tue Sep 14 10:59:59 2010 From: mbergen at obsglobal.com (Bergen, Mark) Date: Tue, 14 Sep 2010 10:59:59 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: References: <020201cb541c$d50652d0$7f12f870$@telenium.ca> Message-ID: Thank you Dan; Interesting, but the audio has happened on calls to landlines as well. I'm glad you mentioned the cell phones though, at first I thought it was specific to them but just yesterday it happened again on a call to a landline. Mark -----Original Message----- From: Dan Keizer [mailto:ve4drk at gmail.com] Sent: September-14-10 10:50 AM To: Bergen, Mark Cc: Chris Friesen; asterisk at muug.mb.ca Subject: Re: [*] Garbled Outbound Audio One of the more common issues we hear on our phones during conference calls is the persistent interference of cell phones (specifically blackberries) .. it is not uncommon to have people asked to move their BB's away from the units .. this has a warbling type of effect on an audio level. Dan. On Tue, Sep 14, 2010 at 10:11 AM, Bergen, Mark wrote: > Thank you for the input Chris; > > We are using Sangoma A101DE T1 and Sangoma A20400 8 FXS analog card > with 35 Snom 320 handsets. I've heard the jitter earlier on when > talking to a user directly (extension to extension) but assumed it was > a network issue, I'm in Winnipeg and the user was in Calgary with the > phones connecting to each other through a VPN. > > We did have our trixbox on a different vlan than the handsets, putting > them both on the same vlan allowed us to bypass the router completely > and the problem has gotten much better but is still present. It is > generally about 6 to 10 seconds of 'garbled' outbound audio after > about 20min, just long enough to be annoying.? A user described it as > sounding like someone talking underwater. > > I'll contact Sangoma and see if maybe it is a driver or setting issue, > no users have complained yet about this issue happening between > extension, just with outside calls. > > Mark > > > > Mark Bergen > > Network Support Analyst > > ONLINE BUSINESS SYSTEMS > > Explore | Innovate | Lead > > > > 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 > > mbergen at obsglobal.com |? Direct Line: 204.982-0218 > > Office: 204.982.0200? |? Fax: 204.982.0201 > > www.obsglobal.com > > > > From: Chris Friesen [mailto:cfriesen at telenium.ca] > Sent: September-14-10 9:55 AM > To: Bergen, Mark; asterisk at muug.mb.ca > Subject: RE: [*] Garbled Outbound Audio > > > > What card and server are you using to run Trixbox? > > > > We had a similar issue using Switchvox on an HP DL360 and a Digium > PCIe card. We were able to match the random static with error messages > in the /var/log, it turned out. What we were able to determine is that > there was an issue with the digium driver and asterisk regarding > timing that was causing this. We changed cards (which changed drivers) > and the random static stopped. > > > > chris > > > > From: asterisk-bounces at muug.mb.ca [mailto:asterisk-bounces at muug.mb.ca] > On Behalf Of Bergen, Mark > Sent: September-13-10 8:27 PM > To: asterisk at muug.mb.ca > Subject: [*] Garbled Outbound Audio > > > > We have noticed on occasion the outbound audio on our phone system is > garbled. We have checked the network and it seems to be in good order. > It happens randomly and usually only for a few seconds (just long > enough to annoy users). It has also happened when there was only a > light load on the network. > > We configured one of the snom 320 handsets to only use the G711u codec > and no real change (outbound audio was still garbled a couple of times > briefly over a 1.5hr phone call). Our system uses VoIP only inside our > local network, we have an MTS PRI for outbound calls. > > Anyone have any ideas or pointers? > > Thanks; > > Mark > > > > Mark Bergen > > Network Support Analyst > > ONLINE BUSINESS SYSTEMS > > Explore | Innovate | Lead > > > > 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 > > mbergen at obsglobal.com |? Direct Line: 204.982-0218 > > Office: 204.982.0200? |? Fax: 204.982.0201 > > www.obsglobal.com > > > > _______________________________________________ > Asterisk mailing list > Asterisk at muug.mb.ca > http://www.muug.mb.ca/mailman/listinfo/asterisk > > From mbergen at obsglobal.com Tue Sep 14 11:02:44 2010 From: mbergen at obsglobal.com (Bergen, Mark) Date: Tue, 14 Sep 2010 11:02:44 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: References: Message-ID: Thank you for the response Roland; We are bypassing the router, but we hadn't thought about it possibly being a problem with the PRI itself. Oh, troubleshooting can be a pain in the butt :). Mark From: Roland.Gallinera at MCICoach.com [mailto:Roland.Gallinera at MCICoach.com] Sent: September-14-10 9:24 AM To: Bergen, Mark Subject: Re: [*] Garbled Outbound Audio You might want to try a different router or check your physical MTS lines. In some cases, if your physical line has problems, the router chokes and needs a couple of milliseconds to compensate hence an send/receive error occurs and retries occur. Check your router error messages (clear it first or take note of the first occurrence of the send/receive retries, and then check the time when the garbled audio occurred) Regards, Roland Gallinera Senior Analyst Programmer Motor Coach Industries 1475 Clarence Avenue Winnipeg, MB R3T 1T5 Direct Line: (204) 287-GEEK (287-4335) roland.gallinera at mcicoach.com "Bergen, Mark" > Sent by: asterisk-bounces at muug.mb.ca 09/13/2010 08:26 PM To "asterisk at muug.mb.ca" > cc Subject [*] Garbled Outbound Audio We have noticed on occasion the outbound audio on our phone system is garbled. We have checked the network and it seems to be in good order. It happens randomly and usually only for a few seconds (just long enough to annoy users). It has also happened when there was only a light load on the network. We configured one of the snom 320 handsets to only use the G711u codec and no real change (outbound audio was still garbled a couple of times briefly over a 1.5hr phone call). Our system uses VoIP only inside our local network, we have an MTS PRI for outbound calls. Anyone have any ideas or pointers? Thanks; Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com _______________________________________________ Asterisk mailing list Asterisk at muug.mb.ca http://www.muug.mb.ca/mailman/listinfo/asterisk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100914/ee4807de/attachment.html From ve4drk at gmail.com Tue Sep 14 11:25:22 2010 From: ve4drk at gmail.com (Dan Keizer) Date: Tue, 14 Sep 2010 11:25:22 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: References: <020201cb541c$d50652d0$7f12f870$@telenium.ca> Message-ID: well, it's more of an RF Interference issue what I'm talking about - -so it's dependent upon the susceptibility of the device itself, whether it's analog/digital doesn't really matter .. Dan. On Tue, Sep 14, 2010 at 10:59 AM, Bergen, Mark wrote: > Thank you Dan; > Interesting, but the audio has happened on calls to landlines as well. I'm glad you mentioned the cell phones though, at first I thought it was specific to them but just yesterday it happened again on a call to a landline. > Mark > > > -----Original Message----- > From: Dan Keizer [mailto:ve4drk at gmail.com] > Sent: September-14-10 10:50 AM > To: Bergen, Mark > Cc: Chris Friesen; asterisk at muug.mb.ca > Subject: Re: [*] Garbled Outbound Audio > > One of the more common issues we hear on our phones during conference calls is the persistent interference of cell phones (specifically > blackberries) .. it is not uncommon to have people asked to move their BB's away from the units ?.. this has a warbling type of effect on an audio level. > > Dan. > > On Tue, Sep 14, 2010 at 10:11 AM, Bergen, Mark wrote: >> Thank you for the input Chris; >> >> We are using Sangoma A101DE T1 and Sangoma A20400 8 FXS analog card >> with 35 Snom 320 handsets. I've heard the jitter earlier on when >> talking to a user directly (extension to extension) but assumed it was >> a network issue, I'm in Winnipeg and the user was in Calgary with the >> phones connecting to each other through a VPN. >> >> We did have our trixbox on a different vlan than the handsets, putting >> them both on the same vlan allowed us to bypass the router completely >> and the problem has gotten much better but is still present. It is >> generally about 6 to 10 seconds of 'garbled' outbound audio after >> about 20min, just long enough to be annoying.? A user described it as >> sounding like someone talking underwater. >> >> I'll contact Sangoma and see if maybe it is a driver or setting issue, >> no users have complained yet about this issue happening between >> extension, just with outside calls. >> >> Mark >> >> >> >> Mark Bergen >> >> Network Support Analyst >> >> ONLINE BUSINESS SYSTEMS >> >> Explore | Innovate | Lead >> >> >> >> 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 >> >> mbergen at obsglobal.com |? Direct Line: 204.982-0218 >> >> Office: 204.982.0200? |? Fax: 204.982.0201 >> >> www.obsglobal.com >> >> >> >> From: Chris Friesen [mailto:cfriesen at telenium.ca] >> Sent: September-14-10 9:55 AM >> To: Bergen, Mark; asterisk at muug.mb.ca >> Subject: RE: [*] Garbled Outbound Audio >> >> >> >> What card and server are you using to run Trixbox? >> >> >> >> We had a similar issue using Switchvox on an HP DL360 and a Digium >> PCIe card. We were able to match the random static with error messages >> in the /var/log, it turned out. What we were able to determine is that >> there was an issue with the digium driver and asterisk regarding >> timing that was causing this. We changed cards (which changed drivers) >> and the random static stopped. >> >> >> >> chris >> >> >> >> From: asterisk-bounces at muug.mb.ca [mailto:asterisk-bounces at muug.mb.ca] >> On Behalf Of Bergen, Mark >> Sent: September-13-10 8:27 PM >> To: asterisk at muug.mb.ca >> Subject: [*] Garbled Outbound Audio >> >> >> >> We have noticed on occasion the outbound audio on our phone system is >> garbled. We have checked the network and it seems to be in good order. >> It happens randomly and usually only for a few seconds (just long >> enough to annoy users). It has also happened when there was only a >> light load on the network. >> >> We configured one of the snom 320 handsets to only use the G711u codec >> and no real change (outbound audio was still garbled a couple of times >> briefly over a 1.5hr phone call). Our system uses VoIP only inside our >> local network, we have an MTS PRI for outbound calls. >> >> Anyone have any ideas or pointers? >> >> Thanks; >> >> Mark >> >> >> >> Mark Bergen >> >> Network Support Analyst >> >> ONLINE BUSINESS SYSTEMS >> >> Explore | Innovate | Lead >> >> >> >> 200-115 Bannatyne Ave., Winnipeg MB? R3B 0R3 >> >> mbergen at obsglobal.com |? Direct Line: 204.982-0218 >> >> Office: 204.982.0200? |? Fax: 204.982.0201 >> >> www.obsglobal.com >> >> >> >> _______________________________________________ >> Asterisk mailing list >> Asterisk at muug.mb.ca >> http://www.muug.mb.ca/mailman/listinfo/asterisk >> >> > From funk.jeff at gmail.com Tue Sep 14 13:58:27 2010 From: funk.jeff at gmail.com (Jeff Funk) Date: Tue, 14 Sep 2010 13:58:27 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: References: <020201cb541c$d50652d0$7f12f870$@telenium.ca> Message-ID: On Tue, Sep 14, 2010 at 11:25 AM, Dan Keizer wrote: > well, it's more of an RF Interference issue what I'm talking about - > -so it's dependent upon the susceptibility of the device itself, > whether it's analog/digital doesn't really matter .. > > Dan. That's probably the GSM mosquito you're hearing. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100914/6335150f/attachment.html From mbergen at obsglobal.com Tue Sep 14 14:03:46 2010 From: mbergen at obsglobal.com (Bergen, Mark) Date: Tue, 14 Sep 2010 14:03:46 -0500 Subject: [*] Garbled Outbound Audio In-Reply-To: References: <020201cb541c$d50652d0$7f12f870$@telenium.ca> Message-ID: Actually I thought it might be GSM, as a test we changed all the codec options on the snom 320 handset to G711u with no change. I'm not sure if it would be interference, at my desk sure, you should see what the compass on my iPhone does!! Server room is right next to me (which does explain the constant humming my head :)), but the last couple of times there was no know interference. Mark Mark Bergen Network Support Analyst ONLINE BUSINESS SYSTEMS Explore | Innovate | Lead 200-115 Bannatyne Ave., Winnipeg MB R3B 0R3 mbergen at obsglobal.com | Direct Line: 204.982-0218 Office: 204.982.0200 | Fax: 204.982.0201 www.obsglobal.com From: Jeff Funk [mailto:funk.jeff at gmail.com] Sent: September-14-10 1:58 PM To: ve4drk at gmail.com Cc: Bergen, Mark; asterisk at muug.mb.ca Subject: Re: [*] Garbled Outbound Audio On Tue, Sep 14, 2010 at 11:25 AM, Dan Keizer > wrote: well, it's more of an RF Interference issue what I'm talking about - -so it's dependent upon the susceptibility of the device itself, whether it's analog/digital doesn't really matter .. Dan. That's probably the GSM mosquito you're hearing. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100914/7f477d04/attachment.html From billreid at shaw.ca Tue Sep 14 17:27:45 2010 From: billreid at shaw.ca (Bill Reid) Date: Tue, 14 Sep 2010 17:27:45 -0500 Subject: [*] CIRA elections Message-ID: <4C8FF6E1.6060100@shaw.ca> Hi All, I wanted to let you know that I am running for the CIRA Board in the upcoming election Sept 22nd-29th. If you are a member I encourage you to vote. If you are not a CIRA member but hold a .CA domain name you still have time to become a member. You must submit your registration by this Sunday, Sept 17th. https://elections.cira.ca/2010/en/faq.html#q35 For the next week there is an online forum for discussing CIRA issues. https://elections.cira.ca/2010/campaign/welcome/en -- Bill From ve4drk at gmail.com Tue Sep 21 08:07:35 2010 From: ve4drk at gmail.com (Dan Keizer) Date: Tue, 21 Sep 2010 08:07:35 -0500 Subject: [*] Interesting codecs Message-ID: A snippet on /. caught my eye -- me, and a few on this list, being ham radio guys, thought this might be of interest as well :-) http://news.slashdot.org/story/10/09/21/0428259/Codec2-mdash-an-Open-Source-Low-Bandwidth-Voice-Codec?from=rss *At a 20ms update rate 51 bits/frame is 2550 bits/s* which points to the author's page ... http://www.rowetel.com/blog/?page_id=452 which points to Bruce K6BP's project setup: http://codec2.org/ and parts of his discussion: http://www.rowetel.com/blog/?p=128 Interesting reading ... Dan. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.muug.mb.ca/pipermail/asterisk/attachments/20100921/70d97d1e/attachment.html From billreid at shaw.ca Tue Sep 28 09:47:04 2010 From: billreid at shaw.ca (Bill Reid) Date: Tue, 28 Sep 2010 09:47:04 -0500 Subject: [*] Tues Oct 5 meeting cancelled Message-ID: <4CA1FFE8.5030204@shaw.ca> Hi All, Epic's presentation rooms are not set up yet and there is nothing on the agenda at this time so cancellation seems the best option. Also next Tues is Epic's Tech day. http://www.epic.ca/seminars/sem_reg_login.asp?sid=107 c u in Nov, Bill PS If you are a CIRA member voting closes tomorrow Sept 29th. I was pleased to see Michael Geist voted for me. http://www.michaelgeist.ca/