[asterisk-users] basic asterisk knowledge
I have question concerns asterisk 1-What is difference between G.729 and G.729A? 2-How can I know the requirement hardware for 150 extension on asterisk 1.4.4 making 50 simultaneous call? 3-Do asterisk have a codec conversion? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Going to VON Stockholm? Meet you at the Asterisk BOF!
The Asterisk BOF session will be tomorrow at 4.30 PM, VON Stockholm at the Stockholm Fairgrounds in Älvsjö. http://tinyurl.com/3degv5 From time to time you will find me in the Voop stand in the Digium/ Asterisk Pavillion in the exhibition. See you! /Olle - Sponsor Codename Pineapple - the Chan_sip3 development! http://www.codename-pineapple.org___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best format for audio via asterisk...
On Sun, 10 Jun 2007, Matthew Pease wrote: Hi all - We are using voicepulse connect asterisk together. We'd like to record our own outgoing messages, to be played back people that will be dialing into our voicepulse connect supplied DID. (am I getting this lingo right?) What is the best audio format to ensure the highest level of audio quality? Record them in the best quality you can (eg. 16-bit WAV), then use something like sox to transcode them into alaw, ulaw, gsm, and whatever else it can do. (What do people use to trascode into g729?) Then hopefully asterisk will pick the one with the least CPU cost to transcode when playing them down the line... If live transcoding isn't an issue for you then make them all g711 (ulaw if in the US, alaw elsewhere) If you don't have a good microphone, etc. then one of the easiest ways to record, is to use the Record() application built-in to Asterisk. Maybe not the best quality as it's already gone through a phnoe once, but it's good enough for testing Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] basic asterisk knowledge
On Sun, 10 Jun 2007, Khaled Chehab wrote: I have question concerns asterisk 1-What is difference between G.729 and G.729A? The letter A. http://en.wikipedia.org/wiki/G.729 ... says that G279A uses slightly less CPU to do the compression at the expense of sound quality. Digium appear to supply G279 rather than G729A. (at least they don't mention A) 2-How can I know the requirement hardware for 150 extension on asterisk 1.4.4 making 50 simultaneous call? Google or search the voip-wiki for asterisk scaling, etc. However these days you don't really have much choice - it's a 2.8-3.4GHz Pentium/something or a 1.8-3GHz Xeon/something, or a 3GHz AMD/something. (and their dual/quad processor versions) Basically any modern server class box will do for your needs unless you're transcoding every call. A 3GHz processor and 1GB or RAM will be fine - but you need to be careful with other issues - like making sure disk IO (if doing a lot of call recording/voicemail) won't interfere with Ethernet/Zap/TelcoInterface traffic... I know that 50 simulataneous calls will work fine on a 1GHz processor as long as you're not transcoding. Also, see this: http://www.digium.com/en/products/voice/g729codec.php where they have done some tests themselves and mention the transcoding numbers vs. CPU speed. 3-Do asterisk have a codec conversion? Asterisk will transcode between different codecs, if the codecs are compiled in, or licensed (g729) but transcoding comes at a big CPU cost. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bridged PRI calls - processor involvement?
This is the io wait figure from vmstat. If I run a vmstat 2 whilst I'm on a call I can see that the wa figure gets very high when the missing audio problem occurs. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 19:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? iowait time? I'm not familiar with that. Where are you seeing that? Also, is it a reproducible problem? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote: It probably did but we run in updates every week and nobody can state exactly when the problem started only a few weeks ago - not very helpful. I can see that when I hear the issue the iowait time is high on the processor. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? Did it accompany an update you made? If you can find out what version the problem started occurring, that would help in fixing the problem. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote: The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk: 1:1.4.0-34.fc5.at Libpri: 1:1.4.0-16.fc5.at Wildcard details: Found TE4XXP at base address fe3ffc00, remapped to f88bec00 TE4XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x377bb400 Reg 1: 0x377bb000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a TTE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (3rd Gen) TE4XXP: Span 1 configured for CCS/HDB3/CRC4 TE4XXP: Span 2 configured for CCS/HDB3/CRC4 The problem: At random points during calls we lose 1-3 seconds of speech (both ways both callee and caller), this can be replicated (or at least a very good approximation!) by generating a high level of interrupt/cpu activity (for instance copying data from a USB caddy as we tried the other day in an attempt to reproduce this more reliably). The calls are bridged PRI:PRI calls, no VOIP involvement. This was not a problem until approx 3-4 weeks ago, but I can't tie it down to an exact date. Steve Interrupt sharing is not a problem anymore with those cards. What version of zaptel did you try installing? Can you explain more about your problems? Also, your configuration and setup would help out as well. --- Matthew Fredrickson Digium, Inc. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339
RE: [asterisk-users] Bridged PRI calls - processor involvement?
Hi Steve, No, nothing like that, it has various updated from an 8mb Internet link and that's about it, I feel now that it's more down to disk I/O with the mpt driver than network. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: 09 June 2007 13:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement? Are you running recording on your box or FTPing large recording files or PDFs or anything other than just voice traffic? Has voice traffic spiked in conjunction with your problems? Are you doing any kind of port monitoring/mirroring on your switch? Most people look at the 100mb or 1Gb figure but there is also another very important spec to look at when evaluating a switch. It is Frame Forwarding Rate measured by Mpps. Take a look at your switch's docs and let us know what your FFR is and if you are doing any mirroring or link aggregation. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: Friday, June 08, 2007 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement? It probably did but we run in updates every week and nobody can state exactly when the problem started only a few weeks ago - not very helpful. I can see that when I hear the issue the iowait time is high on the processor. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? Did it accompany an update you made? If you can find out what version the problem started occurring, that would help in fixing the problem. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote: The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk: 1:1.4.0-34.fc5.at Libpri: 1:1.4.0-16.fc5.at Wildcard details: Found TE4XXP at base address fe3ffc00, remapped to f88bec00 TE4XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x377bb400 Reg 1: 0x377bb000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a TTE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (3rd Gen) TE4XXP: Span 1 configured for CCS/HDB3/CRC4 TE4XXP: Span 2 configured for CCS/HDB3/CRC4 The problem: At random points during calls we lose 1-3 seconds of speech (both ways both callee and caller), this can be replicated (or at least a very good approximation!) by generating a high level of interrupt/cpu activity (for instance copying data from a USB caddy as we tried the other day in an attempt to reproduce this more reliably). The calls are bridged PRI:PRI calls, no VOIP involvement. This was not a problem until approx 3-4 weeks ago, but I can't tie it down to an exact date. Steve Interrupt sharing is not a problem anymore with those cards. What version of zaptel did you try installing? Can you explain more about your problems? Also, your configuration and setup would help out as well. --- Matthew Fredrickson Digium, Inc. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by
RE: [asterisk-users] Bridged PRI calls - processor involvement?
On Mon, 11 Jun 2007, Steve Hanselman wrote: This is the io wait figure from vmstat. If I run a vmstat 2 whilst I'm on a call I can see that the wa figure gets very high when the missing audio problem occurs. I once looked after a Dell 2850 that exhibited some odd behaviour that I never got to the bottom of. It would seem to lock-up or just crawl for 2-3 seconds every now then. Nothing logged, noting on the console. It had 6 SCSI drives fitted. I rebuilt the server twice, rebuilt the s/w RAID arrays twice, even put all 6 drives in another box (which appeared towork OK), but never got to the bottom of it. Each disk would benchmark really fast individually, Ethernet performance was good, but overall, when everything was used together, it just didn't feel right. (compared to other Dells and other servers, biger smaller that I've built and used over the years). I'd see processes hung in a D state (waiting for IO to complete) for what seemed like an overly long time, (waiting on disk), but ... I suspected a BIOS pproblem, but never had a chance to get to the bottom of it. (It was a live server doing *everything* for a small company - DNS, NIS, NFS, Intranet/WiKi, Samba, etc, etc, etc,... so taking it offline for tests was problematic) So I wonder if looking at the BIOS and seeing if there are any Dell upgrades avalable for it might help? Gordon Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 19:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? iowait time? I'm not familiar with that. Where are you seeing that? Also, is it a reproducible problem? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote: It probably did but we run in updates every week and nobody can state exactly when the problem started only a few weeks ago - not very helpful. I can see that when I hear the issue the iowait time is high on the processor. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? Did it accompany an update you made? If you can find out what version the problem started occurring, that would help in fixing the problem. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote: The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk: 1:1.4.0-34.fc5.at Libpri: 1:1.4.0-16.fc5.at Wildcard details: Found TE4XXP at base address fe3ffc00, remapped to f88bec00 TE4XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x377bb400 Reg 1: 0x377bb000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a TTE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (3rd Gen) TE4XXP: Span 1 configured for CCS/HDB3/CRC4 TE4XXP: Span 2 configured for CCS/HDB3/CRC4 The problem: At random points during calls we lose 1-3 seconds of speech (both ways both callee and caller), this can be replicated (or at least a very good approximation!) by generating a high level of interrupt/cpu activity (for instance copying data from a USB caddy as we tried the other day in an attempt to reproduce this more reliably). The calls are bridged PRI:PRI calls, no VOIP involvement. This was not a problem until approx 3-4 weeks ago, but I can't tie it down to an exact date. Steve Interrupt sharing is not a problem anymore with those cards. What version of zaptel did you try installing? Can you explain more about your problems? Also, your configuration and setup would help out as well. --- Matthew Fredrickson Digium, Inc. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current
RE: [asterisk-users] Bridged PRI calls - processor involvement?
I checked for BIOS upgrades the other week and there were none. I'm starting to suspect kernel changes as being the reason for this so I guess I'm going to have to remove some of the patchy disk activity to smooth the load and then start researching!!! Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: 11 June 2007 09:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement? On Mon, 11 Jun 2007, Steve Hanselman wrote: This is the io wait figure from vmstat. If I run a vmstat 2 whilst I'm on a call I can see that the wa figure gets very high when the missing audio problem occurs. I once looked after a Dell 2850 that exhibited some odd behaviour that I never got to the bottom of. It would seem to lock-up or just crawl for 2-3 seconds every now then. Nothing logged, noting on the console. It had 6 SCSI drives fitted. I rebuilt the server twice, rebuilt the s/w RAID arrays twice, even put all 6 drives in another box (which appeared towork OK), but never got to the bottom of it. Each disk would benchmark really fast individually, Ethernet performance was good, but overall, when everything was used together, it just didn't feel right. (compared to other Dells and other servers, biger smaller that I've built and used over the years). I'd see processes hung in a D state (waiting for IO to complete) for what seemed like an overly long time, (waiting on disk), but ... I suspected a BIOS pproblem, but never had a chance to get to the bottom of it. (It was a live server doing *everything* for a small company - DNS, NIS, NFS, Intranet/WiKi, Samba, etc, etc, etc,... so taking it offline for tests was problematic) So I wonder if looking at the BIOS and seeing if there are any Dell upgrades avalable for it might help? Gordon Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 19:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? iowait time? I'm not familiar with that. Where are you seeing that? Also, is it a reproducible problem? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote: It probably did but we run in updates every week and nobody can state exactly when the problem started only a few weeks ago - not very helpful. I can see that when I hear the issue the iowait time is high on the processor. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? Did it accompany an update you made? If you can find out what version the problem started occurring, that would help in fixing the problem, Matthew Fredrickson Software/Hardware Engineer Digium, Inc. On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote: The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk: 1:1.4.0-34.fc5.at Libpri: 1:1.4.0-16.fc5.at Wildcard details: Found TE4XXP at base address fe3ffc00, remapped to f88bec00 TE4XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x377bb400 Reg 1: 0x377bb000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a TTE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (3rd Gen) TE4XXP: Span 1 configured for CCS/HDB3/CRC4 TE4XXP: Span 2 configured for CCS/HDB3/CRC4 The problem: At random points during calls we lose 1-3 seconds of speech (both ways both callee and caller), this can be replicated (or at least a very good approximation!) by generating a high level of interrupt/cpu activity (for instance copying data from a USB caddy as we tried the other day in an attempt to reproduce this more reliably). The calls are bridged PRI:PRI calls, no VOIP involvement. This was not a problem until approx 3-4 weeks ago, but I can't tie it down to an exact date. Steve Interrupt sharing is not a problem anymore with those cards. What version of zaptel did you try installing? Can you explain more about your problems? Also, your configuration and setup would help out as well. --- Matthew Fredrickson Digium, Inc. The information contained in this email is intended for the
Re: [asterisk-users] No Audio with Gtalk
Dear Michael, I got the same problem for a long time, but noboday give me some tips. Do you solve it? Best regards, Charles 2007/4/1, Michael Zoller [EMAIL PROTECTED]: I configured my * with the instructions found here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk to work with gtalk. The Phone rings and connects - but no audio! I am using a self-compiled asterisk 1.4.2 There is a lot of output on the CLI but I can't make sense of it. Perhaps somebody can help? Michael Output from the CLI: JABBER: gtalk_account OUTGOING: iq type='result' from='[EMAIL PROTECTED]/TalkB0AA717E' to='[EMAIL PROTECTED]/Talk.v10402D9EB78' id='317'/ atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=d type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=e type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ [Apr 1 09:50:28] NOTICE[20781]: chan_gtalk.c:1333 gtalk_indicate: Don't know how to indicate condition '-1' JABBER: gtalk_account OUTGOING: iq type='set' to='[EMAIL PROTECTED]/Talk.v10402D9EB78' from='[EMAIL PROTECTED]/TalkB0AA717E' id='f'session xmlns='http://www.google.com/session' type='accept' initiator='[EMAIL PROTECTED]/Talk.v10402D9EB78' id='2926563865'description xmlns='http://www.google.com/session/phone' xml:lang='en'payload-type id='0' name='PCMU' clockrate='8000' bitrate='64000'/payload-type id='100' name='EG711U' clockrate='8000' bitrate='64000'/payload-type id='106' name='telephone-event' clockrate='8000'//descriptiontransport xmlns='http://www.google.com/transport/p2p'//session/iq atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=f type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ atlas*CLI JABBER: gtalk_account OUTGOING: iq type='set' from='[EMAIL PROTECTED]/TalkB0AA717E' to='[EMAIL PROTECTED]/Talk.v10402D9EB78' id='g'session type='terminate' id='2926563865' initiator='[EMAIL PROTECTED]/Talk.v10402D9EB78' xmlns='http://www.google.com/session'//iq atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=g type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH Problems.
All, I am using Asterisk 1.4.4 and it is not playing any MOH. I think the underlying problem is the following error: [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/moh/asterisk' [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player Now it does not matter what I change in the directy= to in the heading [default] in the file musiconhold.conf. [default] mode=files directory=/var/lib/asterisk/moh/klavo I still get the error: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/moh/asterisk' which does not make sense to me. I don't have any other MOH defined As soon as MOH is initiated is immediately stop with no error. -- Executing [EMAIL PROTECTED]:2] MusicOnHold(SIP/202-0895d428, default) in new stack -- Started music on hold, class 'default', on channel 'SIP/202-0895d428' -- Stopped music on hold on SIP/202-0895d428 It's like the musiconhold.conf file is not read. I have rebooted and reloaded with no chance to the above. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio with Gtalk
Well I found out what the reason is. When the gtalk client is behind a NAT it will not work (at least for me it doesn't). Citing from : http://www.google.com/support/talk/bin/answer.py?answer=27930query=udptopic=type= gtalk needs either UDP connections to anywhen on any port OR TCP Connections to anywhen on port 443. For me the only way to get it to work has been to open the UDP ports, which is unacceptable for day-to-day use. Check out these bug reports for more on the subject: #7686 #8193 and #8655. None of the proposed bug fixes have worked for me though - and so I have given up for the time being. Michael Charles Wang wrote: Dear Michael, I got the same problem for a long time, but noboday give me some tips. Do you solve it? Best regards, Charles 2007/4/1, Michael Zoller [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: I configured my * with the instructions found here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk to work with gtalk. The Phone rings and connects - but no audio! I am using a self-compiled asterisk 1.4.2 There is a lot of output on the CLI but I can't make sense of it. Perhaps somebody can help? Michael Output from the CLI: JABBER: gtalk_account OUTGOING: iq type='result' from='[EMAIL PROTECTED]/TalkB0AA717E' to='[EMAIL PROTECTED]/Talk.v10402D9EB78' id='317'/ atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=d type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=e type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ [Apr 1 09:50:28] NOTICE[20781]: chan_gtalk.c:1333 gtalk_indicate: Don't know how to indicate condition '-1' JABBER: gtalk_account OUTGOING: iq type='set' to='[EMAIL PROTECTED] /Talk.v10402D9EB78' from='[EMAIL PROTECTED]/TalkB0AA717E' id='f'session xmlns='http://www.google.com/session' type='accept' initiator='[EMAIL PROTECTED]/Talk.v10402D9EB78' id='2926563865'description xmlns='http://www.google.com/session/phone' xml:lang='en'payload-type id='0' name='PCMU' clockrate='8000' bitrate='64000'/payload-type id='100' name='EG711U' clockrate='8000' bitrate='64000'/payload-type id='106' name='telephone-event' clockrate='8000'//descriptiontransport xmlns='http://www.google.com/transport/p2p'/ http://www.google.com/transport/p2p'//session/iq atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED] /TalkB0AA717E id=f type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ atlas*CLI JABBER: gtalk_account OUTGOING: iq type='set' from=' [EMAIL PROTECTED]/TalkB0AA717E' to='[EMAIL PROTECTED]/Talk.v10402D9EB78' id='g'session type='terminate' id='2926563865' initiator='[EMAIL PROTECTED] /Talk.v10402D9EB78' xmlns='http://www.google.com/session'/ http://www.google.com/session'//iq atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED] /TalkB0AA717E id=g type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH Problems.
On Monday 11 June 2007, Klaverstyn, David C wrote: I think the underlying problem is the following error: You are right. You have to put some mp3 files to /var/lib/asterisk/moh/asterisk according to your configuration. regards t. -- knowledgeTools® ... managing complexity. -- knowledgeTools International GmbH Wallstraße 15 / 15 a 10179 Berlin Fon: +49 30 726 169 20 Fax: +49 30 726 169 249 [EMAIL PROTECTED] www.knowledgetools.de Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378 Geschäftsführer: Oliver Seyboldt, Reinhard Kunz -- This eMail communication (and any attachment/s) may contain confidential or privileged information and is intended only for the individual(s) or entity named above and to others who have been specifically authorized to receive it. If you are not the intended recipient, please do not read, copy, use or disclose the contents of this communication to others. Please notify the sender that you have received this e-mail in error by reply e-mail, and delete the e-mail subsequently. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR on transfers of called ZAP channel
Hello list,
I have a problem with called ZAP channels making an attended-transfer
or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk
is wrong.
At the moment there is a bristuffed Asterisk 1.2.18 running with
bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit:
[default]
exten = 0123456789,1,Macro(dialpstn,${EXTEN})
[macro-dialpstn]
exten = s,1,Set(TRANSFER_CONTEXT=transfer)
exten = s,2,Set(FORWARD_CONTEXT=transfer)
exten = s,3,Set(CIDNUMORIG=${CALLERID(num)}) ; save callerid-num
exten = s,4,Set(CALLERID(num)=98765) ; dial out using msn 98765
exten = s,5,Dial(Zap/g1/${ARG1}|30|t)
exten = h,1,Set(CALLERID(num)=${CIDNUMORIG}) restore callerid-num for
CDR
[transfer]
exten = _X.,1,Set(CALLERID(all)=External 0123456789)
exten = _X.,1,Dial(SIP/${EXTEN})
So I call 0123456789 with SIP phone 10. The callee dials *1 20 for
attended transfer and SIP phone 20 (I have *1 for attended transfer in
features.conf). The called SIP-phone shows the caller-information I
set in context transfer. But the CDR is wrong, it has 98765 in MySQL
field src. So it seems I can't overwrite the ${CALLERID(num)} for cdr,
but for the called channel.
Anybody who can explain that? Or any solution for called Zap channels
making an attended transfer?
--
Best regards,
Gunnar Schaller
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[asterisk-users] Searchable List Archives?
I'd like to be able to search the list archives when I'm reading someone's message to put what they say in context based on what they've said, and what others have said in conversation with them, in the past. It would help me figure out whether to trust some submitters on some issues, and just learn more from the community's collective/cumulative research and discussion. Is there list server Web SW that lets me look at a message in the archives, then click on it to get every message (*across all months*) sent by that author, then every message in the thread (by Message-ID and same/similar subject)? Based on searches by regexp in each message field, including Body. Maybe Digium could upgrade the list SW, or let me do it for them. Or I could set it up at my website, then import the list archive data and parse it into my DB for a searchable mirror. Does the SW with those features exist already, or do I have to write it? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help on text entry. using asterisk.
Hi, please help me in developing and reading Text through IVR application using asterisk. can any one help me at highlevel on this, other than using SPANDSP application. Regards K.Rajesh. _ Marriage? Get Detailed Profiles only at Shaadi.com. www.shaadi.com/ptnr.php?ptnr=mhottag ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi, please help me in developing and reading Text through IVR application using asterisk. can any one help me at highlevel on this, other than using SPANDSP application. Regards K.Rajesh. _ Tried the new MSN Messenger? Its cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searchable List Archives?
Matthew Rubenstein wrote: Maybe Digium could upgrade the list SW, or let me do it for them. Or I could set it up at my website, then import the list archive data and parse it into my DB for a searchable mirror. Assuming google is indexing the list archives at http://lists.digium.com/pipermail/asterisk-users you should be able search the archives using google. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] change moh during a call?
Hello. Is it possible to change the defined moh sound file within an extension? I have: exten = 18,1,Answer exten = 18,n,Wait(3) exten = 18,n,SetMusicOnHold(durchwahl) exten = 18,n,Dial(SIP/118,15,m) exten = 18,n,Hangup Now i have the situation someone calls and my phone is ringing while moh (durchwahl) is playing. When i pickup the call and press the hold button during the call moh (durchwahl) is playing as moh. Thats not what i want. Can i define another file within this extension? regards t. -- knowledgeTools® ... managing complexity. -- knowledgeTools International GmbH Wallstraße 15 / 15 a 10179 Berlin Fon: +49 30 726 169 20 Fax: +49 30 726 169 249 [EMAIL PROTECTED] www.knowledgetools.de Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378 Geschäftsführer: Oliver Seyboldt, Reinhard Kunz -- This eMail communication (and any attachment/s) may contain confidential or privileged information and is intended only for the individual(s) or entity named above and to others who have been specifically authorized to receive it. If you are not the intended recipient, please do not read, copy, use or disclose the contents of this communication to others. Please notify the sender that you have received this e-mail in error by reply e-mail, and delete the e-mail subsequently. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searchable List Archives?
Per Jessen wrote: Matthew Rubenstein wrote: Maybe Digium could upgrade the list SW, or let me do it for them. Or I could set it up at my website, then import the list archive data and parse it into my DB for a searchable mirror. Assuming google is indexing the list archives at http://lists.digium.com/pipermail/asterisk-users you should be able search the archives using google. Search for SIP in the archives: http://www.google.ch/search?num=100hl=ensafe=offq=site%3Alists.digium.com+SIPbtnG=Searchmeta= /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searchable List Archives?
On Mon, Jun 11, 2007 at 08:50:43AM -0400, Matthew Rubenstein wrote: I'd like to be able to search the list archives when I'm reading someone's message to put what they say in context based on what they've said, and what others have said in conversation with them, in the past. It would help me figure out whether to trust some submitters on some issues, and just learn more from the community's collective/cumulative research and discussion. Is there list server Web SW that lets me look at a message in the archives, then click on it to get every message (*across all months*) sent by that author, then every message in the thread (by Message-ID and same/similar subject)? Based on searches by regexp in each message field, including Body. Maybe Digium could upgrade the list SW, or let me do it for them. Or I could set it up at my website, then import the list archive data and parse it into my DB for a searchable mirror. Does the SW with those features exist already, or do I have to write it? http://gmane.org/find.php?list=asterisk http://dir.gmane.org/gmane.comp.telephony.pbx.asterisk -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to tell what codec is used for each end of a call MD110-H323-SIP
[EMAIL PROTECTED] wrote: Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk for SIP I use 'sip show channels' I'm not sure what the equivilent h323 command is. And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Try 'rtp debug' and the rtp packets should scroll by. Thank you for your time and effort to respond. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Different ECs in Asterisk
Does anybody have done some analysis on the different ECs come with Zaptel Driver? If so, can somebody post some summery? Thanks, Yonghua - Need a vacation? Get great deals to amazing places on Yahoo! Travel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as an SCCP client
Hi all, Has anyone tried using Asterisk as an SCCP client? My company has just signed up a 2 year agreement with M5 (fools!!) but are having intellectual issues with things like intra office phone calls and voice mail etc. They suddenly realized after M5 was installed that ALL their calls go out to the Internet and back and they don't like it. M5 uses SCCP. Could an Asterisk box be configured to run as an SCCP client (or many clients) so as to emulate the M5 handsets? At least then we would be in control of our own calls and voice mail. Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes
Hi, The system seems to be IO bound for some reason. Reading at the older posts you mentioned that there is no significant disc activity so it could be ethernet i/o and/or interrupts that are causing this (old or insuficient ethernet driver maybe ?) Usually this kind of i/o wait is present on machines that have run out of memory and need to swap to disk Also with regard to the higher system usage on multicore systems, its very probable that its due to task migration from core to core Here is something we recently noticed that may explain why the dual-core server is under-performing at high call volumes. The following numbers were collected off both servers while they were in production. Note that while they have similar cumulative idle values, the ratio of system time to user time on the single-core server is roughly 2.3 to 1, but on the dual-core server it is roughly 19.6 to 1. I'm not quite sure what to make of this, but it seems to be very relevant to the problem. Mon Apr 2 12:15:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (4core.imminc.com) 04/02/07 12:24:01 CPU %user %nice %system %iowait %idle 12:25:02 all 14.97 0.03 34.25 0.92 49.82 12:25:020 8.83 0.05 33.60 1.28 56.24 12:25:021 17.50 0.02 34.60 0.57 47.32 12:25:022 19.94 0.02 33.52 1.31 45.22 12:25:023 13.62 0.02 35.29 0.52 50.55 Thu May 10 15:30:01 EDT 2007 Idle (sar -P ALL 60 14) (60 seconds 14 slices) Linux 2.6.12-1.1376_FC3smp (8core.imminc.com) 05/10/07 15:38:01 CPU %user %nice %system %iowait %idle 15:39:01 all 2.47 0.01 48.29 0.00 49.23 15:39:010 2.92 0.00 53.17 0.00 43.91 15:39:011 2.98 0.00 48.68 0.02 48.33 15:39:012 2.47 0.02 48.61 0.00 48.91 15:39:013 2.27 0.00 48.35 0.00 49.38 15:39:014 2.38 0.02 47.38 0.00 50.22 15:39:015 2.37 0.02 46.94 0.00 50.67 15:39:016 2.23 0.02 46.63 0.00 51.12 15:39:017 2.17 0.02 46.54 0.00 51.27 Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR on transfers of called ZAP channel
Gunnar--
CDR generation that covers transfers is an umimplemented feature in
Asterisk, in any version.
I have been working on a solution, but unfortunately, my solution is
radical enough that I dare not apply it to 1.2 or even 1.4. It will most
likely break every working implementation of billing that has been built
on Asterisk by end users/developers. Unpleasant visions of angry mobs of
developers armed with baseball bats, who want nothing more than to drag
me out of my home and share their pain and frustration over my
fixes. you get the idea.
Actually, I have TWO solutions! One, is to modify the current CDR
engine, the other is to provide an entirely different solution that is
single-event driven, kinda along the lines of manager events, but more
streamlined for CDR billing purposes.
The first solution somewhat reorganizes CDRS by no longer posting them
to the backend db's when a hangup occurs. Rather, it will post them when
a bridge between channels is finished, or ends. Since a Local
channel acts as a sort of bridge, I think I will have to do the same
thing there. I'm in the middle of it now. I spent/wasted a good amount
of time generating extra CDR's that would describe time in different
parts of a transfer, but as I traveled further down that road, I see
that this will only make things unnecessarily complex. So, I'm not going
to do it. What this means is that a CDR will get generated for each
chunk of a conversation involved in a transfer, but these pieces will
not tell you much about how the chunks relate to each other. The channel
originating the conversation will be the source, and the channel
originally connected to will be the destination. Time spent in 3-way
conferences, music on hold, etc. etc. will most likely not be available.
My theory is that, in most cases, it won't matter. All you REALLY want
to know is who to bill, and for how much time. If a transfer occurs, it
involves someone internally dialing another party. This second
conversation, will generate another CDR, and the guy who dialed it
will be assigned that call, even if he hung up before the call was
answered (blind xfer). For example, picture this: a switch in Modesto
gets a call from Sacramento, and extension 151 gets this call, and dials
Shanghai, and blind transfers the Sacramento call to Shanghai, and then
Sacramento and Shanghai talk for an hour. Two CDR's will be generated.
One will cover the incoming call from Sacramento, and will be little
over an hour. The other CDR that will come out will say 151 dialed
Shanghai and talked an hour. That's it.
The second solution, the event-based one, will generate an event record
for each significant event in the life of each channel. So, START
events when a channel is born; ANSWER events when someone answers a
call; END events when somebody hangs up. There will also be Park,
and Transfer, and MOH, and 3-WAY, Conference-Join, and several
others. Just enough information will be included with each event to
thread together billable sequences. Along with each event record will be
the time the event happened, and channel info. This approach will be
very much more fine-grained, and allow you to do fancy things like
figure out that Sacramento was the only person talking to Shanghai, and
allow you to bill the call to the guy/gal in Sacramento. Trouble with
this approach is that threading together the event records is a
non-trivial operation! But I hope to provide some tools that will make
this easier to do.
So, the bad news is: you will not see any solutions for this problem, in
1.2, or 1.4. the CDR fix (first solution) will most likely end up in
1.6, the event-based solution will probably not be available until 1.8
or 1.10; we shall see.
murf
On Mon, 2007-06-11 at 14:26 +0200, Gunnar Schaller wrote:
Hello list,
I have a problem with called ZAP channels making an attended-transfer
or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk
is wrong.
At the moment there is a bristuffed Asterisk 1.2.18 running with
bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit:
[default]
exten = 0123456789,1,Macro(dialpstn,${EXTEN})
[macro-dialpstn]
exten = s,1,Set(TRANSFER_CONTEXT=transfer)
exten = s,2,Set(FORWARD_CONTEXT=transfer)
exten = s,3,Set(CIDNUMORIG=${CALLERID(num)}) ; save callerid-num
exten = s,4,Set(CALLERID(num)=98765) ; dial out using msn 98765
exten = s,5,Dial(Zap/g1/${ARG1}|30|t)
exten = h,1,Set(CALLERID(num)=${CIDNUMORIG}) restore callerid-num for
CDR
[transfer]
exten = _X.,1,Set(CALLERID(all)=External 0123456789)
exten = _X.,1,Dial(SIP/${EXTEN})
So I call 0123456789 with SIP phone 10. The callee dials *1 20 for
attended transfer and SIP phone 20 (I have *1 for attended transfer in
features.conf). The called SIP-phone shows the caller-information I
set in context transfer. But the CDR is wrong, it has 98765 in MySQL
field src. So it seems I can't overwrite the ${CALLERID(num)} for cdr,
but for the called channel.
Re: [asterisk-users] IAX Peers show command
Ronaldo Z. Afonso wrote: Hi all, What does (T) mean on the output of iax2 show peers? The following my output. darkstar*CLI iax2 show peers Name/UsernameHost Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0 online, 2 offline, 0 unmonitored] Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users T is for TCP, U would be UDP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Peers show command
Hi Anthony, It doesn't make sense. This peer is an IAX peer. It was supposed to use UDP. Does Asterisk also use TCP for IAX? Thanks Ronaldo. Anthony Francis wrote: Ronaldo Z. Afonso wrote: Hi all, What does (T) mean on the output of iax2 show peers? The following my output. darkstar*CLI iax2 show peers Name/UsernameHost Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0 online, 2 offline, 0 unmonitored] Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users T is for TCP, U would be UDP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Peers show command
On Mon, 2007-06-11 at 09:59 -0600, Anthony Francis wrote: T is for TCP, U would be UDP ___ Actually T stands for TRUNK. IAX2 is always UDP. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Peers show command
Ronaldo Z. Afonso wrote: Hi all, What does (T) mean on the output of iax2 show peers? The following my output. darkstar*CLI iax2 show peers Name/UsernameHost Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0 online, 2 offline, 0 unmonitored] Ronaldo. FYI, (T) stands for Trunked peer. Basically means that the communication with that particular host is optimal, with all of the channels using the same packet envelope. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] change moh during a call?
Add a r option with that extension so that way when it is called it will ring and then when you put it on hold it will play your moh. You can also set the MOH in sip.conf, but you will still get the same behavior you have now. Thomas Stein wrote: Hello. Is it possible to change the defined moh sound file within an extension? I have: exten = 18,1,Answer exten = 18,n,Wait(3) exten = 18,n,SetMusicOnHold(durchwahl) exten = 18,n,Dial(SIP/118,15,m) exten = 18,n,Hangup Now i have the situation someone calls and my phone is ringing while moh (durchwahl) is playing. When i pickup the call and press the hold button during the call moh (durchwahl) is playing as moh. Thats not what i want. Can i define another file within this extension? regards t. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Peers show command
T = Trunking. If it's present then trunking is enabled. Ronaldo wrote: Hi Anthony, It doesn't make sense. This peer is an IAX peer. It was supposed to use UDP. Does Asterisk also use TCP for IAX? Thanks Ronaldo. -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Calls being dropped
Where do I get oej's patch, and how do I install it? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Brodmann Sent: Tuesday, June 05, 2007 2:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls being dropped We have a similar problem at our place, since a few months. oej, mentioned a patch he has made after the release of asterisk-1.4.4. So we're all desperately waiting for asterisk-1.4.5 to be released; unless you want to install from svn. 2007/6/4, Compnet Bobby [EMAIL PROTECTED]: We have the latest version of asterisk, on a xeon dell server (2gb ram), with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest stable firmware) and are having a few problems. We have a basic menu that transfers calls to different extensions. The problems can be found on all extensions. We have 2 different incoming providers and the problem happens on both providers. I want your input on 2 problems, they are the following: 1. 60% of the time everything works fine and there are no problems, 40% of times when the calls are transferred to an extension, after a few seconds , the call drops. The log from the server is below(this is from pickup to hangup, the main area of concern is where it says warning). -- Executing [EMAIL PROTECTED]:1] Answer(SIP/9097406868-09e110f8, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/9097406868-09e110f8, menus/welcome-to-exec) in new stack -- SIP/9097406868-09e110f8 Playing 'menus/welcome-to-exec' (language 'en') == CDR updated on SIP/9097406868-09e110f8 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/9097406868-09e110f8, SIP/103|50|m) in new stack -- Called 103 -- Started music on hold, class 'default', on SIP/9097406868-09e110f8 -- SIP/103-09dedd68 is ringing [May 29 09:05:22] WARNING[2678]: chan_sip.c:1899 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1 (Critical Response) [May 29 09:05:22] WARNING[2678]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. -- Stopped music on hold on SIP/9097406868-09e110f8 == Spawn extension (from-sip, 103, 1) exited non-zero on 'SIP/9097406868-09e110f8' 2. When a call comes in or is transferred(not on outgoing), there is a delay until the person on the incoming line can hear you. We can hear them, but they can't hear us. Sometimes there is no delay, sometimes for person calling in cant hear you for 6 seconds. Thanks for the help in advance!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR on transfers of called ZAP channel
On Mon, 2007-06-11 at 09:11 -0600, Steve Murphy wrote:
Gunnar--
CDR generation that covers transfers is an umimplemented feature in
Asterisk, in any version.
I have been working on a solution, but unfortunately, my solution is
radical enough that I dare not apply it to 1.2 or even 1.4. It will most
likely break every working implementation of billing that has been built
on Asterisk by end users/developers. Unpleasant visions of angry mobs of
developers armed with baseball bats, who want nothing more than to drag
me out of my home and share their pain and frustration over my
fixes. you get the idea.
Actually, I have TWO solutions! One, is to modify the current CDR
engine, the other is to provide an entirely different solution that is
single-event driven, kinda along the lines of manager events, but more
streamlined for CDR billing purposes.
The first solution somewhat reorganizes CDRS by no longer posting them
to the backend db's when a hangup occurs. Rather, it will post them when
a bridge between channels is finished, or ends. Since a Local
channel acts as a sort of bridge, I think I will have to do the same
thing there. I'm in the middle of it now. I spent/wasted a good amount
of time generating extra CDR's that would describe time in different
parts of a transfer, but as I traveled further down that road, I see
that this will only make things unnecessarily complex. So, I'm not going
to do it. What this means is that a CDR will get generated for each
chunk of a conversation involved in a transfer, but these pieces will
not tell you much about how the chunks relate to each other. The channel
originating the conversation will be the source, and the channel
originally connected to will be the destination. Time spent in 3-way
conferences, music on hold, etc. etc. will most likely not be available.
My theory is that, in most cases, it won't matter. All you REALLY want
to know is who to bill, and for how much time. If a transfer occurs, it
involves someone internally dialing another party. This second
conversation, will generate another CDR, and the guy who dialed it
will be assigned that call, even if he hung up before the call was
answered (blind xfer). For example, picture this: a switch in Modesto
gets a call from Sacramento, and extension 151 gets this call, and dials
Shanghai, and blind transfers the Sacramento call to Shanghai, and then
Sacramento and Shanghai talk for an hour. Two CDR's will be generated.
One will cover the incoming call from Sacramento, and will be little
over an hour. The other CDR that will come out will say 151 dialed
Shanghai and talked an hour. That's it.
The second solution, the event-based one, will generate an event record
for each significant event in the life of each channel. So, START
events when a channel is born; ANSWER events when someone answers a
call; END events when somebody hangs up. There will also be Park,
and Transfer, and MOH, and 3-WAY, Conference-Join, and several
others. Just enough information will be included with each event to
thread together billable sequences. Along with each event record will be
the time the event happened, and channel info. This approach will be
very much more fine-grained, and allow you to do fancy things like
figure out that Sacramento was the only person talking to Shanghai, and
allow you to bill the call to the guy/gal in Sacramento. Trouble with
this approach is that threading together the event records is a
non-trivial operation! But I hope to provide some tools that will make
this easier to do.
So, the bad news is: you will not see any solutions for this problem, in
1.2, or 1.4. the CDR fix (first solution) will most likely end up in
1.6, the event-based solution will probably not be available until 1.8
or 1.10; we shall see.
murf
On Mon, 2007-06-11 at 14:26 +0200, Gunnar Schaller wrote:
Hello list,
I have a problem with called ZAP channels making an attended-transfer
or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk
is wrong.
At the moment there is a bristuffed Asterisk 1.2.18 running with
bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit:
[default]
exten = 0123456789,1,Macro(dialpstn,${EXTEN})
[macro-dialpstn]
exten = s,1,Set(TRANSFER_CONTEXT=transfer)
exten = s,2,Set(FORWARD_CONTEXT=transfer)
exten = s,3,Set(CIDNUMORIG=${CALLERID(num)}) ; save callerid-num
exten = s,4,Set(CALLERID(num)=98765) ; dial out using msn 98765
exten = s,5,Dial(Zap/g1/${ARG1}|30|t)
exten = h,1,Set(CALLERID(num)=${CIDNUMORIG}) restore callerid-num for
CDR
[transfer]
exten = _X.,1,Set(CALLERID(all)=External 0123456789)
exten = _X.,1,Dial(SIP/${EXTEN})
So I call 0123456789 with SIP phone 10. The callee dials *1 20 for
attended transfer and SIP phone 20 (I have *1 for attended transfer in
features.conf). The called SIP-phone shows the caller-information I
set in context
Re: [asterisk-users] Console duplicate output problem
I guess he might mean don't include the -g on the command line? I'm wondering if asterisk is running in the background of the console you're logged in at, so it's dumping messages to the console, AND you've connected with -r? Moj Barton Fisher wrote: Eric ManxPower Wieling wrote: This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? Stop running in graphics mode. OK, that's a great clue, but can you tell me how to disable now? Bart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Console duplicate output problem
On MY distro (Mandrive) you edit /etc/lilo.conf and set the default kernel to linux-nofb, then rerun lilo. You would have to find out how to disable the framebuffer on YOUR distro. If you use my method chances are your machine won't boot. Mojo with Horan Company, LLC wrote: I guess he might mean don't include the -g on the command line? I'm wondering if asterisk is running in the background of the console you're logged in at, so it's dumping messages to the console, AND you've connected with -r? Moj Barton Fisher wrote: Eric ManxPower Wieling wrote: This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? Stop running in graphics mode. OK, that's a great clue, but can you tell me how to disable now? Bart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream 4104 - Asterisk (Incoming Calls problem)
Hello..I have a Grandstream 4104 (4 FXO) gateway connected to an Asterisk server and a traditional PBX..Asterisk users are able to call the PBX users but PBX users dont have access in Asterisk..Does anyone know if specific configurations in Asterisk and in Grandstream have to be done? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple ENUM entries and Asterisk fails to dial
Hi, I use Asterisk 1.2.17 in my site, and now I'm trying to configure ENUM lookup in my server. When someone calls a number that has multiple ENUM entries, randomly Asterisk seems to fail to return a correct answer, and dial by ENUM fails. I've goggled a bit on this, but didn't get any good conclusion. There is some RFC Compliant ENUM Macro that can be used that is announced to solve this problem, but also it can be read that It seems that Asterisk 1.2.0 comes with a new powerful ENUMLOOKUP. So there is probably no need to use this script anymore.. Should I still use that macro? It is already closed a Digium Issue Tracker concerning handle multiple records with the same order and priority, so, this problem shouldn't be arising anymore, shouldn't it? Does Asterisk 1.4 already solves this issue? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] change moh during a call?
r should never be used on the Dial line. Anthony Francis wrote: Add a r option with that extension so that way when it is called it will ring and then when you put it on hold it will play your moh. You can also set the MOH in sip.conf, but you will still get the same behavior you have now. Thomas Stein wrote: Hello. Is it possible to change the defined moh sound file within an extension? I have: exten = 18,1,Answer exten = 18,n,Wait(3) exten = 18,n,SetMusicOnHold(durchwahl) exten = 18,n,Dial(SIP/118,15,m) exten = 18,n,Hangup Now i have the situation someone calls and my phone is ringing while moh (durchwahl) is playing. When i pickup the call and press the hold button during the call moh (durchwahl) is playing as moh. Thats not what i want. Can i define another file within this extension? regards t. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID issues
All, I have run into some CallerID issues. It seems to have happened as a result of just moving my config from 1.2.12 to 1.4.4 (although I am not sure of this). Therefore I am sure its just a misconfiguration somewhere, I just don't know where. I have throughout the office either Cisco 7961G or Polycomm Soundpoint SIP 430 IP phones. The problem is with CallerID showing up in some places, but not in others. For instance, in the CDR, if a call comes in as unavailable but still displays the number, the phone will only show Unavailable and then display the phone's extension. Most of the time, the callerid on the phone just displays the extension of the phone itself when there is an incoming call. The extension of this part of the issue is that when I make internal calls, it will show the callerid (name) of the user originating the call, but the callerid (number) will again show the extension of the phone receiving the call. Another part of the issue is that the email that comedian mail sends says that the callerID and phone number are unavailable. But when you listen to the voicemail, it will read the number to you (and again it is in the CDR properly). The last part of the issue (that I have been to see) is that FollowMe can also never pass along the number. It always says, You have an incoming call from number unavailable. And again, even if the callerID doesn't come up, the number is there 99% of the time. I have a feeling that all this interconnected somehow. Any help would be greatly appreciated. Thanks. Eric -- Eric Lubow LinkExperts, Inc. Systems Administrator e: [EMAIL PROTECTED] w: www.linkexperts.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] change moh during a call?
On 6/11/07, Thomas Stein [EMAIL PROTECTED] wrote: [snip] Now i have the situation someone calls and my phone is ringing while moh (durchwahl) is playing. When i pickup the call and press the hold button during the call moh (durchwahl) is playing as moh. Thats not what i want. Can i define another file within this extension? You can try to call dial with M option, and set MOH for hold on macro. something like this: Dial(SIP/118,15,mM(call-answer)) [macro-call-answer] SetMusicOnHold(other) Regards, Atis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Introduction to AGI programming
I wrote an introduction to AGI programming paper as an exercise to learn more about the process involved. You can find a copy of it herehttp://mocker.org/papers/. I welcome any comments or corrections to improve upon it. As I said, it was mainly done to force myself to research the topic so there are probably errors! :) -- Kyle Sexton ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event
Kenneth Padgett wrote: My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've I'd love to be notified when you release the Polycom admin program!! What language are you developing it in? If it's PHP, I could help test or develop... -Kenneth Its something I'm doing in my spare time. Sorry, I'm writing it in Freepascal/Lazarus (we are primarily Delphi/Freepascal shop here) but you're welcome to get yo pascal on if you like ;) At any rate, we'll host it initially on subversion. It will be released under LGPL, I think. I will post updates to its status to the .biz list which I think would be more appropriate. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi with java?
Lenz wrote: Hi Lee, we are a Java shop and our experience with Java has been much the one you say - it does scale pretty well and it is very solid. What I was trying to say is that Java is not very well suited to the classic, Unix-style, fire-up-process-and-let-it-die that goes for CGI/AGI programming. On the other side, I have no doubt that with an application server and FastAGI you can get quite a lot of bang for the buck. :) l. On Fri, 08 Jun 2007 18:07:50 +0200, Lee Jenkins [EMAIL PROTECTED] wrote: We have found that generally speaking, running the FastAGI server on the same machine as Asterisk yields better performance than launching separate exe processes through the dial plan. Completely anecdotal of course. This is careful research conducted over our entire 5 customer base... I get what you are saying, I was agreeing with you. :) We *were* writing all of our AGI as binary executables and even then, the FastAGI server that we eventually built still gets better performance vs. when we launched separate AGI per call from the dialplan. My guess is that it is easier on the system for an existing executable (FastAGI Server) to spawn threads of execution for short periods of time to handle (Fasg)AGI requests than it is to run separate executable AGI's instead. We're hoping that performance will be improved even more when we introduce pooling of common objects (db access for example). -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Introduction to AGI programming
Kyle Sexton wrote: I wrote an introduction to AGI programming paper as an exercise to learn more about the process involved. You can find a copy of it here http://mocker.org/papers/. I welcome any comments or corrections to improve upon it. As I said, it was mainly done to force myself to research the topic so there are probably errors! :) -- Kyle Sexton Kyle, I liked it. Maybe you could also cover how the initial vars are pushed to the application one right after another initially and to look for an empty line to indicate end of initial vars coming in. Have you considered putting it on the wiki? That would be an ideal place for a nice white paper like that, IMO. Once google indexes it, it should be fairly easy to find for new Asterlings... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI RECORD FILE for a video message
I have used the AGI call RECORD FILE before for an only audio message. I pretty much always used RECORD FILE filename gsm before. What paramater do I use for gsm so that if video is present it will record video also and still work for only audio. Will this recorded file (with video) play on a video phone when calling out with a call file in outgoing spool? Thanks, I am using 1.4.4 Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple ENUM entries and Asterisk fails to dial
[EMAIL PROTECTED] wrote: Hi, I use Asterisk 1.2.17 in my site, and now I'm trying to configure ENUM lookup in my server. When someone calls a number that has multiple ENUM entries, randomly Asterisk seems to fail to return a correct answer, and dial by ENUM fails. I've goggled a bit on this, but didn't get any good conclusion. There is some RFC Compliant ENUM Macro that can be used that is announced to solve this problem, but also it can be read that It seems that Asterisk 1.2.0 comes with a new powerful ENUMLOOKUP. So there is probably no need to use this script anymore.. Should I still use that macro? enumlookup works greak, as far as I can tell. It is already closed a Digium Issue Tracker concerning handle multiple records with the same order and priority, so, this problem shouldn't be arising anymore, shouldn't it? Does Asterisk 1.4 already solves this issue? just as well as enumlookup in 1.2 Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip show registry shows nothing
Hi, In installed a new Trixbox-based system on Friday and initially had some problems with the system registering with our service provider (voip.co.uk), which I put down to the router's configuration. Anyway, the system was finally 'fixed' and worked well until Monday and then just stopped registering with voip.co.uk and I have spent the whole day trying to encourage it back online. Everything else works - inter-extension calls are OK, I have a Snom 360 at home registered via broadband and I have even put the server in the DMZ without any effect. I have two other servers with similar configurations successfully registered so I am a bit puzzled. I have deleted and remade the trunk to voip.co.uk several times. Thing is that 'sip show registry' shows nothing and I cannot see any evidence in the debug text to indicate that the server is even attempting to register (although I will admit to not being an asterisk guru). One of my checks was to configure a Snom 190 on site to register with voip.co.uk and this worked OK. The trunk does, however show in 'sip show peers': ELY/XX 80.249.108.21N 5060 OK (42 ms) 4851 (Unspecified)D N 0UNKNOWN 4850/4850 my ext at home D N 37731OK (127 ms) 4806/4806 192.168.113.207 D N 2054 OK (16 ms) 4805/4805 192.168.113.206 D N 5060 OK (7 ms) 4804/4804 192.168.113.205 D N 5060 OK (7 ms) 4803/4803 192.168.113.204 D N 5060 OK (7 ms) 4802/4802 192.168.113.203 D N 5060 OK (7 ms) 4801/4801 192.168.113.202 D N 5060 OK (7 ms) 4800/4800 192.168.113.201 D N 5060 OK (8 ms) The server can ping and traceroute to voip.co.uk and I am getting a bit lost for things to try. My final attempt has been to install the latest 1.2 from svn and so I am now running SVN-branch-1.2-r68732. Before I rush headlong into debug files, are there any specific modules or settings that I can check to see that the server really is trying to register the trunk? Happy to post logs etc. here if someone lets me know what is best to show. Thanks Nigel Kendrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak Matthew Fredrickson [EMAIL PROTECTED] wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] which Wifi SIP phones are the good ones
We are planning to buy a wifi SIP phones to work with Asterisk 1.2-18 setup. I would like to get feedback views regarding Linksys WIP300 WIFI IP Phone or any other wifi phones which has been stable. Thanx for any updates. -- Deepak - The all-new Yahoo! Mail goes wherever you go - free your email address from your Internet provider.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
The echo cancellation card is for SIP-Zap calls only, no echo cancellation is done in Asterisk for SIP only calls. SIP to SIP, media is just passed through the server untouched (using media flow through, which is the option in sip.conf of canreinvite=no) if you are not handling any translation, even when handling translation between SIP calls there shouldn't be any echo cancellation done in Asterisk for SIP only calls. The place to look at would be the remote SIP devices which is typically what is adding the echo, this is usually a gain issue of some sort depending on which handsets you are using. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu Sent: Monday, June 11, 2007 16:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak Matthew Fredrickson [EMAIL PROTECTED] wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Answers - Get better answers from someone who knows. Try it now http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Once upon a time I used to have a lot of SIP-SIP calls issues, which not always but sometimes included echo problems. There were no zap devices on the server. Googling and struggling to fix it, I found out that it was because of timing issues and ztdummy was not working properly. It had to do something with the kernel and USB modules and something needed to be fixed in BIOS and zaptel settings somewhere (not in zapata or zaptel confs) so that it can have a properly working timing source. I don't remember the details now but I remember I managed to fix it by building a different kernel version on that server after installaing some other version of zaptel, disabling USB modules on the motherboard, fixing something in zaptel Makefile, disabling unused modules in /etc/sysconfig/zaptel. I don't remember what else I did. but echo and other problems disappeared after whatever I did. It was about 2 years ago and I remember how frustrating it was. Anyways, I guess once you upgraded your hardware, something changed in zaptel settings somewhere which is now effecting the SIP-SIP calls and resulting in echo. Do you have the backup of old setup without this card, which you can install and check what exactly the settings were before. Also I recommend going with Sangoma. I hear a lot of bad stories about digium cards imcompatibility with certain motherboards and conflicts with USB modules on the motherboard, and conflicts with IRQs. Thats why When I went for PRI, I used Sangoma. I've used their A101c and A101d cards, and there have never been any issues. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement
Do the Digium cards have a built in CSU? Is a CSU an FCC requirement? or just a carrier requirement? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Hey thanx for sharing your troubleshooting. Ya over days I kind of did some QA. There are SIP--SIP echo's between random phones. We have 75 phones of Polycom 501. I think might be the network or combination of network polycom creating this. Do you have the backup of old setup without this card, which you can install and check what exactly the settings were before. This is an entie new setup by me, the old one was using 1.4 build I am using 1.2 build both are different server. -- Deepak Zeeshan Zakaria [EMAIL PROTECTED] wrote: Once upon a time I used to have a lot of SIP-SIP calls issues, which not always but sometimes included echo problems. There were no zap devices on the server. Googling and struggling to fix it, I found out that it was because of timing issues and ztdummy was not working properly. It had to do something with the kernel and USB modules and something needed to be fixed in BIOS and zaptel settings somewhere (not in zapata or zaptel confs) so that it can have a properly working timing source. I don't remember the details now but I remember I managed to fix it by building a different kernel version on that server after installaing some other version of zaptel, disabling USB modules on the motherboard, fixing something in zaptel Makefile, disabling unused modules in /etc/sysconfig/zaptel. I don't remember what else I did. but echo and other problems disappeared after whatever I did. It was about 2 years ago and I remember how frustrating it was. Anyways, I guess once you upgraded your hardware, something changed in zaptel settings somewhere which is now effecting the SIP-SIP calls and resulting in echo. Do you have the backup of old setup without this card, which you can install and check what exactly the settings were before. Also I recommend going with Sangoma. I hear a lot of bad stories about digium cards imcompatibility with certain motherboards and conflicts with USB modules on the motherboard, and conflicts with IRQs. Thats why When I went for PRI, I used Sangoma. I've used their A101c and A101d cards, and there have never been any issues. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement
Quoting C F [EMAIL PROTECTED]: Do the Digium cards have a built in CSU? Is a CSU an FCC requirement? or just a carrier requirement? if you expect things to work you need one regardless of regulations, yes the digium cards have it built in, as do most modern t1 cards. if the T1 terminates in something that looks like a scsi connector you have an hssi interface most likely, if it terminates in an rj45, especially if it has status lights, you most likely have yourself a csu built in, sometimes you'll have a db15 instead of the rj45 depending on the country it was designed for but it still works the same if you just get a passive adapter to get to the connector type you need (or make one, t1 speed is a 1/8th of the slowest ethernet so construction technique is not too critical if you ever made an ethernet cable) coming in from the raw copper pair this is what needs to be there : telco supplied pairgain box which is normally an HDSL modem that gets you from a type of dsl circuit to a 2 pair T1 / DS1 circuit (don't confuse DSL and DS ONE in this sentence) that is the actual demarcation point. then comes your csu/(dsu) This is the point where remote loopback tests can be done without actually talking to the guts of your hardware, telco can normally do it to their box as well but when they do a line test they loop to your csu normally. next comes a serial interface of some sort, in a more modern setup its indivisible from the csu, in the old days you had a physical synchronous serial cable between running at t1 clock speed. Where its separate the serial port is also known as an hssi connection or high speed serial interface. So without the csu in the mix converting the t1 channel frame encoding down to the actual serial data, you have no way to talk to the channel. its like saying I have a usb port, do I really need the ethernet dongle in order to plug it into an ethernet jack ? Then again some hardware has an ethernet jack right on it, but it still has all the same ethernet hardware as the dongle in there somewhere even if there is no physical usb path between the pci bus and the ethernet, it still accomplishes the same thing. the csu is sort of like the part of the modem where the start and stop bits are added into the actual data before hitting the actual modem proper where the bits are converted to tones, we don't generally make the distinction on that part of the circuit since the rest is useless without it. TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement
On 6/11/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting C F [EMAIL PROTECTED]: Do the Digium cards have a built in CSU? Is a CSU an FCC requirement? or just a carrier requirement? if you expect things to work you need one regardless of regulations, yes the digium cards have it built in, as do most modern t1 cards. I disagree with this, I have several T1s that don't use Digium equipment and are directly connecting to T1 cards that DONT have a CSU and work fine. The reason this thing came up was because I was going thru documentation for such a card and it mentioned it's an FCC requirement. if the T1 terminates in something that looks like a scsi connector you have an hssi interface most likely, if it terminates in an rj45, especially if it has status lights, you most likely have yourself a csu built in, sometimes you'll have a db15 instead of the rj45 depending on the country it was designed for but it still works the same if you just get a passive adapter to get to the connector type you need (or make one, t1 speed is a 1/8th of the slowest ethernet so construction technique is not too critical if you ever made an ethernet cable) coming in from the raw copper pair this is what needs to be there : telco supplied pairgain box which is normally an HDSL modem that gets you from a type of dsl circuit to a 2 pair T1 / DS1 circuit (don't confuse DSL and DS ONE in this sentence) that is the actual demarcation point. then comes your csu/(dsu) This is the point where remote loopback tests can be done without actually talking to the guts of your hardware, telco can normally do it to their box as well but when they do a line test they loop to your csu normally. next comes a serial interface of some sort, in a more modern setup its indivisible from the csu, in the old days you had a physical synchronous serial cable between running at t1 clock speed. Where its separate the serial port is also known as an hssi connection or high speed serial interface. So without the csu in the mix converting the t1 channel frame encoding down to the actual serial data, you have no way to talk to the channel. its like saying I have a usb port, do I really need the ethernet dongle in order to plug it into an ethernet jack ? Then again some hardware has an ethernet jack right on it, but it still has all the same ethernet hardware as the dongle in there somewhere even if there is no physical usb path between the pci bus and the ethernet, it still accomplishes the same thing. the csu is sort of like the part of the modem where the start and stop bits are added into the actual data before hitting the actual modem proper where the bits are converted to tones, we don't generally make the distinction on that part of the circuit since the rest is useless without it. That specific T1/PRI card I'm talking about has an JR45 connector and does not have a built in CSU. Which brings me back to the second part of my original question, is it required by law. Thank you TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement
On Mon, 11 Jun 2007, C F wrote: I disagree with this, I have several T1s that don't use Digium equipment and are directly connecting to T1 cards that DONT have a CSU and work fine. The reason this thing came up was because I was going thru documentation for such a card and it mentioned it's an FCC requirement. That's not possible, unless the handoff you're getting is not actually T1. However, the card almost certainly has a very seamless, inline/onboard CSU of which you aren't even aware of. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Crashes with Spandsp, app_rxfax.c, and asterisk 1.4.4
Hi everybody, I have a Fedora Core 4 x86 32 bit install, which I recently upgraded from asterisk 1.2 to the office 1.4.4 tarball. In the process of doing that I had to upgrade some autoconf/automake stuff, but it worked fine, and my new asterisk works fine. Except that anytime I receive a fax with spandsp and app_rxfax, asterisk seg faults. I have applied the spandsp patch of course, and I used the newer app_rxfax.c and app_txfax.c from soft-switch.org for 1.4. I have tried numerous versions of spandsp in turn, recompiling the rxfax and txfax application after installing each one. In particular, I am still getting this when using the latest spandsp snapshot from June 8th. I my searching, I found a few other people who mentioned the same problem, but they either didn't say if they solved it or were vague about how they did. My verison of libtiff is 3.7.1. One of the crashes printed out this stack trace information, although that doesn't come out most of the time: XXX -- Executing [EMAIL PROTECTED]:2] Set(Zap/1-1, [EMAIL PROTECTED]) in new stack -- Executing [EMAIL PROTECTED]:3] RxFAX(Zap/1-1, /var/spool/asterisk/fax/1181537898.0.tif|debug) in new stack linux40*CLI *** glibc detected *** /usr/sbin/asterisk: free(): invalid next size (normal): 0x09d5a908 *** === Backtrace: = /lib/libc.so.6[0x5471e0] /lib/libc.so.6(__libc_free+0x77)[0x54772b] /usr/lib/asterisk/modules/app_rxfax.so[0xebbc41] /usr/sbin/asterisk[0x80c3cc8] /usr/lib/asterisk/modules/app_macro.so[0xa330ea] /usr/sbin/asterisk[0x80c3cc8] /usr/sbin/asterisk[0x80c5002] /usr/sbin/asterisk[0x80c5d3e] /usr/sbin/asterisk[0x80f1f99] /lib/libpthread.so.0[0x656bd4] /lib/libc.so.6(__clone+0x5e)[0x5ae4fe] === Memory map: 00111000-00118000 r-xp 03:01 5330798 /usr/lib/asterisk/modules/res_musiconhold.so 00118000-00119000 rwxp 7000 03:01 5330798 /usr/lib/asterisk/modules/res_musiconhold.so 00119000-0011f000 r-xp 03:01 5330792 /usr/lib/asterisk/modules/res_config_pgsql.so 0011f000-0012 rwxp 5000 03:01 5330792 /usr/lib/asterisk/modules/res_config_pgsql.so 0012-00218000 r-xp 03:01 196501 /opt/lumenvox/engine_6.5/lib/libcrypto.so.0.9.7f 00218000-0022a000 rwxp 000f8000 03:01 196501 /opt/lumenvox/engine_6.5/lib/libcrypto.so.0.9.7f 0022a000-0022d000 rwxp 0022a000 00:00 0 X The memory map continues on for several pages but I can supply it if anyone thinks it would be useful. Most of the time the *CLI prompt just shows asterisk disconnecting, and the /var/log/asterisk/full just shows RxFax as the last thing ran, and the message of asterisk starting up again, like this: X [Jun 10 18:26:12] VERBOSE[13094] logger.c: -- Executing [EMAIL PROTECTED]:2] Set(Za p/1-1, [EMAIL PROTECTED]) in new stack [Jun 10 18:26:12] DEBUG[13094] app_macro.c: Executed application: Set [Jun 10 18:26:12] VERBOSE[13094] logger.c: -- Executing [EMAIL PROTECTED]:3] RxFAX( Zap/1-1, /var/spool/asterisk/fax/1181517967.7.tif|debug) in new stack [Jun 10 18:26:17] VERBOSE[13163] logger.c: Asterisk Event Logger Started /var/log/asterisk /event_log [Jun 10 18:26:17] VERBOSE[13163] logger.c: Asterisk Dynamic Loader Starting: [Jun 10 18:26:17] VERBOSE[13163] logger.c: == Parsing '/etc/asterisk/modules.conf': [Jun 10 18:26:17] VERBOSE[13163] logger.c: Found X Is anyone else out there seeing this ? Does anyone have any suggestions, even if it is only how to get more debuging information out ? From the reading and searching I have been doing the last several hours, it appears that the best thing to do in the long run might be to install iaxmodem and HylaFax. However, right now I have my asterisk invoking some custom scripts and uploading the faxes into a database based on DID and CallerID numbers, and I would much rather get this working now this way instead of have to learn how to integrate the same thing with HylaFax. Thanks in advance, --Rob -- http://rgr.freeshell.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
