[asterisk-users] Temporarily placing confbridge participants on hold - two way muting
Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E can't hear, is there a way to suspend D and E for a while (whilst they are played music or whatever) and later join them back in? Failing that, I was considering kicking them and using an AGI script to rejoin them to the conference but I wasn't sure how to do that from the script (the rejoin, not the kick)? Any pointers or suggestions welcomed. (In a nutshell it's for a situation where certain participants need to have privacy in the conference from a group of others and it all needs to be driven from an AGI script). Regards Steve 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 'techni...@brendata.co.uk' 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 http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to find the CDR call start time value
Can anyone suggest how I can find the value of the call start time that will be logged by CDR in the dialplan? I've taken a look through the variables but I can't see anything that seems to hold this? 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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find the CDR call start time value
That's exactly what I was looking for, I'd found this
http://www.voip-info.org/wiki/view/Asterisk+variables which seems to be
a partial copy of the same thing.
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Krunal
Patel
Sent: 02 October 2008 11:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to find the CDR call start time value
HI Steven,
You can get call start time by ${CDR(start)} .
For more information of asterisk variables , please check out
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
Thanks,
Krunal Patel
On Thu, Oct 2, 2008 at 3:08 PM, Steve Hanselman [EMAIL PROTECTED]
wrote:
Can anyone suggest how I can find the value of the call start time that
will be logged by CDR in the dialplan?
I've taken a look through the variables but I can't see anything that
seems to hold this?
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[asterisk-users] Lone worker system
Has anybody got any scripts for a lone worker system using Asterisk before I write them? Something along the lines of a regular phonecall with some kind of random question (e.g. press 1 then 5) to provide monitoring of lone workers with alerts? Steve 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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lone worker system
Spot on (except for the shitty way, it's pretty standard, in building there are paging systems that start an escalating tone, beyond the building these don't work, so if you're away then we'd be dialling the mobile). Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: 12 May 2008 16:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lone worker system He wants a randomly generated phone call to be generated to a specific extension. Eg once an hour for the midnight to dawn shift at a random time per hour. When the person picks up they are asked a question using an audio file. (or text to speech). Then the person has to enter the correct dtmf answering the question (eg 1 - 5) If the person fails to answer the phone (I'm guessing here but a second call will be placed 2 mins later). If this call is also 'fail to answer' an escalation call to a supervisor or something similar will occur indicating that the 'lone worker' failed to respond and is either - dead from a stabbing, or 2 jerking off in the bathroom and not at his post. Cheers, Dean P.S. Hope your lone worker is paid a lot to be working for a shitty company checking up on them like that :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 12, 2008 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lone worker system On Mon, May 12, 2008 at 10:28 AM, Steve Hanselman [EMAIL PROTECTED] wrote: Has anybody got any scripts for a lone worker system using Asterisk before I write them? Something along the lines of a regular phonecall with some kind of random question (e.g. press 1 then 5) to provide monitoring of lone workers with alerts? Steve I think a little more elaboration would get you more helpful advice. I have read your message a couple of times and still don't really understand what you need. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.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 See our current vacancies at www.brendata.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.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?
Tracked this down (or more to the point found the issue causing it), it was high levels of bursty disk activity. The iowait went through the roof (30-40%). The disks are scsi serviced by an MPT-Fusion controller in a Dell Poweredge 2850. We're using LVM to bind the disks into a JBOD set. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 11 June 2007 10:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: 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
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
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
RE: [asterisk-users] Bridged PRI calls - processor involvement?
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
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 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 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
[asterisk-users] Bridged PRI calls - processor involvement?
On a zaptel TE410p, when a call is bridged PRI - PRI how much involvement does the processor have? We're now seeing chunks of missing audio and I can't tell whether this is due to a kernel upgrade or to a zaptel/libpri/asterisk upgrade. I'm not seeing missed interrupts (from a cat of the proc/zaptel files), any other ideas on how I could go about tracking this down? I'm thinking of enabling the debug options on the module to see if this can throw any further light on the problem. As I say, all was fine, the te410 lives on its own interrupt and all was fine until a few weeks back. 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
RE: [asterisk-users] Cutted audio or 2/3s blankson EuroISDN- Asterisk1.4
We're running 1.4.0 of asterisk 1.4.2.1 of zaptel And kernel 2.6.20-1.2316.fc5smp 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
RE: [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN- Asterisk1.4
We're also seeing the same thing, our calls are bridged zaptel calls between ISDN30 PRI interfaces on a single TE410P. We don't' appear to have any lost interrupts. Same as stated, 2-3 second gaps in audio. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Administrator TOOTAI Sent: 11 May 2007 09:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN- Asterisk1.4 Steve Totaro a écrit : Hi Steve Your Zap conf files would be helpful. Zttest results? Cat /proc/interrupts. Sharing interrupts? No. Zap con files should not be relevant as we are using ISDN. [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/zaptel.conf loadzone = us defaultzone=us [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [trunkgroups] [channels] ; context=default ; switchtype=national ; signalling=fxo_ls ; rxwink=300 ; Atlas seems to use long (250ms) winks ; usecallerid=yes ; hidecallerid=no ; callwaiting=yes ; usecallingpres=yes ; callwaitingcallerid=yes ; threewaycalling=yes ; transfer=yes ; canpark=yes ; cancallforward=yes ; callreturn=yes ; echocancel=yes ; echocancelwhenbridged=yes ; rxgain=0.0 txgain=0.0 ; group=1 ; make these both the same. Groups range from 0 to 63. ; callgroup=1 pickupgroup=1 ; immediate=no [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 109917508 0 0 0IO-APIC-edge timer 1: 12365 0 0 0IO-APIC-edge i8042 8: 444560118 0 0 0IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 11367 0 0 0IO-APIC-edge i8042 14:3944731 0 0 0IO-APIC-edge ide0 58: 0 0 0 0 IO-APIC-level uhci_hcd:usb1, uhci_hcd:usb3, ehci_hcd:usb5 66: 0 0 0 0 IO-APIC-level uhci_hcd:usb2, uhci_hcd:usb4 74:4552211 0 0 0 IO-APIC-level libata 90: 18418187 0 0 0 PCI-MSI eth0 98: 27358592 0 0 0 IO-APIC-level HFC-multi 106: 27358571 0 0 0 IO-APIC-level HFC-multi NMI: 14333691827 1273 LOC: 109917988 109917975 109917950 109917910 ERR: 0 MIS: 0 We use ztdummy for Meetme: [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ sudo ./zttest Opened pseudo zap interface, measuring accuracy... 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.951172% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.951172% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% --- Results after 87 passes --- Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952721 lsmod, zttranscode was loaded, I remove it: [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ lsmod Module Size Used by ztdummy10056 0 tcp_diag6400 0 inet_diag 16784 1 tcp_diag mISDN_dsp 201384 1 hfcmulti 79884 1 mISDN_capi107116 1 l3udss146744 1 mISDN_l2 44616 1 mISDN_l1 17560 1 mISDN_core 88224 6 mISDN_dsp,hfcmulti,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1 capi 23616 0 capifs 11152 2 capi kernelcapi 56640 2 mISDN_capi,capi zaptel197608 7 ztdummy crc_ccitt 6784 1 zaptel ipv6 285664 34 ppdev 14088 0 parport_pc 41640 0 lp 17736 0 parport44684 3 ppdev,parport_pc,lp button 12192 0 ac
RE: [asterisk-users] Audio going blank for a few seconds and thencomes back. What could be the reason?
I think this is more related to the PRI, we've been seeing this for a few weeks now, and our environment is bridged PRI-PRI on the same board, Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: 10 May 2007 01:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audio going blank for a few seconds and thencomes back. What could be the reason? I have Grandstream and Aastra phones. It happens on both of them. 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
RE: [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason?
You can use tcpdump or ethereal (wireshark now) to capture the stream and then see if there was loss during the call, just leave a capture going then get your users to mark out the time at which they encountered the silence, compare this to the server time (e.g. their watch to the server) to get a time difference, then figure out what time you need to look at in the trace. Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: 01 June 2007 13:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason? There are some remote extensions connected on this system, and calling long distance is purely on voip. These remote extensions also face the same thing, i.e. audio going blank for a few seconds, when dialing long distance. So in this case, no PRI is involved. Its either the server, or the network. Now I don't know how to find out what is it and why? On 6/1/07, Steve Hanselman [EMAIL PROTECTED] wrote: I think this is more related to the PRI, we've been seeing this for a few weeks now, and our environment is bridged PRI-PRI on the same board, Steve 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
RE: [asterisk-users] Audio going blank for a few seconds andthencomesback. What could be the reason?
There seem to be two threads here that mention multi-second loss with the common part being a PRI, certainly for my situation it's purely PRI as the asterisk box sits in between the telco and another PRI enabled PBX and the calls are bridged between the two. There is no network traffic involved in this case. Not sure where to go with mine though, the load average is nice and low, I don't see any missed interrupts and it's only started happening in the last few weeks since an asterisk upgrade. Latest FC6 kernel, latest yum'd asterisk, zaptel etc Not sure whether it's worth pulling a SVN version down and building that, the only issue is I can't currently reproduce this on demand. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: 01 June 2007 14:36 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audio going blank for a few seconds andthencomesback. What could be the reason? On Friday 01 June 2007 9:24 am, Rob Schall wrote: comcast high-speed, thinking that would be more than enough. Turned out though, with most high speed solutions, there is some limited packet loss and its just to be expected. You internet browsers, etc, would Limited packet loss != **EIGHT SECONDS** of network breakage. Jitter buffers and PLC takes care of most normal network indiscretions, but period dropouts of that big of a time aren't normal and indicate a bigger issue, either with the hardware or the link itself. normally just re-request the packet and move on, but with a stream, you're out of luck. The only real solution is to have a dedicated T1 or mpls connection or something like that for perfect quality. We have solid connections between our offices and haven't had a problem yet. I have numerous installations using standard telco (Bell Canada and Telus) DSL, and at least one on Rogers cable here in Ontario. No real problems. The odd problem if the pipe gets saturated but careful design and monitoring can take care of most of these problems. I agree with Mr. Hanselman; get a packet logger on the link and see what's really going on. Until that's done, everything here is just speculation. I have seen bugs in the IAX2 and SIP jitter buffers on Asterisk which cause dropouts like this, and I'd like to see what's actually going on before pointing any fingers. -A. ___ --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 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
RE: [asterisk-users] Trouble with rxfax multi-page printing with cups
Fax2ps is what we use, works fine. Yum tells me it comes from libtiff Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Artifex Maximus Sent: 08 September 2006 11:05 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Trouble with rxfax multi-page printing with cups Hello, Cups unfortunately don't support multi-page tiff printing. http://www.cups.org/str.php?L1117 I have tried tiff2ps and tiff2pdf but both just embed original tiff file and give the first page only. Is there any solution for printing multi-page tiff easily? More likely an alternative lp than bash or any other script. bye, Zsolt ___ --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 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
RE: [asterisk-users] Unable to match on CallerID in an include block
I'll run some more tests but it's not very different from the posting?
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 22 August 2006 18:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to match on CallerID in an include
block
I suspect that your dialplan is more than you show ;)
It works just fine for me with svn trunk
[from-sip]
include = common
[common]
exten = 1234,1,NoOp(Hmm ${CALLERID(num)})
exten = 1234/7708,1,NoOp(Here)
If I dial 1234 from my 7708 extension, I get the NoOp(Here)
If I dial 1234 from my 7701 extension, I get the NoOp(Hmm 7701)
Julian.
Steve Hanselman wrote:
Hi Julian,
Ah, a very good point, I put that in my first cut but had completely
forgotten in this one!
1.2.10
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 22 August 2006 17:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to match on CallerID in an
include
block
What version of asterisk ?
Julian
Steve Hanselman wrote:
Is there any reason why I can't use the xxx/callerid format in an
include section?
It doesn't seem to work, but if I paste the lines into the main
section
where I include the block it does?
E.g. this doesn't work
[telewest]
Include = spamblock
[spamblock]
_X./12345,s,macro(spamcall)
Whereas this does:
[telewest]
_X./12345,s,macro(spamcall)
Any ideas?
Steve
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RE: [asterisk-users] Unable to match on CallerID in an include block
Hi Julian, Ah, a very good point, I put that in my first cut but had completely forgotten in this one! 1.2.10 Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 22 August 2006 17:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to match on CallerID in an include block What version of asterisk ? Julian Steve Hanselman wrote: Is there any reason why I can't use the xxx/callerid format in an include section? It doesn't seem to work, but if I paste the lines into the main section where I include the block it does? E.g. this doesn't work [telewest] Include = spamblock [spamblock] _X./12345,s,macro(spamcall) Whereas this does: [telewest] _X./12345,s,macro(spamcall) Any ideas? Steve 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.brendataco.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 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
[asterisk-users] Unable to match on CallerID in an include block
Is there any reason why I cant use the xxx/callerid format in an include section? It doesnt seem to work, but if I paste the lines into the main section where I include the block it does? E.g. this doesnt work [telewest] Include = spamblock [spamblock] _X./12345,s,macro(spamcall) Whereas this does: [telewest] _X./12345,s,macro(spamcall) Any ideas? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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
RE: [asterisk-users] rx_fax problem
Rxfax has no ECM, try hylafax and iaxmodem. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paradise Dove Sent: 01 August 2006 21:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] rx_fax problem hi, rx_fax fails to get fax on a bit noisy lines but real fax devices can do that on the same line with no problem! what's the problem? 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 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
RE: [Asterisk-Users] sangoma card test
Create yourself a crossover cable and loop the spans, set one to provide clock and you should quickly see them come up, this will provide a very basic test of hardware. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 16 June 2006 10:20 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] sangoma card test Is there any way of running a diagnostic on a Sangoma A102 card ? Our lines have gone down and I want to avoid the usual BT It must be your equipment line with technical proof. At the moment I have a BLUE/RED alarm on zap show status Julian. ___ --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 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
RE: [Asterisk-Users] FAX over PRI
Sorry for the late reply but both of these are fine, we use spandsp to print some faxes and email others. We also route via a PRI to our other phone system to hylafax on an analog modem and also to an analog fax. So what you want to do is fine and will work. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Gaudette Sent: 21 March 2006 20:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FAX over PRI Hmmm, Im not so sure I can apply this to me though. I just want to do Fax-To-Email using PRI channels as the incoming lines. Not so much transfer to a real fax. I am assuming that this is easily done with Asterisk? (I did it before with Asterisk SIP, but it only worked once every 10 tries or so) Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: March 21, 2006 3:25 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FAX over PRI On Tuesday 21 March 2006 15:09, Michael Gaudette wrote: How should I consider Fax over PRI channels with Asterisk? Is the quality and reliability good, or should I be prepared for alot of grief? I'm having good success doing fax over PRI using a TE405; one span to the PRI, the other to an FXS channel bank that is almost obscenely underutilized (3 channels). I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2 link is a 1-hop SDSL (VOIP only) data link. This works well too. -A. ___ --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 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
RE: [Asterisk-Users] FAX over PRI
We get the occasional bad fax, but it really is an occasional one, other than that, its fine. We dont get any CRC errors or clock slips on the PRI, Id certainly say that it would be a good starting point to check the counters on these, Id also check that your drives are using DMA depending on your hardware, we had a customer a while ago who ended up doing a self install and none of his drives were enabled for DMA. Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Christensen Sent: 19 May 2006 17:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FAX over PRI I have had nothing but problems receiving faxes over PRIs with spandsp. I currently have 4 systems, 4 PRIs from 4 different providers... none of them get better than 50% success rates receiving faxes in spandsp, I constantly get cut off pages. No body seems to have a fix for it, and it is really frustrating. Supposedly it is caused by frame slips on the PRI, but if that is the case, I am 4 for 4 getting crappy PRIs that can't keep time. These same boxes work fine when receiving faxes over fxo ports, or if I plug a fax machine into an fxs port and call in to a spandsp extension the fax will be received just fine, so I am left thinking it must be the PRIs, but if all PRIs are this bad, how can anybody be using them? Tom On 5/19/06, Steve Hanselman [EMAIL PROTECTED] wrote: Sorry for the late reply but both of these are fine, we use spandsp to print some faxes and email others. We also route via a PRI to our other phone system to hylafax on an analog modem and also to an analog fax. So what you want to do is fine and will work. Steve -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Michael Gaudette Sent: 21 March 2006 20:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FAX over PRI Hmmm, Im not so sure I can apply this to me though.I just want to do Fax-To-Email using PRI channels as the incoming lines.Not so much transfer to a real fax. I am assuming that this is easily done with Asterisk? (I did it before with Asterisk SIP, but it only worked once every 10 tries or so) Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Andrew Kohlsmith Sent: March 21, 2006 3:25 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FAX over PRI On Tuesday 21 March 2006 15:09, Michael Gaudette wrote: How should I consider Fax over PRI channels with Asterisk?Is the quality and reliability good, or should I be prepared for alot of grief? I'm having good success doing fax over PRI using a TE405; one span to the PRI, the other to an FXS channel bank that is almost obscenely underutilized (3 channels). I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2 link is a 1-hop SDSL (VOIP only) data link.This works well too. -A. ___ --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 receivedthis 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 1BXUK 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 The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England
[Asterisk-Users] asterisk no longer compiles on gcc 2.95
Throwing errors relating to utils.h: /usr/include/asterisk/strings.h:264: parse error before `__extension__'/usr/include/asterisk/strings.h:264: parse error before `;'/usr/include/asterisk/strings.h:264: warning: type defaults to `int' in declaration of `__retval'/usr/include/asterisk/strings.h:264: `__len' undeclared here (not in a function)/usr/include/asterisk/strings.h:264: warning: initialization makes integer from pointer without a cast/usr/include/asterisk/strings.h:264: initializer element is not constant/usr/include/asterisk/strings.h:264: warning: data definition has no type or storage class/usr/include/asterisk/strings.h:264: parse error before `if'/usr/include/asterisk/strings.h:264: warning: type defaults to `int' in declaration of `__retval'/usr/include/asterisk/strings.h:264: redefinition of `__retval'/usr/include/asterisk/strings.h:264: `__retval' previously defined here/usr/include/asterisk/strings.h:264: parse error before `const'/usr/include/asterisk/strings.h:264: warning: data definition has no type or storage class/usr/include/asterisk/strings.h:264: warning: type defaults to `int' in declaration of `__retval'/usr/include/asterisk/strings.h:264: warning: data definition has no type or storage class/usr/include/asterisk/strings.h:264: parse error before `}'/usr/include/asterisk/strings.h:276: conflicting types for `strtoq'/usr/include/stdlib.h:328: previous declaration of `strtoq'In file included from /usr/include/asterisk/module.h:35, I seem to recall there being a statement that gcc = 3.00 is required, is this now being enforced? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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
RE: [Asterisk-Users] AgentCallbackLogin pre-# announcement?
Yes, there is a patch for this (search mantis), it's static in that it's a single announcement that doesn't currently relate to the queue. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 03 January 2006 10:00 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AgentCallbackLogin pre-# announcement? Is there a way to have AgentCallbackLogin make an announcement before requiring the callee to press #? I can not find anything in the documentation or other sites (voip- info etc). And at the moment the way i have it setup AgentCallbackLogin calls the agent and waits till # is pressed, it then plays the queue greeting. What i would like is for AgentCallbackLogin to play an announcement before requiring # so the agent can decide wether to answer the call based on time of day/workload etc. Example: Agent gets a call back and when answered they hear you have a sales/support/billing call, please press # to accept Is this possible? Thanks Adam ___ --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 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
RE: [Asterisk-Users] CID lookup from an Exchange Public folder
My point exactly. I'll take a look at that script though, if I could automate that each night then it might be fine, tag the imports, clear out those then re-import again. Thanks Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: 17 December 2005 18:29 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CID lookup from an Exchange Public folder Exchange contacts != AD entries. Contacts in Exchange are basically email messages with metadata. Now, if all of your contacts WERE in AD, you could do a script to query AD through LDAP (that's what AD is - LDAP with MS extensions) and you would solve latency problems when Asterisk would query AD instead of clunky MAPI. Here's a cool script to export contacts in a public folder to AD: http://www.msexchange.org/articles/Migrating-Contacts-Distribution-Lists -Out look-Active-Directory.html The problem with this is maintenance, since now you have 2 contact databases. Making sure they are sync'd wouldn't be an automatic process and invariably would mean that an admin would have to fire up ADSI Edit every once in a while. This is mitigated by how often you change contacts. In an org where contacts change rarely, or never this isn't a problem. Where I work, contacts nmber in the THOUSANDS and change EVERY DAY. The administrative overhead of maintaining those guys in AD is brutal, and that's why at my work I have basically banned using public folders as a contact manager and insisted that we use SQL server with a web front-end, this makes things simple for the maintainer, extensible and fast, and SQL server plugs into everything. hth -Original Message- From: Steve Hanselman [mailto:[EMAIL PROTECTED] Sent: Saturday, December 17, 2005 3:18 AM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] CID lookup from an Exchange Public folder We have a public folder full of contacts, but I understood that you could only access this if the contacts were contacts in AD? I was planning on doing a match on telephone number, mobile number and fax. And then pulling a shortened version of the name as the caller ID, Steve -Original Message- From: Jason SJOBECK [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 16 December 2005 21:48 To: asterisk-users@lists.digium.com Cc: Steve Hanselman Subject: Re: [Asterisk-Users] CID lookup from an Exchange Public folder Steve, You can get to anything in Exchange via LDAP. What is and or is not working? Where are you entering the callerID info you want pulled? Please see attachment for where you might want to enter this. Please share if you get this. Cheers. Jason - Message: 1 Date: Fri, 16 Dec 2005 18:29:12 - From: Steve Hanselman [EMAIL PROTECTED] Subject: [Asterisk-Users] CID lookup from an Exchange Public folder To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Has anybody done this? I looked at LDAP but you can't get to them that way, I'm considering either a timed export, or some other way (can you access them via IMAP? Or by wget on the owa web structure?) Steve 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
RE: [Asterisk-Users] CID lookup from an Exchange Public folder
We have a public folder full of contacts, but I understood that you could only access this if the contacts were contacts in AD? I was planning on doing a match on telephone number, mobile number and fax. And then pulling a shortened version of the name as the caller ID, Steve -Original Message- From: Jason SJOBECK [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 16 December 2005 21:48 To: asterisk-users@lists.digium.com Cc: Steve Hanselman Subject: Re: [Asterisk-Users] CID lookup from an Exchange Public folder Steve, You can get to anything in Exchange via LDAP. What is and or is not working? Where are you entering the callerID info you want pulled? Please see attachment for where you might want to enter this. Please share if you get this. Cheers. Jason - Message: 1 Date: Fri, 16 Dec 2005 18:29:12 - From: Steve Hanselman [EMAIL PROTECTED] Subject: [Asterisk-Users] CID lookup from an Exchange Public folder To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Has anybody done this? I looked at LDAP but you can't get to them that way, I'm considering either a timed export, or some other way (can you access them via IMAP? Or by wget on the owa web structure?) Steve 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
[Asterisk-Users] CID lookup from an Exchange Public folder
Has anybody done this? I looked at LDAP but you cant get to them that way, Im considering either a timed export, or some other way (can you access them via IMAP? Or by wget on the owa web structure?) Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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
[Asterisk-Users] function cut()
I thought that app_cut was deprecated in favour of function cut(), but I cant see this in the list or the code as of SVN-trunk-r7472M? Seeing as Ive just edited the dial plan, can anybody shed any light on this, or should I revert back to app_cut? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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
RE: [Asterisk-Users] function cut()
Ok, looks like app_cut is also function cut, but is missing from the Makefile in apps? If you add it in it doesnt compile, due to cut_synopsis not being defined. Seems like its in a state of flux, whos working on this? Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 15 December 2005 14:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] function cut() I thought that app_cut was deprecated in favour of function cut(), but I cant see this in the list or the code as of SVN-trunk-r7472M? Seeing as Ive just edited the dial plan, can anybody shed any light on this, or should I revert back to app_cut? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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
RE: [Asterisk-Users] function cut()
Added it to the Makefile, amended the reference to cut_synopsis and it compiles, installs and works fine. Logged it as a bug on mantis. Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 15 December 2005 16:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] function cut() Ok, looks like app_cut is also function cut, but is missing from the Makefile in apps? If you add it in it doesnt compile, due to cut_synopsis not being defined. Seems like its in a state of flux, whos working on this? Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 15 December 2005 14:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] function cut() I thought that app_cut was deprecated in favour of function cut(), but I cant see this in the list or the code as of SVN-trunk-r7472M? Seeing as Ive just edited the dial plan, can anybody shed any light on this, or should I revert back to app_cut? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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
[Asterisk-Users] Sending a recorded message to voicemail
Hi, We have an IVR application which produces a gsm file (its appended at various points, so I cant just drop them in voicemail), I want to send this to a users mailbox, but I cant see a way to do this, I presume that merely dropping the file into the directory isnt going to trigger off the usual notifications? Any ideas? Regards Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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
[Asterisk-Users] Possible bug in record?
Im trying to get record to append to a file, Im
using this:-
exten = 2,n,record(/tmp/${UNIQUEID}.gsm|5|0|a)
And its creating a new file?
If I check /tmp I can see the same filename being reused
each time, but the file jjust contains the latest recording.
Can anybody else confirm this?
Steve
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[Asterisk-Users] Goldstar GDK 186 voicemail
Are there any GDK users on the list? Do you know if its possible to disable voicemail on a per extension basis such that it returns busy rather than diverting to voicemail? I have the manuals but I cant see any reference to this. Regards Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.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
RE: [Asterisk-Users] callfile: How to invoke SetCallerPres ?
Probably easiest to set a variable to the number to be called and then jump to an extension to do whatever you want to do? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruno Voigt Sent: 13 September 2005 23:37 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] callfile: How to invoke SetCallerPres ? Hi, how may I define in a callfile the CallerID presentation to be used for the requested call, eg. set it to prohibited? TIA, Bruno 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 sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting retries in call files
Can anybody see a way of detecting the current number of retries remaining to a call file in the extension context that it is calling? E.g. If I want to schedule a fax and I want to feed an email back to the sender stating that the number is busy 2/5 retries remaining? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] /etc/init.d/asterisk barfing
Try doing an strace on it and seeing what the last section shows you. i.e. strace asterisk -vvvc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 31 August 2005 22:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] /etc/init.d/asterisk barfing Ok, starting to get cheesed off and feeling rather silly. cvs head as of 5 minutes ago. #root asterisk -vvvc works, no problem. #root safe_asterisk works no problem #root service asterisk start Starting asterisk: [ OK ] #root asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) /var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after the run). Can't find any reasons or errors for this not working - does anyone have any clue on where to start looking - I need * to automatically start on init. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] /etc/init.d/asterisk barfing
Sorry, went of at a tangent (that's what you get for half reading emails I guess!) Ok, guess the easiest thing to do is to check in the contrib directory, diff your one against the redhat (are you running redhat?) Mine is the same and it works fine, maybe you're running an outdated init script. Still worth trying the strace against the service command, at least it'll give you an idea of what it's trying to do. One last thing, did it work and then stop working or is this a fresh install and it's never worked? Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 01 September 2005 10:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] /etc/init.d/asterisk barfing Hi Steve :) The problem is not with the asterisk command, nor with safe_asterisk but with the /etc/init.d/asterisk script if I manually run /etc/init.d/asterisk start all's ok if I manually run service asterisk start it says that it has started, but hasn't :) Julian Steve Hanselman wrote: Try doing an strace on it and seeing what the last section shows you. i.e. strace asterisk -vvvc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 31 August 2005 22:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] /etc/init.d/asterisk barfing Ok, starting to get cheesed off and feeling rather silly. cvs head as of 5 minutes ago. #root asterisk -vvvc works, no problem. #root safe_asterisk works no problem #root service asterisk start Starting asterisk: [ OK ] #root asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) /var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after the run). Can't find any reasons or errors for this not working - does anyone have any clue on where to start looking - I need * to automatically start on init. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LG Goldstar GDK-186/162 question on voicemail
Are there any other GDK users out there with Asterisk? Ive got all the integration working, except voicemail. Does anybody know a way of disabling the forward to voicemail on a per extension or per DDI basis (I can disable the voicemail hunt group but then I cant light the MWI indicators as it seems that only ports marked in the voicemail group can issue the MWI on/off commands). Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Re: SpanDSP rxfax, no tiff
Add the debug option to the rxfax line
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Danz
Sent: 15 July 2005 13:13
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Re:
SpanDSP rxfax, no tiff
Ive tried
both with and without Answer as the first line, same result. When I was
searching through the archives, I believe there was a post from Steve Underwood
that said to always use Answer as the first line.
---
Yes, the
permissions are okay for getting to that folder.
/var/spool/asterisk is writable (voicemail works that's a subdirectory
under the
same path that has the same permissions as the subdirectory
'asterisk-fax'
As the same
user that runs asterisk I did a 'touch
/var/spool/asterisk/asterisk-fax/test.tif' just to be sure I could write
to that
directory. Permissions are fine.
[...]
[custom-fax]
exten =
s,1,Answer
exten =
s,2,Macro(faxreceive)
exten =
s,3,SetVar(>
exten =
h,1,system(/usr/local/sbin/mailfax ${FAXFILE}
[EMAIL PROTECTED] ${CALLERIDNUM} ${CALLERIDNAME} ${ONZENID})
[...]
I do not have the
system perform an answer. Try removing the line,
perhaps the
system is getting confused.
B
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RE: [Asterisk-Users] Voicemail = SMS
A little off topic, but I'm on orange, what's the domain and what is the format
e.g. 07973 or +447973...
From: [EMAIL PROTECTED] on behalf of Wilson Pickett
Sent: Fri 01/07/2005 6:56
To: Mark Charlton; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail = SMS
I have been trying for a while to find a way to get an SMS send when I
receive a voicemail into my asterisk system. I don't want to send an
SMS if the caller doesn't leave a message. I have voicemail.conf set
up to email and delete.
I use a backward solution to this problem, but it works. Orange, my
cell provider offers free SMS alerts for email sent to
[EMAIL PROTECTED] I send my vmail messages to my regular email
server which keeps them for online email retrieval. A procmail recipe
on the server then makes up an email without the vmail attachment to
my orange address with the callerid in the subject. Orange sends an
SMS that tells me I have a vmail message from ${CALLERID}. Although it
seems like a silly solution it does _exactly_ what you asked about.
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RE: [Asterisk-Users] hidecallerid on analog line
It depends on your telco, in the UK on an analog line we can prefix it with 141, so in that case yes, Asterisk can do it. You to find out from your telco whether a caller with a standard handset can do anything to control callerid with your telco. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chawki hammoud Sent: 30 June 2005 09:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] hidecallerid on analog line In the ISDN case, setcallerid or hidecallerid can be configured and I am aware that Asterisk doesn't support that on analog line. My question is whethere there is something like add-on script or hardware that will do the job. The teleco company provide the callerid service, but no private number service. --- Robert Webb [EMAIL PROTECTED] wrote: On Wed, 29 Jun 2005 13:56:00 -0700 (PDT) chawki hammoud [EMAIL PROTECTED] wrote: Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; It would have to be done through who ever provides your POTS service. They provide the caller ID to who you are calling. Some have the option to block it. Asterisk cannot be configured to do this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Setting Caller ID after Dial
And the UK although the PRI provider can either override or supply it for you and you are normally limited (unless you've signed an agreement) to DDI numbers directly provided by the PRI provider. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: 30 June 2005 00:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Setting Caller ID after Dial On 6/29/05, Bryce Chidester [EMAIL PROTECTED] wrote: The CallerID that is seen by others on calls originating from your PRI is set by your PRI provider; you have no control from Asterisk about this as it gets overridden by the provider. You must contact your carrier and ask them to set the CallerID for all PRI lines to the desired name/number. Really? you should really do your home work before you state something like this. In fact for most PRIs in the US you are wrong. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list Searchability
Might be worth asking the owner of voip-info.org if the mailing list link can go on the left sidebar permanently? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: 28 June 2005 16:26 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] list Searchability Great points Steve. I think the best we can do is all throw the newbies a bone ounce in a while. Redirection to the content that is relevant is enough to get most people on the path. Like you said, the hardest part is not seeing the trees for the forest. This is the whole teach a man to fish parable. It is pretty easy to tell someone A) How to search and where to look B) The basics of what Asterisk can do C) How to be a good list citizen With those tools, almost anyone can get their start here and beat the learning curve. Like it has been pointed out, there is not much else we can do with so much information in a free flowing format. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of steve szmidt Sent: Tuesday, June 28, 2005 8:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt On Monday 27 June 2005 20:04, Robert Webb wrote: I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need to be fully indexed. In a Google search box: site:lists.digium.com What you are searching for The problem many newbies faces is TOO MUCH information. Not being able to see the trees because of the forest basically. It does not matter either if it has been discussed until someone went crazy or died. The reason it keeps coming up is because it has not been solved. I totally agree with why. I sure don't want to be the one babysitting them. These posts were simply pointing out what I think, as a former educator, is part of the problem. Something which is not that hard to do. And indeed during some spare times I may put together something which is a lower gradient for those who have a hard time getting it. I sure would like to. Now it could very well be that many of these people never get anywhere because it's just too hard for them. But I know when I started a few years back, that a lot of the howto's have a stiff gradient. It skips pieces of information, assumes knowledge which is hard to come by and so on. Standard stuff. I'm not assuming or expecting that anyone is going to act on what I'm saying. If it was easy someone would have already implemented it. But I am saying that I see there are things that CAN be done which will make it easier. And if it makes it easier, this list will have less stupid and repetitive questions. More people will win using Asterisk and we should all win. (Except those who prefer fewer people competed in this arena. And there are a few here who are happy it's hard for others to take part of the fruit. There always are.) -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nasty little incident ...
I doubt they do, if they are marked as being there, but happen to be down then the numbers would stay the same. Sounds more likely that something happened with the clock source. You'd need to reproduce it out of hours and look at the output of pri show span x and cat /proc/zaptel/* From: [EMAIL PROTECTED] on behalf of Rich Adamson Sent: Wed 15/06/2005 5:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nasty little incident ... We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the meridian was turned off, * would no longer allow any outbound or inbound calls through spans 1 and 3 (although these are connected to the pstn). When I turned the meridian back on - in a hurry I might add ;) (had no time to play with configurations) and restarted *, then everything was ok again ... Should I comment out span 2 and 4, run a ztcfg, unplug the cables in 2 and 4, and then turn off the meridian ? Julian. /* zaptel.conf */ span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 span=3,2,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 loadzone=uk defaultzone=uk Just a wild guess When the two meridian links disappeared, the channel numbers probably changed. Instead of channels 1 through 124, you probably have channels 1 through 62 and your supporting dialplan (and other channel specific items) likely don't match. I thought that the definitions in the zaptel.conf and zapata.conf (see below) defined the channel numbers, not the physical channels themselves ? I use Dial(zap/g3) to call on the zap channels. /* zapata.conf */ context=isdn32-b prilocaldialplan=national internationalprefix = 00 nationalprefix = 0 localprefix = 01702 group=1 signalling=pri_cpe switchtype=euroisdn channel=1-15,17-31 context=meridian-b group=2 signalling=pri_net switchtype=euroisdn channel=32-46,48-62 context=isdn32-a pridialplan=unknown group=3 signalling=pri_cpe switchtype=euroisdn channel=63-77,79-93 context=meridian-a group=4 signalling=pri_net switchtype=euroisdn channel=94-108,110-124 I'm sure there are others on this list that can add to this, but when the card drivers are loaded and ztfg run, the channels that are discovered have to be mapped to what's in zaptel.conf one way or another. (Moving card driver load around changes the discovered order and one must manually modify zaptel.conf to match.) Then each zap channel is defined in zapata.conf, and those definitions have to match the channel numbers resulting from the above zaptel.conf stuff. So, what happens when two E1s disappear? Do the avaiable channel numbers change at the zaptel.conf level? My best guess is they do, but I don't have E1s around to play with to prove it. So, that's my best guess and it certainly can be an incorrect guess on my part. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.ukwinmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfers on PRI connected channel banks and legacy PBX's
Hi, Were using our legacy PBX as a channel bank with asterisk sitting between the pbx and our telco provider spliced by a TE410P. If it were a straight analog FXS card then wed use a hook flash to break into asterisk for transfers etc, does anybody know what the equivalent is for the PRI zaptel support? Regards Steve Steve Hanselman Brendata (UK) Ltd Tel: +44 (0)1268 466111 Fax: +44 (0)870 1387283 Mob: +44 (0)7973 750993 The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization
Jumping in very late to this thread... Is the solution not to change the voicemail system to enable it to utilise other entities as the store, e.g. a pop3 server or an imap server rather than just flat files on disk (which should remain an option). That way it doesn't matter where they listen to them or delete them from? Steve From: [EMAIL PROTECTED] on behalf of Race Vanderdecken Sent: Sat 11/06/2005 12:52 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization Aye, there's the rub. Now having said that, obviously we can't delete the message from the local store of the POP3 client after it has been already downloaded, but we are not talking about that, are we? 1. Thou shall not require any brain cells on the part of the end-user. 2. Thou shall not require any settings to be set on the user's equipment. ... More rules to follow. Rule #3 Thou shall not require the user to delete voicemail messages stored in their email account program by the voicemail server after they have deleted it from their voicemail account, unless they have told the administrator that they will do it, because the user thinks all of their messages (voice, email, fax, paper, phone) are all stored in ROM somewhere on the internet... You will drive your users nuts if they can't delete it from their message from one place. They will not understand they have to delete the same message twice, trust me. Race -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iassen Hristov Sent: Friday, June 10, 2005 7:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization -- Message: 4 Date: Fri, 10 Jun 2005 10:03:04 -0400 From: David Brodbeck [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voicemail and MS Exchange To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 IMAP is no good. Outlook, at least in older versions, cannot handle both an IMAP account and an Exchange account at the same time. (They can do POP3 and Exchange together, though.) Does this matter? All we are saying is that Exchange supports IMAP and we would use IMAP as the protocol to delete the message from the user's mailbox. How does the user access his mailbox is his choice. Now having said that, obviously we can't delete the message from the local store of the POP3 client after it has been already downloaded, but we are not talking about that, are we? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.ukwinmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk to Cisco Unity
With call manager V4 and above it's extremely easy, just connect a SIP trunk to *. BTW Unity is the Cisco voicemail system, Call Manager (CCM) is the actual PBX so your terminology may be confusing some people. From: [EMAIL PROTECTED] on behalf of Simone Sent: Fri 10/06/2005 10:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity I understand what you're saying, but I am not the one who makes the decisions. That decision is made already, so since I am actually getting your point and I agree with that, the only thing I can try to do right now, is try to avoid having Cisco Unity in the other 3 offices. I would love to implement Asterisk in these ones, but if it cannot be connected to Cisco this won't be an option at all, they won't consider it. So, back to the question, is it possible to connect Asterisk to Cisco and have all the functionality expected, and is it hard? Thanks, have a nice day Simone William Boehlke wrote: By the time you install the Asterisk server you have more features than Cisco delivers with Unity, for half the cost and without those annoying viruses. So instead of thinking about connecting Asterisk, consider disconnecting Unity. They make excellent landfill. Regards, William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simone Sent: Thursday, June 09, 2005 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity Hi, just wondering if my question is just unusual or if it is a quite stupid one. Thought there would be someone having this kind of scenario, but maybe I'm wrong. btw, have a nice day Simone Simone wrote: Hi all, first post. My company's office in the UK is soon going to get a Cisco VoIP solution system. What I am interested in, and couldn't find googling, is if it is possible to connect an Asterisk solution to the Cisco system and have all the nice advantages of it (mainly calling the extensions and directly reach the other office). Thanks, have a nice day Simone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.6 - Release Date: 6/8/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.ukwinmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can I hide caller id on the fly (per eachusesetting) on Bristuffed * and quadbri
We do this, you need to ensure that you are allowed to control your callerID (we had to request this from our telco) You should then be able to use the SetCallerPres and SetCallerID to control what (if any) number you give out. Steve -Original Message- From: Robert Rozman [mailto:[EMAIL PROTECTED] Sent: 05 May 2005 09:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can I hide caller id on the fly (per eachusesetting) on Bristuffed * and quadbri - Original Message - From: Peer Oliver Schmidt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 11:14 PM Subject: Re: [Asterisk-Users] Can I hide caller id on the fly (per each usesetting) on Bristuffed * and quadbri Robert Rozman wrote: I wonder if I can hide caller id for just certain users. Can I override caller id setting for show or hide on the fly from dialplan ? Did you try setcallerid()? -- I tried but this will work if calling internal line. I'm after dynamically hiding caller id on QuadBRI outgoing ISDN calls... I guess this is possible with settings in zapata.conf, but only per channel - I wonder if it is possible to set this up by user or do it from dialplan with some command Thanks in advance, Rob. Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan_sccp - status
I think it's displaying the name of the line that the call is coming in on, but you're expecting the name of the calling party (as I was!) Steve -Original Message- From: Mark Johnson [mailto:[EMAIL PROTECTED] Sent: 03 May 2005 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Chan_sccp - status Mark Johnson wrote: Julien Goodwin wrote: Then why haven't you sent a backtrace? If I can see why it's crashing then I can fix it. Thanks, Julien chan_sccp project lead The general consensus was that I needed to be running HEAD to make this work properly. I upraded last night to HEAD and my SCCP stuff seems to working perfect!! Thank you!! Also, I saw you are in need of a 7910 from your announcement. If you email me your shipping info offlist, I will make sure you get a couple of them. Mark While on the topic, I'm having some weird issues with the 7910's and the callerid. I got them to display the outgoing calls correctly, but if I call from an internal SIP phone to an internal SCCP 7910, the display shows that that SCCP phone is calling itself until you answer. After you pick up, it changes to read Unknown Number to sccp ext# Anyone have luck getting this to work? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering without ringing from PRI
Have you tried the latest CVS, there was a bug relating to ALERTING which was fixed yesterday... -Original Message- From: Ugur GUNCER [mailto:[EMAIL PROTECTED] Sent: 08 April 2005 04:54 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Answering without ringing from PRI I made that but still same no ringing for pri coming calls -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mathew McKernan Sent: Friday, April 08, 2005 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Answering without ringing from PRI Hi, Where you have your 1st priority, I suspect you have it set to Answer. Try changing this to Wait(1). Then on priority 2 put answer. i.e. Exten = s,1,Wait(1) Exten = s,2,Answer Exten = blah blah Hope that covers it, Thanks Mathew -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ugur GUNCER Sent: Friday, 8 April 2005 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Answering without ringing from PRI Importance: High How can i set asterisk for when call came from pri ring once then answer pri call. In now call cames from pri then asterisk directly answering pri call without ringing. Then my carries hangup call because they said your box is answer without ringing Iyi Calismalar Saygilarimla Ugur GUNCER Sistem Yoneticisi Telebizz Tel. ve Int. Hizm. Office= +90 212 347 6959 Gsm = +90 544 535 9737 Fax = +90 212 347 6949 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] re: Problem: Compiling error for SpanDSP
Which version of spandsp are you using, current versions check on ASTERISK_VERSION_NUM and handle the callerid accordingly. Some time ago the makefile changed, but yours seems fine for that as you are carrying CCFLAGS through, ensure that you're running a recent copy (and that you've also copied not just the spandsp but the apps as well). Steve -Original Message- From: Justin Newman [mailto:[EMAIL PROTECTED] Sent: 30 March 2005 04:15 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] re: Problem: Compiling error for SpanDSP Date: Tue, 29 Mar 2005 21:43:06 -0500 From: KMZ Enterprises [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem: Compiling error for SpanDSP app_rxfax After resolving my earlier problem in updating the apps Makefile with the patch for SpanDSP, I encountered another problem when I executed the Make utility from /usr/src/asterisk. I obtained an error as shown below. Not sure on how to resolve the problem. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../incl ude -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD- 03/29/05-21:11:20\ -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -D ASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run/asterisk\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/ asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fomit-frame-pointer -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c: In function `phase_e_handler': app_rxfax.c:86: structure has no member named `callerid' make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Regards, Kerry For a while, the structure switched over. If you are getting an error with callerid in your build, try searching the code (app_rxfax.c) and replace the occurances with cid.cid_num. Justin Newman Newman Telecom, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] missing ring-tone
We've got the same issue, I'm just starting to investigate it. Have you resolved your issue now? If not, I'll keep you updated on what I find. Steve From: Lars L. Christensen [mailto:[EMAIL PROTECTED] Sent: 27 March 2005 16:14 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] missing ring-tone Hi there I've got a rather irritating problem with my Asterisk server... Whenever someone tries to call me, the don't get the usual "ring-tone" when they wait for me to pickup the phone. I don't know if I've disabled this feature somewhere in my configuration files. Since I'm in Denmark, I've got an entry in the indications.conf file pointing to Denmark (country=dk). Any ideas where to start trouble shooting? Cheers Lars Christensen The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] missing ring-tone
Yes, Asterisk isn't sending an ALERTING indication on the PRI, this is our problem, sounds as though this isn't yours though, as by the sound of it you're pure VOIP? -Original Message- From: Lars L. Christensen [mailto:[EMAIL PROTECTED] Sent: 05 April 2005 17:56 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] missing ring-tone Unfortunately I haven't resolved the issue yet, so at the moment I've bypassed the Asterisk and connected the Sipura directly to the internet. I've tried to reset the Sipura with no luck. I've also noticed I do get a ring-tone when I call from line 1 to line 2 on the sipura, so the problem must be the Asterisk. L Looking forward to hear from you... Cheers, Lars From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 5. april 2005 18:03 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] missing ring-tone We've got the same issue, I'm just starting to investigate it. Have you resolved your issue now? If not, I'll keep you updated on what I find. Steve From: Lars L. Christensen [mailto:[EMAIL PROTECTED] Sent: 27 March 2005 16:14 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] missing ring-tone Hi there I've got a rather irritating problem with my Asterisk server... Whenever someone tries to call me, the don't get the usual ring-tone when they wait for me to pickup the phone. I don't know if I've disabled this feature somewhere in my configuration files. Since I'm in Denmark, I've got an entry in the indications.conf file pointing to Denmark (country=dk). Any ideas where to start trouble shooting? Cheers Lars Christensen The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.ukThe information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileg! ed information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DASS II cards supported
Get a converter to Q.931, we use one called an IQ200 from (I vaguely recall) Teltrend, search the web, works fine, easy to setup, we've used them at two customers now with no problems at all. Steve From: Stephen Owen hosted [mailto:[EMAIL PROTECTED] Sent: 08 February 2005 14:35 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DASS II cards supported I know Q931 cards are supported, does anybody know how to go about supporting DASS II ? Thanks Stephen The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_skinny and firmware upgrade
Nothing to do with skinny, drop the new file(s) in your tftp directory and edit the .xml file to specify the new version, the phone will upgrade itself when it loads the config. Steve -Original Message- From: Subhi S Hashwa [mailto:[EMAIL PROTECTED] Sent: 23 January 2005 06:33 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] chan_skinny and firmware upgrade Hello all, I am trying to upgrade the firmware on my cisco 7910 without using CCM. I was told that chan skinny is possibly capable of doing that and would like to make sure. I have P00405000600 firmware which I have put in version in skinny.conf. the phone basiclaly stops at verifying load. tcpdump shows nothing happening apart from small amount of traffic to port 2000 (skinny). Does anyone have any ideas on how to get the new firmware into the phone? cisco instructions arent very helpful. PS unlike the bigger brother of the phone, this one does not request PS OS79XX.TXT file and is not SIP capable. -- Best regards, Subhi S Hashwa mailto:[EMAIL PROTECTED] When everything is heading your way, you're in the wrong lane. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_skinny and firmware upgrade
From the very early days of Cisco skinny the phones have all requested XMLDefault.cnf.xml, you just need to pop it in there (either run a tcpdump on the tftp port or run the daemon in logging mode and you'll see). Steve -Original Message- From: Subhi S Hashwa [mailto:[EMAIL PROTECTED] Sent: 24 January 2005 14:46 To: Steve Hanselman Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] chan_skinny and firmware upgrade Monday, January 24, 2005, 9:23:50 AM, Steve Hanselman wrote: Nothing to do with skinny, drop the new file(s) in your tftp directory and edit the .xml file to specify the new version, the phone will upgrade itself when it loads the config. the firmware I have doesn't request xml file it requests SEPMAC.cnf I udnerstand the new versions of firmware request SEPMAC.cnf.xml. Not sure where to go from here, any ideas? -- Best regards, Subhi S Hashwamailto:[EMAIL PROTECTED] When everything is heading your way, you're in the wrong lane. 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: chan_skinny and firmware upgrade
Stick that on the Wiki
-Original Message-
From: Tom Ivar Helbekkmo [mailto:[EMAIL PROTECTED]
Sent: 24 January 2005 16:31
To: Subhi S Hashwa
Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Steve Hanselman
Subject: Re: chan_skinny and firmware upgrade
Subhi S Hashwa [EMAIL PROTECTED] writes:
The xml request is a feature of new firmware, that is my guess.
Yup. Older, skinny capable, phones request the SEP...cnf file, which
is in a binary format. I don't know how to get them to update their
firmware, but I do know how to build the SEP file to configure the
phones correctly. I use the below program to write a simple one, to
give the phone the name, address and TCP port number (2000) to use for
the skinny protocol (I haven't bothered to make something that reads
the configuration data from the terminal -- I edit, compile, and run):
/*
* Program to write a SEPDefault.cnf for Cisco SCCS phones.
* Records in the file have two bytes that tell what data
* is supplied, then the data (either fixed length or as
* a zero-terminated character string.
*/
#include stdio.h
#include stdlib.h
int main(int argc, char **argv) {
FILE *SEPDefault;
unsigned char SEPData[BUFSIZ];
SEPDefault = fopen(SEPDefault.cnf, w);
/* change the string to be the FQDN of the call manager: */
sprintf(SEPData, %c%c%s, 1, 1, pbx.company.com);
fwrite(SEPData, 1, strlen(SEPData) + 1, SEPDefault);
/* change the last four numbers to the IP address of the call manager: */
sprintf(SEPData, %c%c%c%c%c%c, 1, 2, 192, 168, 1, 42);
fwrite(SEPData, 1, 6, SEPDefault);
/* the last four numbers are 2000 as a four byte little-endian integer: */
sprintf(SEPData, %c%c%c%c%c%c, 1, 3, 208, 7, 0, 0);
fwrite(SEPData, 1, 6, SEPDefault);
/* this is a special end of file marker: */
sprintf(SEPData, %c%c, 1, 255);
fwrite(SEPData, 1, 2, SEPDefault);
fclose(SEPDefault);
}
/*
* eof
*/
-tih
--
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
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prohibited. If you have
received this communication in error, please notify Brendata immediately on:
+44 (0)1268 466100, or email '[EMAIL PROTECTED]'
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RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
Has anyone also logged a support call with Digium, it has to be either the card, Linux or the Zaptel drivers. Steve -Original Message- From: Joshua McAdam [mailto:[EMAIL PROTECTED] Sent: 14 January 2005 06:30 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server Has anyone logged a support issue with HP on this one? I still haven't been able to get it working so far, So I'm going to log a support issue here in australia to see what HP can do about this and was wondering if anyone else has. Josh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Monday, 10 January 2005 4:22 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server Make sure you has a span defined for each port on the TE410P. With out signaling it would not take interrupts. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl H. Putz Sent: Monday, January 10, 2005 12:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server I have been having this exact problem with a Tatung dual EMT-64 server as well. I have been trying to get a TE410P running and all looks great, driver loads, runs ztcfg OK, etc. but no interrupts are ever processed. One additional piece of info that I have not seen in this thread is that I am able to successfully start and run a T100P card in this system. In the same PCI slot, wct1xxp driver built from the same CVS HEAD version as the wct4xxp. Just hoping this might shed some light on the problem for any Digium folks monitoring the forum. Karl Putz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
I'm assuming that other non Digium cards work in it, but yes, you're right. Has anybody run any other PCI cards in those slots under Linux and seen interrupts from those cards? Steve -Original Message- From: Adam Goryachev [mailto:[EMAIL PROTECTED] Sent: 14 January 2005 09:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server On Fri, 2005-01-14 at 09:23 +, Steve Hanselman wrote: Has anyone also logged a support call with Digium, it has to be either the card, Linux or the Zaptel drivers. You missed the obvious or the HP Compaq DL380 G4 server Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
Any interrupts would be useful, that's the issue, the interrupt count is zero. -Original Message- From: Matt Riddell [mailto:[EMAIL PROTECTED] Sent: 14 January 2005 11:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server Steve Hanselman wrote: I'm assuming that other non Digium cards work in it, but yes, you're right. Has anybody run any other PCI cards in those slots under Linux and seen interrupts from those cards? You'd be hard pressed to find a standard card requiring accurate interrupts 1000 times per second... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax to email
Has anybody looked into implementing a fax send interface for Asterisk using the FSP code, that way it would plug straight into outlook and all the other windows bits'n'pieces? 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] phones with two ethernet ports
Some of the Cisco phones do (7940,7960 etc, also the 7910+SW but this is skinny only). Steve -Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED] Sent: 02 January 2005 21:35 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] phones with two ethernet ports Hi there, what phones are available that have two ethernet ports? I want to do some cabling at a new installation and i heard there are such phones (SIP i guess) out there. That way i dont have to run two cat5 to the user desktop. I think 3COM had one but can't find the web site reference for the two port phone thanks, erick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Final call for departments
Accounts by itself would be useful. -Original Message- From: David Boyd [mailto:[EMAIL PROTECTED] Sent: 30 December 2004 00:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Final call for departments HOw about : development Dave On Wed, 2004-12-29 at 04:51, Alspach Family wrote: I am getting ready to submit a list of department names to be recorded. This is what I have so far: Accounting Accounts payable Accounts receivable Administration Billing Collections Complaint Customer Service Engineering Facilities Help desk Human Resources Information Technology Inside Sales Investor Relations Legal Mail room Marketing Printing Projects Public Relations Purchasing Receiving Sales Sales Floor Shipping Shop Support Systems Technical Support Travel If any one has additional suggestions, please e-mail them to me ([EMAIL PROTECTED] or [EMAIL PROTECTED]). I am fairly sure that none of the above exist (I was only able to search through the WIKI list, so if there are other prompts in the CVS that are not listed there, I do not know about them.) If I have made a dupe, please let me know so that I can remove it. I was fairly certain that 'Operator' was already available but I was unable to find it by its self. Thanks for your help. I plan on sending these off on Friday the 31st so please try to get them to me by then. Thanks; James ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
I'm guessing he wants to do it the other way around, i.e. the external calling party hears music, not the internal calling party making an external call. -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: 28 December 2004 21:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Music instead of Tunes On Tue, 28 Dec 2004, Marc Storck wrote: more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart Currently I will have to answer the line to do that. Is there a way to do this with asterisk? See the help for dial: 'm' -- provide hold music to the calling party until answered. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
The difference is that you'd have to answer the call, my guess is that it can't be done (by a Joe Average like ourselves), otherwise we'd provide useful information to callers at no charge. -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: 29 December 2004 11:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Music instead of Tunes On Wed, 29 Dec 2004, Steve Hanselman wrote: On Tue, 28 Dec 2004, Marc Storck wrote: more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart Currently I will have to answer the line to do that. Is there a way to do this with asterisk? See the help for dial: 'm' -- provide hold music to the calling party until answered. I'm guessing he wants to do it the other way around, i.e. the external calling party hears music, not the internal calling party making an external call. Are the two cases different in any way? The external call comes in, goes to a context which eventually leads to a Dial(...) calling the internal user. That Dial call provides music to the external caller while the internal call is in progress. Asterisk has no concept of external or internal callers, only channles and contexts. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
So we could provide caller position announcements without the callers actually incurring charges? Has anybody tried this (in the UK)? -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: 29 December 2004 14:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Music instead of Tunes On Wed, 29 Dec 2004, Steve Hanselman wrote: Are the two cases different in any way? The external call comes in, goes to a context which eventually leads to a Dial(...) calling the internal user. That Dial call provides music to the external caller while the internal call is in progress. The difference is that you'd have to answer the call, my guess is that it can't be done (by a Joe Average like ourselves), otherwise we'd provide useful information to callers at no charge. For pots lines this is true. For isdn lines there is no need to answer prior to sending data. The reverse path (from the called party towards the calling party) is opened when (this is form memory, it may be another IE) PROGRESS is transmitted. You can use Playback and a host of other connads on an unanswered line. Some of these will automatically answer the line unless given an option not to. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music instead of Tunes
Thinking about it we may well be able to do this in the UK as one of the complaints I get about Asterisk is that our ring tone has changed to external callers, (the zone is set correctly for zaptel, but it's different from the normal ring tone), so the tones must be coming from the TE405, not just generated as a result of the accept (unless some data in the accept signifies the tones to generate?) -Original Message- From: Paul Crick [mailto:[EMAIL PROTECTED] Sent: 29 December 2004 18:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Music instead of Tunes So we could provide caller position announcements without the callers actually incurring charges? Has anybody tried this (in the UK)? Maybe.. but probably not.. In the UK (and most European countries), the ACCEPT message triggers generation of ringback tone at the calling party's exchange (central office). This is as opposed to the North American way of doing things where the ACCEPT message opens up a one way speech path from the called party to the calling party (originally for providing inband call progress tones I believe). Also, there's a timer on how long you can be in that state without issuing an ANSWER and thus tripping answer supervision/billing commencement. I think technically it IS possible to get UK kit to work in the US fashion, but you have to talk to a switch tech that knows what he's doing, and of course you may get bitten with the Yeah, it's doable, but we don't have that software feature pack installed on our switch line. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo on one E1 line, but not the other
It looks like this is a splice between a couple of ISDN-30 lines and one or more PBX's? Are they both with the same provider, or with different providers? We ended up adjusting the gain our ours as we would hear a distinct echo on certain calls. Other than that, you'll need to do the usual tests, check for shared interrupts, also, see if disk activity causes a problem. (check /proc/interrupts for shared interrupts). Have you also checked the output of the pri commands to ensure that you're not getting line errors? Steve -Original Message- From: Asterisk [mailto:[EMAIL PROTECTED] Sent: 13 December 2004 12:04 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Echo on one E1 line, but not the other We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p EuroISDN. We have 2 ISDN lines, one we had for testing, and one for general (40+ users) use. During the testing phase, we had 10 phones linked to the second ISDN line, and there were no problems with echo at all. Lucky me. However, since we have started rolling out, we've had quite loud complaints that there is a terrible echo. If I direct my 7960 to use the primary line, there is an echo. If I use the second line (dialling the same external number) there is no echo at all. What could be the issue ? I have noticed on the primary line there is a detected rx/tx on channel xx, echo cancellation disabled (or something like that. Is this the cause ? We have several fax / modems going through the line - should I always dedicate a channel to them ? I've included my zaptel.conf and zapata.conf file below. Any help / comments appreciated. # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 span=3,2,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 loadzone=uk defaultzone=uk # ;zapata.conf [general] [trunkgroups] [channels] musiconhold=default language=en rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callerid=asreceived callwaiting=yes usecallingpres=yes callwaitingcallerid=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 ;relaxdtmf=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 ;adsi=yes context=isdn32-b pridialplan=unknown group=1 signalling=pri_cpe switchtype=euroisdn channel=1-15,17-31 context=meridian-b group=2 signalling=pri_net switchtype=euroisdn channel=32-46,48-62 context=isdn32-a pridialplan=unknown group=3 signalling=pri_cpe switchtype=euroisdn channel=63-77,79-93 context=meridian-a group=4 signalling=pri_net switchtype=euroisdn channel=94-108,110-124 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo on one E1 line, but not the other
More by trial and error, we backed off the gain until it disappeared but with no detriment to the call quality (didn't want people to sound like a whisper). Our situation was somewhat different to yours though, we were seeing the issues on calls from our PBX, not on calls through the IP phones. Who hears the echo, the IP phone user or the remote user? Steve -Original Message- From: Asterisk [mailto:[EMAIL PROTECTED] Sent: 13 December 2004 12:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Echo on one E1 line, but not the other Both ISDN lines are going into the same * box - span 1 is the test isdn line and span 3 is the live isdn line. The two ISDN lines are situated right next to each other! As mentioned there is no problem with the test line, so there isn't a problem with * as such (I don't think!). Perhaps I haven't got the configuration quite right. When you say that you adjusted the gain, is that the tx/rx settings in zapata.conf ? How did you determine the correct settings: by placing a call and monitoring using ztmonitor ? Thanks. Julian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 13 December 2004 12:13 To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Echo on one E1 line, but not the other It looks like this is a splice between a couple of ISDN-30 lines and one or more PBX's? Are they both with the same provider, or with different providers? We ended up adjusting the gain our ours as we would hear a distinct echo on certain calls. Other than that, you'll need to do the usual tests, check for shared interrupts, also, see if disk activity causes a problem. (check /proc/interrupts for shared interrupts). Have you also checked the output of the pri commands to ensure that you're not getting line errors? Steve -Original Message- From: Asterisk [mailto:[EMAIL PROTECTED] Sent: 13 December 2004 12:04 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Echo on one E1 line, but not the other We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p EuroISDN. We have 2 ISDN lines, one we had for testing, and one for general (40+ users) use. During the testing phase, we had 10 phones linked to the second ISDN line, and there were no problems with echo at all. Lucky me. However, since we have started rolling out, we've had quite loud complaints that there is a terrible echo. If I direct my 7960 to use the primary line, there is an echo. If I use the second line (dialling the same external number) there is no echo at all. What could be the issue ? I have noticed on the primary line there is a detected rx/tx on channel xx, echo cancellation disabled (or something like that. Is this the cause ? We have several fax / modems going through the line - should I always dedicate a channel to them ? I've included my zaptel.conf and zapata.conf file below. Any help / comments appreciated. # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 span=3,2,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 loadzone=uk defaultzone=uk # ;zapata.conf [general] [trunkgroups] [channels] musiconhold=default language=en rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callerid=asreceived callwaiting=yes usecallingpres=yes callwaitingcallerid=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 ;relaxdtmf=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 ;adsi=yes context=isdn32-b pridialplan=unknown group=1 signalling=pri_cpe switchtype=euroisdn channel=1-15,17-31 context=meridian-b group=2 signalling=pri_net switchtype=euroisdn channel=32-46,48-62 context=isdn32-a pridialplan=unknown group=3 signalling=pri_cpe switchtype=euroisdn channel=63-77,79-93 context=meridian-a group=4 signalling=pri_net switchtype=euroisdn channel=94-108,110-124 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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
RE: [Asterisk-Users] XML to monitor queues on Cisco display ?
You can have a refresh interval on the XML though which achieves the same thing. Also, you can do a push, there are examples in the developers kit available on the Cisco web site. -Original Message- From: Wayne Sheppard [mailto:[EMAIL PROTECTED] Sent: 04 December 2004 18:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] XML to monitor queues on Cisco display ? Henry Devito wrote: I attempted this but I got stuck on one issue. Cisco phones pull data so I couldn't get them to autoupdate. In other words push data to them. I am working on an app to run on a windows desktop that will show the queues, the amount of calls in each queue, the longest wait time and the average wait time. I am also planning on creating the app with alarm thresholds. When the app is minimized it will go to the task bar and if the queue gets too full it will popup the window on the desktop and/or make the icon in the taskbar turn red. Henry Henry, that is a very useful app indeed! Do you plan to share that, sell it, ?? Love to get more info or help.. Cheers, Wayne ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream phone price
You multiply to get the dollar price. Careful where you go on holiday, it could be costing more than you think!! -Original Message- From: David J Carter [mailto:[EMAIL PROTECTED] Sent: 11 October 2004 08:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Grandstream phone price $1.64 to the £1 I think this morning so $35 stands. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wolf N. Paul Sent: 11 October 2004 07:40 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream phone price Except that £55 is more like $75-80 and not $35. Regards, Wolf David J Carter [EMAIL PROTECTED] writes: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave Grandstreams are availabe for $65 quanity one, so its not hard to believe that you could get them for $55 for larger quantities ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mpg123 - multiple instances, taxing CPU
check your musiconhold.conf, for each one you define you'l get an instance. -Original Message- From: Matthew Boehm To: [EMAIL PROTECTED] Sent: 03/09/04 15:04 Subject: [Asterisk-Users] mpg123 - multiple instances, taxing CPU Is there any reason why there should ever be more than 1 instance of mpg123 running on a * server? I just did an 'uptime' and noticed all 3 of my loads where over 3.00. 'top' showed 8 mpg123 processes all processing the same 3 songs (our background music). I tried to kill one of them but another one spawned in its place. Any ideas? Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Closing bug reports without fixing the repor ted problem
[vested interest: I reported that bug!] I agree with this, a system that has x bugs still has x bugs even if you close them, I think they should be closed by houskeeping if they can't be reproduced and the person that reported them can't provide a way to replicate, but they shouldn't be closed because nobody has fixed them for a while. This seems to be the case on a number of entries. Steve -Original Message- From: Rob Wise To: [EMAIL PROTECTED] Sent: 25/08/04 07:01 Subject: [Asterisk-Users] Closing bug reports without fixing the reported problem While this is more a question for the bug system admins, it does impact on the whole Asterisk community so I'm going to ask it here. Why does it seem to be normal procedure to close bug reports which have not been resolved? Twice now bug 1915 has been closed off without a solution being found, much to the protest of those who are suffering from the reported problem. I am curious to know why this is being done as it does not seem beneficial in any way to the Asterisk project to hide problems by closing the ticket rather than leaving them open so they can be worked on. I also am curious to know what other members of the community think of this. Rob Disclaimer: I am one of those affected by the bug, and it is still present in today's CVS head. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] App.c
Delete it and cvs update will retrieve it. -Original Message- From: AJ Grinnell [mailto:[EMAIL PROTECTED] Sent: 02 August 2004 17:33 To: Asterisk Subject: [Asterisk-Users] App.c Can someone tell me where I can get just app.c from. Mine somehow got corrupted, and no updates or anything else will fix it. I just need the one file from the latest cvs. 8-1-04. Please help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE405P and E1
Our ISDN 30 is delivered that way, but we were also supplied with a balun that takes the two balanced coaxs and turns them into a single RJ45, maybe your telco needs to supply you with some extra kit? Steve -Original Message- From: Kim Esben Jørgensen [mailto:[EMAIL PROTECTED] Sent: 26 July 2004 10:40 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TE405P and E1 Hello Im from Denmark and i've just got my Digium TE405P. But i have some problems when i connect it to my E1 connection (ISDN30). My telco delivered a alcatel box witch have a G.703 120ohm (DB9 with a serial to rj45) and a 75ohm coax connection. I've tried to connect using the 120ohm with rj45 and a ordinary utp cable. But it dosent seem to work. I've tried several zaptel.conf setting, but none of them made the led stop blinking. my current zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=us defaultzone=us /my current zaptel.conf Do you have any hints. regards Kim Esben Joergensen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Daytime - Nighttime
Yes, you'd have a dialplan entry that set a value in the database, then acted upon that. You'd probably want some nice voice prompts The system is currently in [Day/Night/Holiday] mode, press 1 to set to day, 2 to set. Steve -Original Message- From: Massimo De Nadal [mailto:[EMAIL PROTECTED] Sent: 22 July 2004 13:57 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Daytime - Nighttime Is it possible to build a dialplan in which shifting from daytime to nightime is not hour based but phone driven ??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller based routing
In your dialplan for your voip routing you'd put a gotoif that jumped to your PSTN context if it matched your criteria (e.g. EXTEN = faxextension) Steve -Original Message- From: GIBERT Frédéric To: [EMAIL PROTECTED] Sent: 21/07/04 13:58 Subject: [Asterisk-Users] Caller based routing Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture, My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn't work! Is there any possibilities to do caller based routing in asterisk, in order that when a fax try to send a fax, the call is automatically routed through the PSTN and not through the VoIP. Thanks. GIBERT Frédéric Mobile: +33 6 72 08 35 16 Fax : +33 1 30 71 39 33 Mail : mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Adding voice mail box
They only get created as they are used and voicemail left, try leaving a message and you should see that the structure etc is created. Steve -Original Message- From: Steve [mailto:[EMAIL PROTECTED] Sent: 19 July 2004 08:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Adding voice mail box On Monday 19 July 2004 01:23 am, Brian K. West wrote: Dont have to.. just add it to the voicemail.conf and it will auto do everything for you. bkw Well, after having restarted * a few times, and rebooted once, I can say that no mailboxes were created automatically. I'm running a week old HEAD. Brian, what version were you running when you observed this nice feature? - Original Message - From: Steve [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 9:58 PM Subject: [Asterisk-Users] Adding voice mail box -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA+zjhljK16xgETzkRAh8jAKCJ7iJhFBVRxBFzbl8cGziqbnUjoQCdEzbb oTA7sXW1EXmmDGpUXrPf174= =zANK -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spa ndsp fails to decode
On the /proc/zaptel it was lost interrupts, you haven't got any so that's good! On opencall.org there's a known issues link and that mentions some fax machines that have issues, might be worth a quick check there. I can receive from our fax machine this end, if you'd like I'll send you a test fax if you send me your fax number, if you can receive from us and tpc then it's more likely that the issue is with your fax machine and you may have to wait for Steve to release his latest version. You'll then have the choice of rxfax or hylafax. One last question, how is your fax machine connected, direct to a Telco pots line or via an fxo card in asterisk? Steve -Original Message- From: Stephen J. Wilcox [mailto:[EMAIL PROTECTED] Sent: 14 July 2004 15:43 To: Steve Hanselman Cc: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spa ndsp fails to decode Hi Steve, not boring at all, I'm out of ideas and I'm not clued up about how to look into this problem further so your suggestion is appreciated. Looking into /proc/zaptel/1 what am I looking for? # more /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS/CRC4 ClockSource 1 WCT1/0/1 ClearChannel (In use) snip repeated 2-15 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 ClearChannel snip repeated 18-31 Listening to audio isnt something that I think I will be able to diagnose, I'll give it a go but it'll sound like a fax machine to me ;) [now, give me v90 any day and i'll tell you whats going on! ;) ] I'm testing from the office fax machine which is used a lot all the time and never has problems, I tried the tpc.int and the fax comes thro fine. Hmm so what is that telling me? The office fax really does send dozens of faxes per day with no failures Looking at the debug output of the two, heres the ey differences: Slow carrier up Slow carrier down - prior to start receiving, altho slow carrier up is shown in both a couple times once transmission begins. The good one goes into 'start rx document' but the bad one seems to keep trying to train.. i have put a diff in below .. is there anything useful here? Thanks for any help! Steve # diff fax-good-tpc.int fax-bad-office 1c1 -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/faxes/20040714-140202.tif) in new stack --- -- Executing RxFAX(Zap/2-1, /var/spool/asterisk/faxes/20040714-143316.tif) in new stack 2a3,4 Slow carrier up Slow carrier down 22c24 TSI: 43 74 65 6e 72 65 74 6e 49 20 6e 6f 6d 65 44 20 20 20 20 20 20 --- TSI: 43 31 37 31 31 36 35 34 35 34 38 30 20 20 20 20 20 20 20 20 20 24,25c26,27 Remote fax gave TSI as: Demon Internet DCS: 83 00 46 f0 00 --- Remote fax gave TSI as: 08454561171 DCS: 83 00 86 90 00 31c33 R8x7.7lines/mm and/or 200x200pels/25.4mm OK --- 2D coding OK 34c36 Minimum scan line time: 0ms --- Minimum scan line time: 5ms 39a42,49 Coarse carrier frequency 1698.91 (60) Training error 93.312809 Training succeeded (constellation mismatch 48.327239) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 23 FTT: 44 Fast carrier up 44,46c54,56 Coarse carrier frequency 1700.20 (56) Training error 5.448260 Training succeeded (constellation mismatch 13.584253) --- Coarse carrier frequency 1699.55 (60) Training error 61.497972 Training succeeded (constellation mismatch 40.535409) 49,55c59,60 Changed from phase 5 to 4 Start rx document - compression 1 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 --- Trainability test failed - longest run of zeros was 31 FTT: 44 61,63c66,68 Coarse carrier frequency 1700.15 (56) Training error 5.155357 Training succeeded (constellation mismatch 11.403019) --- Coarse carrier frequency 1700.09 (60) Training error 46.800474 Training succeeded (constellation mismatch 33.977520) 65,71d69 Fax3Decode1D: Warning, (FakeInput): Premature EOL at scanline 2155 (got 0, expected 1728). Page 1 of /var/spool/asterisk/faxes/20040714-140202.tif: 2156 rows received 0 total bad rows 0 max consecutive bad rows Rx page end detected Changed from phase 5 to 3 73,81c71,72 Slow carrier up MPS: 4f MPS with final frame tag In state 5 Changed from phase 3 to 4 Start rx page MCF: 8c HDLC underflow in state 7 Changed from phase 4 to 5 --- Trainability test failed - longest run of zeros was 77 FTT: 44 87,113c78,80 Coarse carrier frequency 1700.20 (56) Training error 6.299062 Training succeeded (constellation mismatch 14.358997) Fast carrier trained Fax3Decode1D: Warning, (FakeInput): Premature EOL at scanline 2155 (got 0, expected 1728). Page 2 of /var/spool/asterisk/faxes/20040714-140202.tif: 2156 rows received 0 total bad rows 0 max consecutive bad rows Rx page end detected Changed from phase 5 to 3 Slow carrier up EOP: 2f EOP with final frame tag In state 5 Changed from phase 3 to 4 MCF: 8c HDLC
RE: [Asterisk-Users] Flag Bad PRI Channel
You can shut down the span in its entirety, or just exclude some channels in zaptel.conf. Steve -Original Message- From: Shawn Lawrence [mailto:[EMAIL PROTECTED] Sent: 16 July 2004 08:03 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Flag Bad PRI Channel Is there a way to flag a bad or noisy PRI channel in Asterisk so it will be skipped over? Shawn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD Issues
Your priority assignment will probably cause that, if you have 2 people at different priorities asterisk will only send to the other priority if the best 1 is busy. I'm guessing you really want a %age split? Whereby a is guaranteed 70% of the calls and b 30%? We used to achieve this on our old system by putting people into the queues multiple times and then having it round robin. We use the priorities to allow other agents to service calls rather than just leaving people hanging. For instance our network sales team can also service the server sales queue but at a lower priority, they only get the calls if all the agents in the server sales queue are busy. Steve -Original Message- From: Robert Jackson [mailto:[EMAIL PROTECTED] Sent: 14 July 2004 23:08 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ACD Issues That worked great! Thanks for the help. Any ideas on the uneven distribution problems? Right now the agent with the lowest agent number is getting 45% of the calls. She is going crazy! Just trying to figure out what I screwed up. Thanks, Robert Jackson Pro-Medical, Inc. -Original Message- From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 14, 2004 12:56 PM To: Robert Jackson Subject: Re: [Asterisk-Users] ACD Issues On 05:01 AM 7/14/2004, Robert Jackson wrote: 1) Our agents for the main call center are responsible to make calls when they have not already received an ACD call. However it seems that if they make an outbound call asterisk is still routing inbound calls to them. The ACD call beeps at them via the call waiting features then if the agent does not answer the ACD call it logs the agent out. I am just trying to figure out how I can tell the system that the extension is busy. Should I be using the new replacements to incominglimit? In your cisco phones, set call waiting to off. This way when you have a phone call in progress, asterisk will be aware your phone is busy and won't send the call, versus, sending a call, and not having the agent answer it. Otherwise, you have to use incoming limit because there is no way to use set/checkgroup with agents as they aren't handled as devices you can attach a setgroup/checkgroup to. Chris A. Icide 332 Valdez Ave. Half Moon Bay, CA 94019 650-712-8223 voice 212-400-1698 IP voice 650-712-8995 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD Issues
Sorry, my mistake, I thought you'd said you had assigned priorities, let me go back and search for your original posting Steve -Original Message- From: Robert Jackson [mailto:[EMAIL PROTECTED] Sent: 16 July 2004 14:10 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ACD Issues That would certainly make sense, but I am not sure how to set an Agent's priority. The only information that I have been able to find is setting a QUEUE_PRIO value when queuing the calls (New as of July 2004). Thanks, Robert Jackson -Original Message- From: Steve Hanselman [mailto:[EMAIL PROTECTED] Sent: Friday, July 16, 2004 3:59 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] ACD Issues Your priority assignment will probably cause that, if you have 2 people at different priorities asterisk will only send to the other priority if the best 1 is busy. I'm guessing you really want a %age split? Whereby a is guaranteed 70% of the calls and b 30%? We used to achieve this on our old system by putting people into the queues multiple times and then having it round robin. We use the priorities to allow other agents to service calls rather than just leaving people hanging. For instance our network sales team can also service the server sales queue but at a lower priority, they only get the calls if all the agents in the server sales queue are busy. Steve -Original Message- From: Robert Jackson [mailto:[EMAIL PROTECTED] Sent: 14 July 2004 23:08 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ACD Issues That worked great! Thanks for the help. Any ideas on the uneven distribution problems? Right now the agent with the lowest agent number is getting 45% of the calls. She is going crazy! Just trying to figure out what I screwed up. Thanks, Robert Jackson Pro-Medical, Inc. -Original Message- From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 14, 2004 12:56 PM To: Robert Jackson Subject: Re: [Asterisk-Users] ACD Issues On 05:01 AM 7/14/2004, Robert Jackson wrote: 1) Our agents for the main call center are responsible to make calls when they have not already received an ACD call. However it seems that if they make an outbound call asterisk is still routing inbound calls to them. The ACD call beeps at them via the call waiting features then if the agent does not answer the ACD call it logs the agent out. I am just trying to figure out how I can tell the system that the extension is busy. Should I be using the new replacements to incominglimit? In your cisco phones, set call waiting to off. This way when you have a phone call in progress, asterisk will be aware your phone is busy and won't send the call, versus, sending a call, and not having the agent answer it. Otherwise, you have to use incoming limit because there is no way to use set/checkgroup with agents as they aren't handled as devices you can attach a setgroup/checkgroup to. Chris A. Icide 332 Valdez Ave. Half Moon Bay, CA 94019 650-712-8223 voice 212-400-1698 IP voice 650-712-8995 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/as terisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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
RE: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spa ndsp fails to decode
Sorry to bore you more with the clock issue, but have you check /proc/zaptel/span to make sure it's not missing interrupts? There's also an option to record the audio for the fax, you could listen to that vs a recorded file that will receive correctly on a fax machine and see whether there is an obvious difference? (Good luck, that'll be really scraping the barrel!!) Does it matter where you're faxing from? (I'm wondering whether there's an issue with a specific machine? (You can use http://www.tpc.int to send test faxes, you'll get some extra info if a fax fails as to why in the transmission report). Can you successfully send faxes out from your system, what do they look like at the remote end? Steve -Original Message- From: Stephen J. Wilcox [mailto:[EMAIL PROTECTED] Sent: 13 July 2004 18:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spandsp fails to decode Okay having taken in some suggestions and googled this topic to death I'm still stuck - anyone got any ideas? To recap, the faxes are coming in via a digium E1 card but failing to train properly or if they manage it sending a garbled and very truncated fax. A number of folks have suggested clock sync issues.. my zaptel.conf is set to use the PRI as primary clock, i have no evidence of issues altho dont know how to check (other than the call quality is fine, no clicks, no pri down/ups). What can i try? Steve On Mon, 12 Jul 2004, Stephen J. Wilcox wrote: Hi, I just sent this to Steve Underwood, but then found a bunch of posts on the mailing list about similar issues.. does anyone have the fix? I'm running asterisk CVS-HEAD-06/28/04-18:13:13, spandsp 0.0.1k, libtif 3.5.7 one thing i just noticed is that calls come in with format '72' which is G711A-law or LinearPCM.. it uses PCM for the call, i assume this is ok the results of RxFAX vary, it sometimes saves the file in which case i get errors: Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 0 (got 2383, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 1 (x 137). and the resulting tif looks to be only a few rows long or more commonly it just fails entirely.. i paste the output below so you can see. is there anything obvious i'm doign wrong here? TIA! Steve. -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/faxes/20040712-183339.tif) in new stack Changed from phase 0 to 1 Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 31 37 31 31 36 35 34 35 34 38 30 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: DCS: 83 00 86 90 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 5ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1699.90 (64) Training error 56.874846 Training succeeded (constellation mismatch 44.212022) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 14 FTT: 44 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.33 (64) Training error 51.989152 Training succeeded (constellation mismatch 37.988826) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 15 FTT: 44 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.32 (64) Training error 60.898646 Training succeeded (constellation mismatch 46.138793) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 17 FTT: 44 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1795.61 (4) Fast carrier down Fast carrier up Coarse carrier frequency 1789.60 (4) Fast carrier down -- Channel 0/1, span 1 got hangup -- Hungup 'Zap/1-1' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] Dell 6450 / TE405p
If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
Do the modules load (lsmod). Are the interrupts assigned and unique (cat /proc/interrupts) -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going to try that one as well. Thanks. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:01 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 6450 / TE405p
What have you got in /proc/pci? Do you have to do anything funny on the Dell to tell it that a card is there? Maybe it has some kind of health monitoring that you can switch off? -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:36 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p No, the modules don't load (no such device) and it doesn't show up on the interrupts. I think it's more basic than that - on the 6450 each slot has a led indicating that there is a card present. Whatever slot I put the card in on the 6450, it flashes on boot, and then goes out. If I put *any* of the other PCI cards I have (remote access / avm isdn / eicon isdn / network etc) then the led goes on and stays on. It's as if the 6450 doesn't like the TE405P and disables the slot. On the 4400, the led goes on and stays on, and the TE405P flashes red on each channel. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:16 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Do the modules load (lsmod). Are the interrupts assigned and unique (cat /proc/interrupts) -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going to try that one as well. Thanks. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:01 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p If the card plugs into the slot then it's not a power (3.3v/5v) issue as the cards are physically different. -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dell 6450 / TE405p I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz) I cannot get the card working in any of the slots. When I power the server up, the digium card blinks red for a second, and the dell pci light goes out. However, if I plug it into a standard dell dimension, I get the 4 channels flashing red when I power the machine up. My understanding of the pci bus is (obviously) limited - why doesn't the card work in the server ? Should I have bought the te410p instead ? Help! Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies
RE: [Asterisk-Users] Dell 6450 / TE405p
Well, you should see an entry like this: Bus 0, device 11, function 0: Communication controller: PCI device 10ee:0314 (Xilinx Corporation) (rev 1). IRQ 5. Master Capable. Latency=64. Non-prefetchable 32 bit memory at 0xda001000 [0xda00107f]. Any curious messages in dmesg when the machine is booted, any settings in the bios related to PCI? At least from this point you can discount any zaptel issues as this shows regardless of whether zaptel is loaded or not. Steve -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 15:30 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p I've attached the /proc/pci below, but I think it's hardware related, not os - the dell does not seem to recognise that there is a card in the slot. Or any slot I put it in :( Thanks for the help, though. Julian. [EMAIL PROTECTED] root]# cat /proc/pci PCI devices found: Bus 0, device 0, function 0: Host bridge: ServerWorks CNB20HE Host Bridge (rev 33). Master Capable. Latency=32. Bus 0, device 0, function 1: Host bridge: ServerWorks CNB20HE Host Bridge (#2) (rev 1). Master Capable. Latency=32. Bus 0, device 0, function 2: Host bridge: ServerWorks CNB20HE Host Bridge (rev 0). Master Capable. Latency=32. Bus 0, device 0, function 3: Host bridge: ServerWorks CNB20HE Host Bridge (#2) (rev 0). Master Capable. Latency=32. Bus 0, device 4, function 0: VGA compatible controller: ATI Technologies Inc 3D Rage IIC (rev 122). Master Capable. Latency=32. Min Gnt=8. Prefetchable 32 bit memory at 0xfc00 [0xfcff]. I/O at 0xec00 [0xecff]. Non-prefetchable 32 bit memory at 0xfbeff000 [0xfbef]. Bus 0, device 8, function 0: Ethernet controller: Intel Corp. 82557/8/9 [Ethernet Pro 100] (rev 8). IRQ 26. Master Capable. Latency=32. Min Gnt=8.Max Lat=56. Non-prefetchable 32 bit memory at 0xfbefe000 [0xfbefefff]. I/O at 0xe8c0 [0xe8ff]. Non-prefetchable 32 bit memory at 0xfbd0 [0xfbdf]. Bus 0, device 15, function 0: ISA bridge: ServerWorks OSB4 South Bridge (rev 80). Bus 0, device 15, function 1: IDE interface: ServerWorks OSB4 IDE Controller (rev 0). Master Capable. Latency=64. I/O at 0x8b0 [0x8bf]. Bus 3, device 9, function 0: PCI bridge: Intel Corp. 21154 PCI-to-PCI Bridge (rev 0). Master Capable. Latency=32. Min Gnt=6. Bus 4, device 0, function 0: PCI bridge: Intel Corp. 21154 PCI-to-PCI Bridge (#2) (rev 0). Master Capable. Latency=32. Min Gnt=6. Bus 4, device 1, function 0: SCSI storage controller: QLogic Corp. ISP12160 Dual Channel Ultra3 SCSI Proc essor (rev 6). IRQ 27. Master Capable. Latency=32. Min Gnt=64. I/O at 0xcc00 [0xccff]. Non-prefetchable 32 bit memory at 0xfaaff000 [0xfaaf]. Bus 5, device 0, function 0: RAID bus controller: American Megatrends Inc. MegaRAID (rev 32). IRQ 23. Master Capable. Latency=32. Prefetchable 32 bit memory at 0xf000 [0xf7ff]. [EMAIL PROTECTED] root]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 14:14 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p What have you got in /proc/pci? Do you have to do anything funny on the Dell to tell it that a card is there? Maybe it has some kind of health monitoring that you can switch off? -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:36 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p No, the modules don't load (no such device) and it doesn't show up on the interrupts. I think it's more basic than that - on the 6450 each slot has a led indicating that there is a card present. Whatever slot I put the card in on the 6450, it flashes on boot, and then goes out. If I put *any* of the other PCI cards I have (remote access / avm isdn / eicon isdn / network etc) then the led goes on and stays on. It's as if the 6450 doesn't like the TE405P and disables the slot. On the 4400, the led goes on and stays on, and the TE405P flashes red on each channel. Julian. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 09 July 2004 13:16 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Do the modules load (lsmod). Are the interrupts assigned and unique (cat /proc/interrupts) -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 13:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dell 6450 / TE405p Yeah, that's what I thought. Some more developments - I've just plugged it into a Dell 4400 server, worked first time. So what's with the 6450 ? I've got another spare 6450 kicking around somewhere - I'm going
RE: [Asterisk-Users] E1 config help and guidance
Doesn't matter which span you use, but the Telco span should be set as clock 1 and CPE, the Meridian span should be set as clock 0 and NET. Use only the channels you have been assigned on the Telco end, you as many channels as you want on the Meridian end. Take a look to see the order that inbound calls come from the Telco and outbound calls come from the Meridian (just place 2 calls sequentially with pri debug span x), then make sure you allocate the channels accordingly (there's an option in the Group for zaptel for top down etc down with the character that precedes the group number). That's about it. Steve -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: 09 July 2004 19:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E1 config help and guidance I've googled / voip-info'd / searched until my eyes are blurry, but couldn't see the info I was looking for. I've turned here for help! Asterisk CVS head (9/7/04) Fedora Core 2 (updated to 2.6.6 kernel) DE405P (jumpers set to E1) I want to put asterisk in the middle of our current pbx (Meridian Option11) Currently the meridian has a 2MB pri EuroISDN card linked via a rj-45 into a euroISDN bearer. This bearer only has 10 channels activated (out of the 30). Obviously, this works - handsets make external calls. What I wanted to do was to add * to the mix, in the middle so that it can intercept inbound / outbound calls and do what it needs to do, as well as providing all the extra functionality that this wonderful product provides. In order to achieve this, I assumed that I needed to take rj45 from the bearer box and plug that into span 2, and take a cable from span 1 into the bearer box. My problem (and blurry eyes) come from not understanding the various protocols to assign to each span. I want the meridian to think that it's still plugged into the EuroISDN bearer. So span 2 should be set up as a EuroISDN link ? What should span 1 be set up as ? What channels should be configured ? Any guidance (I'm not looking for the solution (would be nice!) but for pointers in the right direction). I have previously been able to set up asterisk using the x100p and graduated to BRI isdn. I just got the 405 today and wanted to play! Thanks in advance. Julian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Penalty in queues.conf
It's so you can have agents that are less likely to take calls (e.g. imagine a sales queue, you'd have the sales people with no penalty, you might have the receptionists with a penalty of 1 and us propeller heads in technical support with a penalty of 2). The technical support people would only be offered a call from the sales queue if all the sales people and the receptionists were busy. Steve -Original Message- From: Isamar Maia [mailto:[EMAIL PROTECTED] Sent: 04 July 2004 10:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Penalty in queues.conf I have already read explanation about that in some places but I don't have still a clear image about the meaning of Penalty parameter inside of queues.conf What means that? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sending SABME continuosly. Urgent help needed!
Hi David,Sorry for the very late reply to this, but it looks as though you're in PRI_NET rather than PRI_CPE, I believe that the network sends to a SABME and you reply with an unnumbered response, Q921 is then established?RegardsSteveDavid Morillo wrote:Hi, I'm trying to install an E1 PRI, and I need it working byMonday, but although everything seems ok, I get no response to calls. When I make a pri extense debug on span 1, I repeatedly get thefollowing: Sending Set Asynchronous Balanced Mode Extended [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 999 EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balancedmode extended] 0 bytes of data. And nothing else. When making a call to that E1, I see the message D-Channel onspan 1 up 4 times, and then a Informational frame, with TEI:000 EA:1 and anything else with zero (13 bytes of data). Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) Channel ID (len= 5) [Ext: 1 IntID: Implicit, PRI Spare: 0,Exclusive Dchan:0 Chan Sel Reserved Ext: 1 Coding:0 Number Specified Channel Type: 3 Ext: 1 Spare: o Resetting Inidicated Channel (0) ] Then D-Channel on span 1 downn, and finally, after a while: (...) Warning[11276]: chan_zap.c:5993 zt_pri_error: PRI: Read on46 failed: Unknown error 500 (...) Notice[11276]: chan_zap.c:6708 pri_dchannel: PRI gotevent: 8 on span 1 I think I have Asterisk stable version 1.0, CVS updated today Can anyone help me? Please! :S Zaptel.conf -- span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone = es defaultzone=es Zapata.conf: -- [channels] language=es context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no jitterbuffers=4 group = 1 channel = 1-15,17-31 I have also tried with span=1,0,0,ccs,hdb3 immediate = yes The line has not CRC activated (I have asked) Thanks! Steve Hanselman Brendata (UK) Ltd Tel: +44 (0)1268 466111 Fax: +44 (0)870 1387283 Mob: +44 (0)7973 750993 The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk
RE: [Asterisk-Users] Cisco 7960G and *
It'll work, either as a SIP phone with the SIP image, or as skinny using wither chan_sccp or chan_skinny (check the wiki). Steve -Original Message- From: Matt Davies | MattDavies.Net [mailto:[EMAIL PROTECTED] Sent: 02 July 2004 15:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960G and * I have been doing so much reading on phones lately that I have completely lost track of some things. I seem to remember that there was one series of Cisco IP phones that required Cisco's call manager. Does anyone know if the 7960 will work with Asterisk or does it require call manager? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Providing Telewest in the UK with per extens ion outbound callerID
Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, SH Is anybody in the UK using Telewest as a PRI Telco provider? SH Are you sending them caller ID? Just a quick point of clarification before commenting further, do you wish to make calls via Telewest's network and send the CLI of your own DDI number range or do you wish to send other numbers as your CLI? If you are seeking toachieve the latter, what sort of numbers do you wish to propagate asthe CLI for your calls? Regards Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve Hanselman Sent: 30 June 2004 18:57 To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain that some of our customers use Telewest and they provide them with caller ID) So, does anybody do this, and if so, what did you have to request from them in order to enable it, and what do you provide to them (how many digits and in what format). Regards Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk
RE: [Asterisk-Users] Providing Telewest in the UK with per extens ion outbound callerID
When the original PBX was installed we asked them to override the CLI and provide a single number as the PBX couldn't provide the DDI number, now the contact at Telewest believes it's somewhere between illegal and impossible to provide DDI numbers to the outside world. -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 10:13 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, Telewest should already allow the CLI transmission of your DDI range, without further datafill changes. If it doesn't work you should check that you are sending the appropriate number of digits. Try sending: -3 digit CLI -the whole number (minus the leading zero) If the comments above don't help please post a trace of an outgoing call and detail the number, if any, that is presented to theCalled Party. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve Hanselman Sent: 01 July 2004 09:57 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve, SH Is anybody in the UK using Telewest as a PRI Telco provider? SH Are you sending them caller ID? Just a quick point of clarification before commenting further, do you wish to make calls via Telewest's network and send the CLI of your own DDI number range or do you wish to send other numbers as your CLI? If you are seeking toachieve the latter, what sort of numbers do you wish to propagate asthe CLI for your calls? Regards Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Steve Hanselman Sent: 30 June 2004 18:57 To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain that some of our customers use Telewest and they provide them with caller ID) So, does anybody do this, and if so, what did you have to request from them in order to enable it, and what do you provide to them (how many digits and in what format). Regards Steve 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 The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk
[Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID
Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) Telewest can't do this. (this is rubbish, I'm certain that some of our customers use Telewest and they provide them with caller ID) So, does anybody do this, and if so, what did you have to request from them in order to enable it, and what do you provide to them (how many digits and in what format). Regards Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk
RE: [Asterisk-Users] How to test E1 interfacing?
1. Can't answer, sorry! 2. Yes, although you have to be clear about what you're proving, you're proving that the cards work, not that asterisk is correctly configured and will eventually talk exactly correctly to the entity that you'll connect it to be a PBX or a telco. You'd configure one as net and the other as cpe, and you could then back to back them. We did this. Steve (based in the UK) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 29 June 2004 10:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How to test E1 interfacing? Hi, I have a project coming up which will need to interface Asterisk to E1 trunks in the UK. I have a couple of questions which I hope someone can answer, or give me some pointers: 1. If I want two E1 trunks, is there anything to choose, performance-wise, between using two ports on a single TE405P, and using two E100P cards? 2. How can I test the E1 operation in the lab, which doesn't have an E1 line available, before taking the unit to the installation site? Can I run two Asterisks back-to-back? Can I run one port into another on a single TE405P? I couldn't find anything on the above in the Wiki; if I didn't look hard enough, please tell me where I missed. Thanks. Any advice would be gratefully received! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue hold time in seconds
I'm going to modify the queue announcements to allow for rounded seconds (e.g. we want to know to the tens of seconds. E.g. Average wait 1 minute 20 seconds). I'm going to add the optional announce of seconds to the queue config and a rounding factor (e.g. 10 in our case). The following parameters will be added Queue-announce-seconds (default is off) Queue-seconds (default will be an as yet unrecorded "queue-seconds") Queue-rounding-seconds (default will be 10) Have I missed anything? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat 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) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk
RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk
Ok, may have got to the bottom of this. The te405P was sharing interrupts with a via82cxxx audio chip, that was being used to generate the music on hold for our existing pbx. Having now shut down the music on hold the system will now run correctly with telewest as the master and the gdk as the slave. This came about due to the high number of missed IRQ's, it seems as though the 2.4.22 via driver is none to selective about what it grabs. I'll leave it running over the weekend and see how it holds up. Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 20 June 2004 18:23 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Hi Steve, How bizarre, your config doesn't look like it should work too well and certainly doesn't look like it should improve your fax problem! I assume that pri_cpe is set for span1 and pri_net for span2 ? Maybe, just maybe, Telewest reconfigured your PRI to look for clocking from your CPE but I've not encountered that configuration before. Try and leave the current config up for as long as you can before you return it to production mode and watch the CLI/logs to see if you get any sporadic clock slips (within a couple of hours I'd expect at least one episode of messages). One last thought, did you bounce the system after you made the changes to zaptel.conf or did you just reload * ? HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Hanselman Sent: 20 June 2004 16:48 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk They look odd to me for sure, I'm certain (99.9%) that Telewest would not clock off of us, but as far as I can see, the current config (which allows the GDK to send and receive faxes) has no external clocking??? Here's the current config: span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 Here's the original config which I took to mean that Telewest provided clock and span2 clocked off span1? span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 (Span1 goes to Telewest - our Telco provider, span2 goes to our current PBX, a GDK-186) Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 20 June 2004 16:34 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Steve, your config description (timing) does sound odd. Could you re-post your revised config files? Thanks Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Hanselman Sent: 20 June 2004 16:18 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk I've changed the zaptel.conf to set both as internal, and it now seems to work, which is backwards to the config I thought it should have been, I would have thought that the Telewest PRI would have been 1 and the GDK 0? Can somebody confirm that this is the correct definition for timing, if it's a +ve number then it's external clocking with the lowest 1 being the highest priority. All spans are clocked relative to the external source and the external source selected is the lowest priority number that is currently being clocked? I'll experiment some more. -Original Message- From: Yifang Dai [mailto:[EMAIL PROTECTED] Sent: 19 June 2004 03:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Let's try again, missed a line in the last reply... On Fri, Jun 18, 2004 at 09:55:24PM -0400, Yifang Dai wrote: On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote: LG GDK-186 PBX --PRI--- TE405P/Asterisk ---PRI--- Telewest (Telco provider) --snip--- Any ideas on where to start? This is most likely to be a timing issue. You need to make sure your asterisk is get timing from your telco, and provide timing for you gdk pbx. /etc/asterisk/zaptel.conf is the place to look. -- Yifang Dai Senior System Administrator Yarde Metals Inc 45 Newell St, Southington, CT 06010 (Phone) 860-406-6107; (FAX) 860-406-4060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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
