Re: [asterisk-users] Help setting IAX variables.
On Tuesday 08 September 2009 00:14:53 Asterisk User wrote: Thanks Tilghman for your quick reply. I know that we should set variables through IAXVAR on source server to access them on Destination server. I just wanted to know the reverse case, where IAX channel variables set on destination server are accessible on Source server or not. Thanks again for your inputs. They are not. IAXVARs are only sent during the NEW, which is a one-way packet. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Help setting IAX variables.
On Monday 07 September 2009 05:55:12 Asterisk User wrote: I am new to Asterisk and want to perform following on my test project. I have 2 Asterisk servers SA and SB and an IAX trunk from SA to SB. Now I set some variables on SB in the same context where an IAX call lands. My question is , is it possible to access these variables in dialplan of SA? If yes then how? I know about IAXVAR application where variables set in source server of IAX channel can be access from destination server... For the variable to be accessed on the destination server, you must explicitly set the IAXVAR variable on the source server. This method does not merely access arbitrary variables on the source server but only variables which have been sent through this mechanism. Just as a version note, you need to be running 1.6.0 or higher to get the IAXVAR mechanism. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Help setting IAX variables.
Thanks Tilghman for your quick reply. I know that we should set variables through IAXVAR on source server to access them on Destination server. I just wanted to know the reverse case, where IAX channel variables set on destination server are accessible on Source server or not. Thanks again for your inputs. --- Asterisk user ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Help with IAX Trunk
thanks for reply. I've same setup with siml. incoming calls 10-12 it works fine but some time my machies get hang and gives same IAX max data space error. thanks On 6/27/07, Jared Smith [EMAIL PROTECTED] wrote: On 6/27/07, Arun Kumar [EMAIL PROTECTED] wrote: so , how much bandwidth I need for 30 simul. calls ? If you're using IAX2 trunking, the bandwidth requirements will be much less than if you're not using IAX2 trunking. Make sure you have trunk=yes in the peer definition in iax.conf. Off the top of my head (without actually running the numbers), I would guess that 30 simultaneous calls using the g.729 codec and using IAX2 trunking would take less than 512kbit/sec in each direction. to support 30 calls over IAX2 do I've to change some setting during compile time or not ? No, just make sure you have a suitable timing source (Digium card, ztdummy, etc.) for the IAX2 trunk. -Jared ___ --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
Re: [asterisk-users] Help with IAX Trunk
On 6/27/07, Arun Kumar [EMAIL PROTECTED] wrote: so , how much bandwidth I need for 30 simul. calls ? If you're using IAX2 trunking, the bandwidth requirements will be much less than if you're not using IAX2 trunking. Make sure you have trunk=yes in the peer definition in iax.conf. Off the top of my head (without actually running the numbers), I would guess that 30 simultaneous calls using the g.729 codec and using IAX2 trunking would take less than 512kbit/sec in each direction. to support 30 calls over IAX2 do I've to change some setting during compile time or not ? No, just make sure you have a suitable timing source (Digium card, ztdummy, etc.) for the IAX2 trunk. -Jared ___ --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] Help with IAX
I finally got a chance to investigate this further. The fundamental
problem seemed to be that I was using a context name of iax-trunk.
When I changed this to intrunk, it worked. What are the rules for
context names? Was it the length or the special character that caused me
problems?
Thanks for your help.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ruggles
Sent: Wednesday, May 30, 2007 3:10 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Help with IAX
Well this may not feel like progress, but it is. You no longer have an
authentication issue, you now have a routing issue. Could you attach a
copy of the extension.conf file on 192.168.253.21?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Malcom
Kemp
Sent: Wednesday, May 30, 2007 3:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Help with IAX
From 192.168.253.20:
*CLI [May 30 14:39:03] NOTICE[6051]: chan_iax2.c:7146
socket_process: Rejected connect attempt from 192.168.253.21, request
'[EMAIL PROTECTED]' does not exist
From 192.168.253.21:
[May 30 14:39:03] WARNING[28640]: chan_iax2.c:6959
socket_process: Call rejected by 192.168.253.20: No such
context/extension
I even changed the extension to take the pattern off:
exten = 205,1,Macro(voicemail,${E205})
+
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Re: [asterisk-users] Help with IAX
Can you send IAX.conf of both the systems Regards, Sanjay Rajdev - Original Message - From: Malcom Kemp [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 30 May, 2007 9:05:34 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Help with IAX ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX
The IAX.CONF are both the sample configs, with the addition of the two pieces that I added and posted in the email. But here they are -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sanjay Rajdev Sent: Wednesday, May 30, 2007 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help with IAX Can you send IAX.conf of both the systems Regards, Sanjay Rajdev - Original Message - From: Malcom Kemp [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 30 May, 2007 9:05:34 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Help with IAX ___ --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 + This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415 or visit www.tecinfo.net + 192.168.253.21.iax.conf Description: 192.168.253.21.iax.conf 192.168.253.20.iax.conf Description: 192.168.253.20.iax.conf ___ --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] Help with IAX
In both iax.conf files change [iax-trunk] to [tecinfo] the [name] in iax.conf is what is looked for when a connection is established and you're telling it to connect with tecinfo on the username= line HTH Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --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] Help with IAX
I did that. Got same results. I also changed the extensions on the .21 box to: exten = 205,1,Dial(IAX2/tecinfo:[EMAIL PROTECTED]/[EMAIL PROTECTED]) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Wednesday, May 30, 2007 12:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Help with IAX In both iax.conf files change [iax-trunk] to [tecinfo] the [name] in iax.conf is what is looked for when a connection is established and you're telling it to connect with tecinfo on the username= line HTH Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] + This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415 or visit www.tecinfo.net +___ --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] Help with IAX
I have made some changed to your config:
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/ mailto:IAX2/192.168.253.20/[EMAIL PROTECTED]
tecinfo1/205)
iax.conf from 192.168.253.21:
[tecinfo1]
type=peer
username=tecinfo2
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[tecinfo2]
type=user
context=iax-trunk
username=tecinfo
secret=secret
host=192.168.253.21
Try this and respond with error messages (if any) from both systems
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
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RE: [asterisk-users] Help with IAX
From 192.168.253.20:
*CLI [May 30 14:39:03] NOTICE[6051]: chan_iax2.c:7146 socket_process:
Rejected connect attempt from 192.168.253.21, request '[EMAIL PROTECTED]'
does not exist
From 192.168.253.21:
[May 30 14:39:03] WARNING[28640]: chan_iax2.c:6959 socket_process: Call
rejected by 192.168.253.20: No such context/extension
I even changed the extension to take the pattern off:
exten = 205,1,Macro(voicemail,${E205})
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ruggles
Sent: Wednesday, May 30, 2007 1:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Help with IAX
I have made some changed to your config:
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/tecinfo1/205
mailto:IAX2/192.168.253.20/[EMAIL PROTECTED] )
iax.conf from 192.168.253.21:
[tecinfo1]
type=peer
username=tecinfo2
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[tecinfo2]
type=user
context=iax-trunk
username=tecinfo
secret=secret
host=192.168.253.21
Try this and respond with error messages (if any) from both systems
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
+
This e-mail was checked by the TecInfo Content Scanning Service for potentially
harmful content, such as viruses or Spam For more information, call
800.863.5415 or visit www.tecinfo.net
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RE: [asterisk-users] Help with IAX
Got the same thing when I removed the username from the
192.168.253.20.iax.conf...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ruggles
Sent: Wednesday, May 30, 2007 1:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: FW: [asterisk-users] Help with IAX
(missed one thing)
I have made some changed to your config:
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/tecinfo1/205
mailto:IAX2/192.168.253.20/[EMAIL PROTECTED] )
iax.conf from 192.168.253.21:
[tecinfo1]
type=peer
username=tecinfo2
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[tecinfo2]
type=user
context=iax-trunk
secret=secret
host=192.168.253.21
Try this and respond with error messages (if any) from both systems
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
+
This e-mail was checked by the TecInfo Content Scanning Service for potentially
harmful content, such as viruses or Spam For more information, call
800.863.5415 or visit www.tecinfo.net
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RE: [asterisk-users] Help with IAX
Well this may not feel like progress, but it is. You no longer have an
authentication issue, you now have a routing issue. Could you attach a copy
of the extension.conf file on 192.168.253.21?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Malcom Kemp
Sent: Wednesday, May 30, 2007 3:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Help with IAX
From 192.168.253.20:
*CLI [May 30 14:39:03] NOTICE[6051]: chan_iax2.c:7146 socket_process:
Rejected connect attempt from 192.168.253.21, request '[EMAIL PROTECTED]' does
not exist
From 192.168.253.21:
[May 30 14:39:03] WARNING[28640]: chan_iax2.c:6959 socket_process: Call
rejected by 192.168.253.20: No such context/extension
I even changed the extension to take the pattern off:
exten = 205,1,Macro(voicemail,${E205})
___
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RE: [asterisk-users] Help with IAX Trunk
Yes. That was the solution. Not sure why that 'r' is there in the first place David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Saturday, December 02, 2006 11:57 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help with IAX Trunk On 09:48, Sat 02 Dec 06, Dave Morrow wrote: H.interesting thought. Not sure how to do it though... I found this this morning. I think it might be the answer I seek http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum =2 Probably yeah. The r option in the dial command will not pass early media but instead generates it's own. I find the r flag for dial and queue the wrong thing to do. In dial it will disable stuff like 'this call will cost you 300 euro a minute and that's something I really wanna hear. In queue() it will kill the periodic announcements. annoying as well. I removed them from everywhere in my extensions.conf and my system is much more usable. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IAX Trunk
On 11:02, Mon 04 Dec 06, Dave Morrow wrote: Yes. That was the solution. Not sure why that 'r' is there in the first place It's there to provide 'ringing' indication on links that do not provide it. (voip-voip connections or voip-pri|pstn connections without early-media passthru) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IAX Trunk
Dave Morrow wrote: My long distance provider requires that a billing code be entered after dialing a long distance call. From the directly attached Asterisk server, these calls work when the user enters their PIN after dialing. From the second server (connected via an IAX trunk), I never get the tone to enter the long distance PIN..all I get is a steady ringtone. Instead of having the user enter the billing code, maybe you could program it to be sent via the dial plan? Or, is the code different each time? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Help with IAX Trunk
Unfortunately, the codes are private for the individual. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, December 02, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with IAX Trunk Dave Morrow wrote: My long distance provider requires that a billing code be entered after dialing a long distance call. From the directly attached Asterisk server, these calls work when the user enters their PIN after dialing. From the second server (connected via an IAX trunk), I never get the tone to enter the long distance PIN..all I get is a steady ringtone. Instead of having the user enter the billing code, maybe you could program it to be sent via the dial plan? Or, is the code different each time? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IAX Trunk
Dave Morrow wrote: Unfortunately, the codes are private for the individual. Then I would suggest that you prompt the user for that code, before the actual dial. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Help with IAX Trunk
H.interesting thought. Not sure how to do it though... I found this this morning. I think it might be the answer I seek http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum=2 David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, December 02, 2006 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with IAX Trunk Dave Morrow wrote: Unfortunately, the codes are private for the individual. Then I would suggest that you prompt the user for that code, before the actual dial. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IAX Trunk
On 09:48, Sat 02 Dec 06, Dave Morrow wrote: H.interesting thought. Not sure how to do it though... I found this this morning. I think it might be the answer I seek http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum=2 Probably yeah. The r option in the dial command will not pass early media but instead generates it's own. I find the r flag for dial and queue the wrong thing to do. In dial it will disable stuff like 'this call will cost you 300 euro a minute and that's something I really wanna hear. In queue() it will kill the periodic announcements. annoying as well. I removed them from everywhere in my extensions.conf and my system is much more usable. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage.
Alexey Ostrovsky wrote: Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) System details: - Linux Slackware 9.1 kernel 2.4.6 - CPU PIII 800 - RAM 500 Mb - motherboard Asus TUSL-2c - hard drive IDE - Asterisk last cvs update - 3 TDM400 cards - codec g729 voiceage Help please. May be somebody have same problem. Thank you. Some more information about that problem. Our system was tested with various codecs and protocols. The results are as follows: All calls one after another between two asterisk system (one in UA another in US). IAX2/ulaw-5 calls20-25 min durationAll OK IAX2/gsm - after 10 min - one-way audio - channels up 30 min - OK 5 min - one-way audio - channels up 20 min - one-way audio - channels up 55 min - one-way audio - channels up 30 min - OK IAX2/g729 3 min - one-way audio - channels up 15 min - one-way audio - channels up 65 min - one-way audio - channels up 30 min - OK 50 min - OK IAX/ilbc 30 min - OK 35 min - OK 10 min - noise 30 min - OK 30 min - OK SIP/gsm30 min - OK 30 min - OK 10 min - one-way audio - channels up 30 min - OK 7 min - one-way audio - channels up SIP/g729 30 min - one-way audio - channels up 25 min - OK 10 min - one-way audio - channels up 30 min - OK 12 min - one-way audio - channels up So, as you see in my case ulaw is better than others. But it is too much for my net link. I have 256 kbit DSL now. For now I am using ilbc fore iax2. It doesn't have a one-way audio problem (for now) on our system. May be this will help to understand what is wrong. Thanking you in advance, -- Best regards, Alexey Ostrovsky ___ 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] Help with IAX , voice Distortion or Breakage.
On Mon, 2004-05-24 at 08:27, Alexey Ostrovsky wrote: Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) System details: - Linux Slackware 9.1 kernel 2.4.6 - CPU PIII 800 - RAM 500 Mb A sysadmin that is so inaccurate as to say 500Mb instead of the obvious 512 Mb? - motherboard Asus TUSL-2c - hard drive IDE - Asterisk last cvs update - 3 TDM400 cards - codec g729 voiceage Help please. May be somebody have same problem. Can you verify you are using g729. Also what kind of network are you doing the IAX call over? What else is on the network at the time of the call. Most of what you say sounds a lot like you are running short of bandwidth. Sounds like the classic problem of running out of the buffered content and decoding what you have when you get it. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Help with IAX , voice Distortion or Breakage.
Steven Critchfield wrote: On Mon, 2004-05-24 at 08:27, Alexey Ostrovsky wrote: Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) System details: - Linux Slackware 9.1 kernel 2.4.6 - CPU PIII 800 - RAM 500 Mb A sysadmin that is so inaccurate as to say 500Mb instead of the obvious 512 Mb? I think there is no differents for my problem. Thank you a lot. - motherboard Asus TUSL-2c - hard drive IDE - Asterisk last cvs update - 3 TDM400 cards - codec g729 voiceage Help please. May be somebody have same problem. Can you verify you are using g729. Also what kind of network are you doing the IAX call over? What else is on the network at the time of the call. Most of what you say sounds a lot like you are running short of bandwidth. Sounds like the classic problem of running out of the buffered content and decoding what you have when you get it. Yes of course I am using g729b codec. I have SHDSL line with 128 bit/c bandwidth. Nothing else on the network in testing time. -- Best regards, Alexey Ostrovsky Sysadmin Ionidea UA ___ 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] Help with IAX , voice Distortion or Breakage.
We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) Without more technical data, its hard to guess. Might consider... 1. if you're using cisco phones, upgrade * to current Head or Stable on both ends of the iax link. The Stable code was just fixed on Friday. (There could be other phones/adapters that are impacted by the iax/gsm timestamp problems.) I've also noticed a fair number of other fixes that were just recently applied to Head. 2. if not cisco phones, check the config's on whatever phone you're using to ensure transmit silence is enabled. If you are using the xten soft phone, the parameter is in the Advanced Settings area. 3. Check to ensure all ethernet nic adapters (and associated switch ports) are running in full-duplex mode, etc. 4. If none of the above apply, it would be helpful to see a packet trace (using ethereal) at about the time the distorting/failure is occurring. ___ 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] Help with IAX , voice Distortion or Breakage.
On Mon, 2004-05-24 at 09:39, Alexey Ostrovsky wrote: Steven Critchfield wrote: On Mon, 2004-05-24 at 08:27, Alexey Ostrovsky wrote: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) - Asterisk last cvs update - 3 TDM400 cards - codec g729 voiceage Can you verify you are using g729. Also what kind of network are you doing the IAX call over? What else is on the network at the time of the call. Most of what you say sounds a lot like you are running short of bandwidth. Sounds like the classic problem of running out of the buffered content and decoding what you have when you get it. Yes of course I am using g729b codec. I have SHDSL line with 128 bit/c bandwidth. Nothing else on the network in testing time. I guess I should have been more specific in asking you to provide a actual call log so we can see for sure that asterisk choose to use the G729 codec. It also may shed light on the problem in case it is something else. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Help with IAX , voice Distortion or Breakage
Title: Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage I am having almost the exact same problem. I have the following setup: Debian Woody Kernel 2.4.18 CPU: P4 1.2GHz RAM: 1GB Asterisk Latest CVS 1 TDM400P card Codec GSM Ive been chasing down bandwidth issues, but have had no luck. We are still pursuing those issues. I just started configuring IAX, so I assumed it was related to my IAX configs. We just noticed this morning that SIP is having the same issue, but that it isnt as severe. We are sending our calls over the regular Internet, but that hasnt been a problem in the past. Besides the problem is too regular to be Internet related (unless it is at my service provider). Regards, John
Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage.
Rich Adamson wrote: We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) Without more technical data, its hard to guess. Might consider... 1. if you're using cisco phones, upgrade * to current Head or Stable on both ends of the iax link. The Stable code was just fixed on Friday. (There could be other phones/adapters that are impacted by the iax/gsm timestamp problems.) I've also noticed a fair number of other fixes that were just recently applied to Head. OK I will update asterisk. Yes we are using Cisco Phones. And simple phones connected to digium TDM cards. But problem was happend even with simple phones. 2. if not cisco phones, check the config's on whatever phone you're using to ensure transmit silence is enabled. If you are using the xten soft phone, the parameter is in the Advanced Settings area. 3. Check to ensure all ethernet nic adapters (and associated switch ports) are running in full-duplex mode, etc. 4. If none of the above apply, it would be helpful to see a packet trace (using ethereal) at about the time the distorting/failure is occurring. I have dump file, but it is about 3 Mb in archive. So, how I can send it to you? ___ 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 -- Best regards, Alexey Ostrovsky Sysadmin Ionidea UA ___ 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
