[Asterisk-Users] plainvoip - IAX2 call rejected
Is anybody using plainvoip provider with IAX2? They seem to support IAX but it rejects my calls. -- Executing Dial(SIP/11-cb98, IAX2/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] May 14 00:33:32 WARNING[26580]: chan_iax2.c:5553 socket_read: Call rejected by 66.199.240.2: No authority found My registration goes through OK. My dial plan: exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED] -- #Joseph ___ --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] Contract Work : On-site NYC
Hi, We require a technical person to do some on-site installation work for us in New York, must be proficient with Linux and Cisco. Regards, Sahil Gupta VoiceValley ___ --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] plainvoip - IAX2 call rejected
Is anybody using plainvoip provider with IAX2? They seem to support
IAX but it rejects my calls.
-- Executing Dial(SIP/11-cb98, IAX2/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
May 14 00:33:32 WARNING[26580]: chan_iax2.c:5553 socket_read: Call
rejected by 66.199.240.2: No authority found
My registration goes through OK.
My dial plan:
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]
Try
exten = _1NXXNXX,2,Dial,IAX2/plainvoip/${EXTEN}
or
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Benchev
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Re: [Asterisk-Users] [SOLVED] plainvoip - IAX2 call rejected
On Sun, 2006-05-14 at 09:58 +0300, Benchev wrote:
Is anybody using plainvoip provider with IAX2? They seem to support
IAX but it rejects my calls.
-- Executing Dial(SIP/11-cb98, IAX2/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
May 14 00:33:32 WARNING[26580]: chan_iax2.c:5553 socket_read: Call
rejected by 66.199.240.2: No authority found
My registration goes through OK.
My dial plan:
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]
Try
exten = _1NXXNXX,2,Dial,IAX2/plainvoip/${EXTEN}
or
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Benchev
Thanks for quick reply.
Yes, you are correct. It is working with:
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
--
#Joseph
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Re: [Asterisk-Users] Confused !
thanks for your replay, after i disallow all codec except g723 i also confused how a2billing is working then what i did , i removed all the codec from /usr/lib/astersik/module without codec_g723.so . then i saw in my log while user calling to my ivr access number a2b is looking for gms codec as all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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 -- Got my Gmail ? ___ --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] Bristuffed Asterisk: Hangup problems
On Friday, May 12, 2006 12:38 PM stoffell wrote: I have sent an email to junghanns.net about this, but haven't received an answer yet. If I do receive anything, I'll post it back to the thread. Friendly piece of advice: Call them! :-) Mail is most of the times not answered... Regards, JP ___ --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] Confused !
I'm not an authority but why don't you get some g729 codecs (10 or so) and use g729 all around. Not allowing for ADSL overheads you can calculate your own requirements on http://www.asteriskguru.com/tools/bandwidth_calculator.php (please double the results since each call is turned around to your service provider.) I would have thought it would be better if you could use reinvite to let your clients speak directly to your service providers. Someone who knows better ought to be able to tell if this would work. Your restriction is what the service provider allows. Most (that I've used) allow g729. I know it uses more bandwidth than g723 but nothing like G711 (ulaw or alaw) and from my experience, the quality is quite reasonable. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 11:27 Subject: Re: [Asterisk-Users] Confused ! thanks for your replay, after i disallow all codec except g723 i also confused how a2billing is working then what i did , i removed all the codec from /usr/lib/astersik/module without codec_g723.so . then i saw in my log while user calling to my ivr access number a2b is looking for gms codec as all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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 -- Got my Gmail ? ___ --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] Confused !
how to use reinvite in my asterisk setup ? thanks Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: I'm not an authority but why don't you get some g729 codecs (10 or so) and use g729 all around. Not allowing for ADSL overheads you can calculate your own requirements on http://www.asteriskguru.com/tools/bandwidth_calculator.php (please double the results since each call is turned around to your service provider.) I would have thought it would be better if you could use reinvite to let your clients speak directly to your service providers. Someone who knows better ought to be able to tell if this would work. Your restriction is what the service provider allows. Most (that I've used) allow g729. I know it uses more bandwidth than g723 but nothing like G711 (ulaw or alaw) and from my experience, the quality is quite reasonable. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 11:27 Subject: Re: [Asterisk-Users] Confused ! thanks for your replay, after i disallow all codec except g723 i also confused how a2billing is working then what i did , i removed all the codec from /usr/lib/astersik/module without codec_g723.so . then i saw in my log while user calling to my ivr access number a2b is looking for gms codec as all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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 -- Got my Gmail ? ___ --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 -- Got my Gmail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
[Asterisk-Users] IAX/SIP to germany with own callerid?
Hi, does someone knows a IAX2 provider that allows to freely set the USER_NUMBER to the german pstn? It's no problem if setting the NETWORK_NUMBER is restricted, i'm using this for call-forwarding with the correct number, not something illegal... (i'm not sure about the terminology here, but i think USER_NUMBER is callerid and NETWORK_NUMBERis ANI in the US, correct?) regards, Andreas _ Read the latest Hollywood gossip @ http://xtramsn.co.nz/entertainment ___ --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] plainvoip - IAX2 call rejected
Use this:
exten = _1NXXNXX,2,Dial,IAX2/username:[EMAIL PROTECTED]/${EXTEN}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Sunday, May 14, 2006 2:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] plainvoip - IAX2 call rejected
Is anybody using plainvoip provider with IAX2? They seem to support
IAX but it rejects my calls.
-- Executing Dial(SIP/11-cb98, IAX2/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
May 14 00:33:32 WARNING[26580]: chan_iax2.c:5553 socket_read: Call
rejected by 66.199.240.2: No authority found
My registration goes through OK.
My dial plan:
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]
--
#Joseph
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Re: [Asterisk-Users] Re: Call parking from legacy PBX over PRI??
On Friday 12 May 2006 17:38, Steven wrote:
Does anyone have a version that talks back during the transfer like Park()
does?
ParkAndAnnounce can do this, can you not specify the correct channel?
Also, I added a feature to ParkAndAnnounce which is now part of trunk which
gives the variable ${PARKEDAT} to the announced channel.
-A.
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Re: [Asterisk-Users] ATXFER
On Friday 12 May 2006 21:00, Josué Conti wrote: Eric, but I to continue using version 1.0.9 of asterisk, would have some solution? Yes, find a consultant to try and backport this feature to the version of Asterisk you desire. -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
Re: [Asterisk-Users] fc5 and link to sources?
If it's a question, the answer is YES. Cheers Fifou Damon Estep a écrit : Are dependencies resolved when using yum to install form the ATrpms http://www.voip-info.org/wiki/view/ATrpms --Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phi Fou Sent: Saturday, May 13, 2006 11:55 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] fc5 and link to sources? Why not installing Asterisk and zaptel from the atrpms repo. It works great even if you need to recompile... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: h323.conf and realtime
Problem solved there was a wrong name of h323.conf. (h323.conf.org) Working on weekends is bad )) Sunday, May 14, 2006, 5:37:13 PM, you wrote: Dears, Does anyone have success story with chan_h323's h323.conf storage in database. I have configuration from res_mysql.conf is working. # cat extconfig.conf [settings] h323.conf = mysql,ast,ast_config # database -- -- Dumping data for table `ast_config` -- Best regards, Vahrammailto:[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
[Asterisk-Users] [patch] fix for redirect manager action with BRIstuffed Asterisk
Hi, BRIstuff contains two bugs in its implementation of the Redirect manager action: 1. If the property ExtraUnqiueId is used, the Priority property is used to redirect the extra channel (instead of ExtraPriority) 2. If the property ExtraChannel is used, 0 is used to redirect the extra channel regardless of the Priority and ExtraPriority properties. A patch for manager.c is available at http://www.reucon.net/~srt/bristuff_redirect.patch as a result to a bug filed against Asterisk-Java at http://jira.reucon.org/browse/AJ-34 I've sent a notice to kpj. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] signature.asc Description: OpenPGP digital signature ___ --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] Confused !
1. In the extension definition, insert canreinvite=yes for each of your clients. 2. In the trunk definition, insert canreinvite=yes Read about reinvite at http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite Apparently some hardware does not like it, and obviously, both the client and the service provider with have to be able to use the same codec (for them to be able to talk to each other) but better if Asterisk is restricted to that codec on both sides to start with. Please understand, I am trying to help and I don't know which parts (of what I'm saying) are not entirely accurate but normally if I say something wrong there are enough people who clamour to correct me. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 14:16 Subject: Re: [Asterisk-Users] Confused ! how to use reinvite in my asterisk setup ? thanks Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: I'm not an authority but why don't you get some g729 codecs (10 or so) and use g729 all around. Not allowing for ADSL overheads you can calculate your own requirements on http://www.asteriskguru.com/tools/bandwidth_calculator.php (please double the results since each call is turned around to your service provider.) I would have thought it would be better if you could use reinvite to let your clients speak directly to your service providers. Someone who knows better ought to be able to tell if this would work. Your restriction is what the service provider allows. Most (that I've used) allow g729. I know it uses more bandwidth than g723 but nothing like G711 (ulaw or alaw) and from my experience, the quality is quite reasonable. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 11:27 Subject: Re: [Asterisk-Users] Confused ! thanks for your replay, after i disallow all codec except g723 i also confused how a2billing is working then what i did , i removed all the codec from /usr/lib/astersik/module without codec_g723.so . then i saw in my log while user calling to my ivr access number a2b is looking for gms codec as all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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] Confused !
thanks i alreday did that , you r very helpfull thanks again /Salaque On 5/15/06, AR Tarzi [EMAIL PROTECTED] wrote: 1. In the extension definition, insert canreinvite=yes for each of your clients. 2. In the trunk definition, insert canreinvite=yes Read about reinvite at http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite Apparently some hardware does not like it, and obviously, both the client and the service provider with have to be able to use the same codec (for them to be able to talk to each other) but better if Asterisk is restricted to that codec on both sides to start with. Please understand, I am trying to help and I don't know which parts (of what I'm saying) are not entirely accurate but normally if I say something wrong there are enough people who clamour to correct me. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 14:16 Subject: Re: [Asterisk-Users] Confused ! how to use reinvite in my asterisk setup ? thanks Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: I'm not an authority but why don't you get some g729 codecs (10 or so) and use g729 all around. Not allowing for ADSL overheads you can calculate your own requirements on http://www.asteriskguru.com/tools/bandwidth_calculator.php (please double the results since each call is turned around to your service provider.) I would have thought it would be better if you could use reinvite to let your clients speak directly to your service providers. Someone who knows better ought to be able to tell if this would work. Your restriction is what the service provider allows. Most (that I've used) allow g729. I know it uses more bandwidth than g723 but nothing like G711 (ulaw or alaw) and from my experience, the quality is quite reasonable. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 11:27 Subject: Re: [Asterisk-Users] Confused ! thanks for your replay, after i disallow all codec except g723 i also confused how a2billing is working then what i did , i removed all the codec from /usr/lib/astersik/module without codec_g723.so . then i saw in my log while user calling to my ivr access number a2b is looking for gms codec as all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___
Re: [Asterisk-Users] ATXFER
Andrew, thank´s for the attention, but necessary to decide this my problem. You he could help me? Regards Josué 2006/5/14, Andrew Kohlsmith [EMAIL PROTECTED]: On Friday 12 May 2006 21:00, Josué Conti wrote: Eric, but I to continue using version 1.0.9 of asterisk, would have some solution?Yes, find a consultant to try and backport this feature to the version ofAsterisk you desire.-A.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER
Hi all I have SER installed and running But ser send all voice message to email But i would like to integrate with Asterisk IVR in the like , did some one integrated this kind of setup if so kindly guide me how can i do that ram ___ --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] VoipBuster issues?
Hi All, Any VoipBuster SIP users on this list that'd be willing to test VoipBuster outbound VoIP to PSTN? All numbers I tried from my (*) server are supposedly being connected, but no phone rings! Also their new WebStart function doesn't cause my phone to ring either... TIA! -- Francesco Peeters ___ --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] fc5 and link to sources?
Earl Terwilliger wrote:
On Saturday 13 May 2006 09:02, Rich Adamson wrote:
Earl Terwilliger wrote:
On Friday 12 May 2006 18:30, Rich Adamson wrote:
Carlos Alperin wrote:
Rich,
Check what is the content of /lib/modules/2.6.15-1.2054_FC5/build?
I see:
[EMAIL PROTECTED] build]# ls
arch crypto initMAINTAINERSREADME usr
blockDocumentation ipc Makefile REPORTING-BUGS
configs driversKbuild Makefile.orig scripts
COPYING fs kernel mm security
CREDITS includelib netsound
[EMAIL PROTECTED] build]#
which I thought was correct.
If it is empty, then you need to do yum install kernel-devel again.
Also you can check running uname -a to see if you have the same release
that the one that you're checking.
Its a new install, no updates, so only a single kernel installed.
[EMAIL PROTECTED] build]# uname -r
2.6.15-1.2054_FC5
which exactly matches the kernel source download/install.
About six of the source files in the zaptel directory compile, however
it barfs with several hundred errors when compiling zaptel.c.
Rich
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson Sent: Friday, May 12, 2006 4:30 PM
To: Asterisk Users-List
Subject: [Asterisk-Users] fc5 and link to sources?
Just installed fc5, installed correct kernel source, and trying to
compile zaptel-1.2. Changed the link in /lib/modules/2.6.15-1.2054_FC5
to point to /usr/src/redhat/SOURCES. Like:
lrwxrwxrwx 1 root root 23 May 12 15:21 build -
/usr/src/redhat/SOURCES
A 'make install' still complains with:
make -C /lib/modules/2.6.15-1.2054_FC5/build
SUBDIRS=/usr/src/zaptel-1.2 modules
make[1]: Entering directory `/usr/src/redhat/SOURCES'
make[1]: *** No rule to make target `modules'. Stop.
make[1]: Leaving directory `/usr/src/redhat/SOURCES'
make: *** [linux26] Error 2
What am I missing here? (must be pretty simple or I need more caffeine)
Rich
Rich,
the original FC5 kernel is broken.. it is a known issue.
Update to the latest kernel and re-compile zaptel again and you will be
fine
earl
Well... the best I can determine is that svn checkout for the zaptel-1.2
code is broken. I updated this fc5 system to 2.6.16-1.2111_FC5 (latest)
and installed the matching source, tried each of the steps referred to
in http://kb.digium.com/entry/21/4/, and the make still fails for zaptel.
The first compile error encountered is:
cc1: error: include/linux/autoconf.h: No such file or directory
In file included from /usr/src/zaptel-1.2/zconfig.h:9,
from /usr/src/zaptel-1.2/zaptel.c:40:
In file included from /usr/src/zaptel-1.2/zaptel.c:40:
/usr/src/zaptel-1.2/zconfig.h:10:27: error: linux/version.h: No such
file or directory
/usr/src/zaptel-1.2/zconfig.h:72:5: warning: LINUX_VERSION_CODE is not
defined
/usr/src/zaptel-1.2/zconfig.h:72:27: warning: KERNEL_VERSION is not
defined
/usr/src/zaptel-1.2/zconfig.h:72:41: error: missing binary operator
before token (
with about 900+ other lines of errors, of which the majority relate to
zaptel.c.
Thoughts?
Rich
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Hello Rich,
Here is what worked for me..
1) install FC5 (selected most everthing i could think of.. i miss the install
ALL option GRIN) Notice this link too with the options you need for FC5 and
more info on the FC5 broken kernel
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Fedora
2) yum -y update
3) yum install kernel-devel
then i installed Asterisk including Zaptel and it all went ok... Notice that
you don't need the kernel source just the development package which has all
the headers, etc. needed to compile Asterisk +
HOWEVER.. i did not check out from svn.. I use my handy/dandy install script
of which the 1st part is as follows:
#!/bin/bash
AST='1.2.7.1'
ZAP='1.2.5'
PRI='1.2.2'
ADD='1.2.2'
SND='1.2.1'
rm -rf /opt/asterisk/install
mkdir /opt/asterisk/install
cd /opt/asterisk/install
wget http://ftp.digium.com/pub/asterisk/asterisk-${AST}.tar.gz
wget http://ftp.digium.com/pub/zaptel/zaptel-${ZAP}.tar.gz
wget http://ftp.digium.com/pub/libpri/libpri-${PRI}.tar.gz
wget http://ftp.digium.com/pub/asterisk/asterisk-addons-${ADD}.tar.gz
wget http://ftp.digium.com/pub/asterisk/asterisk-sounds-${SND}.tar.gz
RESOLVED...
Problem was a broken link at /lib/modules/2.6.16-1.2111_FC5/build, not
pointing to the correct location of the kernel source/headers.
Rich
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Re: [Asterisk-Users] Music on Hold restart at beginning for each call
On Fri, 12 May 2006, Tim Sharp [EMAIL PROTECTED] wrote
I am using the m option on the dial command to play a message instead
of ringing. The message is something like please wait while I try to
locate your party so I need it to start at the beginning for each
call. I think there might be a way in 1.2.x be we are not ready to
upgrade yet so a solution for 1.0.9 is what I am after. Thanks.
The 'm' option is for music on hold, not really announcements. Wouldn't
it be simpler to play the announcement first, then dial? eg
exten = s,1,Playback(local/please_hold_locate)
exten = s,2,Dial(${FOO},20,tm)
--
Chris Hastie
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Re: [Asterisk-Users] Dial Command Reference for SIP channel
On Fri, 12 May 2006, Dave Morrow [EMAIL PROTECTED] wrote Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) Specifically, what is the |20|Ttr ? I cannot seem to find any reference which would indicate this is even a valid format for the SIP channel. Well to my inexpert eye it looks wrong. Are you sure about the those stray commas in the first example? Arguments can be separated by either a comma (,) or a pipe (|). So in your first you are dialling the SIP/100 with a 20 seconds timeout. What's left I think should be TtrwW, not Ttr,,wW. These are the 'options' and are as listed at http://www.voip-info.org/wiki-Asterisk+cmd+Dial. So if you loose those rogue commas, the answer to your what's the difference question is a specific time out value, the ability for both parties to perform transfers and a ringing tone. -- Chris Hastie ___ --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] voice mail notification
you can ask voicemail to run an executable (so also every kind of script, from bash to perl) when new mail is dropped in a mailbox. Check something like vm-new-script config directive in voicemail.conf, or something similar. (now i don't remember exactly) 2006/5/12, Ever Zalazar [EMAIL PROTECTED]: Hello, there is a way to send notification(not email) when it's received an voice mail? Maybe a SIP message to inform? Best REgards Ever Zalazar ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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] Dial Command Reference for SIP channel
Dave Morrow wrote: Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) Specifically, what is the |20|Ttr ? I cannot seem to find any reference which would indicate this is even a valid format for the SIP channel. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+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] Dial Command Reference for SIP channel
Doug Lytle wrote: Dave Morrow wrote: Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) Specifically, what is the |20|Ttr ? I cannot seem to find any reference which would indicate this is even a valid format for the SIP channel. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial Or show application dial in the Asterisk CLI. Dial(SIP/100|20|Ttr,,wW) is NOT a valid Dial command. In Asterisk a | is the same as a ,. In fact , is translated into a | internally, as you can see by watching the CLI. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] 911 @ Zap Channel Breakin
Ok here is one for you. I know we all do the this for 911: exten = _911,1,Dial(Zap/1/911) exten = _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a least cost routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is durring a 911 call test if the Zap channel is Available (probably using ChanIsAvail() ) to test the line. IF the channel is in use I want to barge in with an announcment saying that the line is needed for an emergency and the call we be disconnected. Then immediately drop the call capture the line so noone else can use it, wait about 5 seconds for the telco to clear the far end and place the 911 call. Is this possible? Thnaks Mark C [EMAIL PROTECTED] FWD: 293625 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager interface
I am currently writing some tools that work with the Asterisk Manager interface. Part of the issue is number of socket connections that the client opens back to the manager itnerface. Most of these connections are short lived. Is this is a problem from a design perspective? Or is the management interface designed to handle this. Devraj ___ --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] Asterisk Manager interface
Devraj Mukherjee wrote: Part of the issue is number of socket connections that the client opens back to the manager itnerface. Most of these connections are short lived. Is this is a problem from a design perspective? Or is the management interface designed to handle this. no it is not designed to handle this. Have a look at http://www.voip-info.org/wiki-Asterisk+Manager+Proxy =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] signature.asc Description: OpenPGP digital signature ___ --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] 911 @ Zap Channel Breakin
Mark Coccimiglio wrote: Ok here is one for you. I know we all do the this for 911: exten = _911,1,Dial(Zap/1/911) exten = _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a least cost routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is durring a 911 call test if the Zap channel is Available (probably using ChanIsAvail() ) to test the line. IF the channel is in use I want to barge in with an announcment saying that the line is needed for an emergency and the call we be disconnected. Then immediately drop the call capture the line so noone else can use it, wait about 5 seconds for the telco to clear the far end and place the 911 call. Is this possible? Thnaks Mark C [EMAIL PROTECTED] FWD: 293625 ___ --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 I seem to recall a script that did something similar on voip-info.org. Look through their E911 stuff, perhaps it's still there. Good luck. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] Getting Realtime running (1.2.7.1)
I've got my res_mysql.conf stating: [general] dbhost = 127.0.0.1 dbname = switchref dbuser = asteriskuser dbpass = xxx dbport = 3306 and my extconfig.conf stating: sipusers = mysql,switchref,sip_buddies sippeers = mysql,switchref,sip_buddies When Asterisk starts, and I show peers and show users, I don't see anything that is in the database. When looking at the traffic between Asterisk and MySQL, it's obvious that we are not actually sending a select for any data. Can anybody see what I'm doing wrong? 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
