[asterisk-users] g729 with Cisco gateways?
Does anyone have Asterisk 1.4 using g729 with a Cisco gateway? So far I'm 0 for 2 when trying to get my Asterisk to use g729 with Cisco gateways. I do not have a problem with g729 when using our IP phones or talking with our non-Cisco VOIP platform. The problem only seems to exist, at least in my environment, when talking with Cisco devices. Thanks, Scott ___ -- 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
[asterisk-users] Any SLA alternatives?
I have a group of people who have distinct phone numbers plus a shared number. The shared number is actually a group that rings through to all of their direct numbers. I want them to: 1) be able to make outgoing calls as the shared number and 2) be able to make outgoing calls as their direct number. Currently it's working in a non-ideal state. Each user has a phone with two lines. One line has the shared number and the other has their distinct number. Only one phone has the registration for the shared number at any point in time. The outgoing calls from either line works great from all of the phones. The problem is when they need to forward the shared line somewhere else. Only the phone that's currently registered as the shared line can do the forwarding. The users have no way to know, without logging into all of the phone GUIs, to know which one actually is holding the registration. I researched using shared line appearance (SLA), but several people may need to be able to make outgoing calls using the shared number at the same time. From what I read about SLA, it makes the line appear busy on the other phones. This is not what I'm looking for. Is there a solution for what I'm wanting to do? [phone1] line1=5000 line2=5001 [phone2] line1=5000 line2=5002 [phone3] line1=5000 line2=5003 That is a brief overview of the phone configs. I want any/all of them to be able to make outgoing calls from the 5000 extension, but also maintain the ability to make calls from their 500x extensions. I thought about overwriting all of their Caller ID values, but that defeats the purpose of having distinct lines per user. Thanks, Scott ___ -- 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
[asterisk-users] Asterisk Appliance Logs?
My phone isn't registering with my Asterisk appliance, but I'm not sure where to find any logs to see what is going on? Does the appliance not support log viewing? Thanks, Scott ___ --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] My G729 problem re-visited
Asterisk is not crashing. It sends back OKs to the gateway but doesn't include any codec for the RTP, so the call gets closed. For whatever reason, Asterisk won't talk g729 with any of my gateways, but it will talk (and even transcode) g729 for the phones. Scott On 10/15/07, Power, Paul C. [EMAIL PROTECTED] wrote: Have you figured out if asterisk is crashing or not? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: Friday, October 12, 2007 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] My G729 problem re-visited Gateway sends Asterisk an INVITE (using g729) Asterisk sends Phone an INVITE (using g711 or g729) Phone sends Asterisk an OK (using g711) Asterisk sends Gateway an OK (with no RTP choice) Gateways ends the conversation I can setup the Phone to use g729 and it will reply with an OK for g729, but the OK to the Gateway will still stay empty. Only when I enable g711 on the Gateway will this work. I have experienced this on 2 different models of gateways so far. I included my config for both the Gateway and the Phone in my original message, hoping that maybe I was configuring the Gateway wrong in Asterisk? But no one has said anything so I'm assuming its okay. Phone (g729) to Phone (g729) works Phone (anything) to Gateway (g711) works Phone (anything) to Gateway (g729) does NOT work I'm licensed for g729 (although I'm told I should not need it for pass through). And it will transcode when the phone is g729 and the gateway is g711. But for whatever reason I can't use g729 on the gateway side of the calling process? Thanks, Scott On 10/12/07, Power, Paul C. [EMAIL PROTECTED] wrote: Is the call being dropped or is Asterisk takng a core dump? I have core dump issues with g729 and asterisk 1.4.11, but my set up is a little different than yours... -Original Message- From: Scott Moseman Sent: Friday, October 12, 2007 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] My G729 problem re-visited No ideas on this one from anyone? I suppose I'm going to need to pay for some Digium support because this is a really unusual problem. Does anyone else have a gateway that speaks g729 to Asterisk and works? For whatever reason, Asterisk refuses to reply back to any of my gateways using g729. Phone (g729) to phone (g729) works. Phone (g729) to Asterisk to gateway (g711) works. But attempt g729 between Asterisk and a gateway and it fails -- every time. Asterisk responds to the gateway but never includes any codecs in the packet, unless it's g711. My configurations are shown below. Thanks, Scott On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729
[asterisk-users] Display channels and codecs
Is there an easy way to show all active channels AND the codecs they're using? Other than going through EACH channel individually to check the codec which is, obviously, not a very efficient process. Thanks, Scott ___ --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] My G729 problem re-visited
No ideas on this one from anyone? I suppose I'm going to need to pay for some Digium support because this is a really unusual problem. Does anyone else have a gateway that speaks g729 to Asterisk and works? For whatever reason, Asterisk refuses to reply back to any of my gateways using g729. Phone (g729) to phone (g729) works. Phone (g729) to Asterisk to gateway (g711) works. But attempt g729 between Asterisk and a gateway and it fails -- every time. Asterisk responds to the gateway but never includes any codecs in the packet, unless it's g711. My configurations are shown below. Thanks, Scott On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ --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] My G729 problem re-visited
Gateway sends Asterisk an INVITE (using g729) Asterisk sends Phone an INVITE (using g711 or g729) Phone sends Asterisk an OK (using g711) Asterisk sends Gateway an OK (with no RTP choice) Gateways ends the conversation I can setup the Phone to use g729 and it will reply with an OK for g729, but the OK to the Gateway will still stay empty. Only when I enable g711 on the Gateway will this work. I have experienced this on 2 different models of gateways so far. I included my config for both the Gateway and the Phone in my original message, hoping that maybe I was configuring the Gateway wrong in Asterisk? But no one has said anything so I'm assuming its okay. Phone (g729) to Phone (g729) works Phone (anything) to Gateway (g711) works Phone (anything) to Gateway (g729) does NOT work I'm licensed for g729 (although I'm told I should not need it for pass through). And it will transcode when the phone is g729 and the gateway is g711. But for whatever reason I can't use g729 on the gateway side of the calling process? Thanks, Scott On 10/12/07, Power, Paul C. [EMAIL PROTECTED] wrote: Is the call being dropped or is Asterisk takng a core dump? I have core dump issues with g729 and asterisk 1.4.11, but my set up is a little different than yours... -Original Message- From: Scott Moseman Sent: Friday, October 12, 2007 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] My G729 problem re-visited No ideas on this one from anyone? I suppose I'm going to need to pay for some Digium support because this is a really unusual problem. Does anyone else have a gateway that speaks g729 to Asterisk and works? For whatever reason, Asterisk refuses to reply back to any of my gateways using g729. Phone (g729) to phone (g729) works. Phone (g729) to Asterisk to gateway (g711) works. But attempt g729 between Asterisk and a gateway and it fails -- every time. Asterisk responds to the gateway but never includes any codecs in the packet, unless it's g711. My configurations are shown below. Thanks, Scott On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ --Bandwidth and Colocation Provided by http
Re: [asterisk-users] My G729 problem re-visited
That was actually a VM. Here's the real server (13ms). CLI show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw- 3-12 21 3 13 - 152- alaw- 31-2 21 3 13 - 152- g729- 5444 43 5- - 174- # dmesg | grep 'Xeon(TM)' CPU0: Intel(R) Xeon(TM) CPU 2.80GHz stepping 03 CPU1: Intel(R) Xeon(TM) CPU 2.80GHz stepping 03 Thanks, Scott On 10/12/07, Mike Lynchfield [EMAIL PROTECTED] wrote: How do you get 11ms translation time on ulaw 729 ? we have 12ms and its dual xeons 2.6.. On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ --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] Asterisk Redundancy
On 9/26/07, SIP [EMAIL PROTECTED] wrote: No. It's not. But there still exists the possibility even in a relatively stable situation that the software could crash or that hardware could fail. It's best, when planning a highly-available solution, to plan for the unforeseen and not assume you can avoid all mishaps. Let's assume, for the sake of argument, that the software will NEVER fail. Hardware still might, and that would still mean a lost call unless there's a way to switch running calls over to a new server seamlessly. Also be sure that you have a very redundant network configuration. Too often I see people spend a great deal of time and money to get redundant servers when their switches, firewalls, routers, etc are not even capable of handling a failed network element. Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] My G729 problem re-visited
Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
I'm trying a simple Echo test and it's failing for g729... exten = 1267,1,Answer() exten = 1267,2,Echo() Test #1 (failure) gateway33 codecs g729a, g729b [gateway33] type=friend host=gateway33 context=default-inbound disallow=all allow=g729 gateway33 INVITE = g729b Asterisk 200 OK = no media Asterisk sends (one way) g711u RTP Test #2 (failure) gateway33 codecs g729a [gateway33] type=friend host=gateway33 context=default-inbound disallow=all allow=g729 gateway33 INVITE = g729a Asterisk 200 OK = no media gateway33 ends the session gateway33 INVITE = g729a Asterisk 200 OK = no media gateway33 ends the session ... Test #3 (success) gateway33 codec g729a, g729b, g711u [gateway33] type=friend host=gateway33 context=default-inbound disallow=all allow=ulaw allow=g729 gateway33 INVITE = g729b, g711u Asterisk 200 OK = g711u Asterisk sends/receives g711u RTP Does any of this point to a specific problem? I even have a licensed g729 channel. CLI show g729 0/0 encoders/decoders of 1 licensed channels are currently in use What information can I provide to help troubleshoot? This is making no sense. When I setup my desk phone to use G729 and make the test call directly (bypassing the gateway), the call completes fine and media is sent using G729 successfully. I'm not sure why it would work any differently from a Cisco gateway? The only difference that I'm aware of is that my phone (Polycom 430) seemed to ask for G729, while the gateway was either G729a or G729b specifically. In the instance of my phone, Asterisk came back with G729a in the 200 OK message. Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
On 9/20/07, Luke Groeneveld [EMAIL PROTECTED] wrote: I'm getting frustrated simply trying to get this g729 working. For what it is worth, I had a similar issue to you, and managed to get g729 working by installing the binary files from http://asterisk.hosting.lv Thanks for the suggestion. Looks like I'm having the same problem, though. What's odd is that I can make phone to phone G729 calls through Asterisk, but G729 calls from my gateway do not work. Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
On 9/18/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: However, in Test #3 the call will fail. Why? Because Asterisk will attempt to use ulaw in preference to G.729 if possible, and the other endpoint offered to support ulaw. The format(s) supported by the eventual call destination are not relevant, because at the time Asterisk is making a format decision for the incoming call leg, it has no clue what the destination is going to be or what formats it will support. We purchased a 1 channel G729 license for testing purposes. However, I'm still having problems and G729 calls do not work. [src_gateway] disallow=all allow=g729 [dest_phone] disallow=all allow=ulaw allow=alaw The INVITE from the gatway to Asterisk contains g729b. The INVITE from Asterisk to the phone has g711u|g711a. The dest phone rings, but when picked up, the call is gone and the source phone gets a busy response. What's up? CLI show g729 0/0 encoders/decoders of 1 licensed channels are currently in use I'm getting frustrated simply trying to get this g729 working. Where do I need to increase debugging in order to find out more detailed and useful info about what's going wrong??? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
Here's what I'm showing in the logs... [Sep 18 09:52:09] VERBOSE[2786] logger.c: == Registered file format g729, extension(s) g729 [Sep 18 09:52:09] VERBOSE[2786] logger.c: format_g729.so = (Raw G729 data) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This module is supplied under a commercial license granted by Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Please see the full license text supplied by the accompanying [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: register utility, or ask for a copy from Digium. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This product includes software developed by the OpenSSL Project [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Copyright (C) 1998-2006 The OpenSSL Project [Sep 18 09:52:09] VERBOSE[2786] logger.c: == G.729 Host-ID: x:x:x:etc [Sep 18 09:52:09] WARNING[2786] codec_g729.c: Failed to initialize G.729 copy protection! [Sep 18 09:52:09] VERBOSE[2786] logger.c: codec_g729a.so = (Annex A/B (floating point) G.729 Codec (optimized for i686)) Any ideas where this points me? Thanks, Scott On 9/17/07, Scott Moseman [EMAIL PROTECTED] wrote: What's the best way to debug what's going on within Asterisk? I turned up the 'core debug', but that did not give me what I was hoping to find. I'm hoping to see some kind of error that explains why it will not pass through the g729 codec. Thanks, Scott On 9/14/07, Scott Moseman [EMAIL PROTECTED] wrote: I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question. I even attempted to setup the phone with only the allow=g729, but in that instance it won't even complete the call. We had to add g711 support to the gateway in question for now to get it up and running, but we want to get it back to using only g729. CLI show modules like g729 Module Description Use Count format_g729.so Raw G729 data 0 codec_g729a.so Annex A/B (floating point) G.729 Codec ( 0 2 modules loaded I downloaded the pre-compiled g729 module from Digium. The sip.conf references g729 and the codec module is loaded. Unless there's anything else I need to do that I'm missing? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
Fyi... [myphone] disallow=all allow=g729 canreinvite=no [otherphone] disallow=all allow=g729 canreinvite=no I attempted this setup and it works. Media routed through the Asterisk. Thanks, Scott On 9/18/07, Jeremy Mann [EMAIL PROTECTED] wrote: Does G.729 phone - asterisk - G.729 phone work with reinvite turned off? -Original Message- From: [EMAIL PROTECTED] Sent: Tuesday, September 18, 2007 1:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: September-18-07 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
The gateway is transcoding the PSTN into g729 and passing it to Asterisk. The Asterisk never sees the PSTN from the outside. I have watched the INVITE requests, they contain a request for a g729 only call. But the INVITE to the phone does not include g729. However, as previously stated, I did get a g729 phone to talk to another g729 phone. So I assume that means pass-through *can* work, but something is not working right? Thanks, Scott On 9/18/07, Matt Watson [EMAIL PROTECTED] wrote: PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
Follow me on this, it seems odd (or maybe I don't undertand)... Test #1 [src_phone] disallow=all allow=g729 [dest_phone] disallow=all allow=g729 I can make the call (src to dest) and it will work using g729. Both the call handling and media are going through Asterisk. Test #2 [src_phone] disallow=all allow=g729 allow=ulaw [dest_phone] disallow=all allow=g729 I can make the call (src to dest) and it will work using g729. Both the call handling and media are going through Asterisk. Test #3 [src_phone] disallow=all allow=ulaw allow=g729 [dest_phone] disallow=all allow=g729 The above call attempt will fail, and this is what I'm seeing: chan_sip.c:2944 sip_call: No audio format found to offer. In every test, the source INVITE includes ulaw, alaw and 729. That is the codecs that I configured on the phone themselves. However, in Test #3 the call will fail. Why? This does not necessarily have to do with my g729 gateway, but I'm curious what's wrong with this scenario, maybe using this situation to understand will help me with my gateway... (Although I tried setting only g729 on the gateway and the gateway's peer in the Asterisk and it did not appear to help.) Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] g729 on 1.4.10.1
What's the best way to debug what's going on within Asterisk? I turned up the 'core debug', but that did not give me what I was hoping to find. I'm hoping to see some kind of error that explains why it will not pass through the g729 codec. Thanks, Scott On 9/14/07, Scott Moseman [EMAIL PROTECTED] wrote: I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question. I even attempted to setup the phone with only the allow=g729, but in that instance it won't even complete the call. We had to add g711 support to the gateway in question for now to get it up and running, but we want to get it back to using only g729. CLI show modules like g729 Module Description Use Count format_g729.so Raw G729 data 0 codec_g729a.so Annex A/B (floating point) G.729 Codec ( 0 2 modules loaded I downloaded the pre-compiled g729 module from Digium. The sip.conf references g729 and the codec module is loaded. Unless there's anything else I need to do that I'm missing? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] g729 on 1.4.10.1
I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question. I even attempted to setup the phone with only the allow=g729, but in that instance it won't even complete the call. We had to add g711 support to the gateway in question for now to get it up and running, but we want to get it back to using only g729. CLI show modules like g729 Module Description Use Count format_g729.so Raw G729 data 0 codec_g729a.so Annex A/B (floating point) G.729 Codec ( 0 2 modules loaded I downloaded the pre-compiled g729 module from Digium. The sip.conf references g729 and the codec module is loaded. Unless there's anything else I need to do that I'm missing? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
