Re: [asterisk-users] G.729 Codec problem.
Hi, I just purchased an additional license from Digium but the problem is still there. The output g729 show licenses command when not in a call #g729 show licenses 0/0 encoders/decoders of 2 licensed channels are currently in use *The output *g729 show licenses command* when there is a outgoing call.* #g729 show licenses 1/2 encoders/decoders of 2 licensed channels are currently in use The Asterisk log showing this while on a call: /var/log/asterisk/full [Apr 8 18:12:30] WARNING[5742] translate.c: g729tolin did not update samples 0 [Apr 8 18:12:30] WARNING[5742] codec_g729a.c: out of G.729 decoder licenses Please Help me.. Thanks, Arun s. On Wed, Mar 24, 2010 at 1:48 AM, Arun Sasidhar arun.sasid...@cabotsolutions.com wrote: Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of called party, but he can't hear me. And also we have sip trunks from callcentric.com, but it is functioning as normal. Also the sip to sip local extension calls works fine. When I make a call through PSTN, the Asterisk log showing the following error: r 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 Please suggest a solution. Do we need additional licence? Thanking you in anticipation, * * *Arun Sasidhar* * * * * * * * * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 Codec problem.
We have been experimenting with how many licenses are needed when making calls, recording calls and using chanspy to listen in on calls when G729 is involved. I can tell you that way more licenses are needed then I had understood previously. We are making calls via AMI originate and both legs of the calls are controlled by extensions in a dial plan. The outbound leg computes a few things then does a dial to a sip provider that does G729. The other leg is an extension that queues the call to be passed off to an agent that has logged in from a G729 SIP phone. When the call is setup we use 1 encoder and 1 decoder. If we start recording with a monitor AMI action this jumps to 3 encoders and 7 decoders. If we then use a G729 SIP phone to call an extension that allows the caller to use chanspy to listen in this goes up to 4 encoders and 7 decoders. If we stop recording but keep listening in this goes down to 4 encoders and 5 decoders. The value of transcode_via_sln in asterisk.conf does not seem to effect the number of licenses used. ;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of directly I would think that once the agent gets bridged to the called leg there would be no need for any licenses as both the SIP phone and SIP provider are using G729. I sort of understand why maybe 2 encoders and 2 decoders would be needed if one was recording the call to a non G729 file. You would need to decode each leg of the call, do the recording, and encode each leg of the call. I have no idea why 3 encoders and more important 7 decoders are needed. This is all a long email to say that it is not at all clear to me how the software figures out what and when to decode and encode in the internals of Asterisk. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 8, 2010, at 6:24 AM, Arun Sasidhar wrote: Hi, I just purchased an additional license from Digium but the problem is still there. The output g729 show licenses command when not in a call #g729 show licenses 0/0 encoders/decoders of 2 licensed channels are currently in use The output g729 show licenses command when there is a outgoing call. #g729 show licenses 1/2 encoders/decoders of 2 licensed channels are currently in use The Asterisk log showing this while on a call: /var/log/asterisk/full [Apr 8 18:12:30] WARNING[5742] translate.c: g729tolin did not update samples 0 [Apr 8 18:12:30] WARNING[5742] codec_g729a.c: out of G.729 decoder licenses Please Help me.. Thanks, Arun s. On Wed, Mar 24, 2010 at 1:48 AM, Arun Sasidhar arun.sasid...@cabotsolutions.com wrote: Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of called party, but he can't hear me. And also we have sip trunks from callcentric.com, but it is functioning as normal. Also the sip to sip local extension calls works fine. When I make a call through PSTN, the Asterisk log showing the following error: r 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 Please suggest a solution. Do we need additional licence? Thanking you in anticipation, Arun Sasidhar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for
Re: [asterisk-users] G.729 Codec problem.
Jim- We have been experimenting with how many licenses are needed when making calls, recording calls and using chanspy to listen in on calls when G729 is involved. I can tell you that way more licenses are needed then I had understood previously. We are making calls via AMI originate and both legs of the calls are controlled by extensions in a dial plan. The outbound leg computes a few things then does a dial to a sip provider that does G729. The other leg is an extension that queues the call to be passed off to an agent that has logged in from a G729 SIP phone. When the call is setup we use 1 encoder and 1 decoder. At this point I'm confused. It seems like both ends of the call are G729 at this stage. If so, and you're in pass-thru mode, then no encoders and decoders should be in use. If we start recording with a monitor AMI action this jumps to 3 encoders and 7 decoders. If we then use a G729 SIP phone to call an extension that allows the caller to use chanspy to listen in this goes up to 4 encoders and 7 decoders. If we stop recording but keep listening in this goes down to 4 encoders and 5 decoders. The value of transcode_via_sln in asterisk.conf does not seem to effect the number of licenses used. ;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of directly I would think that once the agent gets bridged to the called leg there would be no need for any licenses as both the SIP phone and SIP provider are using G729. Agree... it seems like a good idea to get that working first. No licenses should be needed, maybe there is a config or setup issue. I sort of understand why maybe 2 encoders and 2 decoders would be needed if one was recording the call to a non G729 file. You would need to decode each leg of the call, do the recording, and encode each leg of the call. I have no idea why 3 encoders and more important 7 decoders are needed. I know very little about what Asterisk offers in the way of real-time monitoring features. But I can say that the moment you want to monitor in real-time, some type of conferencing or mixing must be performed and so both ends of the call must be decoded. At minimum that would be 2 encoders and 2 decoders, but that's assuming the monitor leg can be set to listen only -- not sure if that's possible. -Jeff On Apr 8, 2010, at 6:24 AM, Arun Sasidhar wrote: Hi, I just purchased an additional license from Digium but the problem is still there. The output g729 show licenses command when not in a call #g729 show licenses 0/0 encoders/decoders of 2 licensed channels are currently in use The output g729 show licenses command when there is a outgoing call. #g729 show licenses 1/2 encoders/decoders of 2 licensed channels are currently in use The Asterisk log showing this while on a call: /var/log/asterisk/full [Apr 8 18:12:30] WARNING[5742] translate.c: g729tolin did not update samples 0 [Apr 8 18:12:30] WARNING[5742] codec_g729a.c: out of G.729 decoder licenses Please Help me.. Thanks, Arun s. On Wed, Mar 24, 2010 at 1:48 AM, Arun Sasidhar arun.sasid...@cabotsolutions.com wrote: Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of called party, but he can't hear me. And also we have sip trunks from callcentric.com, but it is functioning as normal. Also the sip to sip local extension calls works fine. When I make a call through PSTN, the Asterisk log showing the following error: r 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path
[asterisk-users] G.729 Codec problem.
Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of called party, but he can't hear me. And also we have sip trunks from callcentric.com, but it is functioning as normal. Also the sip to sip local extension calls works fine. When I make a call through PSTN, the Asterisk log showing the following error: r 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 [Mar 24 13:59:27] WARNING[18090] translate.c: No translator path from alaw to unknown [Mar 24 13:59:27] WARNING[18090] codec_g729a.c: out of G.729 decoder licenses [Mar 24 13:59:27] WARNING[18090] translate.c: g729tolin did not update samples 0 Please suggest a solution. Do we need additional licence? Thanking you in anticipation, * * *Arun Sasidhar* * * * * * * * * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
