Dan Austin: Thanks for your suggestion, on one side of the link im using Asterisk-1.2.1, in the app_conference side im using Asterisk-1.2.12.1. But it seems that is not relevant. You were right, the Linksys SPA im using sends 30ms RTP packetization, using a grandstream phone made it work.
Tony: Thank you very much for your help too Well, now knowing the fundamental cause, im going to the hardest part, finding a solution for the issue on app_conference code, I dont think is a good idea rely on 20ms frames. Or is it possible to make Asterisk use only 20ms frames? Kind Regards On 11/5/06, Tony Mountifield <[EMAIL PROTECTED]> wrote:
In article <[EMAIL PROTECTED]>, Moises Silva <[EMAIL PROTECTED]> wrote: > Tony, thanks for the suggestion. Yes, I remembered an issue with ILBC, > but the phone on the other side is using ULAW as well. I tried to > avoid any transcoding to see if that helped, but so far, no look. Im > still looking the code to find a solution. > > Any other ideas out there? A couple: 1) Does your iax.conf on the rogue IAX box have a setting for trunkfreq? If it was set at trunkfreq=30 (I suppose it should really be called trunkperiod, not trunkfreq), and trunking is enabled, that could explain the 240-sample frames. 2) Try running ethereal or tcpdump on that box to capture both the IAX stream and the stream from the phone on the other side. It's not the codec that is at issue with the remote phone, but rather the phone's setting for frame size. If it is sending 30ms frames, even in uLaw, it may be that they are getting relayed through at the same size. Hope this helps Cheers Tony > On 11/4/06, Tony Mountifield <[EMAIL PROTECTED]> wrote: > > In article <[EMAIL PROTECTED]>, > > Moises Silva <[EMAIL PROTECTED]> wrote: > > > Hi everyone. I know app_conference is not formal part from Asterisk, > > > but I think you may help me a bit to understand what is happening > > > here. > > > > > > app_conference is working just fine, except when receives frames > > > with 240 samples, it seems is somehow hardcoded to expect frames of > > > 160 samples, so when receives 240, a buffer overflow ocurrs on the mix > > > buffer and crashes Asterisk. Frames with 240 samples, so far, are just > > > generated by my IAX2 connection with other server, so, if I enter a > > > conference with a SIP channel using ULAW, ZAP channel using SLIN, and > > > IAX2 channel (kiax softphone) with ULAW, everything works fine, but at > > > the moment the other IAX2 server enters the conference (ULAW also), > > > everything crashes. > > > > > > I can stop asterisk from crashing modifing the app_conference code > > > to NOT mix 240 samples frames, and works, but obviously the voice from > > > the IAX2 server is not received by the other parties. > > > > > > Can somebody just give me a hint about where to look? > > > > > > Why just that IAX2 server generates frames with 240 samples? > > > > One possible idea - is the phone on the other side of that server > > negotiating the iLBC codec with 30ms packet size? > > > > Cheers > > Tony > > -- > > Tony Mountifield > > Work: [EMAIL PROTECTED] - http://www.softins.co.uk > > Play: [EMAIL PROTECTED] - http://tony.mountifield.org > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-dev mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > > -- > "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
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