search for: speextolin_framein

...g0000 format: wav49, 0x8133390 -- x=1, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg0000 format: gsm, 0x8132f48 -- x=2, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg0000 format: wav, 0x8157988 Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec...

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

Hello everyone. I am having problems getting speex support. It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install

..._read: RTP: Received packet with bad UDP checksum Keeps outputting the notice till hangup. sip show channel # shows Speex as the codec used. 2. All codecs disabled in X-Lite/Pro except iLBC: X-Lite/Pro shows Speex used even it is disabled. Sep 27 12:17:32 WARNING[135169024]: codec_speex.c:166 speextolin_framein: Out of buffer space Sep 27 12:17:32 WARNING[135169024]: codec_ilbc.c:172 lintoilbc_framein: Out of buffer space Sep 27 12:17:32 WARNING[135169024]: codec_speex.c:166 speextolin_framein: Out of buffer space Sep 27 12:17:32 WARNING[135169024]: codec_ilbc.c:172 lintoilbc_framein: Out of buffer spa...

...etrans_pkt: Maximum retries exceeded on call 333ab105721da3172443a8582d1d5ae9@192.168.0.201 for seqno 102 (Non-critical Request) ................................ ] Asterisk Ready. [Thread 245776 (LWP 28302) exited] [New Thread 262161 (LWP 28305)] Sep 30 10:53:40 WARNING[262161]: codec_speex.c:166 speextolin_framein: Out of buffer space Sep 30 10:53:40 WARNING[262161]: codec_speex.c:166 speextolin_framein: Out of buffer space Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 262161 (LWP 28305)] iax2_send (pvt=0xffffffff, f=0xffffffff, ts=4294967295, seqno=-1, now=-1, transfer=-1...

Hi, I'm running Asterisk 1.0.7 and would like to add Speex support. I downloaded Speex 1.0.5, installed and recompile Asterisk again. Now trying from X-Lite to connect using Speex but getting lot of weird erros on Asterisk console: Sep 7 15:03:25 WARNING[28605]: codec_speex.c:166 speextolin_framein: Out of buffer space I was trying to setup Speex on my second Asterisk server and wanted to use this codec for IAX between these two boxes. But I'm getting unable to negotiate codecs. Other codecs works like a charm. Any ideas? Thank you. -- David

...een googling forever and whatever solutions are offered don't work for me. I am using x-lite (the latest build) and trying to use Speex. When I do call from the x-lite to another SIP phone or PSTN (through Cisco gateway) My asterisk fills up with this message: WARNING[1007]: codec_speex.c:196 speextolin_framein: Out of buffer space The x-lite client can hear the remote end (SIP or PSTN call) quite clearly, but what comes from the X-Lite is completely garbled and mixed with DTMF tones. I had tried the registry fix (which only changes the magic number from 97 to 110 and apparently didn't do anything e...