search for: ast_best_codec

...;b" A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at <IP HIDDEN> port 14958 Adding codec 0x2000 (amr) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to B, it only offers amr: [May 6...

Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video attachment while using any video phone. This does work in my 1.8.23.1 installation. The file is skipped with the ast_streamfile error (and moved to OLD), and the prompts following that message display the ast_best_codec error. [Apr 7 16:05:50] WARNING[17497][C-00006fdd]: file.c:1017 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/2036/INBOX/msg0000 (format (ulaw|h264)): No such file or directory [Apr 7 16:05:50] WARNING[17497][C-00006fdd]: app_voicemail.c:8609 play_message: Playback of mess...

hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan->nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta

...<52997aa1-eb00-481c-8c56-e26d78d01515> -- Started music on hold, class 'default', on channel 'PJSIP/7000-00000000' -- Channel PJSIP/7000-00000000 joined 'softmix' base-bridge <52997aa1-eb00-481c-8c56-e26d78d01515> [May 7 16:21:32] WARNING[20789]: channel.c:834 ast_best_codec: Don't know any of (h263|h263p|h264) formats [May 7 16:21:32] WARNING[20789]: channel.c:834 ast_best_codec: Don't know any of (h263|h263p|h264) formats > 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to 192.168.8.203:17200 > 0x7f46f4187280 -- Probation passed -...

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

...to Gentoo using the same configs from FreeBSD on my Linux machine, except the new Linux machine is running 1.0.3 where the old machine was running 1.0.2. Whenever I try to dial into one of my DIDs, I get this in the debugs and the call gets dropped. Jan 18 13:35:15 WARNING[5735]: channel.c:270 ast_best_codec: Don't know any of 0xf800 formats Jan 18 13:35:15 ERROR[5735]: chan_iax2.c:5738 socket_read: No best format in 0xf800??? Jan 18 13:35:15 NOTICE[5735]: chan_iax2.c:5740 socket_read: Rejected connect attempt from 192.168.1.1, requested/capability 0x100/0xf900 incompatible with our capability...

...0x80100 (g729|h263), peer - audio=0x43f (g723|gsm|ulaw|alaw|g726|adpcm|ilbc)/video=0xc0000 (h261|h263), combined - 0x80000 (h263) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) ... ... Oct 29 16:20:58 WARNING[3098]: channel.c:506 ast_best_codec: Don't know any of 0x80000 formats Does anyone know how to resolve this problem? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061029/f27f19f0/attachment.htm

...ension(callbk, callbk_real_context,xferto, 1, cid_num)) { snprintf(dialstr, sizeof(dialstr), "%s@%s/n", xferto, callbk_real_context); } callback_request_and_dial(callbk, "Local", ast_best_codec(callbk->nativeformats), dialstr, 15000, &outstate, cid_num, cid_name); } static struct ast_channel *callback_request_and_dial(struct ast_channel *caller, const char *type, int format, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name) { int cause = 0;...

...ll, I get the following message on SERVER-A and the call disconnects. -- Executing [777 at from-sip-UK:1] Dial("SIP/ADRIANSPHONE-09dd5178", "SIP/abc-777:mypassword at someip.no-ip.info:5070/777|40|trw") in new stack [2011-04-12 15:18:03] WARNING[13926]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats -- Called abc-777:mypassword at someip.no-ip.info:5070/777 [2011-04-12 15:18:03] NOTICE[17058]: chan_sip.c:12108 handle_response_invite: Failed to authenticate on INVITE to '"Adrian Marsh" <sip:ADRIANSPHONE at 82.XXX.XXX.26>;tag=as0ff3...