search for: codec1

...hout me doing anything..Has anyone observed this thing before... Called 810 -- SIP/810-b6dc is ringing -- SIP/810-b6dc answered SIP/910-6c4e -- Attempting native bridge of SIP/910-6c4e and SIP/810-b6dc WARNING[1227879616]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 = 524302 is not codec1 = 524302, can't do reinvite -- Got SIP response 482 "Loop Detected" back from 129.82.44.226 WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 3c2706bad222-n2s56u19hj1l@129-82-44-226 for seqno 1 (Response) WARNING[1142106560]: File chan_...

...quot;, "SIP/jeremy|20") in new stack -- Called jeremy -- SIP/jeremy-b7a9 is ringing -- SIP/jeremy-b7a9 answered SIP/andrew-89e3 -- Attempting native bridge of SIP/andrew-89e3 and SIP/jeremy-b7a9 Jul 25 16:49:36 WARNING[851980]: rtp.c:1392 ast_rtp_bridge: codec0 = 12 is not codec1 = 256, cannot native bridge. == Spawn extension (default, 801, 1) exited non-zero on 'SIP/andrew-89e3' -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 192.168.200.226 Jul 25 16:49:42 WARNING[114695]: chan_sip.c:2820 process_sdp: No compatible codecs!...

Hello. Voice quality when calling - this is one of the most important in the PBX. You need to record the quality parameters for each call to improve. Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11. In asterisk 13 I did not find a handler after the call, but before

...ecvip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `from_u` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `uri` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `useragent` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `codec1` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `codec2` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `llp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL COMMENT 'lost packets by local end', `rlp` varchar(10) CHARACTER SET utf8 COL...

...tion for SDP (m = '', c = 'IN IP4 123.123.123.123') Sep 16 08:27:47 WARNING[245775]: chan_sip.c:2679 process_sdp: Insufficient information for SDP (m = '', c = 'IN IP4 123.123.123.123') Sep 16 08:27:47 WARNING[360465]: rtp.c:1382 ast_rtp_bridge: codec0 = 268 is not codec1 = 0, cannot native bridge. == Spawn extension (sip, 88888888, 1) exited non-zero on 'SIP/105-1559' (123.123.123.123 is the IP of our VoIP-provider, 88888888 is my cell phone, and 105 is the asterisk-connected phone). Regards, Evert

...0 and Cisco 7905. If g729 is the only available codec for 7905's configuration, then call from 7960 to 7905 goes without any problem and both phones use g729. But if I call from 7905 to 7960 the following is displayed on * console: WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot native bridge. And * does transcoding from g729 to g711. Both phones have reinvite turned on. Why everything works only way and does not work other way? Question #2: What approach should be used to have an * as a MoH server? For example, I want to have 100 simultaneous SIP calls....

Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like <tftpserver-dir> <mac-address> firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _________________________________________________________________ Rethink your

Dear Tariq; About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting? Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).