search for: ast_channel_make_compatible

...ipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing Dial("SIP/217.168.168.49:5060", "SIP/723@216.52.153.207") in new stack -- Called 723@216.52.153.207 WARNING[1240577216]: File channel.c, Line 1711 (ast_channel_make_compatible): No path to translate from SIP/216.52.153.207-2e12(1) to SIP/217.168.168.49:5060(4) -- SIP/216.52.153.207-2e12 answered SIP/217.168.168.49:5060 WARNING[1240577216]: File channel.c, Line 1711 (ast_channel_make_compatible): No path to translate from SIP/217.168.168.49:5060(4) to SIP/216.52.153.2...

...rs installing SJphone and it worked fine, but when I install it over a third user with the softphone, the phone dial for 2 seconds and a window alert goes out on the softphone: Busy Call rejected: 486 Busy Here And on my Asterisk server this message: Apr 28 09:05:37 WARNING[8140]: channel.c:2685 ast_channel_make_compatible: No path to translate from SIP/cabecera-af04(256) to SIP/usuario2-d58a(2) I don't know where is the error because I created the extensions as before with the others users and it should work Thanks for any help -------------- next part -------------- An HTML attachment was scrubbed... URL: htt...

I am having a problem with asterisk transcoding GSM and G729 codecs, the error message is below: Jun 14 09:38:12 WARNING[18292]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/3004-fcfb(256) to SIP/3003-c1c3(2) Jun 14 09:38:12 WARNING[18292]: app_dial.c:1586 dial_exec_full: Had to drop call because I couldn't make SIP/3004-fcfb compatible with SIP/3003-c1c3 == Spawn extension (test, 3003, 1) exited non-zero on 'SIP/3004-fcfb'...

...sed, then the call to phone 2 is setup (completed) and when trying to proceed with call from phone1, asterisk stops: *CLI> -- Executing Dial("H323/ip$192.168.1.100:1894/4096", "H323/11@wave") in new stack -- Called 11@wave May 7 11:29:22 WARNING[845]: channel.c:2693 ast_channel_make_compatible: No path to translate from H323/wave-1(-2033656) to H323/ip$192.168.1.100:1894/4096(-2033656) -- H323/wave-1 answered H323/ip$192.168.1.100:1894/4096 May 7 11:29:22 WARNING[845]: channel.c:2693 ast_channel_make_compatible: No path to translate from H323/ip$192.168.1.100:1894/4096(-2033656) to...

...kup=yes disallow=all allow=g729 allow=g723 allow=ulaw allow=ilbc [voip] type=peer host=202.202.202.202 and here is the extension.conf. I have placed in the middle of extension.conf exten => _X.,1,Dial(SIP/${EXTEN}@voip) exten => _X.,2,Hangup Aug 11 10:15:01 WARNING[11260]: channel.c:2127 ast_channel_make_compatible: No path to translate from SIP/isphone-8213(256) to SIP/200-1264(4) Aug 11 10:15:02 NOTICE[11260]: channel.c:1736 ast_set_read_format: Unable to find a path from g723 to g729 Aug 11 10:15:02 NOTICE[11260]: channel.c:1703 ast_set_write_format: Unable to find a path from g729 to g723 -- SIP/ispho...

Hi, I've been using nufone DIDs for months with no problem. Now there are codec problems that prevent any kind of calls working. For example, May 5 13:04:12 WARNING[928]: channel.c:2115 ast_channel_make_compatible: No path to translate from IAX2/NuFone@NuFone/25(256) to SIP/wengo-out-968a(4) May 5 13:04:12 WARNING[928]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make IAX2/NuFone@NuFone/25 compatible with SIP/wengo-out-968a The above worked perfectly recently, how recently can't...

...onfig_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine. [2007-06-03 20:16:10] DEBUG[27424]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '1014' -- Called 1014 [2007-06-03 20:16:10] WARNING[27424]: channel.c:3222 ast_channel_make_compatible: No path to translate from SIP/1014-081e93c0(256) to OSS/dsp(64) [2007-06-03 20:16:10] WARNING[27424]: channel.c:3222 ast_channel_make_compatible: No path to translate from SIP/1014-081e93c0(256) to OSS/dsp(64) ^^^^^^^^^^^^^^^^^^^^^^^^^ ?????? [2007-06-03 20:16:18] NOTICE[27408]: chan_sip.c:2758...

...pe 4, while native formats is 256 (read/write = 256/256) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) -- SIP/to-my-voip-11b955c0 answered SIP/8021-52514588 Nov 12 14:31:30 WARNING[21432]: channel.c:2781 ast_channel_make_compatible: No path to translate from SIP/8021-52514588(4) to SIP/to-my-voip-11b955c0(256) Nov 12 14:31:30 WARNING[21432]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/8021-52514588 compatible with SIP/to-my-voip-11b955c0 Thank you for any help! -- Abdullah -------------...

...isk, but calling US numbers is not working. I CAN call the same US numbers with the service by using a direct connection from a softphone for example. The entries that show up in the log after failed attempts to call the US are: Jul 11 20:04:04 WARNING[25225728]: File channel.c, Line 1851 (ast_channel_make_compatible): No path to translate from SIP/2203-2929 (4) to IAX2[vbx]/1(16) Jul 11 20:04:04 WARNING[25225728]: File app_dial.c, Line 672 (dial_exec): Had to drop call because I couldn't make SIP/2203-2929 compatible with IAX2[vbx]/1 I don't see anything suspicious entries in the CLI logging with...

...Dec 15 14:45:13 WARNING[19794]: chan_sip.c:2572 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Dec 15 14:45:13 WARNING[19794]: translate.c:116 ast_translator_build_path: No translator path from alaw to unknown Dec 15 14:45:13 WARNING[19794]: channel.c:2752 ast_channel_make_compatible: No path to translate from H323/ip$172.z.z.z:4836/14(256) to SIP/193.x.x.x-40455d68(8) Dec 15 14:45:13 WARNING[19794]: app_dial.c:1602 dial_exec_full: Had to drop call because I couldn't make H323/ip$172.z.z.z:4836/14 compatible with SIP/193.x.x.x-40455d68 == Spawn extension (default, 3298, 2...

...8:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a path from g729 to gsm -- IAX2/teliax-1 is ringing -- IAX2/teliax-1 answered Zap/1-1 Jun 17 18:47:18 WARNING[7396]: channel.c:2308 ast_channel_make_compatible: No path to translate from Zap/1-1(68) to IAX2/teliax-1(256) Jun 17 18:47:18 WARNING[7396]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make Zap/1-1 compatible with IAX2/teliax-1 -- Hungup 'IAX2/teliax-1' == Spawn extension (outgoing, 19737228839, 1) exited...

...t; Asterisk <----> IP <----> Patton M-ATA <----> Analog Fax 2 I tried Analog Fax 2 -> Analog Fax but nothing works!! In the Patton configuration I put G711 and no silence suppression. In asterisk I have some errors : [Feb 14 11:28:55] WARNING[10547]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-mg-02-006f37f0(256) to SIP/0625037998-006de430(8) [Feb 14 11:29:04] WARNING[10546]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 14 11:29:04] WARNING[10546]: channel.c:2702 set_format: Unable to find a code...

...26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to find a codec translation path from ilbc to ulaw Jun 26 17:53:52 WARNING[14248]: chan_sip.c:2555 sip_write: Asked to transmit frame type 1024, while native formats is 4 (read/write = 4/4) Jun 26 17:53:52 WARNING[14248]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/53061-92e0(4) to SIP/10.10.10.10-78fa(1024) Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge: Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call: Bridge failed on chann...

...lwrite: write() returned error: Broken pipe -- AGI Script smvoice completed, returning 0 -- Executing [smvoice_dial_goto_voicemail at smvoice-dialout:1] Dial("DAHDI/18-1", "SIP/524|30|tT") in new stack -- Called 524 [May 3 21:47:41] WARNING[21746]: channel.c:3782 ast_channel_make_compatible: No path to translate from SIP/524-00000001(4096) to DAHDI/18-1(4) [May 3 21:47:41] WARNING[21746]: chan_sip.c:3890 sip_write: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:47:41] WARNING[21746]: chan_...

...frame type 256, while native formats is 2 (read/write = 2/2) Jun 8 18:48:51 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) -- IAX2/66.246.246.52:4569-7 answered SIP/201-fbb8 Jun 8 18:48:51 WARNING[6405]: channel.c:2308 ast_channel_make_compatible: No path to translate from SIP/201-fbb8(2) to IAX2/66.246.246.52:4569-7(256) Jun 8 18:48:51 WARNING[6405]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make SIP/201-fbb8 compatible with IAX2/66.246.246.52:4569-7 -- Hungup 'IAX2/66.246.246.52:4569-7' == Spaw...

...rties ?? My Grandstream supports G729, alaw and gsm... in this order. The Zoiper softphone has alaw and gsm as codecs... in that order. Although there should be a matching codec found, my Grandstream can not call the Zoiper softphone. CLI shows : [Mar 11 17:47:21] WARNING[22367]: channel.c:3340 ast_channel_make_compatible: No path to translate from SIP/mygrandstream-09c599e0(2) to SIP/zoiper-09cd57f8(256) [Mar 11 17:47:21] -- Got SIP response 415 "Unsupported Media Type" back from 192.168.1.106 (<-- zoiper) SIP debug : [Mar 11 17:55:57] Peer audio RTP is at port 192.168.1.101:10110 (<-- the Gra...

...6:27:35 NOTICE[1061908]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G723 Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)? Nov 23 16:27:35 WARNING[1061908]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/sip.westend.com-082fd1b8(8) to SIP/xxx-3ef8(1) Nov 23 16:27:35 WARNING[1061908]: channel.c:2633 ast_channel_bridge: Can't make SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 compatible Nov 23 16:27:35 WARNING[1061908]: res_features.c:358 ast_bridge_call: Bridge fai...

...translation path from g729 to ulaw -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8 -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8 -- H323/H323gw-2 is ringing -- H323/H323gw-2 answered SIP/amejia-1fc8 Apr 19 15:23:45 WARNING[75484]: channel.c:2685 ast_channel_make_compatible: No path to translate from SIP/amejia-1fc8(4) to H323/H323gw-2(256) Apr 19 15:23:45 WARNING[75484]: app_dial.c:1553 dial_exec_full: Had to drop call because I couldn't make SIP/amejia-1fc8 compatible with H323/H323gw-2 == Spawn extension (test, 444, 1) exited non-zero on 'SIP/amejia-1fc8&...

On oh323.conf I have: codec=G711U frames=20 But while connecting it gives me in log: ? 1:18.636 ? ? ? ? ?H225 Caller:8111de8 H245 ? ?Capability merge result: ? Table: ? ? G.723.1(5.3k){hw} <1> ? Set: ? ? 0: ? ? ? 0: ? ? ? ? G.723.1(5.3k){hw} <1> Which I don't have, so the connection is dropped. Any known solutions? (remote side has g711 u-Law) -- Witold Kr?cicki (adasi) adasi