search for: ast_channel_make_compat

Displaying 20 results from an estimated 38 matches for "ast_channel_make_compat".

2003 Jun 06
1
more about SIP ...
...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/216.52.1...
2006 Apr 28
1
Warning: No path to translate with SJPhone
...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:...
2006 Jun 14
1
transcoding problem
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...
2007 May 07
0
H323 to H323 bridging ... failed ... also with chan_local
...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$192.168.1.100:1894/4096(-2033656)...
2005 Aug 10
1
Error while calling
...kup=yes disallow=all allow=g729 allow=g723 allow=ulaw allow=ilbc [voip] type=peer host=202.202.202.202 and here is the extension.conf. I have placed in the middle of extension.conf exten => _X.,1,Dial(SIP/${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/i...
2005 May 05
2
Did nufone change allowed codecs?
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...
2007 Jun 03
0
Strange problem with channel allocation
...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:2...
2010 Nov 12
1
Call failed becaus of SIP tanslate
...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 ---------...
2005 Jul 11
2
Unable to dial certain calls
...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...
2006 Dec 15
2
call from h323 to SIP
...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, 329...
2005 Jun 17
1
Unable to find a path from g729 to gsm
...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) ex...
2007 Feb 14
6
Fax with T.38
...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...
2007 Jun 26
1
Asterisk to Cisco 2600 GW DTMF Not Working
...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 c...
2011 May 04
1
asterisk 1.4.35 to 1.4.41
...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]: c...
2005 Jun 08
2
format g729 and Voxee.com
...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' ==...
2010 Mar 11
2
Codec preference
...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...
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
...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...
2006 Apr 19
1
Codec problem from SIP to H323
...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-1...
2007 Jun 12
2
Bridge bug in 1.4?
2003 Oct 23
1
Problems with OH323/codecs
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