search for: ast_read

...asterisk as belows: sangoma card, connected with E1, CAS Signalling. I have two problem. 1. The asterisk don't received any DTMF when caller input to 2. when i dial to system, the caller hear bad sounds. monitor on console. asterisk show error. Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', but no exception handler Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', but no exception handler Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', b...

...hen at the some time. i have trouble then my system is half (CPU instructions = 100) i tested for some case as belows: - When i dial, then my system became answer, the caller hangup. system error message show (loop without condition and half machine) Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', but no exception handler Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', but no exception handler Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', b...

All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 spandsp-0.0.5pre4 The receivefax app works perfectly, ie i am able to receive the faxes, and what not, but...

This is a serious problem! I have brought up this issue in four previous attempts to get some feedback. I find it hard to believe that no one else is having this same problem. Apr 11 13:27:36 NOTICE[4446]: channel.c:1906 ast_read: Dropping incompatible voice frame on Local/103@sip-00f3,2 of format alaw since our native format has changed to slin Apr 11 13:27:36 NOTICE[4446]: channel.c:1906 ast_read: Dropping incompatible voice frame on Local/103@sip-00f3,2 of format alaw since our native format has changed to slin Apr 11 13...

Hi When I make a call from the outside to asterisk and the call is asnwered, all is OK, but when I make a call from the outside to asterisk and hangup before the call is answered, you got this WARNING in the console: Jul 6 19:33:08 WARNING[10037]: channel.c:1521 ast_read: Exception flag set on 'UniCall/2-1', but no exception handler Jul 6 19:33:08 WARNING[10037]: channel.c:1521 ast_read: Exception flag set on 'UniCall/2-1', but no exception handler . . . Jul 6 19:33:08 WARNING[10037]: channel.c:1521 ast_read: Exception flag set on 'UniCall/2-1...

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working with agentcallback and it works fine, i mean i can login and logout, but when an incomming calls gets the phone i can't answer. my iax.conf looks like...

hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines..... it seems the channel.c try to call ast_read(), read some bytes from the channel but there is nothing ... whether it's a loop to check data on the channel ? and man...

Several times a day I get this meesage scrolling on one of our asterisk boxes: Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our native format has changed to alaw Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our native format has changed to alaw Ma...

I have an incoming 800-number over IAX from Teliax and I'm experiencing the large packet loss on connection. When a call comes in there is no ring tone and the first few words of the welcome message are cut off, regardless of the delay I set. Standard call (not 800-number) coming over IAX with the same provider works just fine only the tall free number. So it seems there are some packet loss

...uot;Local/xxxxxxxxxxxx@callout-204e,1", "silence/1") in new stack -- Playing 'silence/1' (language 'en') == Spawn extension (callout, xxxxxxxxxxxxx, 2) exited non-zero on 'Local/xxxxxxxxxxx@callout-204e,2' Sep 3 11:59:22 NOTICE[14350]: channel.c:1287 ast_read: Dropping incompatible voice frame on IAX2/voiptalk/1 of format SLINR since our native format has changed to GSM Then step 7 is ok. Any help? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachme...

...n ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at res_musiconhold.c:180 #5 0x080673ae in ast_deactivate_generator (chan=0x9455ca0) at channel.c:1382 #6 0x08068d4e in generator_force (data=0x9455ca0) at channel.c:1405 #7 0x08061c50 in ast_read (chan=0x9455ca0) at channel.c:1857 #8 0x08069293 in ast_generic_bridge (c0=0xb659fcd0, c1=0x9455ca0, config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c, bridge_end= {tv_sec = 0, tv_usec = 0}) at channel.c:3260 #9 0x080655fd in ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0, config=0xb6c4feb0,...

I've downloaded diax-0.9.6b and configured for IAX2. Calls from Diax to * are perfect. However, when calling from * to Diax, I get the following: channel.c:1097 ast_read: Dropping incompatible voice frame on IAX2[mike]/3 of format GSM since our native format has changed to ULAW In iax.conf I have: allow=all disallow=g723.1 disallow=lpc10 allow gsm Has anyone else seen this? Thanks, Mike

...nd IAX2/4589/5 <> <>One or both client may sometimes segfault. Additionally, when they do get properly connected, I'm seeing this message which I didn't before. Sound quality gets very poor as I'd expect from the message. Nov 18 15:56:23 NOTICE[-184759376]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/4589@4589/3 of format SPEEX since our native format has changed to ULAW Both have: disallow=all allow=ulaw in the iax.conf file. Anyone have any ideas how I messed this up? Thanks, Adam Fineberg -------------- next part -------------- An HTML attach...

Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=... context=interno ;auth=plaintext disallow=all allow=gsm allow=ulaw allow=alaw Any hint? I'...

Sorry I forgot the subject in the last post. When my snom200 receives an inbound SIP external sip call, it somehow rejects the call and with a busy tone. The debug shows the following error: channel.c:1142 ast_read: Exception flag set on 'SIP/sipphone-7796', but no exception handler what does this mean and how can I debug it further?? Thanks JC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/2...

Hi, I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client (IAXPhone): - when I call from Iax to SIP sound works - when I call from Sip to Iax sound doesn't work, I get : Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Any advice, help ? Thanks in ad...

...ht the forwarded number. However, there is one-way voice problem. The caller can hear the destination(the forwarded number), but after the called party answers, the caller can't hear anything. Then the CLI> produce continuous errors as following: Oct 6 10:57:45 NOTICE[11026]: channel.c:1409 ast_read: Dropping incompatible vo ice frame on Local/1604xxx8621@hk-8073,2 of format gsm since our native format h as changed to g729 Oct 6 10:57:45 NOTICE[11032]: channel.c:1409 ast_read: Dropping incompatible vo ice frame on Local/xxx25837550@van-c9ae,2 of format ulaw since our native form at has changed...

...ar in my console is: Executing Set("SIP/something", "FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif") in new stack -- Executing RxFAX("SIP/something", "/var/spool/asterisk-fax/1137692307.5.tif") in new stack Jan 19 12:38:30 NOTICE[12008]: channel.c:1906 ast_read: Dropping incompatible voice frame on SIP/something of format slin since our native format has changed to ulaw "Dropping incompatible voice frame on SIP/something of format slin since our native format has changed to ulaw" This seemed particularly important, but I can't really say w...

Hi guys, I installed asterisk 1.4.17 and mISDN 1.1.7 with AGX addons to test the faxs. If I receive i do not have any problem, but i'm not able to send put any fax, i get always the same error: txfax_exec: transmission done with ast_read(chan) == NULL Anyone has txfax working with asterisk 1.4? I try to download app_rxfax.c and app_txfax.c with the asterisk.patch file but without success. On HYPERLINK "http://www.soft-switch.org"www.soft-switch.org is not longer available. Anyone can help me please ? TIA Giordano...

Hi. I have newely installed vicidial now i am getting thise error anyone can hel me. NOTICE[31819]: channel.c:1972 ast_read: Dropping incompatible voice frame on Local/8600051 at default-ed7b,1 of format gsm since our native format has changed to slin Regard's Vijay Kumar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk...