search for: 245776

...ing spammed by these messages in the console (see below) and sound doesn't work with today's cvs. I rolled back a week, and it works fine. In addition to the sound problems, I had to enable inband dtmf squelch on the dilnk mgcp phones. if not, each pressed key was counted twice NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP a...

...caller number is spoken, but no connection between the caller and the mobile phone is established. This is the output from asterisk -vvvv: == CDR updated on CAPI[contr1/279xxxx]/0 -- Executing capiHOLD("CAPI[contr1/279xxxx]/0", "") in new stack Sep 15 19:09:05 NOTICE[245776]: app_capiHOLD.c:73 capiHOLD_exec: sent FACILITY_REQ PLCI = 0x101 Sep 15 19:09:05 NOTICE[245776]: app_capiHOLD.c:84 capiHOLD_exec: PLCI = 0x101 is on hold now -- Executing capiECT("CAPI[contr1/279xxxx]/0", "279xxxx:017520xxxxx") in new stack Sep 15 19:09:05 NOTICE[245776...

I run an Apple OS X workstation and I've got a server on the same LAN that's both a webserver and an Asterisk PBX. I wanted to be able to originate calls in the OS X Address Book application, and have Asterisk dial them and connect them to the phone on my desk. I've assembled a system that uses AppleScript to connect, via XML-RPC, to a web application that, in turn, connects to

Hi All I am having this problem when setting up a H323 call. Can anybody tell me what is going on? Thanks Serge ------------------ NOTICE[245776]: File chan_oh323.c, Line 1293 (oh323_write): H323:1637: Format changed from 4 to 8. WARNING[245776]: File frame.c, Line 76 (ast_smoother_feed): Smoother was working on 4 format frames, now trying to feed 8? ERROR[245776]: File chan_oh323.c, Line 1380 (oh323_write): H323:1637: Failed to fill smo...

...n console (FWD): vgw3*CLI> -- Executing Dial("SIP/332-552e", "SIP/613@fwd") in new stack -- Called 613@fwd -- SIP/fwd-357f is ringing -- SIP/fwd-357f answered SIP/332-552e -- Attempting native bridge of SIP/332-552e and SIP/fwd-357f Sep 13 11:02:52 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:02:57 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:03:01 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:03:02 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP cod...

...be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing '/etc/asterisk/meetme.conf': Found WARNING[245776]: File app_meetme.c, Line 162 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' (language 'en') i read from the lists, that I need to install zaprtc to solve this problem. when i try to compile zaprtc, which i got from http://www.junghanns.net/ast...

On Wed, 30 Apr 2003 00:24:19 -0400, Uriel Carrasquilla wrote: > >Gary: >I just copied the content from chan->exten to chan->dnis. I am calling from How are you doing this coying ? >one extension to another. >Have you got DynExtenDB to work? nope, haven't got over the first problem yet. Gary .

...blished), while asterisk isn't receiving anything in return? Ignoring this (assuming for some wacko reason, this is OKAY), I put a sample.call file in asterisk outgoing spool (Zap/g1) and asterisk very quickly traverses all channels and then gives the following error: Aug 12 03:13:20 NOTICE[245776]: channel.c:1597 __ast_request_and_dial: Unable to request channel Zap/g1/[PHONENUMBERREMOVED] Aug 12 03:13:20 NOTICE[245776]: pbx_spool.c:235 attempt_thread: Call failed to go through, reason 0 Now the question is: whatchu gonna do, when they come for you? Cheers folks, al. _________________...

...luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes then getting worse again. I've noticed that the following messages look directly connected to sound interruptions: on asterisk console: NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible in debug log: Aug 1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe The music stops playing simultenuously with these messages appearing....

Hi, On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new Numbers. I changed the extensions in extension conf to match the new numbers. But i always get: Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel 'CAPI[contr1/89064934]/0' sent into invalid extension 's' in context 'default', but no invalid handler I only changed the MSNs in the extension.conf. It has worked with the old numbers from German Telekom. Any help? Tom [makeit] exten =...

...erisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack NOTICE[245776]: File app_dial.c, Line 698 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time When I try to call in, usually nothing happens. One time it answered and Asterisk acted like it was going through the dial plan steps, but on the phone I never heard anyth...

..., it works perfectly with inband (it detects the whole mailbox (in my case 10007)), but not with rfc2833 (in this case, it only detects 107 as the mailbox number). With gsm codec, the inband doesn't work, I guess that's due to the frequecy responce of the codec, asterisk just say: WARNING[245776]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames Anyway, I would like to know if is there anybody trying to make Asterisk work correctly with rfc2833 digits transmition. The reason is that I need to use gsm, and I need to use Voicemail. Any ideas?, Thanks in advance,...

...213006 (LWP 28300)] ........................................Sep 30 10:51:12 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call 333ab105721da3172443a8582d1d5ae9@192.168.0.201 for seqno 102 (Non-critical Request) ................................ ] Asterisk Ready. [Thread 245776 (LWP 28302) exited] [New Thread 262161 (LWP 28305)] Sep 30 10:53:40 WARNING[262161]: codec_speex.c:166 speextolin_framein: Out of buffer space Sep 30 10:53:40 WARNING[262161]: codec_speex.c:166 speextolin_framein: Out of buffer space Program received signal SIGSEGV, Segmentation fault. [Switchin...

I have been fooling around with DynExtenDB and run into two glitches. 1) The code is looking for (chan->dnis) and in my case I find (null). I forced (chan-dnis) to be the same as (chan->exten). So far so good. Now I can connect and talk. This lead me to the second glitch. 2) As soon as the call ends by hanging up, the code issues a (ast_spawn_extension). This causes asterisk to drop

I've got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP?

Hi, does anyone have any recommended (read tried and tested) way of making asterisk be able to handle incoming faxes. I've a PC running asterisk with a digium E1 card in and simply want to be able to route a call to some application which will take a fax call and save the fax as an image. I guess the bits I'm unsure of are how to terminate a fax call without a modem (if this is

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type