[asterisk-users] Problem connecting to another server, Failed to authenticate on INVITE
:[EMAIL PROTECTED];tag=as69b98e07 User-Agent: SJphone/1.60.289a (SJ Labs) - --- (9 headers 0 lines) --- -- Executing [EMAIL PROTECTED]:1] Dial(SIP/ccn-081c9260, sip/[EMAIL PROTECTED]) in new stack Audio is at source.asterisk.server.ip port 18784 Adding codec 0x4 (ulaw) to SDP ...snip... Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to destination.asterisk.server.ip:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP source.asterisk.server.ip:5060;branch=z9hG4bK3e9e8861;rport From: Chris N sip:[EMAIL PROTECTED];tag=as2c01a79e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 16 Jun 2008 02:12:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 19975 19975 IN IP4 source.asterisk.server.ip s=session c=IN IP4 source.asterisk.server.ip t=0 0 m=audio 18784 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 ...snip... a=sendrecv --- -- Called [EMAIL PROTECTED] tz*CLI --- SIP read from destination.asterisk.server.ip:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP source.asterisk.server.ip:5060;branch=z9hG4bK3e9e8861; received=source.asterisk.server.ip;rport=5060 From: Chris N sip:[EMAIL PROTECTED];tag=as2c01a79e To: sip:[EMAIL PROTECTED];tag=as6b3765e7 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=destination.asterisk.server, nonce=6cc9de0c Content-Length: 0 - --- (11 headers 0 lines) --- Transmitting (no NAT) to destination.asterisk.server.ip:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP source.asterisk.server.ip:5060;branch=z9hG4bK3e9e8861;rport From: Chris N sip:[EMAIL PROTECTED];tag=as2c01a79e To: sip:[EMAIL PROTECTED];tag=as6b3765e7 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jun 15 21:12:42] NOTICE[1]: chan_sip.c:12253 handle_response_invite: Failed to authenticate on INVITE to 'Chris N sip:[EMAIL PROTECTED];tag=as2c01a79e' -- SIP/destination.asterisk.server-081cfab0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/ccn-081c9260' status is 'CONGESTION' ---End Transcript--- -- Chris Nestrud Email: [EMAIL PROTECTED] http://ChrisNestrud.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI connected to meetme conf gives Failed to write frame error
I am using the Debian package of asterisk which is version 1:1.4.13~dfsg-1. I have created an AGI script which plays files in response to external events and am using a callfile to connect it to a meetme conference. The first two or three files generally play correctly, but after that point almost all of the attempts to play files fail with this error. [Nov 4 22:10:41] WARNING[23300]: file.c:638 ast_readaudio_callback: Failed to write frame Occasionally a file will correctly be played to the meetme conference. The agi debug command shows no errors from the AGI script as it is executing. Through research, it looks like this error is caused by situations such as trying to play a file when the caller has disconnected. In this case the meetme conference is always active, and the AGI script remains active and attempts to play additional files. I welcome suggestions on how to avoid or work around this error. Chris -- Chris Nestrud Email: [EMAIL PROTECTED] http://www.panix.com/~ccn/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with asterisk 1.4.11 and playing files to meetme conference
I am using asterisk Version: 1:1.4.11~dfsg-1 as found in Debian. I'm using a call file to connect a meetme conference to an AGI script which plays files using the stream_file method. I have four files which should play in sequence, though only the first two files actually play. I get these errors in the CLI: [Sep 16 06:20:43] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 276 Buffer size: 320 [Sep 16 06:20:43] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 320 Buffer size: 276 [Sep 16 06:20:44] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 18 Buffer size: 320 [Sep 16 06:20:44] WARNING[18424]: app_meetme.c:1599 conf_run: Unable to set buffering information: Invalid argument [Sep 16 06:20:44] WARNING[18423]: file.c:626 ast_readaudio_callback: Failed to write frame [Sep 16 06:20:44] WARNING[18423]: file.c:626 ast_readaudio_callback: Failed to write frame [Sep 16 06:20:44] NOTICE[18423]: pbx_spool.c:371 attempt_thread: Call completed to Local/[EMAIL PROTECTED] This setup worked without problems under asterisk 1.2. Any idea of what's going on? I'd be happy to provide more information if needed. -- Chris Nestrud Email: [EMAIL PROTECTED] http://www.panix.com/~ccn/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy kills audio
On Fri, 14 Sep 2007 20:46:13 -0400, John Albano [EMAIL PROTECTED] wrote: I'm seeing the problem on both etch and lenny releases. Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux I have a similar problem and am also using Debian (lenny). I'm using an SMP kernel. Could that be the issue? I tested with an AGI script, and the problem is that audio isn't sent. The script receives DTMF digits and otherwise acts as expected. -- Chris Nestrud Email: [EMAIL PROTECTED] http://www.panix.com/~ccn/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy kills audio
On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote: On Fri, 14 Sep 2007 20:46:13 -0400, John Albano [EMAIL PROTECTED] wrote: I'm seeing the problem on both etch and lenny releases. Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux I have a similar problem and am also using Debian (lenny). I'm using an SMP kernel. Could that be the issue? I tested with an AGI script, and the problem is that audio isn't sent. The script receives DTMF digits and otherwise acts as expected. Which kernel exactly? Debian's linux-image-2.6.22-2-686, version 2.6.22-4. What is the output of: uname -a [EMAIL PROTECTED]:~# uname -a Linux a1271.userdns.net 2.6.22-2-686 #1 SMP Fri Aug 31 00:24:01 UTC 2007 i686 GNU/Linux I thought this might be due to the fact that this is a Dual Core Processor, so I tested using maxcpus=1 as a parameter to the kernel. This did not resolve the problem. The CPU is: model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4200+ -- Chris Nestrud Email: [EMAIL PROTECTED] http://www.panix.com/~ccn/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy kills audio
On Sun, 16 Sep 2007 01:29:11 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Sep 15, 2007 at 01:30:26PM +, Chris Nestrud wrote: On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote: On Fri, 14 Sep 2007 20:46:13 -0400, John Albano [EMAIL PROTECTED] wrote: I'm seeing the problem on both etch and lenny releases. Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux I have a similar problem and am also using Debian (lenny). I'm using an SMP kernel. Could that be the issue? I tested with an AGI script, and the problem is that audio isn't sent. The script receives DTMF digits and otherwise acts as expected. Which kernel exactly? Debian's linux-image-2.6.22-2-686, version 2.6.22-4. What is the output of: uname -a [EMAIL PROTECTED]:~# uname -a Linux a1271.userdns.net 2.6.22-2-686 #1 SMP Fri Aug 31 00:24:01 UTC 2007 i686 GNU/Linux I thought this might be due to the fact that this is a Dual Core Processor, so I tested using maxcpus=1 as a parameter to the kernel. This did not resolve the problem. The CPU is: model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4200+ Can you try zaptel from 1.4 SVN, or even ztdummy from 1.4 SVN? http://svn.digium.com/svn/zaptel/branches/1.4/ztdummy.c It should now use high-resolution timers for 2.6.22 users , so this may work around your problem. Thank you. I compiled and installed from the 1.4 branch and this has solved the problem. Audio and conferencing work as expected. --- Results after 22 passes --- Best: 99.998 -- Worst: 99.995 -- Average: 99.996866, Difference: 99.996866 -- Chris Nestrud Email: [EMAIL PROTECTED] http://www.panix.com/~ccn/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel ztdummy module causes playback to fail
I'm using asterisk 1.4.11 and Zaptel version 1.4.5.1 with kernel 2.6.22. I have the ztdummy module loaded, which is using zaptel and rtc. When the ztdummy module is loaded, sounds are not heard when using the asterisk background command. When the ztdummy module is unloaded, which also removes zaptel and rtc, sounds are heard. I've also tested this under kernel 2.6.21 with the same results. The zttest program reports an error when ztdummy and associated modules are not present, and hangs when they are loaded. This is an AMD Athlon(tm) 64 X2 Dual Core Processor 4200+ with 2GB ram. Any idea of what is causing this problem and how it can be solved? Chris -- Chris Nestrud Email: [EMAIL PROTECTED] http://www.panix.com/~ccn/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reliably detecting hangup
Hello list, I have an application which is one large AGI. My extensions.conf answers, calls the AGI, and hangs up the call. A second AGI is set to be called if hangup is detected. exten = s,1,answer exten = s,n,agi(appmain) exten = s,n,hangup exten = h,1,agi(after_hangup) The appmain AGI uses Exec to run other programs, including meetme. If a caller hangs up while the meetme program is running, it appears that control is not returned to the appmain AGI and the after_hangup AGI is not called. What is the best way to reliably detect when a caller has hung up and perform an action at that time? Chris -- Chris Nestrud Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
