Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
On Tue, 2008-07-22 at 13:21 -0400, Jerry Geis wrote: On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote: / // �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: / Call from 'devcentos5x64_to_ebox4300' to extension 'mediaport_audio_visual' rejected because extension not found. Jerry-- from the console, type dialplan show smvoice-mediaport, and let's verify for certain that it's in there. I'll try to reproduce your problem in my test system here. murf Steve, I get this: dialplan show smvoice-mediaport There is no existence of 'smvoice-mediaport' context Command 'dialplan show smvoice-mediaport' failed. my extensions.conf has a context: ; media [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf Then express.dnis.conf has: ; This file is generated from MessageNet EMACS ; Phone Caller ID DNIS Manager screen ; MMAUDIO : EBOX 4300 - exten = mediaport_audio_visual,1,Goto(smvoice-mediaport-audio-visual,s,1) [smvoice-mediaport-audio-visual] exten = s,1,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup Not seeing what the problem is here. especially since 1.2 and 1.4 both work. Jerry Jerry-- I've opened a bug in your behalf at http://bugs.digium.com/view.php?id=13144 Please follow the above link and hit the 'monitor issue' button there, and it will send you an email whenever the issue has updates. I don't know if you created an account on bugs.digium.com, but if you have not, it would be a good idea (and time) to register. I've been pounding my head against the wall with a subtle bug that I *think* I've fixed; I've decided to commit the fix and close the above bug, but I realize full well that it may not be a fix to your problem! So, here is the plan: if after I close 13144, and you update your trunk/1.6 version of asterisk, and you still have the problem, then re-open 13144, and further discussion on this problem will occur via this bug report. The bug I fixed involved a memory leak in the dialplan structures, which has resulted, for me, in: 1. missing contexts 2. crashes on loading 3. crashes during 'stop gracefully' I found the problem on a code review, and valgrind verified that in some circumstances, it was happening. Fixing it cleared up all the weird affects. But then again, I managed to intensify the bug by having lotsa code in both extensions.conf, and in extensions.ael, and having the smvoice-mediaport-audio-visual context in BOTH files. The inclusion did not affect the results; if I included express.dnis.conf, or just pasted its contents in place of the '#include..', it didn't matter. So, please monitor that bug, and let me know if all is well. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote: �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: Call from 'devcentos5x64_to_ebox4300' to extension 'mediaport_audio_visual' rejected because extension not found. Jerry-- from the console, type dialplan show smvoice-mediaport, and let's verify for certain that it's in there. I'll try to reproduce your problem in my test system here. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
dialplan show default There is no existence of 'default' context Command 'dialplan show default' failed. I am getting the same thing for default What gives? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote: / // �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: / Call from 'devcentos5x64_to_ebox4300' to extension 'mediaport_audio_visual' rejected because extension not found. Jerry-- from the console, type dialplan show smvoice-mediaport, and let's verify for certain that it's in there. I'll try to reproduce your problem in my test system here. murf Steve, I get this: dialplan show smvoice-mediaport There is no existence of 'smvoice-mediaport' context Command 'dialplan show smvoice-mediaport' failed. my extensions.conf has a context: ; media [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf Then express.dnis.conf has: ; This file is generated from MessageNet EMACS ; Phone Caller ID DNIS Manager screen ; MMAUDIO : EBOX 4300 - exten = mediaport_audio_visual,1,Goto(smvoice-mediaport-audio-visual,s,1) [smvoice-mediaport-audio-visual] exten = s,1,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup Not seeing what the problem is here. especially since 1.2 and 1.4 both work. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: dialplan show default There is no existence of 'default' context Command 'dialplan show default' failed. I am getting the same thing for default Check the console and logs from when you started Asterisk to see if there were any errors reported when loading/parsing the dialplan. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
I dont see any errors in the dialplan while loading. I tried to past the whole log but it was rejected. Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files. I cant even dialplan show default at this time. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: I dont see any errors in the dialplan while loading. I tried to past the whole log but it was rejected. Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files. I cant even dialplan show default at this time. It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that should have been included in the source code. If you read that you'll realize that dialplan show command was deprecated in 1.4 and be removed in 1.6. Until your read those files you are going to continue to have strange problems. The thing is that they are not strange problems. They are problems you should expect if you don't read the upgrade notes. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files. We aren't disputing that, so you don't need to keep repeating it :-) You'll have to open a bug on bugs.digium.com and attach the log file there; we won't be able to help you any further until we can find out why your dialplan was not loaded when you used 1.6. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Eric ManxPower Wieling wrote: It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that should have been included in the source code. If you read that you'll realize that dialplan show command was deprecated in 1.4 and be removed in 1.6. Until your read those files you are going to continue Eric, I think you're mistaken; show dialplan was depreciated, not dialplan show. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
I sit corrected. He should still be reading the upgrade files. Doug Lytle wrote: Eric ManxPower Wieling wrote: It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that should have been included in the source code. If you read that you'll realize that dialplan show command was deprecated in 1.4 and be removed in 1.6. Until your read those files you are going to continue Eric, I think you're mistaken; show dialplan was depreciated, not dialplan show. Doug -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working. I am getting a SIP/401 Unauthorized error and then a SIP/404 error. I changed nothing in the configs. Is there a particular parameter needed for 1.6 that 1.4 did not care about? If I drop back to 1.4 it starts working again. Thanks Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working. I am getting a SIP/401 Unauthorized error and then a SIP/404 error. I changed nothing in the configs. How are you getting SIP-related errors from Console/DSP? Posting a console log would be most helpful, as many people on the mailing list are not telepathic :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
ow are you getting SIP-related errors from Console/DSP? Posting a console log would be most helpful, as many people on the mailing list are not telepathic :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) Kevin, below is the log your talking about. please note no configuration files were changed from 1.4 to 1.6, going back to 1.4 works again. Jerry -- Asterisk 1.6.0-beta9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': == Found [0;37;40m[1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': [1;30;40m == [0;37;40mFound [0mConnected to Asterisk 1.6.0-beta9 currently running on ebox4300 (pid = 4877) ebox4300*CLI Verbosity is at least 5 [Kebox4300*CLI --- SIP read from UDP://192.168.1.8:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;rport From: Jerry Geis 204 sip:[EMAIL PROTECTED];tag=as7d1f7b71 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 Jul 2008 16:53:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 20475 20475 IN IP4 192.168.1.8 s=session c=IN IP4 192.168.1.8 t=0 0 m=audio 14322 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - �--- (14 headers 14 lines) --- � == Using SIP RTP CoS mark 5 � == Using SIP VRTP CoS mark 6 �Sending to 192.168.1.8 : 5060 (NAT) �Using INVITE request as basis request - [EMAIL PROTECTED] �No user '3175661677' in SIP users list �Found peer 'devcentos5x64_to_ebox4300' for '3175661677' from 192.168.1.8:5060 � --- Reliably Transmitting (no NAT) to 192.168.1.8:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;received=192.168.1.8;rport=5060 From: Jerry Geis 204 sip:[EMAIL PROTECTED];tag=as7d1f7b71 To: sip:[EMAIL PROTECTED];tag=as324df4b6 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-beta9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0e961d2a Content-Length: 0 �Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) � [Kebox4300*CLI --- SIP read from UDP://192.168.1.8:5060 --- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;rport From: Jerry Geis 204 sip:[EMAIL PROTECTED];tag=as7d1f7b71 To: sip:[EMAIL PROTECTED];tag=as324df4b6 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 - �--- (10 headers 0 lines) --- � --- SIP read from UDP://192.168.1.8:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6a460d62;rport From: Jerry Geis 204 sip:[EMAIL PROTECTED];tag=as7d1f7b71 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=devcentos5x64_to_ebox4300, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=0e961d2a, response=1a8e257ae008af4156b1f65be8d4d267 Date: Mon, 21 Jul 2008 16:53:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 20475 20476 IN IP4 192.168.1.8 s=session c=IN IP4 192.168.1.8 t=0 0 m=audio 14322 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - �--- (15 headers 14 lines) --- �Sending to 192.168.1.8 : 5060 (NAT) �Using INVITE request as basis request - [EMAIL PROTECTED] �No user '3175661677' in SIP users list �Found peer 'devcentos5x64_to_ebox4300' for '3175661677' from 192.168.1.8:5060 �Found RTP audio format 0 �Found RTP audio format 8 �Found RTP audio format 3 �Found RTP audio format 101 �Peer audio RTP is at port 192.168.1.8:14322 �Found audio description format PCMU for ID 0 �Found audio description format PCMA for ID 8 �Found audio description format GSM for ID 3 �Found audio
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: �Looking for mediaport_audio_visual in smvoice-mediaport (domain 192.168.1.25) Do you have an extension called 'mediaport_audio_visual' in a context called 'smvoice-mediaport'? If so, can you post that context so we can see how it looks? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Do you have an extension called 'mediaport_audio_visual' in a context called 'smvoice-mediaport'? If so, can you post that context so we can see how it looks? Kevin, I mentioned that 1.4 works - 1.6 did not, going back to 1.4 works again. Here are the pieces: my sip.conf has context pointing to smvoice-mediaport part of extensions.conf: [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf file /etc/asterisk/express.dnis.conf ; MMAUDIO : EBOX 4300 - exten = mediaport_audio_visual,1,Goto(smvoice-mediaport-audio-visual,s,1) ; MMAUDIO : EBOX 4300 - exten = 1054,1,Goto(smvoice-mediaport-audio-visual,s,1) Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
