robert brown
2004-Jul-08 00:39 UTC
[Asterisk-Users] pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING
Hello, Can anyone help with the output shown below? It?s running on RH9, recent CVS of Asterisk and with one X100P card (2 channels), a budget tone 102 and Xlite softphone. CLI> -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:32 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:40 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Due to various SIP errors, where I am unable to authenticate, I decided to blank out all configuration relating to any SIP phones. Robert Brown FWD: 290651 HYPERLINK "mailto:robert.brown01@blueyonder.co.uk"robert.brown01@blueyonder.co.uk --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.716 / Virus Database: 472 - Release Date: 05/07/2004 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040708/91ac7014/attachment.htm
brian
2004-Jul-08 07:02 UTC
[Asterisk-Users] pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING
Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler Honestly I don't know how much more clear this message can be. You need an exten => s,1,something in your [default] context. bkw -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of robert brown Sent: Thursday, July 08, 2004 2:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING Hello, Can anyone help with the output shown below? It’s running on RH9, recent CVS of Asterisk and with one X100P card (2 channels), a budget tone 102 and Xlite softphone. CLI> -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:32 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:40 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Due to various SIP errors, where I am unable to authenticate, I decided to blank out all configuration relating to any SIP phones. Robert Brown FWD: 290651 robert.brown01@blueyonder.co.uk
Chris A. Icide
2004-Jul-08 09:39 UTC
[Asterisk-Users] pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING
On 12:39 AM 7/8/2004, robert brown wrote:>CLI> -- Starting simple switch on 'Zap/1-1' > >Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel >'Zap/1-1' sent into invalid extension 's' in context 'default', but no >invalid handler > >-- Hungup 'Zap/1-1'In english, ---Starting simple switch on 'Zap/1-1' means, 'Hey! The phone line you have plugged into the first X100P card is ringing! I'm going to answer it!' Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler means, 'I answered the line and in zapata.conf you told me to go to context 'default' to start handling a call on this line. So I went to context 'default' in extensions.conf and since I have no signalling to tell me what number was dialed, I'm going to look for extension 's' (which I look for when the number dialed is a null string), but I couldn't find extension 's' in context default, so I started looking for extension 'i' because thats what I do when I can't find the context/enten I'm looking for (i stands for invalid, and I will alwasy look for extension i when I can't find a matching extension for the call), but alas, extension I wasn't there either. At this point I would normally be creative and try very hard to figure out what you really wanted me to do versus what you told me to do, but alas, unfortunately 99.95% of the programming world thinks such behavior is a bug and not a feature, so I am prevented from reading your mind and must follow my rules, so I am allowed to spit out an error message, which of course I did. ------ Hope that helps you understand the error messages you were seeing. What you need to do is create an context/extension in your extensions.conf file like such [default] exten => s,1,Dial(SIP/<your-sip-info-here>,20,t) ; see the wiki for what the 20 and t means (or do 'show application dial' from the cli) exten => i,1,Playback(invalid) exten => i,2,Hangup this should at least get you past the errors, the above entries aren't something you want to leave that way, because they don't handle all the possibilities you might run into (like not answering your phone if it rings for more than 20 seconds) -Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040708/b79cb555/attachment.htm