search for: pbx_find_extension

Yesterday I upgraded from asterisk 1.2.21.1 to 1.2.22. We are running Zaptel-1.2.17.1. After upgrading all of our calls from the phone company (DIDs trunks with wink start on a channelized T1) were not coming in. Looking in the log file '/var/log/messages'. I saw the following error message. pbx.c: Cannot find extension '' in context '(null)' This was confusing since I

Hi, Jan 6 01:43:09 WARNING[12209]: pbx.c:796 pbx_find_extension: No such switch 'Realtime' What does this message mean ? Something wrong with the switch statement in my extensions.conf or maybe is the module net correctly installed ? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.dig...

Hi all, i need to know how to use the switch statement in extensions.conf to throw calls on a secondary asterisk server. I have been trying all day to be able to make it work but its not working. It displays the following error: pbx_find_extension: No such switch 'master:secret at 192.168.0.7' I have read the help material on voip-info for connecting 2 asterisk servers and followed the steps given there but no use. Can anyone help me with this? Also need to know what the switch statement does internally. i mean does the server whic...

hallo, does anybody successfully managed to get swissvoice ip10s with h323 firmware work with asterisk ? mgcp firmware works fine, but with h323 i'm still getting one way audio. regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

...rted by Timo Ter??s) * ASTERISK-23261 - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR variables for subsequent records (Reported by zvision) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim) * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load (Reported by David Brillert) * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set - probably introduced in 11.7.0 (Reported by...

...rted by Timo Ter??s) * ASTERISK-23261 - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR variables for subsequent records (Reported by zvision) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim) * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load (Reported by David Brillert) * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set - probably introduced in 11.7.0 (Reported by...

CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? It appears that RealTime for the extensions.conf file is on a context by context basis, but you have to create each new context in the extensions.conf file then add a "switch => Realtime" line (then reload). I want to be able to add phones without having to edit any files.

...UID included for Redhat, but missing for Debian distros in install_prereq script (Reported by Rusty Newton) * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR variables for subsequent records (Reported by zvision) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim) * ASTERISK-23336 - Asterisk warning "Don't know how to indicate condition 33 on ooh323c" on outgoing calls from H323 to SIP peer (Reported by Alexander Semych) * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set to minrate=...

...UID included for Redhat, but missing for Debian distros in install_prereq script (Reported by Rusty Newton) * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR variables for subsequent records (Reported by zvision) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim) * ASTERISK-23336 - Asterisk warning "Don't know how to indicate condition 33 on ooh323c" on outgoing calls from H323 to SIP peer (Reported by Alexander Semych) * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set to minrate=...

...- [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response (Reported by NITESH BANSAL) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim) * ASTERISK-23336 - Asterisk warning "Don't know how to indicate condition 33 on ooh323c" on outgoing calls from H323 to SIP peer (Reported by Alexander Semych) * ASTERISK-23320 - Preloading pbx_config.so with a CustomPresence hint defin...

...- [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response (Reported by NITESH BANSAL) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim) * ASTERISK-23336 - Asterisk warning "Don't know how to indicate condition 33 on ooh323c" on outgoing calls from H323 to SIP peer (Reported by Alexander Semych) * ASTERISK-23320 - Preloading pbx_config.so with a CustomPresence hint defin...