[Asterisk-Users] pbx or asterisk?
hi all i have a pbx siemens connect via E1 to my asterisk box. the asterisk box can call without problems to pbx extensions. but when y press the numbers form example 402 in the pbx phones asterisk give me this -- Saved useragent X-Lite release 1103m for peer 402 -- Going to extension s|1 because of Complete received -- Executing Playback(Zap/31-1, vm-goodbye) in new stack -- Accepting call from '' to 's' on channel 0/31, span 1 == Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1' -- Hungup 'Zap/31-1' -- Accepting call from '' to 's' on channel 0/31, span 1did not receive any number or i have miss configure somenthing in asterisk box? -- .- Pablo Allietti LACNIC ___ --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
Re: [Asterisk-Users] pbx or asterisk?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Looks like your zap channels are droping into the default context... better to set up a from-pstn context and start there. Pablo Allietti wrote: hi all i have a pbx siemens connect via E1 to my asterisk box. the asterisk box can call without problems to pbx extensions. but when y press the numbers form example 402 in the pbx phones asterisk give me this -- Saved useragent X-Lite release 1103m for peer 402 -- Going to extension s|1 because of Complete received -- Executing Playback(Zap/31-1, vm-goodbye) in new stack -- Accepting call from '' to 's' on channel 0/31, span 1 == Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1' -- Hungup 'Zap/31-1' -- Accepting call from '' to 's' on channel 0/31, span 1did not receive any number or i have miss configure somenthing in asterisk box? -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDjey8y9wPyZpnL2URAiVCAJ4hQCz+eb1/MaABy2gxUMOcMw1AMwCfYEJI VTt9lDiRDMLZhJ2aOL4Qpnw= =KqmL -END PGP SIGNATURE- ___ --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
[Asterisk-Users] pbx to asterisk
Good day all I just want to know if someone tried this and with out any hassles What I want to do is take 4 extension(analog) of a current,old,pabx unit and put them into a asterisk server with a 4port analog card,like the voicetronix openline4 card. (PSTN)(old PABX)---===(4 ports asterisk) Please Help Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pbx to asterisk
On Thu, 14 Apr 2005, Altus Snyman wrote: Good day all I just want to know if someone tried this and with out any hassles What I want to do is take 4 extension(analog) of a current,old,pabx unit and put them into a asterisk server with a 4port analog card,like the voicetronix openline4 card. (PSTN)(old PABX)---===(4 ports asterisk) Just make sure that hangup-detection on the Voicetronix will be compatible with whatever it is that the PBX does at the end of a call. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
