search for: uridi

...) [in] exten => uxbod,1,Dial(sip/uxbod,20) exten => uxbod,2,VoiceMail(5001@incoming) exten => uxbod,3,Hangup() [outbound-local] exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}) exten => _9NXXXXXX,2,Congestion() exten => _9NXXXXXX,102,Congestion() [uri] exten => _[a-z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) exten => _[A-Z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) exten => _X.,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) [macro-uridial] exten => s,1,NoOp(Calling remote SIP peer ${ARG1}) exten => s,n,Dial(SIP/${ARG1},120,tr) exten => s,n,Congestion() When I try dialli...

...t;> >>> I use outgoing URI-dialing for my sip-phones as suggested in >>> http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial >>> >>> The relevant extensions look like this: >>> >>> [dial-uri] >>> exten => _[a-z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) >>> exten => _[A-Z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) >>> exten => _X.,1,Macro(uridial,${EXTEN}@${SIPDOMAIN}) >>> >>> [macro-uridial] >>> exten => s,1,Set(dialuri=${CUT(ARG1,\;,1)}) >>> exten => s,n,Set...

hi all i've just setup an * box and want to test voip calling, initially from sip user to sip user... local sip users can call each other, no issues. problem arises when i try and call a remote sip account, my * box always returns "SIP/2.0 404 Not Found" any ideas ?

Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex -- There is no instance of a country having benefited from prolonged warfare -- Sun Tzu - The Art of War -------------- next part -------------- A

Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian

Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic