similar to: binding asterisk-h323 on two interfaces

Hi, a partner, who exchanges voip traffic with my asterisk box, complains, that asterisk ignores hints about ports to use. Hints about ports to use, seem to be a feature of H323. (I'm not firm enough with H323 to verify this.) The remote party opens the media-in channel: remote-ip:port-A -> local-ip:port-B My local Asterisk-box uses the same channel for media-out: local-ip:port-B ->

hello, i have problem with oh323 channel driver (tryied differnet versions). when i try to make call between oh323 - sip, oh323-isdn, oh323-capi asterisk crash with segmentation fault. Channel driver was compiled with pwlib 1.5.0 and openh323 1.12.0 libs. Does anybody know solution ? WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Executing Dial("H323:31119",

HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling

on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif

I am using gentoo Linux and Asterisk CVS-HEAD-09/23/04-19:57. I have tested both KPhone and IaxComm for linux but receiving no audio from asterisk. sound is working fine, as I can listen playing files using PLAY or APLAY. KPhone is configured with DTMFmode=inband and codec is ulaw and IaxComm is configured with ilbc if somebody can sort out this Thank you regards, -- Atif

Hello there! Somebody tried the meetme|b which initiates the conf-background AGI. Actually I want to originate another call from a conference.my AGI originates the call and connects it to the conference, but the calleeee is nowhere My extension exten => 21,1,meetme(21|pb) and my AGI **************************************************************************** #!/usr/bin/perl -w

Hi all, I am configuring asterisk in a cable modem network, using a motorola TM401A. I can make calls from the MTA but I can receive, display the following error: -- Executing [1500 at alberti:1] Dial("OSS/dsp", "MGCP/aaln/1 at 0-13-11-82-bd-a.ssw.intercal.net|30") in new stack [Sep 2 00:10:53] NOTICE[28062]: chan_mgcp.c:3572 mgcp_request: Asked to get a channel of

hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application thank you regards, -- Atif

this is from my extensions.conf, the first three patterns are for toll-free numbers, and fourth pattern is for other numbers, where an AGI is called for authentication. now when I dial 011448000664327 if falls into the fourth pattern, where as it should be matched by the first pattern. Any suggestions 1 - exten => _01144800XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 2 - exten =>

Hi all, I have 2 question. First: I use dovecot (version 2.2.9) with mdbox mail format. When I run dsync tool with "mirror" or "backup" parameters my source and destination directory synchronize correctly but if I delete some messages in user mailbox, deleted messages does not synced to destination. For example : atif at domain.com path is /mail/domain.com/atif/ and its

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

https://bugzilla.mindrot.org/show_bug.cgi?id=2254 Bug ID: 2254 Summary: Better error message for globs that have too many results. Product: Portable OpenSSH Version: 6.6p1 Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5 Component: sftp

Hi Gurus, We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension. But if we dial the external DID number via this trunk from

My grasp of R and statistics are both seriously lacking, so if this question is completely naive, I apologize in advance. I've hunted for a couple hours on the internet and none of the methods I've found have produced the result I'm looking for. I'm currently a student in a Statistics class and we are learning the ANOVA. We had to do one by hand and then reproduce our work in SAS.

Hi, I am new to R. and even though I've made my code to run and do what it needs to . It is taking forever and I can't use it like this. I was wondering if you could help me find ways to fix the code to run faster. Here are my codes.. the data set is a bunch of 0s and 1s in a data.frame. What I am doing is this. I pick a column and make up a new column Y with values associated with that

During debugging of H323 trunk side (using Jeremy Macnamara's H323 driver in ~/channels/h323) a couple of things come don't quite work as advertised... 1/ the following line in extensions.conf explicitly sets the outgoing caller ID (required in my case for downstream GK processing..) exten => _1NX.,1,SetCallerID,6400047602100 exten => _1NX.,2,Dial,H323/${EXTEN:1} what

Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No

Hi, I am using the asterisk-oh323 wrapper and I am looking to allow registration of h323 endpoints and allow Asterisk to act as a gateway. The idea is simple: H323 endpoints would register with Asterisk. They each would have their own internal extension (like SIP). If a H323 endpoint dials an outbound extension, then the h323 call gets routed to a H323 Gatekeeper which then terminates

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing