similar to: Problems with IAX

Need some help with IAX trunking. I've got six systems: AsteriskM (main) ___________________|____________________ | | | | | Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5 AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk boxes are using ztdummy for timing, they are all using IAX trunking. My calls come in

I feel I'm doing something obviously wrong here and will kick myself when I see the answer!!! The scenario: SIP phone -> Asterisk1 -> IAX -> Asterisk2 -> IAX -> Asterisk3 -> PSTN So I place a call from the SIP phone. A1 picks it up and forwards it to A2 which forwards it to A3. A3 sends the call to the PSTN. I control A1 and A2, but not A3. When a call fails (for

Hi, I'm setting up multiple asterisk servers and trying to do the classic DIAL(IAX2/asterisk1/${EXTEN}&IAX2/asterisk2/${EXTEN}&IAX2/asterisk3/${EXTEN},15) After googling a bit, I fell on a discussion about putting this in a variable so that adding additionnal servers would be easy. I can't seem to find the link anymore, but it went something like this: extensions.conf:

Hi I have the following configuration | UA1 --|------ asterisk1 -----------------------+ UA2 --|------ asterisk2 -----------------------+ DB UA3 --|------ asterisk3 -----------------------+ UA4 --|------ asterisk4 -----------------------+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register

Hi, I am designing a asterisk load balancing model as follow. There are 3 asterisks connected to a single DB and a single server storing all the configuration file and voicemail. Round Robin DNS will distribute the request to asterisks. DNS round robin ---+ asterisk1--------------------------+ DB and file server +---asterisk2-----------------------+

Hi, I am using iax2 trunks between asterisk servers and am having a callerid problem. We are using realtime sip clients distributed between multiple servers. Only in test now but have run into a calleeid problem - the name of the called party is not displayed if the called party is on a different server, it works if the called party is on the same server. On each server sip clients show calleeid

I keep getting this error [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614 __allocate_taskprocessor: Unable to start taskprocessor listener for taskprocessor bbe08c34-9d1c-4e5f-8ae0-0cc75289caca [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245 default_listener_shutdown: pthread_join(): Cannot allocate memory [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614 __allocate_taskprocessor: Unable to

Hi, I recently configured Linux HA for Asterisk service (using Asterisk resource agent downloaded from link: https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk ). As per configuration it is working good but when I include "monitor_sipuri=" sip:42 at 10.3.152.103" " parameter in primitive section it is giving me an errors like listed below; root at

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channel bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward the call to

Hi all. Ive recently setup two Asterisk boxes (running Asterisk@Home to be specific), and Im trying to get a trunk going between them. So far I have tried a combination of IAX and SIP configuring them through AMP and writing the config files manually, but I cant seem to get calls going between the two. I have named each box asterisk1 and asterisk2. Does anyone have some working SIP and/or IAX

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

Hello group, I wanted to try out the asterisk iax protocol between two asterisk machines but have several problems with it. My scenario looks like follows. I am using asterisk 0.9.0 on both machines. SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2 Both SER and asterisk run on a machine with a public IP address. When the telephone on one side makes a call the telephone

Hello, My network is: Client_SS7_1-- -----------asterisk1------asterisk2 Client_SS7_2-- ? I receive a fax from Client_SS7_1 ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Then, asterisk2 forward the fax to Client_SS7_2 I want that the SIP signaling go to asterisk2, But, I need that the RTP don?t go

Hello, I've got a test setup with 2 asterisk boxes: Asterisk1 with: asterisk 11.5.1 dahdi 2.7.0.1 Digium TDM400 with 2 FXO ports Asterisk2 with: asterisk 11.5.1 dahdi 2.7.0 Digium TDM400 with 2 FXS ports Asterisk1 has the following AEL Dialplan: context remote { s => { Answer(); Dial(DAHDI/g1/7005); }; }; When a call from Asterisk2 comes in, it is correctly

Hi, I have 2 Asterisk systems and a unique scenario where I need to play a particular tone on Asterisk1 and identify the same tone on Asterisk2. Following is my call flow, Asterisk1(Plays audiofile1,Wait for 2 Sec,Plays a tone,Plays audiofile2) -> PSTN -> 3rd Party CONFERENCE SYSTEM <- PSTN <- Asterisk2(Record audiofile1,Wait for a tone,Record audiofile2). A few points to keep in

Hi!I've a question: I've 2 asterisk, I want pull the ethernet wire and then reconnect it after 5 second, using the VRRP protocol, where must I set the IP for the connection goes on the second asterisk? I want this: I call to asterisk1, then I pull the ethernet wire down, vrrp makes up the other asterisk but not the audio streaming...the callers are always pointed to asterisk1, but for the

Hi I am trying to make this setup work phone1---g729---asterisk1---sip---asterisk2---g729---phone2 I have tried several configurations but none worked I keep getting transcoding errors I have installed one g729 licence on each asterisk, but I can't verifiy because the show g729 command is not available, I use 1.2.17 Do I need 2 g729 licences per asterisk ? Do I need to register

Hi Kevin, no you didn't miss the reply and I've not resolved it yet. Have you got similar problems? Pamela Kevin Fjelsted wrote: >Pamela, >Did you resolve the problems you described? >I didn't see a reply on the list but I may have missed it. > >-Kevin > >-----Original Message----- >From: Pamela Weis [mailto:peawy@gmx.at] >Sent: Thursday, August 05, 2004

Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |