similar to: IAX2 Trunk Problem

Hi I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk and they were working fine b'coz of bandwidth issue I changed from SIP to IAX now I'm facing a strange problem after some time on the cli of my asterisk box I see lots of messages of IAX2 trunk and b'coz of that my agents are not able to hear any thing and I've restart my * box. Please guide me what I

HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi this is my code for * manager: $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die("Connection to host failed"); fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret:

Hi, I've configured my exten.conf for few exten. But I'm curious to know how long can be my exten like (exten => XXXXXXX.....). Is there any limit for this or not. B'coz I've noticed one strange problem. I'm usnig snom300 as my hard phone to make calls. when my exten length is 14 then calls goes immed. without any problem but when I change length from 14 to 15 call goes but

Hi I've two servers : 1. UK 2. Pakistan Pakistan * server has ISDN30. Pakistan(ISDN30) <====> UK ===> User Im planning to setup an IAX2 trunk between these two server ? so , how much bandwidth I need for 30 simul. calls ? Im planning to use G729 on both my server ? to support 30 calls over IAX2 do I've to change some setting during compile time or not ? pls suggest.

Anybody have any issues running tdmoe on kernel 2.6+? I've got Suse 9.1 + 9.2 running 2.6.5 and 2.6.8 respectively, and when I enable dynamic spans between them, both boxes dump something similar to: Badness in local_bh_enable at kernel/softirq.c:141 [<c0120768>] local_bh_enable+0x48/0x60 [<c02952b0>] dev_queue_xmit+0x230/0x240 [<c02a0980>] eth_header+0x0/0x140

HI Here is my info: Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents this asterisk box is connected to another asterisk box using 5 IAX trunk to load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my cli start flooding with message: Maximum trunk data space exceeded even I've only 3 calls on my asterisk system. asterisk restart option don't work, my

HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 62222 asterisk passes this is ser and then again ser passes this no 2222 (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070603/6564c117/attachment.htm

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

Hi Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=0000000c, dsts=00000101) May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=00000101, dsts=0000000c) May 13 14:56:36 pbx2

On Thu, Sep 1, 2016 at 5:30 PM, <centos-request at centos.org> wrote: > Send CentOS mailing list submissions to > centos at centos.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.centos.org/mailman/listinfo/centos > or, via email, send a message with subject or body 'help' to > centos-request at

Hi I've configured the queue on my asterisk box and everything is working fine. In my queue I've 3 agents logged in the queue. When call comes they are able to receive the calls without any problem. But some time they are on break and there extension rings and no one is there to answer the call (we don't want them to log off from the queue) but we have one normal user in the same

Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding

Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun

Hi, in my dial plan I've configured two trunks to make outbound calls (one for national calls and other international). I want to allow only 2-3 extension to make use of my international trunk to make outbound calls so I want some kind of auth. based on their callerid . Please guide. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm running a php script to generate calls using Asterisk Manager and its working fine. this script call a specified land line number if the phone is answered then It will connect to an extension and play an IVR. But I see in Asterisk CLI its placing the call and it shows channel answered but I don't receive call on my land line and it starts playing the IVR. Please guide me how to

Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in

HI I'm looking for a card that support both PRI and TDM. Please suggest me ? thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070604/cb01d15d/attachment-0001.htm

Asterisk Project Security Advisory - AST-2008-011 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Traffic amplification in IAX2 firmware | | |