similar to: registration not timing out?

Hi All, I am having problems with some SIP peers. I seem to loose registration. If I reload SIP the registration comes back. They usually stay registered for about 2 days before they drop. The problem is not all of them drop usually just the list 2 in the list. The other strange thing is that the 2 the do drop their registration do not occur at the exact same time. It could be many hours

Hi all I am trying to figuure out how to get iconnecthere incoming calls to work outbound works fine but incoming goes nowhere but to my iconnecthere vocemail if I do a sip show registry it shows up as regg'ed nnn=is my iconnect here number xxx is my secret Thank you Jeremy [general] qualify=no register=NNNNNNNNNNN:XXXX@iconnecthere/NNNNNNNNN context=default bind = 0.0.0.0 port=5060

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hi, I've been very puzzled lately. I installed a phone system for a friend a few weeks ago, and they're having a problem that I can't get rid of, actually 2 problems. Before I go into the problems, let me tell you about the setup. It's a pretty small setup with only 4 handsets, all Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual core, 2GHz) and 512MB Ram.

Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|

I just got a TE110P up on an XO PRI - everything looks good so far. We've been given a block of 23 numbers for the PRI. If I explictly set the incoming extension in extensions.conf like: exten => 1153,1,Answer or: exten => _XXXX,1,Answer I can get the incoming call. If I try and do: exten => s,1,Answer I'll see something like this: -- Extension '1153' in context

Hi This is the output from show dialplan dial-sipmnf-sippt-pstn [ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ] 's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config] 2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config] 3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]

I had some complaints today that one of my incoming SIP numbers was failing for several hours. I looked at my console and didn't see anything unusual but SIP show registry confirmed that my registrations were in a failed state. I did a SIP reload and saw this in the output: Sep 22 18:54:48 NOTICE[9997]: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 312896!

My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

The best place for Astri Con 2006 would definatly be Omaha, Nebraska! ;) very central ...ah one could hope. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

Hi, I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on "Accept" button. Greg -------------- next part -------------- An

Greetings I have been running * for about a month now. Configuration. (5) Cisco 79xx IP phones (1) XP100P Pentium III (300mhz) 192meg memory Redat 8.0 (updated) It seems to run for about 3-6 hours, then the process stops. I have noticed, that * does not stop, if I do NOT have it register to other sip servers. (FWD and PCH). Here is are the last few lines in the /var/log/asterisk/messages

I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is

Hi, we a have a SER (OpenSER) in front of 2 real-time Asterisk. SER simply forward SIP messages to 1 of the Asterisks: UA --> SER --> Asterisk We have a problem with REGISTERs: Asterisk answers with 200 OK, but changes the Contact header, inserting the IP of SER instead of the original IP (the IP of the UA). It seems that performs a sort of NAT-traversal, but all the elements are on

I just upgraded my test Asterisk box to the latest CVS HEAD. "show version" only shows "Asterisk CVS HEAD built by root....etc", with no date or version number. I downloaded this version on Monday, Oct 3. About once every minute, I get this while at the CLI> prompt: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 1! This only appeared

Hi all, i have a question: what about a CTI implementation with Asterisk. I've been looking for info in www.voip-info.org <http://www.voip-info.org/> and in google, but There are no precise informations! Thanks a lot stefano -------------- next part -------------- An HTML attachment was scrubbed... URL:

God save this mailling list :) Which is the best settings for the best quality of Audio ? I'use isdn4Linux driver and SIP client but bad quality PSTN <-> Asterisk and SIP <-> PSTN