similar to: Asterisk and Kamailio NAT problem

Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten => _555,1,ChanSpy(Agent) exten => _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which

Hello I'm trying to use an AGI that returns the queues status (numbers of available agents, etc ), but I'm having some problems with it (it's still very buggy). Is there any AGI repository with source code samples? Had anyone used an AGI to check queues and agents status? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip:

Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:password at sip.startel.pt [startel] type=peer host=sip.startel.pt username=username fromuser=username secret=password qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm insecure=very port=5060 nat=yes canreinvite=yes The problem is: Asterisk is registering with its

Hello to all I'm using asterisk 1.4 and dahdi. I had everything working fine, and I could place calls through my R2 channel. But now the channel is always "RED" and Im getting this error message: TE4XXP: Version Synchronization Error! Here is my chan_dahdi.conf------------------------------ [channels] language=en context=incomingr2 signalling=mfcr2 mfcr2_variant=ar

Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone know exactly what settings needed to reach the asterisk server on my home network? Internet ->DSL transparent bridge ->router ->asterisk ->softphone x-lite attempts to login and register, but times out. There must be some setting I'm

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

I wonder if Asterisk's skinny/sccp channel driver could be used as a "client" to register with a Cisco PBX. That is, along with a SIP client, say, have Asterisk and said SIP client stand in for a Cisco phone, or an IP Communicator. Anyone done this? Cheers, b. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hey gang, I'm trying to find a) If you can put channel variables into a Call file and b) what the appropriate syntax is. Any ideas? Thanks, PB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090721/cb8c2656/attachment.htm

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!! <asterisk-users at lists.digium.com> --- (13 headers 16 lines) --- Sending to AA.BBB.CCC.DD : 28127 (NAT) Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk. Found user '701' for '701' Found RTP audio format 107 Found

Hello I just updated the kernel, dahdi-linux and dahdi-tools Im also using now asterisk 1.4.26.1 And im still with a red light (not RED/YELLOW anymore): [root at catumbela ~]# /etc/rc.d/init.d/dahdi status ### Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/ RED 1 PRI CAS RED 2 PRI CAS RED 3 PRI CAS RED 4 PRI

Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as person B hits transfer button again to transfer person A to C, the call is lost. But in the

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how

Hello I have a simple configuration to allow the admins to listen the agents calls: exten => _654,1,ChanSpy(Agent) exten => _654,2,Hangup() The problem is... even when the agents hung up... it seems the channels remain active: asterisk*CLI> show channels SIP/211-b3042018 654 at default:1 Up ChanSpy(Agent) SIP/211-b3fbf768 654 at default:1 Up ChanSpy(Agent)

2017-02-16 14:27 GMT+01:00 Joshua Colp <jcolp at digium.com>: > On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote: > > Hello, > > > > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP > > hardphone. > > > > When a phone has enabled this feature, it would send a SIP PUBLISH to its > > SIP Server letting this server dispatch to

Hi All, I've been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not. I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting "directmediadeny|directmediapermit" to offload media from asterisk for peer-to-peer calls BUT what if someone wants to record a call or engage some feature-code ?

Hi, I want to have Kamailio in front of one or more Asterisk boxes. I don't think it is necessary for Kamailio and Asterisk to register with one another. I'd like for PJSIP to recognise Kamailio by its IP address. I have two boxes, both have public IP addresses, they also have private IP addresses and can communicate with each other. I have a Snom phone accessing Kamailio via its

Hello, I put together a new tutorial about asterisk realtime integration with kamailio (openser). This time the database used is the one of asterisk, also call routing logic is controlled by asterisk, here is the link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Practically is an easier way to scale starting from existing asterisk installations. The other

Works for me.... Thanks, Hristo Benev -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc Sent: Monday, May 17, 2010 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be