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...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: gomespereira at startel.pt

...: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which Asterisk module do I have to enable? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira at startel.pt

...9656535 at 127.0.0.1 Cseq:: 103 User-agent:: Asterisk PBX How can I force Asterisk to register with its public IP? Is it possible to configure STUN in an Asterisk trunk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira at startel.pt

...xxx.xxx.xxx.xxx 5060 OK (890 ms) Is there something missing in my SIP.CONF to improve the compatibility with Kamailio? How can I debug the RTP stream in Asterisk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira at startel.pt

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

...- span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-30:1101 dchan=16 ------------------------------------------- What could be the problem? Why was this working fine and now the channel is RED? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira at startel.pt

...s recordformat=wav monitor-join=yes savecallsin=/var/www/html/recordings/ custom_beep=beep group=1 wrapuptime=19 ackcall=no group=1 agent => 600,1234,Jose agent => 601,1234,Maria Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira at startel.pt

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

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

...nSpy(Agent) SIP/211-b4ba9f88 654 at default:1 Up ChanSpy(Agent) (the agent is using extension 211) I have more then 10 lines like these. Why do the ChanSpy calls dont hang up? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira at startel.pt

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:

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

...ource VPM400: Not Present VPM450: Not Present Completed startup! DMESG -------------------------------------------------------- Could this be a configuration issue or a hardware problem? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira at startel.pt