search for: gxp1200

...=> 80,6,Hangup When I call from one of my internal SIP-clients to let Asterisk play to me the file "/tmp/winkel-gesloten.alaw", which I recorded in alaw-format with the RECORD()-application, I get the following on the Asterisk CLI : -- Executing [11 at intern:1] Answer("SIP/GXP1200-086d6b88", "") in new stack -- Executing [11 at intern:2] NoOp("SIP/GXP1200-086d6b88", "CallerID : "callerid?" <51>") in new stack -- Executing [11 at intern:3] Playback("SIP/GXP1200-086d6b88", "/tmp/welkom-tcs.alaw") in...

...ion with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr = 0.0.0.0 [BT201] type=friend context=intern host=192.168.4.210 secret=testpaswoord [GXP1200] type=friend context=intern host=192.168.4.211 secret=testpaswoord extensions.conf : [root at asterisk asterisk]# cat extensions.conf [intern] exten => 210,1,Dial(SIP/BT201) exten => 211,1,Dial(SIP/GXP1200) Asterisk CLI shows me : asterisk*CLI> sip reload Reloading SIP == Parsing ...

...conHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on 'SIP/BT201-088f93e0' -- Executing [52 at intern:1] Dial("SIP/GXP1200-088f93e0", "SIP/BT201| 30") in new stack -- Called BT201 -- SIP/BT201-088faa00 is ringing -- SIP/BT201-088faa00 answered SIP/GXP1200-088f93e0 -- Packet2Packet bridging SIP/GXP1200-088f93e0 and SIP/BT201-088faa00 == Spawn extension (intern, 52, 1) exited non-zero on...

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

Hi folks, Am new to asterisk pbx systems. I am trying to figure out what to do, I'll list and folks feel free to give feedback and advice. MAIN purpose for usage: 1.exposure to setup an asterisk box 2.get home phone service via VOIP/internet connection. tasks so far ------------------ 1. setup and install asterisk (1.4.x) --> DONE -currently configuring sip.conf