similar to: Incoming SIP getting in, but not ringing.

I've got a strange situation with musiconhold. It works if I dial my extension 6000: >From extensions.conf: exten => 6000,1,Answer exten => 6000,2,MusicOnHold() Debug output if I call 6000: -- Executing Answer("SIP/gs1-b6ee", "") in new stack -- Executing MusicOnHold("SIP/gs1-b6ee", "") in new stack -- Started music on hold,

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Could anyone outline a method for trouble-shooting these messages "SIP/2.0 482 Loop Detected" I'm seeing on a particular peer? There is no call going on when these pop up. Sip read: SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 207.149.XX.XX:5060;branch=z9hG4bK0a322471 From: "asterisk" <sip:asterisk@207.149.241.3>;tag=as395506d6 To:

Hi, I have researched more into the problem of my Asterisk set-up not answering calls. The following error was shown on the CLI, can anyone explain what the problem causing Asterisk to not answer the SIP calls be? Information: I have an Asterisk box on a home LAN, behind a D-Link router/firewall connected to a cable modem. The 82.x.x.x is the IP for my cable modem. 192.168.0.101 is my

Hullo :) I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from sipgate.co.uk to any other extension. My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind transfer, simply dial the number you want to transfer to, and press 'FWD'... This is what

Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of "8|." to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

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