similar to: SIP rejecting calls?

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:

hello ! My problem is: Astriks should create a connection to other members using a german Sip provider (www.sipgate.de). there are no problems with connections to: o Sip- Accounts o national phone numbers o mobile phone numbers but connections to international phone numbers DO NOT WORK (see the attached protokoll). The connection to international phone numbers does work when I directly use

When attempting to call one of the example numbers, like 17474745000, I only get "488 Not Acceptable Here". It works fine when I configure the softphone (Xten X-Lite) to use sipphone's server directly. Am I missing something? Here's my relevant config sections: sip.conf: in [general]: register => 17472442457:mypassword@proxy01.sipphone.com [sipphone] type=friend

Using Asterisk 1.0.9 When I try to make an outgoing call with SIP I get the message " 513 Message too big" back from SER. Any ideas what I am doing wrong? Debug below. SER and Asterisk are running on the same Server SER is on port 5060 Asterisk is on port 5061 In my extension.conf I have the line SERADDRESS=192.219.85.57:5060 in Globals and am using exten

Hi, I am new to Asterisk and am runing asterisk 1.2.9.1 on an OpenBSD box. With a few manuals I was able to set up some SIP providers with which outgoing and incoming calls work. However, there is one provider with which inbound calls don't work at all. The only apparent error/warning message is this WARNING[13688]: chan_sip.c:3527 process_sdp: Unable to lookup host in c= line, 'IN IP4

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header

ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: <sip:test at ekiga.net>;tag=as64618445 To: <sip:test at

Ok, here's an odd one. I would have opened a bug on this but last time I tried that I got flamed.. :) Problem: When proxy requests digest challenge (SIP) Asterisk responds normally with the exception that for some reason it changes the FROM: (Also changes Contact: )to what's in the original TO: line. Why on earth is it doing this?! It must be a bug, I've gone over my extensions.conf

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

> Hello fellow asterisk people! > > I have Asterisk listening on port 5061 and SER on port 5060. > > Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. > > My problems are with SIP. I can make incoming calls from SIP to asterisk > and to any of the other networks, but when I try to make an outgoing call > from Asterisk to SER I see the following in

Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

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

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

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

is it possible to register asterisk in a sip proxy as if it were a terminal (like a cisco ATA)? how? Thanx Jenna ;) ___________________________________________________________ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar