similar to: ekiga sip registration fails; externip no help

I am experiencing a "606 not Acceptable" error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putting stunaddr=stun.ekiga.net into the sip.conf file under [ekiga]. I also tried

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

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

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

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

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

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

In SIP setting on the e71 I set the public user name as 1995 at 10.10.11.180. There is a sip.conf context [1995] On the asterisk CLI I get: Registration from '<sip:%201995 at 10.10.11.180:5060>' failed for '10.10.11.98' - No matching peer found So I changed the sip.conf context to [%201995] Then: [2009-11-19 20:44:28] WARNING[14371]: chan_sip.c:11797 check_auth:

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

I've just got Uverse installed. I had dsl, but ATT insisted I couldn't keep my old dsl, but had to switch to Uverse internet - vdsl. My setup: linux box as router : 10.10.11.252 asterisk box: 10.10.11.180 10.10.11.252 is multihomed and connected to the Uverse Residential Gateway. I've set it up as DMZplus, and it shows the public ip address as eth1. I can ssh into the

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:

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = <mypass> [obi202-aor](!) type = aor max_contacts = 2 ; ===== endpoints ======== [gv-voice](obi202-endpoint) auth = gv-voice aors = gv-voice identify_by=auth_username ;identify_by=username ; I also tried

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

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

greetings, using: CentOS 6.7 current KDE 4.3.4 ekiga 3.2.6 i am having problems getting ekiga to make any type of connection. i have gone thru documentation and troubleshooting manuals with out finding reason other than; ~]$ ekiga -d 4 2>&1 | grep "PDU is likely too large" ~]$ echo 3600 > /proc/sys/net/ipv4/netfilter/ip_conntrack_udp_timeout \ bash:

I'm having a problem getting a good clear output sidnal from Ekiga to a VOIP conference call using the Ekiga.net free conference call system. I'm told that each time I speak, my voice is clear & intelligible for about .5 - 2 seconds, but then it starts to be garbled, sounding like the sounds R2D2 makes. I've used 2 or three mic/headsets - two plug into my audio I/O sockets on my

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

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==