similar to: SIP Registration Failure Logging

-uzzi PS: If you're not seeing any connection information, be sure to double-check the IP address is correct. Learned that lesson the hard way =\ On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg <jr at amanue.com> wrote: > Let's say I have two Asterisk boxes, A and B. I am trying to get A to do > SIP registration on B, so an extension for A can dial SIP phones covered by >

Hi list, I'm in central-europe and signed yesterday a broadvoice account. My Asterisk box is CVS 2005-08-25. Problem I face is: "Failed to authenticate on REGISTER to 'phonenumber@sip.broadvoice.com' (Tries 2)" then "Registration for 'phonenumber@sip.broadvoice.com' timed out" and finaly "Giving up forever to register

Hi All; Asterisk version is: 1.8.5.0 But I see at the consol the following warning and really I did google but did not understand if it is bug or related to settings: [Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468) for sip_reinvite_retry for dialog 3c581fa96f2b-53yysntgjmwb in handle_response_invite But actually, we see some SNOM IP

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

I had a SIP problem on an 11.5.0 system that I look after. It registers with a SIP trunk provider, and at one point the provider had an issue that caused registration to fail. The problem was that Asterisk did not keep retrying, and it was not until it was restarted that registration was re-established. Here are the entries in the full log: [Nov 5 21:19:12] NOTICE[3248] chan_sip.c: --

Anyone else seen this with recent CVS? Asterisk CVS-12/05/03-14:25:13 Dec 9 13:22:46 NOTICE[10251]: File chan_sip.c, Line 2981 (sip_reg_timeout): Registration for '^B@117.0.0.0' timed out, trying again Dec 9 13:22:46 WARNING[10251]: File chan_sip.c, Line 428 (__sip_xmit): sip_xmit of 0x81512ac (len 320) to 117.0.0.0 returned -1: Invalid argument Dec 9 13:33:07 NOTICE[10251]:

Hi Guys, i am using the following config in pbx1: register => pbx1:endopass at 172.16.200.175 <pbx1%3Aendopass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=password context=[default] deny=0.0.0.0/0.0.0.0 permit=172.16.200.175/255.255.255.128 in pbx2: register => pbx2:endopass at 172.16.200.176 <pbx2%3Aendopass at 172.16.200.176> [pbx1] type=friend

Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get "May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite:

I have 2 Polycom SP 500's attached to my system. Both are behind NATs, but both seem to work fine, for the most part. A few weeks ago, I started to notice that I get an error message from one of them: Feb 20 08:54:58 NOTICE[10663]: chan_sip.c:7691 handle_request: Registration from '<sip:Polycom1@xxx.yyy.93.254>' failed for 'zzz.aaa.103.75' However, trying to call the

Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATted over public network, with address 89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) Then I have configured an account as following: [999] type=friend

I've got a problem with a SIP Account I am trying to dial in with. The correct extension rings but when I pick up the call is not made and I get a busy signal. Dialing out works just fine - just calling this number doesn't seem to work. Any pointers? Thx Michael excerpt from sip.conf: [general] context=default port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0

Dear mailing list, I've a Asterisk 1.4.21.2~dfsg-3+lenny1 package installed on my debian and I've a strange behavior. After some days running normally, my asterisk is under heavy attack, however, there is nothing logged in the console (logging from debug -> error) or file (level from notice ->error) I can see that there is also a peak on the network traffic. My first guess is that

I have a Hitachi WirelessIP5000 phone (the original 1st gen hardware). It's running the latest firmware (v2.2.6) and my Asterisk server is running 1.2.10. This setup has been working great for me for a long time, and then last week I started having a problem where the Hitachi phone loses registration. It registers fine when I first power it on, but at some point later the display shows

We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to 999@pbx.itsp.com REGISTER attempt 2 to 999@pbx.itsp.com Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

Hi I'm having trouble routing a call between two A*k servers I admin. SERVER- A: has a simple extensions set, and just needs to Dial to server B, but authenticate as part of the dial: exten => 777,1,Dial(SIP/abc-777:mypassword at someip.no-ip.info:5071/777,40,trw) exten => 777,2,Hangup So that should pass the call to the server listening on port 5071 of someip.no-ip.info,

Hi, I have a problem on my server. To illustrate, I created a new user, unclewiggly, and set up a password. I then used sent an email from my usual account. Logging in with my browser I am able to see the inbox, the mail and contents correctly. I then set up a new instance of Eudora 7.1 on my windows 7 box with the appropriate login id and check the account using the correct password.

Apparently, the realtime system in asterisk is faulty. Implementing realtime, begins a host of seeding messages along with registration messages visible at the CLI prompt. This is not the case with .conf file configuration Unfortunately, it is not clear where the bug originates but is shows it's head while calling the register_verify function, (which there are 2 one in chan_sip.c and one in

a strange problem of multiple sip registrations and peer selection in sip.conf is calling for your suggestions!! let's examine this scenario: some numbers and passwords hidden with HHHs to protect the guilty :) I have 3 distinct sip subscriptions with cordiaip.net provider in US. For each of these i insert in sip.conf (with peer name differences and relefant number/password differences,