similar to: chan_sip.c:5267 sip_reg_timeout Error

Hello, I have a problem with an LAN-Server behind an NAT-router. Asterisk Version 1.2.1 or 1.2.2 doesnt matter 10 minutes after starting Asterisk I loose all registrations at external SIP-proxys. The reason seemed to be that Asterisk send every second an request to every sip-proxy "Request: OPTIONS sip:sip.domain.tld". Every request is responded by the sip-proxy. After some minutes

Hello, I?ve been using Asterisk 1.2.18 for a while, and today, with no apparent changes, I started receiving these messages: Aug 31 13:26:57 NOTICE[27528]: chan_sip.c:5495 sip_reg_timeout: -- Registration for 'user at sipserver' timed out, trying again (Attempt #19) All trunks and extensions went to: sipserver:5060 user 120 Request Sent 011

Hello, We have one of our SIP provider that's is sending incoming sip call without need of registration. Incoming call working fine (as outgoing call), but * still try to register to there sip gateway : chan_sip.c:3159 sip_reg_timeout: Registration for 'phone@50.50.50.50' timed out, trying again -- Got SIP response 404 "User Not Found in data base" back from 50.50.50.50

I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten => 777,1,Answer exten => 777,2,Wait,2 exten => 777,3,DeadAGI,a2billing.php exten => 777,4,Wait,2 exten => 777,5,Hangup I am using 777 as the calling card

I am using DIDx.net as my DID provider but they don't seem to get their act together. A lot of times the phone numbers don't work. How can provide my own DID, my asterisk server is being hosted at a Data center and has a reliable vendor that does my termination and do SIP to SIP and have no T1 channels. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I successfully obtained the Asterisk code and extracted them into /usr/src. When I make and install asterisk, zaptel, libpri etc. Are they supposed to move automatically into their respective directories? I cannot find: /etc/asterisk/ /usr/lib/asterisk/modules/ /var/lib/asterisk Do I have to manually create them or this is failed install? Thanks. -------------- next part -------------- An

I am using Asterisk and A2billing Calling Card Platform and after the 6th call the quality starts to degrade. The way it set up is the user calls into the system then dial out so I have 12 channels being used up but 6 active calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running Linux Fedora 6, Pentium 4 Hyper-Threading, 64 bit, 1GB of RAM, 80 GB Sata Drive, bandwidth 4 Mbps

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is james@james.com.... that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate

Hi all, I tried searching, so if this has already been discussed please point me in the right direction. On separate occasions I've seen cases where Asterisk boxes will be unable to register with each other via SIP or IAX2 when a Firewall is replaced. I'll describe two different cases. In both we have three offices connected via IPsec tunnels. Case 1: High Availability firewall

Just setup *, got a developers kit FXO where the incoming/outgoing pstn is plugged in. I've then got an IAXy that is plugged into a Philips DECT phone. * is setup so that the [bell] section rings the phone - exten => s,1,Dial(IAX2/myuser,30) What's happening when someone calls my number is that the phone rings 3 times before you hear the dect phone ringing. Anybody got any ideas as

All, I signed up for a broadvoice BYOD plan over the weekend (very excited about their offering) and after about an hour I had asterisk registered and was making in and out bound calls. However, the next day (without changing anything) I couldn't call in or out and haven't been able to get it going again. I can connect using a softphone (X-Lite) and make calls in and out

Hi list! Another day, another problem... I'm checking with Nagios my Asterisk at home, and since yesterday I noticed that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours, so that I have a new IP every day), the peer of an VoIP-provider I use is UNREACHABLE. Yesterday I though it was a problem on the line, but today is the same, so I think it is something other...

Hi! I am new to Asterisk too, I got the similar problem and I would like to know how to get * to work behind NAT. When I have the SIP Debug turn on, I got the error 479 from FWD when * try to register with FWD, it looks like * is using the local IP (192.168.x.x) in the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content, but it does not seems to make Asterisk aware the

Still getting the same error. Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request) please advise anyone!!!!!someone!!! jai

I figured it out. The ftp site was not named well and corrected. The other problem I have it after the extraction and make; it was suppose to go under /etc but that did not happen. I am trying to figure out why. On 12/28/07, broadband Voice <broadbandvoice at gmail.com> wrote: > > I successfully downloaded the Asterisk package from Digium but not able > tar zxvf zaptel-*.tar.gz.

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >

Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long