similar to: retransmitting NAT

hi all, just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext but from spa to pap2 i dont see it, i looked

Anyone know what would cause this error? !! Got reject for frame 39, retransmitting frame 39 now, updating n_r! !! Got reject for frame 39, retransmitting frame 40 now, updating n_r! I assume this would cause audio issues as well. Thanks, Steve > Message type: CALL PROCEEDING (2) > [18 03 a9 83 89] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP

I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:02 WARNING[131081]: PRI:

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hi, im trying to build a UAC and I'm coming up with some trouble whenever I receive a SIP 491 Request Pending Response. This happens because I try to place a call on hold using an Invite request rigth before Asterisk sends me a Re-Invite for the same call. I respond to the 491 response with an ACK however for some strange reason Asterisk doesn't accept the ACK and insists on retransmitting

This may be a newbie SIP/NAT question. If so I am sorry. But any help would be appreciated. My Asterisk server is behind an ipchains box and I am trying to connect to Broadvoice. All works fine without the NAT. I have a global nat=yes prior to my register, but the sip debug allows shows "no nat)". Is this "label" issue, and am I barking up the wrong tree? Sip.conf....

Hi All, I have a asterisk box that is now on its own static address on the net.it was originally behind a nat firewall. The problem I have is that the remote SPA-2000's that are behind nat firewalls now fail. here is relevent sip.con entry [2001] type=friend username=2001 host=dynamic defaultip=81.178.77.67 allow=ulaw dtmfmode=rfc2833 mailbox=2001@local context=sip

Hi all. I am having lots of trouble with random calls dropping after 20 seconds, and I finally managed to capture a full sip trace. I'll paste it in full below, but I'll give a summary first. It seems that Asterisk is not recognizing the ACK messages that it receives from the Grandstream ATA. This happens only on the ACK that follows the OK that marks a call as established. This makes

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2

I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs

Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register => 092779077:XXXX at 85.119.188.3 This the output of SIP show peers : asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status

Hi all- HELP! This is actually a revisit of a problem that I had earlier with E400P's at a customer site. Customer still gets locked up channel problem, but has learned to live with it (channels clear themselves after several minutes). The symptoms, which I believe are directly related: I'm having problems with tons of framing and "read" errors on my E1 connections (and

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Fedora: Linux abcde 2.6.11-1.1369_FC4 #1 Thu Jun 2 22:55:56 EDT 2005 i686 i686 i386 GNU/Linux Asterisk: 1.2.4 SIP Problem 1. Asterisk sends SIP messages to Softphone. 2. Softphone receives SIP messages and replys back. 3. Asterisk doesn't receive these replies and keeps on sending. Asterisk: Reliably Transmitting (no NAT) to 192.168.1.4:5060: OPTIONS sip:192.168.1.4 SIP/2.0 Via:

Dear List i have upgrade my * box with the latest CVS version of Asterisk Stable 1.0 and zaptel/libpri my system is MDK9.2 with 1 TE410P and seems work well for now but i have a little amount of traffic (25 IN/OUT calls) i only notice this Warning.. What kind of error is? ------------------------------- Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got reject for

Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729

Hi All, I posted this a couple weeks ago with no response, I'm hoping that someone will see it this time around and be so kind as to offer advice for resolving this issue (or point me in the direction of a better place to ask) "Some" (but not all) calls on one of our Asterisk boxes are being dropped in Voicemail -- only in voicemail -- after about 20 seconds with the error logged

I've been running the 8/1/2004 Head release up until a little over a week ago. I was forced to due to a card failure to upgrade to 1.2.16 without any advance preparation or testing (most of my connections are via satellite to all corners of the globe with high latency). Up until the upgrade I was running with very few issues. Since the upgrade I have been experiencing strange issues