search for: retransmits

Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc32ed at 192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaaec88 at 192.168.1.95 for seqno 101

Hi, We have a Ericsson BP250 Phone system setup witht he following configuration Telco <-> Asterisk E405P <-> BP250 The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded. Currently running Asterisk 1.2.4 Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next

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, What does retransmitting NAT means? I have a client that uses SPA 942, and his phone sometimes cannot be called. i did a sip sebug and i keep on seeing retransmitting NAT. on the realtime it shows that it is registered, so when i try to call it , asterisk thinks it is still online so it tries to reach it instead of saying it's unavailable, [Oct 9 11:10:33] -- Called 103100 it

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

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 have an asterisk system in Costa Rica that connects to a SIP provider in Atlanta. Sometimes SIP packets seem get dropped or retransmitted too quickly. In trying to debug this I turned on SIP debug in Asterisk and the SIP provider enabled packet capture on his end. What I saw was me sending an invite, them sending a 100 Trying, me sending a cancel, me sending a retransmit of the cancel, me

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

I am running a Xen HVM guest with netfront PV drivers. This is running SLES10 SP3 inside the guest. The Dom0 is also SLES10 SP3. Now I am trying to communicate from that HVM guest to a Windows Visa or also Windows 7 machine and I am getting really poor TCP performance. When tracing on the network traffic, I can see that no packets are dropped or missing or anything, but what happens is that the

...act: <sip:800902 at 82.158.83.xxx:5062;user=phone> Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream HT-502 V1.1C 1.0.1.57 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 >>> Asterisk does not recognize and retransmits <-------------> --- (12 headers 0 lines) --- Retransmitting #1 (NAT) to 82.158.83.xxx:5062: SIP/2.0 200 OK Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1226703311;received=82.158.83.xxx;rport=5062 From: "800902" <sip:800902 at 130.117.xxx.xxx;user=phone>;tag=467506068 To...

...t; >> If the concern is more about the reliability instead of throughput, should I add TCPonly = yes in the host configuration to make the VPN runs on TCP? > > The problem with TCP, is that TCP, encapsulated inside a TCP stream, is a recipe for very poor performance, as you could have retransmits, encapsulated in retransmits. > > But then the questions might be more like: Have you read up about why VPNs over TCP isn’t a good idea? > And since you have, what reliability issues are you having with tinc over UDP? > And if you have those reliability problems over UDP, what tests h...

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

...This is happening after: - call is setup, 2 way audio - call can function correctly for up to 5 minutes, with the external provider re-inviting every 1 minute When the problem happens - external peer re-invites asterisk - asterisk sends 200 OK - external peer sends ACK - asterisk retransmits 200 OK - external peer sends ack - .. - asterisk retransmits 200 OK (Retransmitting #6) - external peer sends ack - Asterisk logs the above message about maximum retries exceeded, and sends BYE to the inside SIP UA. The network configuration is as follows: phone <--> alterna...

Hello everybody. I have a problem with an integration between an Asterisk (1.4.24.1) on FreeBSD 7.0 and a Shoretel 7.5 server. To make a very long story short, when someone behind asterisk call an extension behing shoretel everything work as expected. When someone behing the shoretel server call someone behind asterisk the first 10 seconds of the call seems ok but then the line is dropped

Hi, I'm currently investigating a problem with our windows 2008r2 guest on centos 6 hosts. The issue is that the windows system sometimes sees a SYN packet for a tcp connection but doesn't respond. Three seconds later the retransmitted packet arrives and this time windows decides to proceed normally with the connection. This is with the virtio drivers but I have now switched to the

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

Hi, I have set up the latest code * server, and I'm having problems with the sip phone. I'm using the budgetone. I can receive calls just fine. When I try to place outgoing calls, I can see that the rules get followed fine. But once it dials the analog interface (X100P), it starts retransmitting packets in chan_sip.c:retrans_pkt() any ideaas why this would be? thanks TJ

Hello list, I am confronted with the following problem : making a call only leasts for about 30 seconds, then the call is ended. The CLI shows : [May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 954539948-5060-2 at 192.168.1.100 for seqno 11 (Critical Response) -- See doc/sip-retransmit.txt. [May 21 14:31:50] WARNING[25345]:

se?ores alguien le ha presentado este problema al acceder al voicemail o al hacer una llamada a la pstn 1940> Playing 'vm-received' (language 'es') -- <SIP/111-08d91940> Playing 'digits/yesterday' (language 'es') -- <SIP/111-08d91940> Playing 'digits/at' (language 'es') -- <SIP/111-08d91940> Playing

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