search for: retransmitting

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

...= 89 Apr 23 12:25:02 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:03 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:25:09 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 2 now, updating n_r! Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 3, but we have nothing -- resetting! Apr 23 12:25:23 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:26 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:39 W...

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 sa...

...up to 10 channels at once with *no errors*. Above 10 channels, I start to get many (several per second when running 30 channels) framing and "read" errors, with text similar to the following: WARNING[1167272128]: File chan_zap.c, Line 5670 (zt_pri_error): PRI: !! Got reject for frame 26, retransmitting frame 26 now, up_dating n_r! (repeating for each error several times, with ascending retransmitted frame numbers) and also, less often: WARNING[1167272128]: File chan_zap.c, Line 5670 (zt_pri_error): PRI: Read on NN failed: Unknown error 500 (NN is the channel) MY SETUP: Tyan S2723, with dual X...

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, Excl...

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

...e 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 the 491 Response. Asterisk replies with the following 491 response: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 10.110.7.89:5070;branch=z9hG4bK5a668c33f196837c3602266b23b389e0;received=10.110.7.89 From: <sip:30001 at 10.110.7.20:5070>;tag=SIPTester To: <sip:30008 at 10.110.7.20>;t...

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

...wards: 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: <sip:6615xxxxx at 130.117.xxx.xxx;user=phone>;tag=as2e12c79...

I agree with the in-effective of TCP transmission, but I wonder if the the UDP packet is dropped, the tinc VPN itself wouldn’t retransmit, and if the upper level application doesn’t handle the packet loss well, will this be the problem? Or the upper level application have very limited tolerance to packet loss(like RDP application, I guess if the packet loss go to certain threshold, the connection

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

...ly 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 <--> alternative SIP server <--> Asterisk <-NAT-> External peer The alternative SIP serve...

...problem rises in the same way even when people with the shoretel telephone call the conference on the asterisk server, so we can cut out the sip phones from the general picture. I did some SIP debugging and this is basically the problem...: Reliably Transmitting (no NAT) to 192.168.235.6:5060: Retransmitting #1 (no NAT) to 192.168.235.6:5060: Retransmitting #2 (no NAT) to 192.168.235.6:5060: Retransmitting #3 (no NAT) to 192.168.235.6:5060: Retransmitting #4 (no NAT) to 192.168.235.6:5060: I did various amount of empirical testing like forcing nat, setting the shoretel peer as insecure, turning off...

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

...gets 100% of all available CPU (user CPU) for 10~20 seconds and then it keeps about 80% CPU usage.... Thas's just a portion of the asterisk log: -------------------------------------- Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: !! Got reject for frame 67, retransmitting frame 67 now, updating n_r! Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: !! Got reject for frame 67, retransmitting frame 68 now, updating n_r! Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: !! Got reject for frame 67, retransm...

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

...plication/sdp Content-Length: 256 v=0 o=root 27452 27452 IN IP4 10.2.0.2 s=session c=IN IP4 10.2.0.2 t=0 0 m=audio 13256 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Retransmitting #1 (no NAT) to 10.2.0.203:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 From: "Jim Felderman" <sip:1101 at 10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29 Call-ID: 001d45b6-1d4900...