similar to: Call terminated with error message logged

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

We have IAX2 peer between two asterisk and I am getting following error following IAX2 WARNING. IAX calling is functional [May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3030332, seqno=211) [May 10 15:23:44] WARNING[2047]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded

With todays CVS head I am getting the following being sent after a call has been terminated on my Cisco 7960. It eventually gives up with a critical error. chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission 000b46a0-8661000a-4405e325-7e25031f@192.168.123.20 for seqno 102 (Critical Response) Any ideas I am sure it was working ok with cvs head a month ago. Chris ---- sip

I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows "Call answered" when dialing *8 while the

hi i got TOS and retranssmission error on receiving SIP call chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission 10CAED68-0F1D-DF82-DA1E-A76C1CB3D8A3 at 172.18.100.72 for seqno 43156 (Critical Response) -- See doc/sip-retransmit.txt. [May 22 13:42:44] WARNING[18021]: chan_sip.c:2821 retrans_pkt: Hanging up call 10CAED68-0F1D-DF82-DA1E-A76C1CB3D8A3 at 172.18.100.72 - no reply to

Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI> -- Executing Dial("SIP/16008-3d17", "SIP/16007&SIP/16006|20|tr") in new stack -- Called 16007 -- Called 16006

I'm having some trouble with placing some iax calls over an openvpn: Setup A is a 1.8GHz Celeron, T100P attached to a Zhone Zplex. Setup B is a 266MHz P2, T100P attached to a Zhone Zplex. Setup C is a 700MHz P3, T100P attached to an Adtran TA 750. Setup D is a 233MHz Pentium, with an X100P. Setups A and B are on the same physical network. IAX calls routed between them work fine. Setup D is

Running * with default config files except for sip.conf. Any call made is dropped 5 seconds after connection, with the following messages: Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 6C94C1B1-77C4-11D8-91FB- 000A95DA04DA@192.168.1.152 for seqno 48221 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7' Mar

Hi, My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on. I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box. If I try to connect to it from outside I get this error : Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum

I have an issue where * tries to route a call over IAX to another server even if the server is down. I have included the relevant entries from my iax.conf, extensions.conf, and some debug output. If someone could tell me what I have configured incorrectly, I would appreciate it. Thanks, Stephen -----------iax.conf on voip2---------- [voip1] type=friend username=voip1 host=x.x.x.x (ip

Is there a way to increase the number of retries or the time to help with this? WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 0ea2761d6a82fa49221f547c739bde18@192.168.0.200 for seqno 103 (Request) WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 0ea2761d6a82fa49221f547c739bde18@192.168.0.200 for seqno 103

Greetings, DIAX seems to work well placing calls, but I can't actually receive a call . Here, DIAX (x305) "registers", then I use a sip phone to place a call to DIAX (which definitely is not in use by me at debug time, but it is idle on my desktop.I think), and then * goes to vmail. Here's the debug output: affinity*CLI> iax debug IAX Debugging Enabled Rx-Frame Retry[N/A]

Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call 6010532c6fedf9be383872e07e4be70c@192.168.1.2 for seqno 102 (Request) I'm running asterisk with a Cisco 7960G If anyone know's why i'd get this.....Any help would be appreciated!

For whom asked me support for capi devices, that's here: http://www.junghanns.net/asterisk/ I'm using a AVM B1 card. also AVM passive card (FRITZ!PCI) works.... Then is you use SuSe all is configured by yast... Hello, probably is a feature what I'm asking for but because of my inexperience to asterisk this is my question: I've configured CAPI ISDN to receive calls. When I

We're trying to get a SIP peer going between our asterisk box and our provider. It should then ring our phone. The call does come in and it does execute the extension in the dial plan. But the provider says they never get a 200 OK back and therefore they send another INVITE and then after a few seconds drop the call. Here's our setup: sip.conf [ngt-trunk] type=peer qualify=yes port=5060

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/dd10d5ef/attachment.htm -------------- next part -------------- Hello Today I observed this strange problem, as soon as I called from my SNOM IP phone (910) to CISCO IP Phone 7905G (810), I got following warning messages and call didn't connect. But after couple of minutes this

Whilst on a call, I'm getting the following... -- Started music on hold, class 'default', on SIP/phone3-a7d5 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '#' in context 'default' -- Playing 'pbx-invalid' (language 'en') ie - without anyone pushing keys - I hear the music on Hold - as does the

Hi, I've been running tinc for a couple of months and it's great, but I have a periodic problem which maybe you guys can figure out. I operate a 3-node tinc VPN, lets say A, B and C. A / \ B --- C The problem is that after a while, node C can't exchange data with node B. It works fine (ping and other traffic) for about 10 minutes, then fails. Here is some debug

Hi I am experiencing some weird behaviour. Calls get disconnected random. There is no error in the log files. Sometimes I can talk over 30minutes+ and it is fine. Just earlier I was only able to talk 2 minutes per session and get disconnected. All I hear when this happens is a fast busy. My set up is this: 8 * Grandstream Budge Tone 101. 4 * X100P cards. Compaq 1Ghz ML Server. I am running

Good morning, I am trying to use the Windows iax client. My iax.conf looks like this: [general] port=5036 bindaddr=10.1.3.111 bandwidth=high allow=gsm ; Always allow GSM, it's cool :) tos=lowdelay [pos| type=friend context=default auth=plaintext secret=pos deny=0.0.0.0/0.0.0.0 permit=10.1.3.0/255.255.255.0 host=dynamic defaultip=10.1.3.2 In the registrations dialog