similar to: cvs head and seqno 102 (Critical Response) messages for Cisco 7960

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

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Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros: Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call 000ded24-d7000024-5d2ca17a-29c81cf4@65.204.176.54 for seqno 1 01 (Response) Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call

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

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

Hi, I've got Asterisk configured and working (sort of) with an Eicon Diva Server 2M ISDN card (connected to S0 bus of another PBX). This * box is on a 'live', non-nat IP address. I also have a couple of budgetone phones, one behind NAT and one not. When I place an outgoing call, I get the following messages: -- Executing Dial("SIP/filbert-9876", "CAPI/288:333") in

Here's the problem my sipura 2000 is setup on Nat Network in my office and my Asterisk Server is setup also on Nat Network at home the sipura can register and get calls but no audio comes in and out of the sipura and when i dial local extensions on the sipura i get this error message. any suggestions on what i can try as work around. *CLI> NOTICE[1158921008]: File chan_sip.c, Line 5394

Hi, the following logs are being generated while i test sip-to-sip windows software phones. Dec 7 17:05:16 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call 40dedd1535853f17250b4d0854e35c17@200.75.243.237 for seqno 102 (Critical Request) == No one is available to answer at this time Dec 7 17:05:22 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum

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've seen other request info on this config before. I've got the system up and running, Zap/Modem has been working for several days, no errors/warnings. Simply added info for broadvoice and I can dial into my system via the remote, but dial out from my box renders the following: I get one ring on the remote line, then on the console chan_sip.c:675 retans_pkt: Maximum retries exceeded

We have an interesting issue: One of our providers has two softswitches. Calls coming from the first one are handled fine by asterisk, calls coming from the second one and going through the first one are euhm... dropped half a second into the RTP stream. I have opened a ticket at Digium for it: http://bugs.digium.com/view.php?id=10449 The output of "sip debug" is funny from line

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

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

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

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

Hi, I am testing a 7960 in this context: [SIP] --- > VPN ---> [*] ---> [ANY] (ANY == any type of phone: isdn, SIP, IAX, etc.) the call goes through and is dropped after 5 seconds with this message in the log: "File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call <IP address> for seqno 101" when calling from the LAN with the exact same phone:

I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 8f68421-22821e1e at localhost for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call 8f68421-22821e1e at localhost - no reply to our critical packet. any Ideas? Jason

HI all, Recently, I got the following message from CLI and finally the asterisk will hang. Anyone can tell me how to fix the problem or why it will happen. Thanks. Jun 17 14:18:02 DEBUG[24573] channel.c: Avoiding initial deadlock for 'SIP/1127-008d65f0' Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11337 sipsock_read: We could NOT get the channel lock for SIP/1589-0087cdd0! Jun 17

The following message occurs every two minutes in my log file when in diagnostic mode. Aug 22 13:58:00 WARNING[1133735216]: File chan_sip.c, Line 421 (retrans_pkt): Maximum retries exceeded on call 46afc0866a7987591248e29d299e47ee@10.1.1.63 for seqno 102 (Request) Aug 22 14:00:01 WARNING[1133735216]: File chan_sip.c, Line 421 (retrans_pkt): Maximum retries exceeded on call

Hello list! I only can make successful calls if I disable gsm with "disallow=gsm". As soon as I allow gsm the following appears at the console. There are much much more Lines with "File dsp.c, Line 1198" but I cut them for a better survey : --------- Log Start ------------- Asterisk Ready. WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on