search for: __transmit_response

...24-d7000024-5d2ca17a-29c81cf4@65.204.176.54 for seqno 1 01 (Response) Jan 27 21:54:12 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 21:54:14 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unabl e to determine sequence number from '' Jan 27 21:54:18 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unabl e to determine sequence number from '' Jan 27 21:54:22 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unabl e to determine sequence number from...

...calls work ok all the time. The Asterisk box is behind NAT so that does complicate things slightly. However, the Iconnecthere PCPhone client software works perfectly for incoming calls. When an incoming call is attempted Asterisk displays the following: Warning [131081]: chan_sip.c Line 1991: (__transmit_response): Unable to determine sequence number from '' A second or so later the caller is directed to the iconnecthere voicemail service. Is anyone else experiencing a similar problem, or does it just work? Any hints or tips would be welcome :-) Thanks, Nathan.

hello to asterisk start WARNING {98311] : File chan_sip.c, line 388 (retrans_pkt) : Maximum retries exceeded on call xxxxxxx...xxxxxxxxxx@192.168.0.1 for seqno 102 (Request) with a call from x-lite 38113@fwd.pulver.com WARNING {98311] : File chan_sip.c, line 2002 (__transmit_response): Unable to determine sequence number from '' and x-lite hang up the second warning is new since morning and i have not made changes from yesterday thanks

...imum retries exceeded on call 3c2706bad222-n2s56u19hj1l@129-82-44-226 for seqno 1 (Response) WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 3c2706bad222-n2s56u19hj1l@129-82-44-226 for seqno 2 (Response) WARNING[1142106560]: File chan_sip.c, Line 2329 (__transmit_response): Unable to determine sequence number from '' Thanks Tony

...omplains about: > ------------------------------------------------- > WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries > exceeded on call FBD1EC34-9FD3-4855-9DC1-97933EE0F5F3@192.168.1.104 for > seqno 43 (Response) > WARNING[81926]: File chan_sip.c, Line 2002 (__transmit_response): Unable to > determine sequence number from '' > ------------------------------------------------- > > > > Has anyone had the same problem? > > Senad > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@...

...nd this is the output corresponding to the hangup: ##### Jul 30 12:26:22 WARNING[11276]: File chan_sip.c, Line 417 (retrans_pkt): Maximum retries exceeded on call 4a509c9e-c0af-94d5-2cf8-d9a65493ad21@192.168.251.26 for seqno 102 (Request) Jul 30 12:26:27 WARNING[11276]: File chan_sip.c, Line 2156 (__transmit_response): Unable to determine sequence number from '' ##### Any suggestion? I'm quite disoriented in this moment, i initially tought that was a problem due to the "t" command for the Dial app, (somewhere it played "pbx-invalid" and hung up), but that part is now solved......

Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available fro Asterisk. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

...essages in * and the log on the Wellgate shows it doesn't get past inviting. Asterisk: WARNING[65545]: File chan_sip.c, Line 432 (retrans_pkt): Maximum retries exceeded on call 59dfc0-a010103-13c4-3f559705-154cf8-6f0e@10.1.1.52 for seqno 1 (Response) WARNING[65545]: File chan_sip.c, Line 2220 (__transmit_response): Unable to determine sequence number from '' Wellgate: 1-RVSIP_CALL_LEG_STATE_INVITING The funny thing is that if I comment out the above section and reload *, I now can call from Wellgate to * but not the other way around of course as the Wellgate isn't registered anymore. Are ther...

Hi all: This is a warning that appears when * loads. WARNING[1074494336]: File app_voicemail2.c, Line 2889 (load_module): SQL init VoiceMail2 tries to bring up VM SQL service? Regards, Gus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031014/3bad51d0/attachment.htm

...w Polycom Soundpoint IP 500 to test out in Asterisk environment. So far it has been good. Call Hold, Transfer, DMTF etc. However, I do notice every now and then the Polycom fails to register with Asterisk. Asterisk console outputs the following: Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable to determine sequence number from '' Feb 3 13:02:34 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to authenticate user "DavidLiu" <sip:DavidLiu@192.168.0.254>;tag=9F67E426-59D92ED7 Feb 3 13:02:36 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to au...

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured