search for: __sip_ack

I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '6dc8436c7c568eea75fffdc75478ed54@142.55.31.179' of Request 102: Found DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '02b25a634efc4410769f47653c152a71@142.55.31.179' of Request 102: Found DEBUG[1125329600]: File cha...

...ager registered action DBGet == Manager registered action DBPut == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> Sep 13 22:16:37 DEBUG[13167]: chan_zap.c:6294 do_monitor: Message status f or 0 changed from -1 to 0 on 4 Sep 13 22:17:33 DEBUG[13167]: chan_sip.c:1274 __sip_ack: Stopping retransmission on '6a7f127b0d47ebd168678c665f4d2365@192.168.0.17' of Request 102: Match Found *CLI> Sep 13 22:18:10 DEBUG[13167]: chan_sip.c:6350 check_user_full: Setting NAT on RTP to 0 Sep 13 22:18:10 DEBUG[13167]: chan_sip.c:9413 handle_request_invite: Checking SI P...

...e line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '2d57437068f8800f671168c74e01eae1@210.9.49.249' of Request 102: Found Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on ...

...y everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]: chan_sip.c:2426 sip_hangup: update_call_counter(3) - decrement call limit counter Apr 23 12:42:39 DEBUG[7625]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Apr 23 12:42:39 DEBUG[7625]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '1fd7824840123666030e29a70d1d7739@192.168.1.200' of Request 102: Match Found...

...server. When you call the phone it rings, answers, and then goes right to a busy signal. Any ideas? Thanks for your help! Jane At the console in verbose mode I get: *CLI> DEBUG[8501248]: File chan_sip.c, Line 663 (create_addr): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '63e5425660664f565ce2c88b2cdc4d51@ipaddressofasteriskserver' of Request 102: Found *CLI> DEBUG[8501248]: File chan_sip.c, Line 3898 (check_user): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '...

...ate_addr: Setting NAT on RTP to 0 May 26 09:49:02 DEBUG[3242]: chan_sip.c:1487 sip_call: Outgoing Call for 15111111111 May 26 09:49:02 DEBUG[3242]: chan_sip.c:1592 update_user_counter: 15111111111 is not a local user -- Called 15111111111@SIP_PROVIDER May 26 09:49:03 DEBUG[3227]: chan_sip.c:822 __sip_ack: Acked pending invite 102 May 26 09:49:03 DEBUG[3227]: chan_sip.c:840 __sip_ack: Stopping retransmission on '3eb3b5d102a12e1f57a33ac13235ac9f@82.101.145.169' of Request 102: Found May 26 09:49:03 DEBUG[3227]: chan_sip.c:872 __sip_semi_ack: (Provisional) Stopping retransmission (but retainin...

...figuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '746374551@10.0.1.5' of Response 4963: Found DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 NOTICE[81926]: File pbx.c, Line 1171 (pbx_extension_helper): Cannot find extension context 'from-sip' DEBUG[81926]: File chan_si...

...39; Jun 5 01:55:33 DEBUG[1236360496]: File channel.c, Line 1381 (ast_set_write_format): Set channel SIP/5010-d3c4 to write format 2 Jun 5 01:55:33 DEBUG[1236360496]: File rtp.c, Line 838 (ast_rtp_write): Ooh, format changed from 0 to 4 Jun 5 01:55:36 DEBUG[1158913328]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '7a655bae-29f17562-197726d5@62.212.12.21' of Response 26024: Found Jun 5 01:55:36 DEBUG[1158913328]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '7a655bae-29f17562-197726d5@62.212.12.21' of Response 26024: Not Found Jun 5 01:55:36...

...ery low traffic. The problem arises when I try to send a fax: the Asterisk server initiates the call and, after a few seconds, the Linksys hangs the call by sending a BYE message: DEBUG[7416]: chan_sip.c:11375 handle_request: **** Received ACK (6) - Command in SIP ACK DEBUG[7416]: chan_sip.c:1396 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #258 DEBUG[7416]: chan_sip.c:1407 __sip_ack: Stopping retransmission on 'e793a727-98582c99@192.168.6.222' of Response 102: Match Found <-- SIP read from 192.168.6.222:5060: BYE sip:9021111111@192.168.6.220 SIP/2....

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

...Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 400 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 410 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 567 (__sip_ack): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 567 (__sip_ack): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File cdr_addon_mysql.c, Line 123 (mysql_log): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[123437984...

Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '117801284512845hUxv-9991110061--17708185305@63.201.117.76' of Response 20587: Found Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_...

...available to answer at this time WARNING[1209269552]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but no rule 't' in context 'sip' DEBUG[1209269552]: File chan_sip.c, Line 1025 (sip_hangup): find_user(phone1) - decrement inUse counter DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '3c29efbbc5b1-diw483wrl88j@10-1-2-24' of Response 1: Found On FWD I get the following DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1 D...

...ot;) in new stack -- Executing SetVar("SIP/2203-751b", "MONITORDIR=/var/spool/asterisk/monitor") in new stack DEBUG[30737]: File app_macro.c, Line 166 (macro_exec): Extension s, priority 1 returned normally even though call was hung up DEBUG[7176]: File chan_sip.c, Line 503 (__sip_ack): Stopping retransmission on '15125295754753e9727742e77c3d1c51@10.0.1.10' of Request 103: Found *CLI> Output when destination side ends the call first (this works as it should): (normal call setup and progress not shown - I show everything after hangup) *CLI> DEBUG[31761]: Fil...

...ve a T1 card and a dual proc box that has been running just fine up till this weekend. I tihnk some of the format changes affected my install. Jun 27 16:12:38 DEBUG[262161]: File chan_sip.c, Line 612 (create_addr): Setting NAT on RTP to 0 Jun 27 16:12:38 DEBUG[262161]: File chan_sip.c, Line 523 (__sip_ack): Stopping retransmission on '7ceb145123fa12fc73729d134d2820d8@155.97.244.130' of Request 102: Found Jun 27 16:13:18 DEBUG[262161]: File chan_sip.c, Line 3437 (check_user): Setting NAT on RTP to 0 Jun 27 16:13:18 DEBUG[262161]: File chan_sip.c, Line 523 (__sip_ack): Stopping retransmission...

...]: chan_zap.c:1523 update_conf: Updated conferencing on 4, with 0 conference users -- Hungup 'Zap/4-1' pbx*CLI> ---- On Trying to make an outgoing call Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '000f2300-08d000f6-4f620267-55399868 at 192.168.1.161' of Response 101: Match Found Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:10669 handle_request_invite: Checking SIP call limits...

...*CLI> Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 95AB5805-C94F-4C15-AC5A-6DFE5F58D644@192.168.1.101 Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:5200 check_user_full: Setting NAT on RTP to 0 Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:817 __sip_ack: Stopping retransmission on '95AB5805-C94F-4C15-AC5A-6DFE5F58D644@192.168.1.101' of Response 46613: Found Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:5200 check_user_full: Setting NAT on RTP to 0 Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:6991 handle_request: Check for res for 20...

...> > > > > > [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:9196 __find_call: = Looking > for Call ID: 280f68000ff289291b366a1242530ce8 at 192.168.67.4:5060 > (Checking To) --From tag as494dfc4b --To-tag 4155795028 > > [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4419 __sip_ack: Stopping > retransmission on '280f68000ff289291b366a1242530ce8 at 192.168.67.4:5060' of > Request 102: Match Found > > [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying > SIP dialog 280f68000ff289291b366a1242530ce8 at 192.168.67.4:5060 > > [201...

...;, "") in new stack DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format changed from U NKN to ULAW DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling timer at 16 0 sample intervals -- Playing 'vm-login' DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping retransmission on '6E5D898E-492D-400B-A42B-8B25FE25F2EE@192.168.7.3' of Response 1: Found DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling timer at 0 sample intervals DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling timer at 0 sample int...

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking