search for: build_route

...y disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid="Me" host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of build_route: messages from asterisk then the sip client die ....... Stopping retransmission on '569a2a5a77939bca491565ec50d0d3e5@82.51.138.189' of Request 104: Found build_route: Record-Route hop: <sip:421171@192.246.69.223;ftag=as61269cb9;lr=on> build_route: Contact hop: <sip:421171@65.39.20...

...t SUBSCRIBE: SUBSCRIBE sip:1001 at testing.net SIP/2.0 ... CSeq: 1 SUBSCRIBE Call-ID: 3d28dadd-87e5e749-1b7c8e1d at 192.168.1.133 Event: dialog Expires: 3600 Contact: <sip:33F18ADD-554124D20004E0F0-6CB68700 at 10.0.0.32;transport=udp> ... [2015-05-04 16:56:50] DEBUG[1948]: chan_sip.c:16341 build_route: build_route: Contact hop: <sip:33F18ADD-554124D20004E0F0-6CB68700 at 10.0.0.32 ;transport=udp> <-- 401 unauthorized --> 2nd SUBSCRIBE (authenticated): SUBSCRIBE sip:1001 at testing.net SIP/2.0 ... CSeq: 2 SUBSCRIBE Call-ID: 3d28dadd-87e5e749-1b7c8e1d at 192.168.1.133 Event: dialog Aut...

...ort has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username & pw to asterisk when I try to configure it as a client. Eg - Call from a Grandstream (working)- Jul 8 20:53:57 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: Contact hop: <sip:4000@192.168.1.42> -- Executing NoOp("SIP/4000-98ec", "") in new stack -- Executing Goto("SIP/4000-98ec", "intern-post|4001|1") in new stack -- Goto (intern-post,4001,1) -- Executing Dial("SIP/4000-98...

I was able to get app_prepaid working, but unfortunately I am getting one way audio on the phone that I was placing the call from. It is behind NAT. It appears that the app_prepaid is not taking this into consideration since I see: Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route: build_route: Contact hop: <sip:7708183799@192.168.1.101:5060;line=jet7pbic> Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1406 ast_rtp_bridge: Oooh, 'SIP/7708183799-8d6d' changed end address to 192.168.1.101:10094 (format 6) Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1408 ast_rtp_bridge: Oo...

...> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user 'xirak' is 1 out of 0 DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: Contact hop : <sip:192.168.7.3> -- Executing VoiceMailMain2("SIP/xirak-259d", "") 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): Schedu...

...ct = yes [to_GNUGK]] exten => _.,1,Dial(h323/${EXTEN}@10.10.1.12:1720,60,C) [to_SER] exten => _.,1,Dial(SIP/${EXTEN}@10.10.1.170:5060,60) DEBUG File ========== Jul 15 16:14:10 DEBUG[65541]: Check for res for Jul 15 16:14:10 DEBUG[65541]: is not a local user Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop: <sip:15613021234@10.10.1.170;ftag=661806388;lr=on> Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop: <sip:999012020@10.10.1.13:5060;user=phone;transport=udp> Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:10 DEBUG[311316]: type=h323, format=2...

Hi there, do I have a local problem, or is registration at IAXTEL impossible at the moment? "iax2 show registry" permanently shows a TIMEOUT for 69.73.19.178. Philipp

The IAX2 channel is to IAXmodem. The SIP extension is an ATA with a fax attached. Nov 28 15:30:20 DEBUG[2997] chan_sip.c: build_route: Contact hop: Nov 28 15:30:20 VERBOSE[3276] logger.c: -- SIP/2201-090995f0 answered IAX2/24729-2 Nov 28 15:30:20 DEBUG[2995] chan_iax2.c: Ooh, voice format changed to 8 So what does this mean? The fax works just fine. I am just trying to tune up my dialplan.

...e getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> -- Executing Wait("SIP/sip3-768a", "1") in new stack -- Executing AgentLogin("SIP/sip3-768a", "") in new stack Aug 12 16:31:37 DEBUG[1127562160]: rtp.c:1156 ast_rtp_write: Ooh, format chan...

....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 call limits for device Phone3 Sep 13 22:18:10 DEBUG[13167]: chan_sip.c:5497 build_route: build_route: Contact hop: <sip:Phone3@192.168.0.18:5061> -- Executing VoiceMailMain("SIP/Phone3-9d74", "") in new stack Sep 13 22:18:10 DEBUG[13167]: channel.c:1388 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-login' (langu...

...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 DEBUG[1133735216]: File chan_sip.c, Line 973 (find_user): Call from user 'phone1' is 1 out of 0 DEBUG[1133735216]: File chan_sip.c, Line 3307 (build_route): build_route: Contact hop: <sip:phone1@10.1.2.24:5060;line=1> -- Executing Dial("SIP/phone1-2c71", "IAX/user:secretpass/BYEXTENSION@iaxtel") in new stack -- Calling using options 'exten=18007692511;callerid=phone1 <7201>;language=en;context=iaxtel;userna...

...ew SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:6991 handle_request: Check for res for 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:1605 update_user_counter: is not a local user 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:65.67.76.30:5060> 2004-08-09 17:36:29 DEBUG[245775]: pbx.c:1255 pbx_extension_helper: Launching 'Congestion' 2004-08-09 17:36:29 DEBUG[245775]: channel.c:652 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/65.67.76.30-0814e4f0' 2004-08...

...vvvvvvvvvvvv', when I pick up the phone on the gsbt100 and dial '8902': Setting NAT on RTP to 0 Stopping retransmission on '20fc1bc647b262ea@192.168.1.200' of Response 15995: Found Setting NAT on RTP to 0 Check for res for gsbt100 Call from user 'gsbt100' is 1 out of 0 build_route: Contact hop: <sip:gsbt100@192.168.1.200;user=phone> -- Executing Answer("SIP/gsbt100-b25b", "") in new stack -- Executing Answer("SIP/gsbt100-b25b", "") in new stack -- Executing SetCIDNum("SIP/gsbt100-b25b", "55555555")...

...ly to record both sides of the conversation but now we only have the initiating caller channel being recorded. Occasionaly the other caller is also recorded but the speed of the recording is completely wrong causing distortion and out of sync. Here fwiw are the logs. Aug 15 18:31:32 DEBUG[9995]: build_route: Contact hop: <sip:snom@81.58.13.190:5060;line=ikojqrcx> Aug 15 18:31:32 DEBUG[9995]: Device 'SIP/snom' changed to state '2' Aug 15 18:31:32 VERBOSE[9995]: -- Executing SetVar("SIP/snom-7214", "CALLFILENAME=call_to_00NUMBER_HIDDEN_dated_20050815-183132"...

...er_counter: Call from user 'xlite1' is 1 out of 0 -- Called xlite1 Aug 13 10:19:03 DEBUG[245774]: chan_sip.c:840 __sip_semi_ack: -- SIP/xlite1-89a7 is ringing Aug 13 10:19:05 DEBUG[245774]: chan_sip.c:799 __sip_ack: Acked pending invite 102 Aug 13 10:19:05 DEBUG[245774]: chan_sip.c:4411 build_route: build_route: Contact hop: <sip:xlite1@Ip of Xlite1:5060> -- SIP/xlite1-89a7 answered OH323/R27469 Aug 13 10:19:05 DEBUG[524304]: channel.c:2613 ast_channel_bridge: Got a FRAME_CONTROL (4) frame on channel SIP/xlite1-89a7 Aug 13 10:19:05 DEBUG[524304]: channel.c:2675 ast_channel_bridge: Br...

...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 for device 319 Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:6267 build_route: build_route: Contact hop: <sip:319 at 192.168.1.161:5060> Nov 20 20:51:48 DEBUG[5101]: channel.c:775 channel_find_locked: Avoiding initial deadlock for 'SIP/319-081d8e00' -- Executing Dial("SIP/319-081d8e00", "Zap/1/0004479086365389") in new stack Nov 20 20:51...

Hello All I am trying to answer incoming SIP calls, first, by dialing an extension, thence into voicemail, which works; and secondly by going straight into voice mail which does not. The extension.conf that works is like this; [incomingSIP] exten=>_.,1,Dial,Zap/2|1 exten=>_.,2,Voicemail,u5152 exten=>_.,3,Hangup the extension.conf which does not is like this; [incomingSIP]

...8.223.74' Request 102: Found -- SIP/fwdpulvercom-dd5a is ringing Unable to handle indication 3 for 'Phone/phone0' Scheduled a registration timeout # 100 Acked pending invite 102 Stopping retransmission on '1bf802c37038448b795a4dc8300e0627@12.218.223.74' of Request 102: Found build_route: Record-Route hop: <sip:612@69.90.155.70;ftag=as3d6e380d;lr=on> build_route: Contact hop: <sip:612@69.90.168.13:5028> -- SIP/fwdpulvercom-dd5a answered Phone/phone0 No path to translate from Phone/phone0(1) to SIP/fwdpulvercom-dd5a(2) Had to drop call because I couldn't make Pho...

..._full: Setting NAT on RTP to 4 Oct 4 00:53:41 DEBUG[1083546560]: chan_sip.c:7087 handle_request: Check for res for 6568543197 Oct 4 00:53:41 DEBUG[1083546560]: chan_sip.c:1650 update_user_counter: Call from user '6568543197' is 1 out of 0 Oct 4 00:53:41 DEBUG[1083546560]: chan_sip.c:4492 build_route: build_route: Contact hop: +6568543197 <sip:6568543197@192.168.1.103:5060> -- Executing SetVar("SIP/6568543197-86c2", "sip_codec=g729") in new stack -- Executing Dial("SIP/6568543197-86c2", "Zap/g1/91596323") in new stack -- Called g1/915963...

...-pstn Then I went to incoming routes and set it up for any did, any cid, but the telasip connection is taking a different route. Here are the log entries for both. Telasip: May 2 11:11:55 DEBUG[2670] chan_sip.c: Checking SIP call limits for device jlynch May 2 11:11:55 DEBUG[2670] chan_sip.c: build_route: Contact hop: < sip:7707190068@4.79.19.59> May 2 11:11:55 VERBOSE[6343] logger.c: -- Executing Set("SIP/jlynch-cf63", "FROM_DID=6782280738") in new stack May 2 11:11:55 VERBOSE[6343] logger.c: -- Executing Answer("SIP/jlynch-cf63", "") in new...