search for: check_user_full

Displaying 9 results from an estimated 9 matches for "check_user_full".

2007 Nov 20
1
FXO Hangs up automatically
...zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/4-1 Nov 20 20:49:32 DEBUG[6028]: 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:...
2005 Aug 02
0
Hang up as soon as other party picks up call
...ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001 secret=4001 host=dynamic context=callout disallow=all allow=ulaw And below is what i get from Asterisk debug. Aug 2 11:04:17 DEBUG[180235]: chan_sip.c:5320 check_user_full: Setting NAT on RTP to 0 Aug 2 11:04:17 DEBUG[180235]: chan_sip.c:825 __sip_ack: Stopping retransmission on '674C5258-069C-4AF8-8B58-317838C513D3@192.168.1.40' of Response 37605: Found Aug 2 11:04:17 DEBUG[180235]: chan_sip.c:5320 check_user_full: Setting NAT on RTP to 0 Aug 2 11:04:17 D...
2005 Sep 13
1
wctdm, issue w/outbound calls
...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 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(&q...
2004 Aug 26
0
Out Dial Problem
...shows me Connect, Trying, Dialing and then hangup. I've found the log as the following : *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 Au...
2005 Jun 28
2
Trying to get *8 call pickup to work
...a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 13 headers, 15 lines Using latest request as basis request Sending to pickup.phone.ip.addr : 5060 (non-NAT) Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:5441 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c From: "Test SIP" <sip:1310@asterisk-server>;tag=5eba6d75ff7e1e47 To: <sip:*8@asterisk-server>;tag=as114aad8b...
2004 Oct 03
0
Call gets disconnected upon connect
...mmediately after being answered with the debug error message saying something like: "channel.c:2646 ast_channel_bridge: Didn't get a frame from channel: SIP/6568543197-9af5" Any idea why this may be happening? Here is the debug log: Oct 4 00:53:41 DEBUG[1083546560]: chan_sip.c:5269 check_user_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...
2005 Mar 17
3
Realtime Problem = Segmentation faults
Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the
2007 Apr 12
2
Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
...chan_sip.c:1044 parse_sip_options: * SIP extension value: 17 for call 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136 Using INVITE request as basis request - 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136 Sending to 10.0.10.136 : 5060 (non-NAT) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:7291 check_user_full: Setting NAT on RTP to 0 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:7295 check_user_full: Setting NAT on VRTP to 0 Reliably Transmitting (no NAT) to 10.0.10.136:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5;received=10.0.10.136 From:...
2004 Dec 17
6
Realtime and PostgreSQL
Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing