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