Displaying 20 results from an estimated 47 matches for "update_user_counter".
2006 Jun 19
2
Asterisk 1.07 crash under Debian Sarge
...17:44:18 DEBUG[17975]: Hangup: channel: -2 index = 0, normal =
51, callwait = -1, thirdcall = -1
Jun 12 17:44:18 DEBUG[17975]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1321090091
Jun 12 17:44:18 DEBUG[17975]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:19 DEBUG[17975]: update_user_counter(302) - decrement inUse
counter
Jun 12 17:44:19 DEBUG[18016]: Building dynamic conference '999'
Jun 12 17:44:20 DEBUG[18016]: Placed channel SIP/508-af01 in ZAP conf
1023
Jun 12 17:44:20 DEBUG[18016]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:20 D...
2004 Aug 09
1
Inbound Call Errors...
...the CLI:
*CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc:
Allocating new 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...
2006 May 26
0
SIP call problem
...cuting Dial("Zap/3-1",
"SIP/15111111111@SIP_PROVIDER") in new stack
May 26 09:49:02 DEBUG[3242]: chan_sip.c:1309
create_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
Requ...
2006 Jan 12
2
Random Disconnects
...invite 102
Jan 12 19:38:49 DEBUG[1546]: Stopping retransmission on
'66656f196342cc765ba57344169430de@192.168.1.200' of Request 102: Found
Jan 12 19:38:49 DEBUG[1546]: build_route: Contact hop:
Jan 12 19:38:49 VERBOSE[1546]: -- SIP/101-264e answered SIP/119-41c6
Jan 12 19:38:49 DEBUG[1546]: update_user_counter(109) - decrement outUse counter
Jan 12 19:38:49 DEBUG[1546]: Scheduling timer at 0 sample intervals
Jan 12 19:38:49 VERBOSE[1546]: -- Attempting native bridge of
SIP/119-41c6 and SIP/101-264e
Jan 12 19:38:49 DEBUG[1546]: Acked pending invite 102
Jan 12 19:38:49 DEBUG[1546]: Stopping retransmission...
2004 Nov 30
1
realTime configuration help needed
...a Grandstream phone (with the IP
192.168.1.203)
I get :
Nov 30 12:58:03 DEBUG[3096]: chan_sip.c:2370 sip_alloc: Allocating new
SIP call for 659d80f14cb431cd@192.168.1.203
Nov 30 12:58:03 DEBUG[3096]: chan_sip.c:7292 handle_request: Check for
res for
Nov 30 12:58:03 DEBUG[3096]: chan_sip.c:1592 update_user_counter: is
not a local user
Nov 30 12:58:03 DEBUG[3096]: chan_sip.c:1592 update_user_counter: is
not a local user
Nov 30 12:58:03 DEBUG[3096]: chan_sip.c:833 __sip_ack: Stopping
retransmission on '659d80f14cb431cd@192.168.1.203' of Response 50530: Found
--------------------------------------...
2005 May 25
0
FAST BUSY on Back to back ZAP outgoing calls
...005-05-25 17:36:45','Living Room <102>','102','777','default',
'SIP/102-bbbc','','MP3Player','http://64.236.34.196:80/stream/2004',15,1
5,'ANSWERED',3,'3607330988','')
2005-05-25 17:37:00 DEBUG[2691]: update_user_counter(102) - decrement
inUse counter
2005-05-25 17:37:01 DEBUG[2691]: Exception on 22, channel 4
2005-05-25 17:37:01 DEBUG[2691]: Got event Hook Transition Complete(12)
on channel 4 (index 0)
2005-05-25 17:37:02 DEBUG[2691]: Exception on 22, channel 4
2005-05-25 17:37:02 DEBUG[2691]: Got event Dial C...
2004 Aug 13
0
SIP<->H323 "Failed to create smoother"
...-- Executing Dial("OH323/R27469", "SIP/xlite1|10") in
new stack
Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265
create_addr: Setting NAT on RTP to 0
Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500
sip_call: Outgoing Call for xlite1
Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1633
update_user_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_rout...
2005 May 28
0
TDM zap channel Exception on 15, channel 1
...in new stack
May 27 18:08:06 DEBUG[1224]: app_dial.c:497 dial_exec: SIMPLE DIAL (NO URL)
May 27 18:08:06 DEBUG[1224]: chan_sip.c:1309 create_addr: Setting NAT on
RTP to 0
May 27 18:08:06 DEBUG[1224]: chan_sip.c:1487 sip_call: Outgoing Call for
7011
May 27 18:08:06 DEBUG[1224]: chan_sip.c:1620 update_user_counter: Call
from user '7011' is 1 out of 0
-- Called 7011
May 27 18:08:08 DEBUG[1224]: chan_zap.c:3768 __zt_exception: Exception
on 15, channel 1
May 27 18:08:08 DEBUG[1224]: chan_zap.c:3080 zt_handle_event: Got event
Ring/Answered(2) on channel 1 (index 0)
May 27 18:08:10 DEBUG[1224]: cha...
2004 Oct 03
0
Call gets disconnected upon connect
...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_route:
build_route: Contact hop: +6568543197
<sip:6568543197@192.168.1.103:5060>
-- Executing SetVar("SIP/6568543197-86c2", "sip_codec=g729") in new stack
--...
2005 Sep 13
1
wctdm, issue w/outbound calls
...ntervals
Sep 13 22:18:12 WARNING[13167]: app_voicemail.c:4890 vm_authenticate:
Couldn't r
ead username
Sep 13 22:18:12 DEBUG[13167]: pbx.c:2262 __ast_pbx_run: Extension 8500,
priority
1 returned normally even though call was hung up
Sep 13 22:18:12 DEBUG[13167]: chan_sip.c:2315 sip_hangup:
update_user_counter(Ph
one3) - decrement inUse counter
Sep 13 22:18:16 DEBUG[13167]: chan_sip.c:6350 check_user_full: Setting NAT
on RT
P to 0
Sep 13 22:18:16 DEBUG[13167]: chan_sip.c:9413 handle_request_invite:
Checking SI
P call limits for device Phone3
Sep 13 22:18:16 DEBUG[13167]: chan_sip.c:5497 build_route: bu...
2004 Jul 08
0
Problem SIP no audio just noise
...0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Found peer 'phone1010'
Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:4851 check_user: Setting NAT on
RTP to 0
Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:6424 handle_request: Check for
res for damian
Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:1386 update_user_counter:
damian is not a local user
Looking for 99826816 in default
Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:4115 build_route: build_route:
Contact hop: <sip:damian@10.1.1.11:5060>
list_route: hop: <sip:damian@10.1.1.11:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1....
2005 Jan 27
0
Asterisk @ Home & BroadVoice (Outbound) help
...ng invite 102
Jan 27 13:53:45 DEBUG[10539]: Stopping retransmission
on '0653e99a0f007c703b3ed2026efd7ec1@192.168.0.101' of
Request 102: Found
Jan 27 13:53:45 VERBOSE[10539]: -- Got SIP
response 604 "Does not exist anywhere" back from
147.135.0.128
Jan 27 13:53:45 DEBUG[10558]:
update_user_counter(16266981070) - decrement outUse
counter
Jan 27 13:53:45 DEBUG[10558]: 16266981070 is not a
local user
Jan 27 13:53:45 VERBOSE[10558]: == No one is
available to answer at this time
Jan 27 13:53:45 DEBUG[10558]: Exiting with
DIALSTATUS=NOANSWER.
Jan 27 13:53:45 VERBOSE[10558]: -- Executing
Cong...
2004 Dec 20
7
'I'nvalid extension handling problems, even with workaround
...ting native bridge of SIP/gsbt100-b25b and SIP/voiptalk.org-8789
Oooh, format changed to 8
Ooh, format changed from UNKN to ULAW
Ooh, format changed from UNKN to ALAW
Didn't get a frame from channel: SIP/voiptalk.org-8789
Bridge stops bridging channels SIP/gsbt100-b25b and SIP/voiptalk.org-8789
update_user_counter(902) - decrement outUse counter
902 is not a local user
Exiting with DIALSTATUS=ANSWER.
-- Executing Hangup("SIP/gsbt100-b25b", "") in new stack
== Spawn extension (internal, 8902, 6) exited non-zero on 'SIP/gsbt100-b25b'
-- Executing Wait("SIP/gsbt100-b25...
2005 Sep 21
1
Call getting disconnected in queue
...-- Started music on hold, class 'default', on Zap/2-1
-- outgoing agentcall, to agent '1005', on 'Local/1004@from-sip-d281,1'
-- Executing Dial("Local/1004@from-sip-d281,2", "SIP/1004") in new
stack
Sep 21 11:56:39 ERROR[12633]: chan_sip.c:1615 update_user_counter: Call
from user '1004' rejected due to usage limit of 1
-- Couldn't call 1004
== Everyone is busy/congested at this time
-- Called Agent/1005
-- Playing 'agent-incorrect' (language 'en')
== Spawn extension (from-sip, 2002, 1) exited non-zero on '...
2005 Feb 08
1
Can only call VoIP SIP Providers (Weird)
...8 10:03:06 DEBUG[3366]: Registration successful
Feb 8 10:03:06 DEBUG[3366]: Cancelling timeout 29574
Feb 8 10:03:18 DEBUG[4482]: Didn't get a frame from
channel: SIP/BV-522e
Feb 8 10:03:18 DEBUG[4482]: Bridge stops bridging
channels SIP/201-df7a and SIP/BV-522e
Feb 8 10:03:18 DEBUG[4482]:
update_user_counter(16262872020) - decrement outUse
counter
Feb 8 10:03:18 DEBUG[4482]: 16262875555 is not a
local user
Feb 8 10:03:18 DEBUG[4482]: Exiting with
DIALSTATUS=ANSWER.
Feb 8 10:03:18 VERBOSE[4482]: == Spawn extension
(from-internal, 916262872020, 2) exited non-zero on
'SIP/201-df7a'
Feb 8 10:...
2004 Aug 26
0
Out Dial Problem
...C5A-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 2000
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:1633 update_user_counter: Call
from user '2000' is 1 out of 0
Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:4423 build_route: build_route:
Contact hop: <sip:2000@192.168.1.101:5060>
Aug 26 15:54:17 DEBUG[-1260983376]: pbx.c:1255 pbx_extension_helper:
Launching 'ChanIsAvail'
Aug 26 15:54:17 DEBUG[-...
2005 Jun 28
2
Trying to get *8 call pickup to work
...new stack
-- Executing Dial("Zap/24-1", "SIP/1312|20") in new stack
Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0
Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1487 sip_call: Outgoing Call for 1312
Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1620 update_user_counter: Call from user '1312' is 1 out of 0
We're at asterisk.server.ip.addr port 19630
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x1 (g723)
Answering with preferred capability 0x100 (g729)
Answering...
2005 Sep 10
1
False Zap answer problem (Again)
...('2005-09-10 19:22:33','\"Ichat\" <15>','15','19','ichat',
'SIP/15-f784','Zap/5-1','Dial','Zap/5|24|rTtWw',8,0,'NO
ANSWER',3,'','1126369353.4')
Sep 10 19:22:41 DEBUG[27367] chan_sip.c: update_user_counter(15) - decrement
inUse counter
Sep 10 19:22:41 VERBOSE[27367] logger.c: -- Zap/5-2 answered SIP/21-efcb
Sep 10 19:22:45 DEBUG[27367] chan_zap.c: Exception on 11, channel 5
Sep 10 19:22:45 DEBUG[27367] chan_zap.c: Got event Ringer On(10) on channel
5 (index 0)
Sep 10 19:22:47 DEBUG[27367] chan_...
2005 Jan 15
0
Polycom IP600 - Bridge stops because we're zombie or need a soft hangup
...hangup: c0=SIP/302-928e, c1=SIP/sip_proxy-out-f201, flags: No,No,No,Yes
Jan 15 14:26:16 DEBUG[4290]: Bridge stops bridging channels SIP/302-928e and SIP/sip_proxy-out-f201
Jan 15 14:26:16 DEBUG[4290]: Ignoring too old packet packet 661161242 (expecting >= 661161243)
Jan 15 14:26:16 DEBUG[4290]: update_user_counter(12699264242) - decrement inUse counter
Jan 15 14:26:16 DEBUG[4290]: 12699264242 is not a local user
Jan 15 14:26:16 DEBUG[4290]: Exiting with DIALSTATUS=ANSWER.
Call length 1hr 39mins
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2005 May 13
0
Dropped Calls between Sip and Zaptel
...smission on
'660440d07b1155645cabd0ce681609d9@10.125.1.245' of Request 102: Found
May 13 08:37:16 DEBUG[8480]: Didn't get a frame from channel:
SIP/cronus-116-78ed
May 13 08:37:16 DEBUG[8480]: Bridge stops bridging channels Zap/1-1 and
SIP/cronus-116-78ed
May 13 08:37:16 DEBUG[8480]: update_user_counter(cronus-116) - decrement
outUse counter
May 13 08:37:16 DEBUG[8480]: Exiting with DIALSTATUS=ANSWER.
May 13 08:37:16 VERBOSE[8480]: == Spawn extension
(macro-netvoice-stdexten, s, 302) exited non-zero on 'Zap/1-1' in macro
'netvoice-stdexten
May 13 08:37:16 VERBOSE[8480]: == Spawn...