Displaying 20 results from an estimated 24 matches for "sip_alloc".
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shf_alloc
2003 Jul 31
3
Mutex problem in sip?
Hello,
CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ...
grep -e "Error" -e "eventually" p-console
chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
busy
chan_sip.c line 1453 (sip_alloc): Got it eventually...
chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource
busy
chan_sip.c line 1453 (sip_alloc): Got it eventually...
chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: De...
2009 May 18
3
Number of max SIP calls.
...insip other winsip.
But the problem is approx of 60 calls get mature and asterisk give error
for the remaining like shown below.
> May 18 14:57:15] WARNING[8314]: rtp.c:2433 rtp_socket: Unable to allocate RTP socket: Too many open files
> [May 18 14:57:15] WARNING[8314]: chan_sip.c:6710 sip_alloc: Unable to create RTP audio session: Too many open files
> [May 18 14:57:15] ERROR[8314]: acl.c:481 ast_ouraddrfor: Cannot create socket
> [May 18 14:57:15] ERROR[8314]: acl.c:481 ast_ouraddrfor: Cannot create socket
> [May 18 14:57:15] WARNING[8314]: rtp.c:2433 rtp_socket: Unable to alloc...
2006 May 17
1
Deadlocks in 1.2.7.1
...e made and sip
show peers and other commands show no obvious problem. We then
recompiled 1.2.7.1 with all the DEBUG_ turned on in the makefile and
now we see the following messages:
May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236
__ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): Deadlock?
waited 460 sec for mutex '&iflock'?
May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239
__ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor):
'&iflock' was locked here.
May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236
__ast_pthread...
2003 Jun 23
2
Sip too many open files?
Today my pbx stopped responding to my sip phones..
looking into the log, here what I got:
Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new):
Unable to allocate socket: Too many open files
Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc):
Unable to create RTP session: Too many open files
Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 4655
(sip_send_mwi_to_peer): Unable to build sip pvt data for MWI
Jun 23 15:51:07 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new):
Unable to allocate socket: Too many open files
Jun 23 15:51:0...
2004 Aug 09
1
Inbound Call Errors...
...searched all over the web and have not really found anything
related to this error.... The only thing I found is related to a
system stops responding on DTMF, which does not happen here... THe
following is the output from 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]:...
2003 Sep 25
4
SIP Problem
...wd. 3ff9e23356a 00102/00000 00000ms 0000ms UNKN
504 active SIP channel(s)
tail -4 /var/log/asterisk/messages
Sep 25 18:31:00 WARNING[1125329600]: File rtp.c, Line 708 (ast_rtp_new): Unable
to allocate socket: Too many open files
Sep 25 18:31:00 WARNING[1125329600]: File chan_sip.c, Line 1479 (sip_alloc):
Unable to create RTP session: Too many open files
Sep 25 18:31:01 WARNING[1125329600]: File rtp.c, Line 708 (ast_rtp_new): Unable
to allocate socket: Too many open files
Sep 25 18:31:01 WARNING[1125329600]: File chan_sip.c, Line 1479 (sip_alloc):
Unable to create RTP session: Too many open files...
2004 Dec 08
3
Asterisk 1.0.1 Too many open files
My asterisk process produced the following errors this morning:
Dec 8 10:44:07 WARNING[50315282]: rtp.c:829 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files
Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:2352 sip_alloc: Unable to create RTP session: Too many open files
Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:8024 sip_request: Unable to build sip pvt data for 'xxxxxxxxxx@sip0'
Dec 8 10:44:07 NOTICE[50315282]: app_dial.c:743 dial_exec: Unable to create channel of type 'SIP'
I don't think i...
2007 Jun 15
0
Error: Unable to allocate RTCP socket: Too many open files
...t to 3600 seconds.
-- Called iswitch/27117973000
[Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket
[Jun 15 09:22:04] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to allocate
socket: Too many open files
[Jun 15 09:22:04] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create RTP audio
session: Too many open files
[Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket
[Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create socket
[Jun 15 09:22:05] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable to al...
2007 Jun 20
0
Error: Unable to allocate RTCP socket: Too manyopen files
...led iswitch/27117973000
> [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create
> socket
> [Jun 15 09:22:04] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable
> to allocate
> socket: Too many open files
> [Jun 15 09:22:04] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create
> RTP audio
> session: Too many open files
> [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create
> socket
> [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create
> socket
> [Jun 15 09:22:05] ERROR[5306]: rtp.c:1861 ast_...
2006 Apr 04
1
Too many open files
...(s)
Apr 5 00:48:38 NOTICE[14897]: app_dial.c:1042 dial_exec_full: Unable to
create channel of type 'LOCAL' (cause 0 - Unknown)
Apr 5 00:48:38 ERROR[14899]: rtp.c:933 ast_rtp_new_with_bindaddr:
Unable to allocate socket: Too many open files
Apr 5 08:48:38 WARNING[14899]: chan_sip.c:3079 sip_alloc: Unable to
create RTP audio session: Too many open files
ulimit -a
core file size (blocks, -c) unlimited
data seg size (kbytes, -d) unlimited
file size (blocks, -f) unlimited
pending signals (-i) 1024
max locked memory (kbytes, -l) 32
max m...
2004 Nov 30
1
realTime configuration help needed
...DEBUG[3088]: channel.c:199 ast_channel_register_ex:
Registered handler for 'SIP' (Session Initiation Protocol (SIP))
---------------------------
When I try to make a call to * through 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...
2005 Sep 02
0
Unable to create RTP session
...e mechine
here is the asterisk trace
------------------------------------------------
-- Setting call duration limit to 3000 seconds.
Sep 2 15:58:12 WARNING[10334]: rtp.c:852
ast_rtp_new_with_bindaddr: Unable to allocate socket:
Too many open files
Sep 2 15:58:12 WARNING[10334]: chan_sip.c:2313
sip_alloc: Unable to create RTP session: Too many open
files
Sep 2 15:58:12 WARNING[10334]: chan_sip.c:8202
sip_request: Unable to build sip pvt data for
'5000131@192.168.0.11:5060'
Sep 2 15:58:12 NOTICE[10334]: app_dial.c:1084
dial_exec: Unable to create channel of type 'SIP'
== Everyone...
2009 Nov 04
0
memory leak with static users
...m the number
of static users) an OOM and his consequent kill.
Using refcounter debug (uncomment REF_DEBUG macro in chan_sip.c), we
noticed only peer objects (builded by build_peer) have "Refcount not
zero".
Reading /tmp/refs, we noticed destructor is called for all channels
created (with sip_alloc) for to send SIP OPTIONS message.
This channels appear into refcounter debug when message is sent, but not
when timout expires and they are destroyed with **call destructor**.
Using our users.conf, the same thing happens with version 1.6.1.6, but not with 1.4.22.
Do you have any suggestions?
Tha...
2012 Mar 20
0
Outgoing trunk is restricted to g.729 but need ulaw
...s at macro-dialout:36] Dial("SIP/1234-00000039",
"SIP/trunkout/1xxxxxxxxx,60,L(180000:20000)") in new stack
[Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:25057 sip_request_call: Asked to
create a SIP channel with formats: 0x100 (g729)
[Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:7215 sip_alloc: Allocating new
SIP dialog for 4870fab953c16a611b9248584748fe59 at 127.0.0.1:0 - INVITE (No
RTP)
[Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:344 ast_rtp_instance_new:
Using engine 'asterisk' for RTP instance '0x9ded230'
[Mar 19 18:22:56] DEBUG[17418]: res_rtp_asterisk.c:472 ast_rtp...
2006 Jan 25
0
asterisk 1.2 with grandstream ht-496 2nd port registration issues
...CK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64)
Jan 25 22:26:07 DEBUG[41042]: chan_sip.c:3318 parse_request: Header 11:
Content-Length: 0 (17)
Jan 25 22:26:07 DEBUG[41042]: chan_sip.c:3318 parse_request: Header 12:
(0)
--- (12 headers 0 lines)---
Jan 25 22:26:07 DEBUG[41042]: chan_sip.c:3102 sip_alloc: Allocating new
SIP dialog for e62dffffcd4dffff@194.183.145.211 - REGISTER (No RTP)
Jan 25 22:26:07 DEBUG[41042]: chan_sip.c:10933 handle_request: ****
Received REGISTER (2) - Command in SIP REGISTER
Using latest REGISTER request as basis request
Sending to 194.208.44.44 : 34560 (non-NAT)
Transmitt...
2003 Jun 18
0
MP3Player and Ringing (long)
...When it calls ast_write
to write the first stream, ast_release_generator is called and restores
the write format to 2 (GSM); further writes produce codec errors since
MP3Players writes Linear PCM frames.
I extracted some debug log:
Jun 5 01:55:33 DEBUG[1158913328]: File chan_sip.c, Line 1327
(sip_alloc): Allocating new SIP call for
7a655bae-29f17562-197726d5@62.212.12.21
Jun 5 01:55:33 DEBUG[1158913328]: File chan_sip.c, Line 3359
(check_user): Setting NAT on RTP to 0
Jun 5 01:55:33 DEBUG[1158913328]: File chan_sip.c, Line 2899
(build_route): build_route: Contact hop: 5010 <sip:5010@62.212.1...
2004 Dec 13
0
[oh323] sporadic call setup
..., 9205,0721*8 " <8900>'.
Dec 13 13:13:05 DEBUG[-1296254032]: pbx.c:1260 pbx_extension_helper: Launching 'Dial'
Dec 13 13:13:05 DEBUG[-1296254032]: app_dial.c:490 dial_exec: SIMPLE DIAL (NO URL)
Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:2395 sip_alloc: Allocating new SIP call for (null)
Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:1295 create_addr: Setting NAT on RTP to 0
Urgent handler
Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:1536 sip_call: Outgoing Call for 2005
Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:1669 update_user_counter: C...
2006 May 22
2
I've broken voicemail
I went to put in the new sound files over the weekend, but forgot to backup
the custom folder and lost my custom digital receptionist files.
I then had to copy the old files back from a duplicate machine.
The problem is now though that voicemail just hangs up when I dial it.
Other apps work - *70 for example gives me 'call waiting...activated' so I
know it's accessing the files
2010 Oct 28
0
Intermittent failure when placing calls - unable to create channel of type SIP
...tension_helper: Launching
'Set'
[Oct 27 18:46:48] DEBUG[25028]: pbx.c:3696 pbx_extension_helper: Launching
'Dial'
[Oct 27 18:46:48] DEBUG[25028]: chan_sip.c:23241 sip_request_call: Asked to
create a SIP channel with formats: 0x4 (ulaw)
[Oct 27 18:46:48] DEBUG[25028]: chan_sip.c:7381 sip_alloc: Allocating new
SIP dialog for 2ccf324d10670f2c73f478b523f926a4 at 10.15.1.1 - INVITE (With
RTP)
Really destroying SIP dialog '2ccf324d10670f2c73f478b523f926a4 at 10.15.1.1'
Method: INVITE
[Oct 27 18:46:48] WARNING[25028]: app_dial.c:1750 dial_exec_full: Unable to
create channel of type ...
2006 Feb 01
1
Unable to Register to Asterisk through Proxy
...n_sip.c:3322 parse_request: Header 8: Max-Forwards: 69 (16)
Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 9: Expires: 600 (12)
Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 10: (0)
--- (10 headers 0 lines)---
Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3106 sip_alloc: Allocating new SIP dialog for 3deb0da4bada8c3@192.168.1.2 - REGISTER (No RTP)
Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:10945 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER
Using latest REGISTER request as basis request
Sending to yyy.yyy.yyy.yyy : 5060 (non-NAT)
Transmitting...