Displaying 11 results from an estimated 11 matches for "sip_new".
2013 Dec 15
3
Why doesn't Asterisk try to prevent transcoding
Let's say I have two devices configured and the follow call scenarios occur.
[100]
disallow=all
allow=g722&ulaw
Polycom phone with g722,ulaw,alaw,g729
[101]
disallow=all
allow=ulaw
Polycom phone with g722,ulaw,alaw,g729
101 dials 100 -> ulaw to ulaw is chosen
100 dials 101 -> g722 to ulaw is chosen
Ideally when 100 dials 101 ulaw would be chosen since it is the common
format.
2003 Jun 23
2
Sip too many open files?
...many open files
Jun 23 15:51:07 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 channel.c, Line 293
(ast_channel_alloc): Alert pipe creation failed!
Jun 23 15:51:07 WARNING[7176]: File chan_sip.c, Line 1152 (sip_new):
Unable to allocate channel structure
Jun 23 15:51:07 NOTICE[7176]: File chan_sip.c, Line 4414
(handle_request): Unable to create/find channel
Jun 23 15:53:14 WARNING[7176]: File channel.c, Line 293
(ast_channel_alloc): Alert pipe creation failed!
Jun 23 15:53:14 WARNING[7176]: File chan_sip.c, Li...
2012 Mar 20
0
Outgoing trunk is restricted to g.729 but need ulaw
...] DEBUG[17418]: acl.c:715 ast_ouraddrfor: For destination
'192.168.1.6', our source address is '192.168.1.25'.
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting
SIP_TRANSPORT_UDP with address 192.168.1.25:5060
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6557 sip_new: *** Our native
formats are 0x100 (g729)
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6558 sip_new: *** Joint
capabilities are 0x100 (g729)
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6559 sip_new: *** Our
capabilities are 0x10e (gsm|ulaw|alaw|g729)
The receiving asterisk system only does ulaw or ala...
2006 Apr 19
2
Unable to allocate socket: Too may open files
...cp_new: Unable to
allocate socket: Too many open files
Apr 19 14:20:51 WARNING[4045]: acl.c:306 ast_ouraddrfor: Cannot create
socket
Apr 19 14:20:51 WARNING[4045]: channel.c:562 ast_channel_alloc: Channel
allocation failed: Can't create alert pipe!
Apr 19 14:20:51 WARNING[4045]: chan_sip.c:2748 sip_new: Unable to
allocate SIP channel structure
Apr 19 14:20:51 NOTICE[4045]: app_dial.c:1029 dial_exec_full: Unable to
create channel of type 'SIP' (cause 0 - Unknown)
Apr 19 14:20:51 WARNING[4065]: rtp.c:911 ast_rtcp_new: Unable to
allocate socket: Too many open files
Apr 19 14:20:51 WARNING[40...
2004 Sep 29
3
X100P Unstable.
Hello All ,
In some ocasions i?m getting a problem with my X100P board.
I?m trying to trace tre problem , but i didn?t find a possible
answer.
-> I get those messages when trying to use Zap Channel
Sep 29 14:15:46 WARNING[-1094796368]: chan_sip.c:2107 sip_new: Unable to
allocate channel structure
Sep 29 14:15:46 NOTICE[-1094796368]: chan_sip.c:7283 handle_request:
Unable to create/find channel
When i try to make a call to the line gives me a busy signal.
The problem only solves when i reboot the PC. :(
Best Regards,
-Jefferson Carvalho
2007 Jun 15
0
Error: Unable to allocate RTCP socket: Too many open files
...0a67bbd8 answered SIP/sipCloverCSC-b7eba8a8
-- Packet2Packet bridging SIP/sipClCSC-b7eba8a8 and SIP/10.65.138.105-0a67bbd8
[Jun 15 09:21:48] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel allocation
failed: Can't create alert pipe!
[Jun 15 09:21:48] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate AST channel
structure for SIP channel
[Jun 15 09:21:48] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: Unable to
create/find SIP channel for this INVITE
-- SIP/iswitch-0a69fb70 is ringing
-- Call on SIP/iswitch-0a69fb70 left from hold
-- SIP/iswitch-0a69f...
2007 Jun 20
0
Error: Unable to allocate RTCP socket: Too manyopen files
...rCSC-b7eba8a8
> -- Packet2Packet bridging SIP/sipClCSC-b7eba8a8 and
> SIP/10.65.138.105-0a67bbd8
> [Jun 15 09:21:48] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel
> allocation
> failed: Can't create alert pipe!
> [Jun 15 09:21:48] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate
> AST channel
> structure for SIP channel
> [Jun 15 09:21:48] NOTICE[5306]: chan_sip.c:13662 handle_request_invite:
> Unable to
> create/find SIP channel for this INVITE
> -- SIP/iswitch-0a69fb70 is ringing
> -- Call on SIP/iswitch-0a69fb70 left...
2015 May 21
0
Too many open files - 786 000 already specified as max num open files?
...current callers.
I keep getting these types of messages in the CLI:
[May 21 11:39:21] WARNING[18469]: channel.c:1189 __ast_channel_alloc_ap:
Channel allocation failed: Can't create alert pipe! Try increasing max file
descriptors with ulimit -n
[May 21 11:39:21] WARNING[18469]: chan_sip.c:7041 sip_new: Unable to
allocate AST channel structure for SIP channel
[May 21 11:39:21] WARNING[18469]: res_rtp_asterisk.c:459 create_new_socket:
Unable to allocate RTCP socket: Too many open files
[May 21 11:39:21] ERROR[18469]: acl.c:706 ast_ouraddrfor: Cannot create
socket
I have specified this on the comm...
2005 Mar 30
2
Unable to allocate channel structure
Hi there, i have a problem with this error in some of my asterisk
boxes, some days in the morning i found this error in the asterisk
console specifically this:
Unable to allocate channel structure
Unable to create/find channel
When this happens im unable to make and receive calls.
The only way to fix this is restarting asterisk. The asterisk version
im running on all servers is "Asterisk
2007 Sep 20
4
Asterisk 1.2.24 simultaneous call limits.
Hi everyone,
I am running into wall today with simultaneous call limits. I have two
Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a
lot of sip calls from one machine to the other by issuing AMI Originate
commands to one machine. The machine that makes calls plays a message
(demo-intruct) upon the other machine answer. The machine receives the
calls just waits for 40 seconds
2010 May 12
2
asterisk-users Digest, Vol 70, Issue 25
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..
1- nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo