similar to: asterisk 11.6 nat problem

Displaying 20 results from an estimated 1100 matches similar to: "asterisk 11.6 nat problem"

2009 Dec 29
1
identifying channel for softhangup
When I place an outbound call from asterisk 1.6.1.12 to a FXO port on my Cisco 1760V 12.4, the channel changes - seemingly incrementing: e.g., in the first call, below, the channel name is "SIP/vgw1-00000075" -- the second call (on the same FXO port after a soft hangup on the CLI) is "SIP/vgw1-00000077" How can I extract this information in the dialplan so that I can use
2010 Nov 13
2
asterisk 1.8 fax woes
I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. All my custom modules (including swift <thanks darren!>) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip
2011 Feb 04
1
SoftHangup on asterisk 1.8.2.3
I am trying to use SoftHangup in my dialplan, but it's either not working or I'm not using it correctly. when i'm on the console, i see: pbx1*CLI> core show channels Channel Location State Application(Data) SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line)) SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,, 2 active
2009 Dec 18
0
calls ending up in default context
I'm trying to figure out how calls are ending up in my default context (which should never happen). I've got a Cisco 1760V with a VIC-2FXO-M1/VIC-4FXS and 5 Cisco sip phones. When I make a call from one of the FXS ports on the 1760, the call goes into asterisk's default context instead of where i think i'm directing it. Can someone tell me what I have misconfigured? 1760
2010 Aug 29
1
evil disconnect of call with cisco 1760
I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is
2010 Jan 10
2
app_swift 1.6.2 DTMF issue
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. for example consider: 999,1,Swift(some long message that you dont want to wait for|5000|5) 999,n,NoOp(DTMF: ${SWIFT_DTMF}) if while I am listening to the playback, i interrupt and dial: - "12345", SWIFT_DTMF is set to
2010 Nov 15
2
SIP calls destroyed after 1:20
After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this? -- Jeremy Kister http://jeremy.kister.net./
2012 Jan 23
1
ConfBridge details
running Asterisk 1.8.9.0-rc2, what are the ways to interface with ConfBridge ? I see the CLI command 'confbridge' documented for asterisk 10, but i dont see how to interface with confbridge on 1.8 What I'm trying to do is keep track of conferences that are used. I tried something like the below, but not only does Confbridge not return, but i'd need something that erases the
2011 Apr 15
3
sip error logging
I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' - No matching peer found my logger.conf
2013 Feb 12
1
asterisk 11 AGI
I recently upgraded to asterisk 11 from 1.8. I had VXML working via AGI in 1.8 - from extensions.conf: [VXML] exten => s,1,Answer exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})}) exten => s,n,AGI(agi://localhost/url=${ENCODED}) exten => s,n,Hangup Using asterisk 11 on the same host with the same config in extensions.conf: -- Executing [s at VXML:1]
2011 Jun 06
2
issues.asterisk.org
i'm trying to review issues that i'm monitoring and/or have reported at http://issues.asterisk.org when I click on 'My View' or 'View Issues' I get an error: APPLICATION ERROR #401 Database query failed. Error received from database was #1142: DELETE command denied to user 'mantisreadonly'@'localhost' for table 'mantis_tokens_table' for the
2011 May 13
1
asterisk 1.8 + google voice
somewhere along the way, i noticed incoming calls from google voice are no longer working on my asterisk 1.8.3.2 system. When the call comes in, asterisk immediately prints on the console: == Spawn extension (google-in, s, 2) exited non-zero on 'Gtalk/+12153930924-f947' [May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to
2013 Oct 04
1
OT: Asterisk loses Oprah on live TV
just thought this was cute enough to pass along, https://www.youtube.com/watch?feature=player_detailpage&v=GLwct15X_3g#t=135 -- Jeremy Kister http://jeremy.kister.net./
2012 Jul 18
4
asterisk 1.8 on Solaris/sparc
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. The system itself is happy and phone calls (between two parties) seem fine. Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file. To
2011 Mar 29
1
wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes
2009 Oct 28
5
need a local tech
I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can
2011 Jan 16
2
res_fax_digium.so crashing
Since digium is apparently blind to users of their Free Fax for Asterisk, does anyone have advice on how to report a crashing problem with res_fax_digium and Asterisk 1.8.2 ? I have detailed logs/reports and a backtrace ready, but I have no idea who can help. -- Jeremy Kister http://jeremy.kister.net./
2010 Nov 04
2
useless mpg123 processes hanging around
Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3 when i start asterisk, i immediately see two mpg123 processes spawned which sit there forever. I can't imagine it's normal behavior, but if it is, please explain :) # /etc/init.d/asterisk stop stopping asterisk. #[...] # /etc/init.d/asterisk start starting asterisk. # psg aster root 14573 1 0 16:29 pts/2 00:00:00
2011 Jan 26
6
Asterisk 1.8.2.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users
2011 Jan 26
6
Asterisk 1.8.2.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users