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- Need CHANSpy enabled on only three extensions
by aqua_puddles » Mon Jan 18, 2016 8:33 am
- 1 Replies
- 1184 Views
- Last post by ambiorixg12
Mon Jan 18, 2016 9:13 pm
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- Realtime LDAP error - loading sip.conf
by smbrown » Sun Jun 20, 2010 9:21 pm
- 2 Replies
- 2674 Views
- Last post by ventor
Mon Jan 18, 2016 5:26 pm
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- Different ring from internal and external calls
by asyscom » Fri Jan 15, 2016 12:47 am
- 6 Replies
- 2514 Views
- Last post by csyria
Mon Jan 18, 2016 3:31 pm
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- Register problem to Provicer
by joehoe12 » Fri Jan 15, 2016 3:28 am
- 13 Replies
- 5001 Views
- Last post by david55
Mon Jan 18, 2016 3:11 pm
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- Asterisk Amateur Needs Help with Dial-Plan
by jameswendt » Thu Jan 14, 2016 9:03 pm
- 3 Replies
- 1524 Views
- Last post by david55
Mon Jan 18, 2016 6:55 am
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- configure softphone to send SMS to asterisk
by stephend » Sat Mar 21, 2015 4:30 pm
- 1 Replies
- 1589 Views
- Last post by shree_venus
Mon Jan 18, 2016 1:48 am
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- Asterisk 11 - Stopmixmonitor issues
by edestechnical » Fri Jan 15, 2016 4:59 am
- 1 Replies
- 1322 Views
- Last post by abw1oim
Mon Jan 18, 2016 1:43 am
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- One way Audio
by mettichi » Fri Jan 15, 2016 1:01 pm
- 4 Replies
- 2585 Views
- Last post by mettichi
Sat Jan 16, 2016 4:36 am
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- Original Asterisk/Digium AADK s800i
by resultz_1 » Thu Jan 07, 2016 9:39 pm
- 2 Replies
- 1500 Views
- Last post by resultz_1
Fri Jan 15, 2016 12:16 pm
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- Logging into GUI receive Fatal Error
by ascheinman » Thu Jan 14, 2016 9:23 am
- 2 Replies
- 1778 Views
- Last post by ambiorixg12
Thu Jan 14, 2016 8:53 pm
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- Change DTMF mode in dialplan with PJSIP
by phonefxg » Thu Jan 14, 2016 6:21 am
- 2 Replies
- 1612 Views
- Last post by phonefxg
Thu Jan 14, 2016 8:50 am
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- [RESOLVED] PJSIP partitioning endpoint selection
by boucher820 » Wed Jan 13, 2016 10:01 am
- 4 Replies
- 1548 Views
- Last post by jcolp
Thu Jan 14, 2016 8:50 am
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- CDR direct execute failed
by vitormazuco » Wed Jan 13, 2016 7:51 am
- 1 Replies
- 1783 Views
- Last post by abw1oim
Thu Jan 14, 2016 1:54 am
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- System stopped routing calls until Asterisk service restart
by newguise » Thu Jan 07, 2016 9:47 pm
- 2 Replies
- 1772 Views
- Last post by newguise
Wed Jan 13, 2016 5:13 pm
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- Issue while compiling Asterisk Software
by tella » Tue Jan 12, 2016 3:27 pm
- 14 Replies
- 4192 Views
- Last post by tella
Wed Jan 13, 2016 12:36 pm
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- Getting "488 Not Acceptable Here" on inbound call
by marama » Mon Jan 11, 2016 3:45 pm
- 3 Replies
- 2938 Views
- Last post by jcolp
Wed Jan 13, 2016 11:30 am
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- Writing an Asterisk security statistics module for statsd
by nwcoaster6 » Wed Jan 13, 2016 1:42 am
- 0 Replies
- 2140 Views
- Last post by nwcoaster6
Wed Jan 13, 2016 1:42 am
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- Extension voicemail press 1 for cell 2 for voicemail
by opterons » Fri Jan 08, 2016 1:24 pm
- 3 Replies
- 1819 Views
- Last post by opterons
Wed Jan 13, 2016 12:38 am
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- sip2sip registers but incoming calls get "No matching endpoi
by lardconcepts » Wed Jan 06, 2016 5:53 pm
- 1 Replies
- 2303 Views
- Last post by lardconcepts
Tue Jan 12, 2016 4:54 pm
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- Voicemail
by aristech » Tue Jan 12, 2016 2:59 pm
- 1 Replies
- 1320 Views
- Last post by ambiorixg12
Tue Jan 12, 2016 3:09 pm
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- Unknown for CID
by campch5 » Tue Jan 12, 2016 12:06 pm
- 6 Replies
- 2360 Views
- Last post by david55
Tue Jan 12, 2016 1:21 pm
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- A problem about SIP TRUNK. RTP remote media address.
by qqgod » Mon Jan 11, 2016 10:05 pm
- 1 Replies
- 1744 Views
- Last post by david55
Tue Jan 12, 2016 12:29 pm
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- SIP call disconnects after 20 seconds
by campch5 » Wed Dec 23, 2015 4:07 pm
- 5 Replies
- 6604 Views
- Last post by david55
Tue Jan 12, 2016 12:08 pm
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- [RESOLVED] updating libsrtp
by meneses » Tue Jan 12, 2016 7:13 am
- 2 Replies
- 1454 Views
- Last post by meneses
Tue Jan 12, 2016 11:34 am
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- trouble with: call to external, forwarded to internal
by buckman666 » Sat Jan 09, 2016 7:50 am
- 8 Replies
- 3324 Views
- Last post by buckman666
Tue Jan 12, 2016 7:50 am
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- Lead calls to queues equally?
by peter59 » Mon Jan 11, 2016 6:42 am
- 4 Replies
- 1858 Views
- Last post by satish4asterisk
Tue Jan 12, 2016 3:45 am
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- Can't make calls through GSM module
by problemsolved » Sat Jan 09, 2016 12:58 pm
- 5 Replies
- 2182 Views
- Last post by ambiorixg12
Mon Jan 11, 2016 10:37 pm
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- Permanently modify accept header
by dr1esken » Mon Jan 11, 2016 7:38 am
- 3 Replies
- 1520 Views
- Last post by dr1esken
Mon Jan 11, 2016 1:23 pm
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- Sip custom tag in Header
by gerlado » Mon Jan 04, 2016 12:51 pm
- 3 Replies
- 3168 Views
- Last post by gerlado
Mon Jan 11, 2016 6:54 am
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- Disable DTMF logging to specific call
by stuartalt » Fri Jan 08, 2016 2:34 pm
- 4 Replies
- 2283 Views
- Last post by stuartalt
Mon Jan 11, 2016 6:42 am
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- Sometimes phones don't ring, sometimes no audio?
by zhoujianfu » Sat Jan 09, 2016 12:19 am
- 2 Replies
- 2061 Views
- Last post by zhoujianfu
Sun Jan 10, 2016 12:32 am
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- dahdi install under centos 7
by meral » Fri Dec 25, 2015 7:00 pm
- 14 Replies
- 13400 Views
- Last post by sufialhussaini
Sun Jan 10, 2016 12:05 am
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- asterisk odbc realtime sip registration
by system32 » Thu Jan 07, 2016 10:29 am
- 4 Replies
- 2324 Views
- Last post by system32
Fri Jan 08, 2016 3:16 pm
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- Asterisk AMI PlayDTMF to instantiate disconnect key combinat
by jakehallas » Fri Jan 08, 2016 12:59 pm
- 1 Replies
- 1357 Views
- Last post by david55
Fri Jan 08, 2016 2:53 pm
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- Pulling Queue Agent information in order to send email
by pguima » Thu Jan 07, 2016 2:08 pm
- 3 Replies
- 1721 Views
- Last post by csyria
Fri Jan 08, 2016 11:35 am
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- Is it possible capture and play voice with Asterisk ARI
by dartan » Thu Jan 07, 2016 11:56 am
- 4 Replies
- 1910 Views
- Last post by dartan
Fri Jan 08, 2016 7:23 am
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- mgcp for help
by xitu » Fri Jan 08, 2016 3:49 am
- 0 Replies
- 1405 Views
- Last post by xitu
Fri Jan 08, 2016 3:49 am
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- [help] slin16 codec translation wrong behaviour
by victor_sverdlin » Wed Dec 23, 2015 5:52 am
- 1 Replies
- 3285 Views
- Last post by victor_sverdlin
Fri Jan 08, 2016 2:25 am
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- outgoing call drop radon dahdi TDM410P
by jhidalgo_jhp » Thu Jan 07, 2016 7:42 am
- 1 Replies
- 1430 Views
- Last post by david55
Thu Jan 07, 2016 3:55 pm
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- Pri A101 sangoma down/active
by dr.x » Thu Jan 07, 2016 10:05 am
- 0 Replies
- 1399 Views
- Last post by dr.x
Thu Jan 07, 2016 10:05 am
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- Problem in sharing data between EXE and SO file:
by boobalan » Wed Jan 06, 2016 11:07 pm
- 1 Replies
- 1295 Views
- Last post by david55
Thu Jan 07, 2016 7:41 am
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- js200-fx google voice authentication fails
by js200fx » Sat May 30, 2015 1:05 pm
- 3 Replies
- 3238 Views
- Last post by protocold
Wed Jan 06, 2016 11:39 pm
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- how to add c++ program to asterisk app module ?
by nwcoaster6 » Tue Jan 05, 2016 2:09 am
- 3 Replies
- 3168 Views
- Last post by david55
Wed Jan 06, 2016 4:51 pm
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- ATA Recieves SIP CANCEL
by sjk303 » Tue Jan 05, 2016 9:54 pm
- 5 Replies
- 3641 Views
- Last post by sjk303
Wed Jan 06, 2016 2:46 pm
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- Asterisk 13 - Address Family mismatch between source
by dbbrito » Fri Nov 14, 2014 7:16 pm
- 7 Replies
- 11253 Views
- Last post by stevenharzan
Wed Jan 06, 2016 12:41 pm
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- Dial with a specific music on hold
by sebclick » Wed Sep 02, 2015 9:16 am
- 1 Replies
- 2187 Views
- Last post by sebclick
Wed Jan 06, 2016 10:46 am
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- AsteriskNow Installation Problem
by khan_sikki » Fri Jan 01, 2016 8:36 am
- 9 Replies
- 10572 Views
- Last post by khan_sikki
Wed Jan 06, 2016 9:45 am
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- [HELP]: calls do hangup on pickup with chrome
by nth281 » Tue Jan 05, 2016 5:57 pm
- 0 Replies
- 4749 Views
- Last post by nth281
Tue Jan 05, 2016 5:57 pm
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- Asterisk Outbond call issue
by anishmenon » Wed Dec 30, 2015 2:09 pm
- 1 Replies
- 12185 Views
- Last post by david55
Tue Jan 05, 2016 4:09 pm
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- Using Alphanumeric Chars. For The Outgoing Dial Rules
by burhantetik » Tue Jan 05, 2016 1:12 am
- 1 Replies
- 2652 Views
- Last post by david55
Tue Jan 05, 2016 4:01 pm
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