[asterisk-users] Matching asterisk PBX cdrs to Telco's Trunk CDR's
Please see below. --Original Message-- From: sha...@a1telecoms.co.za To: asterisk-users@lists.digium.com ReplyTo: sha...@a1telecoms.co.za Subject: Matching asterisk PBX cdrs to Telco's Trunk CDR's Sent: Feb 20, 2012 17:43 Say, the Telcos CDR's have date, time, duration. number dialed and cost, but no extension number. * has extension, but no cost. I'm looking for some software to marry these to sets of data records. Time and Duration may be out a few seconds and number dialed may be duplicated. Any ideas? Tx Shaun VOIP Telecoms Solution Provider BSc. (Elec. Eng.) UP A1 Telecoms cc Office: 087-940-0188 Mobile: 082-449-6273 Fax: 088-011-640-5633 Email:sha...@a1telecoms.co.za Keeping you connected for less -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speed Dials Management....
Say, Is there any existing add-on / code etc. that manages speed dials. I find myself dialing number repeatedly and think that it would be great to have a system that can be controlled from the telephone instrument and work on the fly to build up a speed dial list. I would like that after I dial a number I can record a tag and have a speed dial no assigned. I should be able to dial using this speed dial and hear my tag played for me and also have the option of keying in a description. It would be great to be able to print this list of speed dials and the no’s assigned to them. I use the TrixBox implementation... This is the closest I’ve found to what I’m looking for: http://www.ietf.org/rfc/rfc3398.txt Tx Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Customizing sip response codes for PBX Sip trunk
Say the PBX is: Mitel-3300-ICP 10.2.0.26_2 Created SIP trunk to * but PBX doesn't see trunk as unavailable when its really unavailable. It simply fails the calls.. How can I change the SIP response code to respond with e.g. All Channels busy? Any suggestions on how to program the Mitel to work? Tx Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk reload, to execute file
Say, When * reloads it changes the file permissions of below file. How can I call an executable which corrects for this? chmod 777 /var/lib/asterisk/agi-bin/dialparties.agi Tx Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing sip response codes
Say, I've a SIP extension. How can I change the SIP response code to match those needed by the registered SIP device? In this case a Mitel PBX.Tx Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sysmon on Centos Asterisk system using 100 perc CPU..How to kill it?????
15906 root 39 19 94988 832 520 R 100.9 0.0 11:22.21 sysmon15913 root 39 19 94988 832 520 R 100.9 0.0 11:21.95 sysmon15905 root 39 19 94988 832 520 R 98.9 0.0 11:24.76 sysmon15908 root 39 19 94988 832 520 R 98.9 0.0 11:20.76 sysmon15909 root 39 19 94988 832 520 R 98.9 0.0 11:21.44 sysmon15910 root 39 19 94988 832 520 R 98.9 0.0 11:23.99 sysmon15916 root 39 19 94988 832 520 R 98.9 0.0 11:13.95 sysmon Any ideas how to kill this?is found in /usr/bin Tx Shaun VOIP Telecoms Solution Provider BSc. (Elec. Eng.) UP A1 Telecoms cc Office: 087-940-0188 Mobile: 082-449-6273 Fax: 088-011-640-5633 Email:sha...@a1telecoms.co.za Keeping you connected for less -Original Message- From: Shaun Wingrin sha...@a1telecoms.co.za Date: Tue, 14 Jun 2011 22:44:40 To: Shaun Wingrinvoi...@gmail.com Subject: sysmon on Centos Asterisk system using 100 perc CPU..How to kill it? asterisk-users@lists.digium.com 15906 root 39 19 94988 832 520 R 100.9 0.0 11:22.21 sysmon 15913 root 39 19 94988 832 520 R 100.9 0.0 11:21.95 sysmon 15905 root 39 19 94988 832 520 R 98.9 0.0 11:24.76 sysmon 15908 root 39 19 94988 832 520 R 98.9 0.0 11:20.76 sysmon 15909 root 39 19 94988 832 520 R 98.9 0.0 11:21.44 sysmon 15910 root 39 19 94988 832 520 R 98.9 0.0 11:23.99 sysmon 15916 root 39 19 94988 832 520 R 98.9 0.0 11:13.95 sysmon Any ideas how to kill this? is found in /usr/bin Tx Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip, Qualify=200 that doesn't qualify. How to signal this state to the Peer
Say, If bandwidth e.g. ADSL goes fuzzy, is there a way to force * to unregister the Peers? I noticed with qualify=200 for example, even if latency goes above and * shows Lagged and then UNREACHABLE The peer's calls are still accepted. Is there a way to automatically prevent this? Thanks Shaun-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attempted SIP connection by foreign host. Help!
Say, I just picked this up on my messages! There are a whole host of these requests! Anyone know whow there people are? Is there a way to report them? Any suggestions as to how to block them? [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:17] NOTICE[1010] chan_sip.c: Registration from '912 sip:1...@41.1.1.1' failed for '184.106.217.112' - Wrong password C:\tracert 184.106.217.112 Tracing route to 184-106-217-112.static.cloud-ips.com [184.106.217.112] over a maximum of 30 hops: 1 2 ms 1 ms 1 ms 192.168.10.199 2 5 ms 3 ms 2 ms 192.168.1.197 311 ms14 ms 8 ms 196-210-138-1.dynamic.isadsl.co.za [196.210.138.1] 414 ms 9 ms11 ms cdsl1-rba-vl2360.ip.isnet.net [196.38.73.133] 510 ms 9 ms 9 ms cdsl1-rba-vl150.ip.isnet.net
[asterisk-users] mISDN install on Asterisk 1.6 failing
Hi, Has anyone had experience installing it? yum install asterisk-chan_misdn I'ts the latest Trixbox Distro version and same issues exists if add in the Trixbox repo. FAILS as per below: I have a ISDN single port PCI BRI card installed and detected. __ Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: www.ftp.saix.net * base: www.ftp.saix.net * extras: www.ftp.saix.net * updates: www.ftp.saix.net Excluding Packages from CentOS-5 - Addons Finished Excluding Packages from CentOS-5 - Base Finished Excluding Packages from CentOS-5 - Extras Finished Excluding Packages from CentOS-5 - Updates Finished Setting up Install Process Resolving Dependencies -- Running transaction check --- Package asterisk-chan_misdn.i386 0:1.4.22-3 set to be updated -- Processing Dependency: libsuppserv.so.0 for package: asterisk-chan_misdn -- Processing Dependency: libmISDN.so.0 for package: asterisk-chan_misdn -- Processing Dependency: libisdnnet.so.0 for package: asterisk-chan_misdn -- Finished Dependency Resolution asterisk-chan_misdn-1.4.22-3.i386 from trixbox has depsolving problems -- Missing Dependency: libmISDN.so.0 is needed by package asterisk-chan_misdn-1.4.22-3.i386 (trixbox) asterisk-chan_misdn-1.4.22-3.i386 from trixbox has depsolving problems -- Missing Dependency: libisdnnet.so.0 is needed by package asterisk-chan_misdn-1.4.22-3.i386 (trixbox) asterisk-chan_misdn-1.4.22-3.i386 from trixbox has depsolving problems -- Missing Dependency: libsuppserv.so.0 is needed by package asterisk-chan_misdn-1.4.22-3.i386 (trixbox) Error: Missing Dependency: libisdnnet.so.0 is needed by package asterisk-chan_misdn-1.4.22-3.i386 (trixbox) Error: Missing Dependency: libmISDN.so.0 is needed by package asterisk-chan_misdn-1.4.22-3.i386 (trixbox) Error: Missing Dependency: libsuppserv.so.0 is needed by package asterisk-chan_misdn-1.4.22-3.i386 (trixbox) You could try using --skip-broken to work around the problem You could try running: package-cleanup --problems package-cleanup --dupes rpm -Va --nofiles --nodigest The program package-cleanup is found in the yum-utils package. Shaun Wingrin VOIP Telecoms Solution Provider BSc. (Elec. Eng.) UP A1 Telecoms cc Office: 010-590-0222 Mobile: 082-449-6273 Fax: 0880-11-640-5633 Email: sha...@a1telecoms.co.za Keeping you TALKING for LESS! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk script to repeat dial of a number
Say, I'm looking for a simple way to dial a number repeatedly for two minutes at a time. The purpose is to busy up a faulty analogue line in an incoming hunt group. Tx Shaun-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Xfer extension to extension call, flash hookpass through by Asterisk needed via quintum and X-lite/Eyebeam
Say, I need to replicate what happens on a wired extension when a call is transfered and transfered back. Asterisk has to detect and pass through the flash hook to the Quintum when its pressed on the Eyebeam. My setup is:PBX--Quintum FXS port -- Asterisk 1.4 Server--Eyebeam 1.5 softphone The Quintum has to recognise the flash hook and pass this on to the PBX. What setup is needed on Asterisk and any ideas on Quintum? Thanks Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Howto see the source ip address of SIP call in cli monitor
Hi, I have qualify = no . if I set sip debugging on I can see it - but this gives many long debug messages. Is there a way to see the source ip in the cli as the calls scroll up? I only see the destination ip in the cli . Tx Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Convert file in GSM codec to G729 codec
Hi, I've tried the link http://www.asteriskguru.com/tools/audio_conversion.php but it returns an error at the moment. Any other ideas most welcome. Tx Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote host can't match request CANCEL to call
Hi, Why does this warning occur and what are the implications of it? I'm concerned about calls never getting hung up.! chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to call '2f197e56611061a678c13b881b269...@411.2.139.106'. Giving up. Tx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial command with r parameter - no ring tone
Hi, Any ideas why? If I leave it out - there is ring tone passed through. Using g729 codec. Sip based call...___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Required:Asterisk Beep tone while call connects
Hi, There is a long call setup time untill the call connects. How can I play a beep tone say every 4 seconds to the caller untill the call connects? Tx. Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G.729 VAD issue
Hi, My setup is SIP Call--Asterisk--VSP1 or VSP2 or VSP3 I'm experiencing an interconnect issue with one of the VSP's that seems to have to do with Asterisk not having any VAD control. The error is: NOTICE[10989]: frame.c:203 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end The VSP has switched off silence suppression on their Quitnum device. Any ideas are most welcome. Thanks Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 VAD issue
Hi, My setup is SIP Call--Asterisk--VSP1 or VSP2 or VSP3 I'm experiencing an interconnect issue with one of the VSP's that seems to have to do with Asterisk not having any VAD control. The error is: NOTICE[10989]: frame.c:203 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end The VSP has switched off silence suppression on their Quitnum device. Any ideas are most welcome. Thanks Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please explain the meaning of the output of lsmod | grep ztdummy?
lsmod | grep ztdummy ztdummy38856 0 zaptel231496 3 ztdummy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] say I wish to run tail command on messages file to pick up if any channels unavailable messages appear.
Can I use grep ? Tried but not working. please help Thanks Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial string required to drop any call not exactly 10 digits long
Hi,
exten = _[0-9]XXX,1,Goto(jump,${EXTEN},1)
seems to allow calls shorter than 10 digits through...
Hope you can help.
Thanks
Shaun___
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Re: [asterisk-users] Dialing plan Question
Hi Can you please help me make this into one statement...
It doesn't work if I say _9000[1-9]0[1-8].
Also would like to be able to achieve _9000[1-9]0[1-8],
Asterisk 1.4
exten = _900010[0-8].,1,Goto(route1,${EXTEN:5},1)
exten = _900010[0-8].,2,Hangup
exten = _900020[0-8].,1,Goto(route,${EXTEN:5},1)
exten = _900020[0-8].,2,Hangup
exten = _900030[0-8].,1,Goto(route,${EXTEN:5},1)
exten = _900030[0-8].,2,Hangup
all the way to ...
exten = _900090[0-8].,1,Goto(route,${EXTEN:5},1)
exten = _900090[0-8].,2,Hangup
Shaun Wingrin
VOIP Telecoms Solution Provider
BSc. (Elec. Eng.) UP
A1 Telecoms cc
Office: 087-940-0188
Mobile: 082-449-6273
Fax: 088-011-640-5633
Email:[EMAIL PROTECTED]
Keeping you TALKING for LESS!___
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[asterisk-users] Inbound calls from Asterisk to Asterisk with SIP Forbidden from 'asterisk
Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret= accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12 File: sip_custom.conf [VoipDirect777821] type=friend host=141.122.139 username=VoipDirect777821 secret=wsPiOov8830 accountcode=5260477782 amaflags=billing context=Incomming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes sip show peers shows both as registered. this is the error when try and place a call from Asterisk 1 to Asterisk 2: - Executing [EMAIL PROTECTED]:1] Dial(Console/dsp, SIP/VoipDirect777821|60|) in new stack -- Called VoipDirect777821 [Dec 1 23:20:21] WARNING[25399]: chan_sip.c:12334 handle_response_invite: Received response: Forbidden from 'asterisk sip:[EMAIL PROTECTED];tag=as070b02e2' -- SIP/VoipDirect777821-0876c360 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Hangup(Console/dsp, ) in new stack == Spawn extension (a1, 582, 2) exited non-zero on 'Console/dsp' Hangup on console I get the same error even if I include this on Asterisk 1: register = VoipDirect777821:[EMAIL PROTECTED] Please help ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail - audio problem
Please help... The 1st voicemail message after a reload has audio to the caller. All subsequent calls have no audio to the caller even though the same voicemail application is being called? Asterisk Version 1.4.21.2 Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/voip-1fd034e0, 910|u) in new stack -- SIP/voip-1fd034e0 Playing 'vm-theperson' (language 'en') == Spawn extension (In, 08792200189, 2) exited non-zero on 'SIP/voip-1fd034e0' voicemail.conf [default] ; Define maximum number of messages per folder for a particular context. ;maxmsg=50 910 = 910,Ext910,[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail - audio problem
Dear David, Thanks for the reply. I have lsmod | grep ztdummy ztdummy38856 0 zaptel231496 3 ztdummy but still the issue persists?! Any ideas really apreciated. - Original Message - From: David A. Bandel [EMAIL PROTECTED] To: Shaun Wingrin [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 19, 2008 10:36 PM Subject: Re: [asterisk-users] VoiceMail - audio problem On Wed, Nov 19, 2008 at 1:07 PM, Shaun Wingrin [EMAIL PROTECTED] wrote: Please help... The 1st voicemail message after a reload has audio to the caller. All subsequent calls have no audio to the caller even though the same voicemail application is being called? make sure you have ztdummy loaded. Not sure why, but I ran into a problem similar to what you're describing with 1.4.21.2 (even though I have a wcte11xp module loaded) and modprobing ztdummy fixed it. Asterisk Version 1.4.21.2 [snip] HTH, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI dial and echo recorder
Say any ideas how to do the following from the cli In order to test I would like to dial my phone from the Asterisk cli and then record my voice on asterisk and have it played back to me? Also how can a I specify a specific callerid? Thanks Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP ACCOUNT CODE not included in CDR when SIP Status is Unknown
Please help with this strange issue. When sip show peers returns status Unknown the CDR does not include the accountcode even though the call is correctly processed. I'm using A2 Billing and it uses the accountcode to determine the authentication. Asterisk version 1.4.21.2 I'm calling from a Quintum device. I'm very puzzeled. Name/username HostDyn Nat ACL Port Status 1532497439/1532497439 (Unspecified)D 0UNKNOWN The SIP settings are: [1532497439] type=friend host=dynamic username=1532497439 secret=wspiov8729 accountcode=1532497439 callerid=90002 regexten=90002 amaflags=billing context=OutboundWS disallow=all allow=g729 trunk=yes qualify=6000 qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 directrtpsetup=no Thanks Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP ACCOUNT CODE not included in CDR when SIP Status is Unknown
Perhaps this is an issue with the SIP registration? Any idea why Asterisk accepts the call if qualify fails? Please help with this strange issue. When sip show peers returns status Unknown the CDR does not include the accountcode even though the call is correctly processed. I'm using A2 Billing and it uses the accountcode to determine the authentication. Asterisk version 1.4.21.2 I'm calling from a Quintum device. I'm very puzzeled. Name/username HostDyn Nat ACL Port Status 1532497439/1532497439 (Unspecified)D 0UNKNOWN The SIP settings are: [1532497439] type=friend host=dynamic username=1532497439 secret=wspiov8729 accountcode=1532497439 callerid=90002 regexten=90002 amaflags=billing context=OutboundWS disallow=all allow=g729 trunk=yes qualify=6000 qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 directrtpsetup=no Thanks Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] show g729 seems to no longer work in latest 1.4 version. What do I use please?
Thanks Shaun___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination
The setup is as follows: SIP phone registers via international link to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2 via Zaptel Channels need to be hairpinned from Box 1 to 2. How is sip.conf configured on Box 1 and 2 so that we don't get an error: Failed to authenticate user when 1's extensions.conf uses SIP to dial Asterisk Box 2 . How do we ensure that RTP traffic flows from SIP phone registering at 1 directly to 2 without first passing through 2? Tx Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I determine if IAX trunking is being used and how many calls are being trunked?
Thanks Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1st call after some time has one way speech, but calls after that are fine..
Hi, Hoping someone can help with this most frustrating situation. I have a Linksys PAP2T registering with ADSL to my asterisk server which also sits behind a Mikrotik router. Thanks Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 and transfer=mediaonly, Error unable to transfer but there is sound.
Hi, The iax.conf is below and the trace. Any ideas please? disallow=all allow=g729 trunk=yes qualify=yes qualifysmoothing=yes nat=yes canreinvite=yes context=OutboundWS transfer=mediaonly Executing [EMAIL PROTECTED]:1] Dial(SIP/919-094d6e60, IAX2/ECom-iax/2782449627|60|) in new stack -- Called ECom-iax/2782449627 -- Call accepted by xxx.xxx.xxx.x (format g729) -- Format for call is g729 -- IAX2/ECom-iax-1 is making progress passing it to SIP/919-094d6e60 -- IAX2/ECom-iax-1 is ringing -- IAX2/ECom-iax-1 stopped sounds -- IAX2/ECom-iax-1 answered SIP/919-094d6e60 -- Channel 'IAX2/ECom-iax-1' unable to transfer -- Hungup 'IAX2/ECom-iax-1' Thanks, Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why does a perfectly fine iax2 host becomes UNREACHABLE?
Hi, I have two asterisk servers and I've created an IAX2 config on both as below. The one server shows host as OK with 20ms and the oterh shows it as unreachable? Please help. disallow=all allow=g729 trunk=yes qualify=yes qualifysmoothing=yes nat=no context=OutboundWS transfer=mediaonly Thanks Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Running asterisk as non root user
Hi, I've followed instructions of the book AsteriskFutureOf TelephonySecEdit on page 295 onwards ) Link to the Asterisk book: http://downloads.oreilly.com/books/9780596510480.pdf) and get an error when running service asterisk start. The error is: cat: /var/run/asterisk.pid: No such file or directory . I can run aserisk fine from the non-root user. Please help Code Snippet: 1: 2: 3: 4: 5: 6: 7: 8: 9: 10: 11: 12: [EMAIL PROTECTED] run]# /etc/init.d/asterisk restartShutting down asterisk: [FAILED]Starting asterisk: [ OK [EMAIL PROTECTED] run]# Asterisk ended with exit status 1Asterisk died with code 1.cat: /var/run/asterisk.pid: No such file or directoryAutomatically restarting Asterisk.mpg123: no process killedAsterisk ended with exit status 1Asterisk died with code 1.cat: /var/run/asterisk.pid: No such file or directoryAutomatically restarting Asterisk. The suggestion to do the following didn't work...: Edit the [directories] section of asterisk.conf and change the line that reads astrundir = /var/run TO: astrundir = /var/run/asterisk Then: mkdir /var/run/asterisk chown theuser /var/run/asterisk Edit /etc/init.d/asterisk And make sure there are no references to /var/run/asterisk.pid you want /var/run/asterisk/asterisk.pid instead Any help most welcome___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZTDUMMY Running but IAX2 message:Unable to support trunking on peer 'XXXXXXXX' without zaptel timing
Hi All, Hope someone can help. Asterisk version 1.4.14 is running and just installed zaptel-1.4.11with only ztdummy selected in menuselect . ztdummy seems to be running (as below) but still get the above error, even though I've stopped and restarted asterisk... Do I need to set ZAP_TIMING=-I ? Where do I do that? lsmod | grep ztdummy ztdummy 9256 0 zaptel190852 1 ztdummy Thanks, Shaun Wingrin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider
Shaun schrieb: Hi All, This is puzzling me greatly. The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to Asterisk are SIP clients. Codec throughout G729 (only have 1 license on Asterisk server loaded though). When calling the SIP clients from PAP2T I can't hear them but they can hear me. If I call from PAP2T through Asterisk using?IAX2 to a VOIP provider there is speach in both directions! Any suggestions? Thanks Shaun check your firewall/nat settings. If your setup will work for around 5 minutes after you have rebooted the pap2t then you have to active the nat keep alive and nat mapping service in the pap2t. best regards steve smith DEAR STEVE, THANKS, I DID CHECK THEM AND THEY NEEDED TO BE ACTIVATED. THIS DOES NOT SOLVE THE PROBLE. STILL ONE WAY SPEECH WHEN CALLING PAP2T --ASTERISK--SIP DEVICE ATTACHED. HOWEVER PAP2T--ASTERISK--SIP PROVIDER WORKS FINE AS WELL AS SIP DEVICE ATTACHED--ASTERISK--SIP RPOVIDER. ANY SUGGESTIOPS WELCOME___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (Newbie)How to reduce security risks in opening IAX Sip Ports
Please direct me to any usefull links to help secure my asterisk server once these ports are opened. Thanks Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterix seperate the signalling and the media ip's with Quintum
New to Asterix and perhaps someone can help. The plnned configuration is that the Quintums are to register to the Asterix and the signalling to be handled by the Asterix but the media (G 729 code) to be directed to the service provider. Thanks Shaun ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
