[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

[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

[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

Re: [asterisk-users] ZTDUMMY Running but IAX2 message:Unable to support trunking on peer 'XXXXXXXX' without zaptel timing

? 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

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 --

[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

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:

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

[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

[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:

[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

[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

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

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

[asterisk-users] VoiceMail - audio problem

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

[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

[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

[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___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To

Re: [asterisk-users] Dialing plan Question

,${EXTEN:5},1) exten = _900020[0-8].,2,Hangup exten = _900030[0-8].,1,Goto(route,${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

[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:

[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:

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

[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

[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

[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:

[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.

[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

[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

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

[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

[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

[asterisk-users] mISDN install on Asterisk 1.6 failing

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

[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

[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

[asterisk-users] sysmon on Centos Asterisk system using 100 perc CPU..How to kill it?????

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

[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

[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

[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

[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

[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