search for: wingrin

Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081211/85bd0069/attachment.htm

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 [0872200189 at In:2] VoiceMail("SIP/voip-1fd034e0", "910|u") in new stack -- <SIP/voip-1fd034e0> Playing 'vm-theperson' (language

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090304/38e17d3e/attachment.htm

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:

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx 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

...w g729' for other possible commands) chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! Please help. Purchased the license file and registered it. Loaded: http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v35_nocona.tar.gz 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:shaunw at a1telecoms.co.za Keeping you TALKING for LESS! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.co...

...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:shaunw at a1telecoms.co.za Keeping you TALKING for LESS! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.co...

Can I use grep ? Tried but not working. please help Thanks Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081212/c856a1f1/attachment.htm

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 '2f197e56611061a678c13b881b2691a9 at 411.2.139.106'. Giving up. Tx -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090423/c491a7b9/attachment.htm

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100410/0d4e92e9/attachment.htm

...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: shaunw at a1telecoms.co.za Keeping you TALKING for LESS! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.c...

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 at 41.1.1.1>' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010]

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

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 [082449627 at private: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

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080821/3d47a67a/attachment.htm

Hi, What is good monitoring software to run to get the above info and more? Tx Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080913/21a3679a/attachment.htm

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 Host

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

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