search for: benshaw

...d voicepulse/14121234567@VPWS WARNING[81926]: File chan_iax2.c, Line 4528 (socket_read): I don't know how to authenticate wSS36eLJ68 to 66.234.228.132 -- Hungup 'IAX2[voicepulse]/16384' wSS36eLJ68 is *not* my voicepulse username. Same thing when I try to call my other * server: [benshaw] type=peer host=my.other.*.server user=USERNAME pass=PASSWORD and the log shows -- Executing Dial("Zap/2-1", "IAX2/benshaw/1234567") in new stack -- Called benshaw/1234567 WARNING[81926]: File chan_iax2.c, Line 4528 (socket_read): I don't know how to authenticate i...

...that "zap show channels" gives me the state of the channel as well. On to the main event: the CLI log when run as "asterisk -vvvvgc": Asterisk Ready. -- Starting simple switch on 'Zap/24-1' -- Executing NoOp("Zap/24-1", "01062004-15:21:12 "BENSHAW CAN CON" <2915794>") in new stack -- Executing System("Zap/24-1", "/usr/bin/perl /usr/local/scripts/astbot.pl '"BENSHAW CAN CON" <2915794>'") in new stack -- Executing Dial("Zap/24-1", "IAX2/benshaw@ak_home/s|20|T...

...ntext=fxs signalling=fxo_ks usecallerid=yes callwaiting=yes callwaitingcallerid=yes hidecallerid=no threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=32 echocancelwhenbridged=no echotraining=yes relaxdtmf=no group=1 callgroup=1 pickupgroup=1,3 immediate=no callerid="Benshaw VOIP" <(256) 848-5432> channel => 1-16 context=fxo callerid=asreceived signalling=fxs_ks transfer=no group=2 callgroup=2 pickupgroup=2 echocancel=128 echocancelwhenbridged=yes echotraining=yes callprogress=no channel => 17-23 context=in_1234567 callerid=asreceived group=3 callgro...

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of t...

...9. Re: USB handsets/headsets?? (Dan) 10. Re: Anyone using * in a live production environment? (Andrew Kohlsmith) 11. Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users] Setcontext based on CID...) (Chris Hirsch) --__--__-- Message: 1 From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> Organization: Benshaw Canada To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] a bit frightened, guys Date: Thu, 6 Nov 2003 10:34:49 -0500 Reply-To: asterisk-users@lists.digium.com > But isn't it likely that many people call 911 simultaneously in case of > an emer...

Hi: I use shared internet bandwidth and the calls are very clear from around midnight till about 4 pm when it goes bad after that. Is there a way to boost the internet bandwidth for Asterisk at the peak time? Thanks Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Sent: Tuesday, 25 November, 2003 08:56 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls > > > > Yep, we use it for international calling. Works great: > > exten => _9011.,1,Dial(Zap/...

...ng tones with specific callers from its contact directory too. Hopefully, someone will chime in with more precise (and helpful) detail before I return to the office, but I hope my reassurance is helpful anyway... Date: Thu, 9 Feb 2006 10:49:08 -0500 From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> Subject: [Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <200602091049.08777.akohlsmith-asterisk@benshaw.com> Content-Type: text/plain; charset="us-...

At 1/16/04 7:25 AM, Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: >That's pure bullshit -- I use software RAID *specifically* because I value >my data. I don't want to buy two hardaware RAID controllers to have one >sit on the shelf just in case the first dies... and if the second dies >you're SOL because they've l...

...telco and they said everything looks good on their end, and since this a production T1 it doesn't need turning up. Now that I know support comes with these cards, I might give Digium a shout. :) Terence Date: Mon, 1 May 2006 07:20:51 -0400 From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> Subject: Re: [Asterisk-Users] PRI Issue: D-Channel woes To: asterisk-users@lists.digium.com Message-ID: <200605010720.51416.akohlsmith-asterisk@benshaw.com> Content-Type: text/plain; charset="iso-8859-1" On Monday 01 May 2006 01:42, Terence Burnard wrote: > Module...

...s and could never reach a human. So far, I am not too impressed with the response that I have received from Voicepulse. Regards, - darnell On Dec 8, 2003, at 3:23 PM, asterisk-users-request@lists.digium.com wrote: > - > > Message: 4 > From: Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> > Organization: Benshaw Canada > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Problems with voicepulse.com > Date: Mon, 8 Dec 2003 16:45:59 -0500 > Reply-To: asterisk-users@lists.digium.com > >> [default] >> exten => _1NXXNXXXXXX,1,Dia...

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To:

On Friday, June 10, 2005 3:16 AM, Andrew Kohlsmith [SMTP:akohlsmith-asterisk@benshaw.com] wrote: > On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote: > > Received: from source ([81.56.129.44]) by exprod5mx8.postini.com > > ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT > > > > Your MTA claimed it was called "SOURCE" but rDNS tells...

RE Message: 5 Date: Sat, 7 May 2005 23:18:46 -0400 From: Andrew Kohlsmith akohlsmith-asterisk@benshaw.com Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice? To: asterisk-users@lists.digium.com On May 7, 2005 11:04 pm, John Stegenga wrote: > Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive > ring - for a reasonable fee... Please do a google search for &quo...

Hi all I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec. I need to get this down to 15 sec (nat /pat firewall issue) any ideas? thanks Liaan

...een colo* and Nufone, but the jitter values between office* and colo* seem to be unsigned but negative: Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 192.168.2.2 benphone 16385/16387 00178/00177 00006ms 6356992ms 0805ms GSM 66.225.202.72 benshaw 16386/00025 00169/00171 00051ms 0012ms 0026ms GSM As a result the jitter buffer grows and grows and grows... and we get horrendous audio. jitterbuffer=off doesn't seem to make a difference. Trunking is enabled but the problem exists with and without trunking, with single calls a...

Sorry to have had to post this, But I need urgent help with configuring one adit 600 I picked up from e-bay. Issues. I cannot access the console port, I am using HyperTerminal with settings VT100, 9600, 8-N-1 I also do not have any user-manual so I am kind of stuck. Any help in getting me started would be really appreciated. Any default settings like Ethernet port address, that can help me

I have been investigating more tight integration between * and the Nortel MICS... it appears that it is at least theoretically possible to have * store voicemail and log which stations call where. Both require a T1 card. The T1 card requires either a clocking module or the 6-port fiber module to provide T1 timing. Naturally a T100P or TE405P is required on the * side. To log which

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

I have an issue with call parking and hope there is some undocumented feature for this. ;-) We are replacing our legacy PBX with asterisk, but to save money over time (handsets and network), I am trying to maintain the use of our legacy PBX. Asterisk extensions can not use the call parking features (not usable over trunk cards) of the old PBX, so I have to get the old PBX to use asterisk's.