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...ug output of a call from my softphone to voicemail (ext 232 does not answer). Can anyone explain the cutoff? Thanks ------------------------------------------------------------------------ ------------ pbx*CLI> sip debug SIP Debugging Enabled pbx*CLI> Sip read: INVITE sip:232@pbx.ocg.ca SIP/2.0 To: <sip:232@pbx.ocg.ca> From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660 Via: SIP/2.0/UDP 172.31.254.106:9330;branch=z9hG4bK-d87543-22694588-1--d87543-;rport Call-ID: 113d5508a72b5176 CSeq: 1 INVITE Contact: <sip:233@172.31.254.106:9330> Max-Forwards: 70 Allow: INVITE...

...h Sendmail without complex rules / virtual user tables). They also include error logs, parameter checking, etc. Let me know if you want them Michelle Dupuis Technical Support Specialist Oxford Consulting Group Ltd. Making IT work for your business... T: (519) 672-8238 E: <mailto:support@ocg.ca> support@ocg.ca W: <http://www.ocg.ca/> www.ocg.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051007/33ee15dc/attachment.htm

After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX protocol went downhill and many carriers (like VoicePulse) are discontinuing support for IAX. Is this correct? We are all heading for SIP? Thanks, MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Quoth: OCG Technical Support <support at ocg.ca> > >Although we've programmed the softkeys per the manuals, they seem to have no >effect (just dead). For example, our 57i is setup like this: I had similar problems and ended up using the speeddial inband functionality. FWIW, my 57i's...

I'm looking for some ideas of people who have setup a sidecar (eg: Aastra 560M). Obviously it's handy for BLF (to see who's on a call)...but what else? Anyone want to share interesting things they've done with a sidecar? -------------- next part -------------- An HTML attachment was scrubbed... URL:

After more investigation, I decided to just recompile asterisk (on my newly upgraded Fedora core 4 system). Make dies with this error: "No rule to make target 'usr/lib/gcc/i386-redhat-linux/3.4.3/include/stddef.h" It seems this directory is gone under FC4, and replaced by No rule to make target 'usr/lib/gcc/i386-redhat-linux/4.0.0/include/ I can't find the

Hello, Is there or can anyone provide a comprehensive guide (designed for Linux/Asterisk novices) to installing/setting up Asterisk in order to support Fax2Mail service? In my case, I would like Asterisk to receive fax calls to predefined numbers (ranges) and to associate each of these numbers to email addresses. Thank you in advance. David --------------------------------- Yahoo!

Anyone tried Asterisk with Fedora 9 (recent release)? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080519/d16cc79d/attachment.htm

Dear, I am looking help with the asterisk pbx, how to setup lynix and asterisk . Thanks -- Talkvoip Telecom Canada Tel:416-893-2089 email: info@talkvoip.ca , talkvoip@gmail.com www.talkvoip.ca <http://www.talkvoip.ca> -------------- next part -------------- An HTML attachment was scrubbed... URL:

...et a ticket with 'kinit administrator at mydomain.com'. But my samba shares don't work. In fact, when I browse (from windows 7 domain member) to the host (lserver), it just times out. Similarly, when I try from another Linux server: smbclient //lserver/test -U administrator at ocg.ca Enter administrator at ocg.ca's password: session setup failed: NT_STATUS_NO_LOGON_SERVERS I've gone in circles adding nmb, windbind, changing smb.conf options, etc. After 3 days I'm pulling my hair out. My exact same configuration works fine on Centos 6. I've included...

...ts.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com> Sent: Sunday, June 28, 2015 9:26 AM To: Asterisk Users List Subject: Re: [asterisk-users] Branch based on call volume On 27Jun, 2015, at 15:34, Michelle Dupuis <mdupuis at ocg.ca<mailto:mdupuis at ocg.ca>> wrote: Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? Do you mean large number of calls or how loud the call is? -- Cheers, Matt Riddell _______________________________________________ htt...

Is that a bios setting (I don't recall seeing it) or an OS setting? I run a lot of Via C3 machines (they are so nifty) but don't remember seeing this. -----Original Message----- From: Technical Support [mailto:support@ocg.ca] Sent: Tuesday, September 06, 2005 10:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; shmaltz@gmail.com Subject: RE: [Asterisk-Users] Asterisk overheating on VIA Epia MSeriesmotherboard You can dramatically reduce the heat from your EPIA board by turning on CPU sc...

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a 7921 wireless phone working with Asterisk, and I want to tighten the wide open port range of my IPTABLES now. I tried allowing only SCCP port (2000) in/out and found that my audio was gone. A quick look at my iptables message shows source port 15886 and dest port 25968 used: FORWARD - Drop: IN=eth1 OUT=eth2 SRC=172.31.253.4 DST=172.31.254.102 LEN=200 TOS=0x18 PREC=0xA0 TTL=63 ID=0

I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x "restart gracefully" However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks

Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150627/6774c750/attachment.html>

Any users of the VocTel VOIP service? (Canadian) How have you found the quality (Choppy / smooth audio)? Any problems registering? (I have been unable to register for hours) After reading about the collapse of a big USA VOIP provider, I'm curious Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050629/41f09bdd/attachment.htm

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone written a LogWatch script for Asterisk? I use logwatch for monitor all my critical services and would like to do the same for Asterisk. LogWatch is very popular, so I'm guessing that someone has created one but hasn't had time to post it somewhere... Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050627/753d72c0/attachment.htm