similar to: RE: [asterisk] VocTel service provider

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:

I had my asterisk server working fine with FWD as a SIP provider, so I now added a second SIP provider (voctel). The addition to my sip.conf file is almost identical to FWD, however, asterisk now generates lots of debug messages for some strange reason! In particular, the line "##### Testing 127.0.0.1 with 172.31.0.0" shows up every two seconds! (See my log below). If I comment out

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

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

All, A bit of background: I work on a QA API on a network that is very choppy (a lot of network interrupts), and we use ssh to do a large part of this automation. This leads to some problems: ssh connections seem to be sensitive to network state, becoming unusable if the choppiness reaches a certain threshold, and either timing out or disconnecting if this happens. Anyways, I stumbled across

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:

Hi, I recently purchased from a friend 2 Cisco 7905G for testing them with Asterisk. I was able to upgrade one of them with the SIP image, the other hang up during the upgrade process and now it won't boot again. When powered up, the red and green lights keep on and the screen is blank. Does any one know a procedure to fix this ? I do not have a contract with Cisco, I have even call a

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:

My dial plan seems to work great - in that when I call extensions 1234 it connects to 1234. Strangely, after the call terminates (the other side hangs up first), Asterisk continues in the same context and then matches to extensions _. which causes an invalid extension error! Why does asterisk not leave the context (called internalmenu) after the remote hangup? Instead, it continues to the

Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules /

The Asterisk Developer Team is proud to announce the Asterisk SPE v1.0 Beta Release for immediate download on tftp.digium.com. The SPE has been developed as a joint project between Digium, the Asterisk Company, Voop, the European Asterisk Dialtone provider and the Asterisk community. The Asterisk Service Provider Edition is focused on the needs for the new breed of Telecom companies - the

Last week I was able to do some debugging of the problem I'm having with IAX2/GSM, residential-grade broadband, and VOIP. To summarize, I am having a great learning experience with * and Zap cards, SIP and IAX2. I hit a wall though, when I registered with iaxtel and tried doing VOIP. I spend the better part of a workday with the jitterbuffer and all sorts of settings and finally started to

I'm thinking of placing Asterisk on an itx motherboard in a tiny case. The ITX motherboards top out around 400Mhz PII (in terms of power relative to a desktop). How much CPU would I need for an office of 50 people? How much disk storage for voicemail + OS? (typical / average) The system will have no PCI cards (no Digium FSO/FXO cards) - everything over the LAN connection. Thanks,

I have asterisk up and running now, and installed XLITE on 2 PC's. Both machines (mistakenly) registered as the same user / extension. Strangely, asterisks allows this and both phones can make calls! But, only the first one to register can receive calls at the extensions. 1. Is this normal behavior? (Why allow 2 phones on same extension) 2. Why is asterisk not showing the second phone when

We've got some replies to questions online about Astricon and we now have a mirror available at: http://astricon.voctel.com/news.php If anyone has any comments about Astricon, please forward them to me and I will put them up on the site so that all the people who didn't go can read them. Cheers, Matt Riddell http://www.sineapps.com/news.php (Daily Asterisk News - html)

Hi all, It appears that something very strange is going on... Here's the deal -- whether someone calls in to my * server or I call out (doesn't matter), I can hear them perfectly: no gaps, no packet loss, nothing ... however, when I speak there seems to be very noticable latency and "choppiness" as if there were packetloss or lots of jitter. I'm using SIP for outgoing

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 setup like so: softkey4 type: speeddial softkey4 label: "*Park" softkey4

?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation). ________________________________ From: asterisk-users-bounces at lists.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:

I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on

I have a new CentOS 7 installation which I joined to my domain using 'realm join mydomain.com'. That worked great. I can get 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