similar to: Sending calls to a particular T1 port.

We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were

This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan

On 2016-02-17 15:32, Richard Mudgett wrote: > On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar <maillist at lightspeed.ca> > wrote: > >> Hi everyone. >> >> We have an Asterisk server running Debian Squeeze, with Asterisk >> v1.8.13.1 (basically, the Debian Stable version for Squeeze, but >> with some minor source code changes specific to our site).

Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers => mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 And here's the error messages I get: voip2*CLI> realtime mysql status

I see that the RubyOnRails.com site has migrated to Mongrel with Apache as a front-end proxy. Is that now the/a recommended setup for Rails apps? We''re still using FCGI but I''m always interested to learn more about other folk''s successful deployment choices. Across all of our sites we''re pumping out about 300k pages per day so anything that saves memory or has

Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run into a snag when compiling res_fax_spandsp (and yes, we really need that module). The old

Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part of a dialled number is the country code and city code ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050303/facb2a2d/attachment.htm

This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error:

We've had several customers report since upgrading them to our new Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer works. No significant changes have been made to their SIP configuration, nor to their ATA configuration. While not exhaustive, these are the ATAs that don't work: Linksys SPA2102 Linksys PAP2T-3.1.15 Thomson 780 Thomson 784 Unfortunately, this

The documentation of how to use the 79xx series' phones and features with Asterisk is really hard to find and put together. The higher end phones like 7970 are more like converged PC+phone, a thin client to telephony and network apps. But it's really hard to target it as a development and deployment platform because the docs and techniques are so obscure. There seems to be a fair amount

Hi, I tried again getting DTMF detection on my ISDN devices with dahdi going again. I used the channel debug to see whether asterisk sees the frames and detects them as DTMF. Interestingly here's what works: 1. GSM phone -> chan_dahdi g1 -> asterisk -> can_sip -> SIP phone Both the GSM phone and the SIP phone can issue DTMF that will be detected as features (transfer) 2.

I'm trying get a couple pieces of software we use at work so we can replace a bunch of aging Windows machines with Linux boxes, but I'm having a little trouble. I'll start with the potentially easiest problem first. One of the programs I'm trying to use is Lightspeed Virtual Terminal. It runs fairly well with just a couple of glitches. If I go to the directory it was installed

Hi! Is drag''n''drop works currently under wxruby? I want an app, where you can drag files from a wxlistctrl to a wxtreectrl. Can anyone show a little example? Gergo -- +-[ Kontra, Gergely<kgergely@mcl.hu> PhD student Room IB113 ]---------+ | http://www.mcl.hu/~kgergely "Olyan langesz vagyok, hogy | | Mobil:(+36 20) 356 9656 ICQ: 175564914 poroltoval

We are running Asterisk version 1.4.23-1, libpri-1.4.9 and zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10. We have a T1 PRI connected to the telco. Over the last 4-5 days, we have getting Yellow/Red alarms coming from the T1 PRI. The other two ports in use are connected to internal test switches (Avaya Legend/Avaya Definity), and are not showing any errors.

I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the

Yes, offcourse, the 172.19 is a "bad" example. I suggest we use the official RFC 1918 zones. 10.0.0.0/8 172.16.0.0/12 192.168.0.0/16 And for example 192.168.0.0/24 as extra, or people get confused. ;-) Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens > Rowland penny via samba > Verzonden: woensdag 7

Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/E&M Wink. I tried to move one span over one

I have a frend in Manilla who is trying to connect to an Asterisk-based VoIP provider here in Western Canada. Has anyone had difficulties with SIP in the Philippines ? I'm having a lot of trouble getting info from the provider there (PLDT) and it seems as if the device can't access a port that will allow it to get out and REGISTER with the switchboard even because the provider never sees