similar to: Using Dial() on SIP and DAHDI connections simultaneously

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).

We have two Asterisk servers. One is a live server supporting our customers, and the other is a backup server that's being upgraded and pressed into service. Both servers have a Digium TE405P T1 card in them, and in order to test the T1 service on the backup server, I've created a T1 crossover cable (as per http://www.voip-info.org/wiki/view/crossover+T1+cable) that goes from port 4 on the

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'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

You need to ensure that traffic to the SIP box is sent to the correct IP. Also if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT and traffic redirection works as is so the Asus router knows it should send the traffic through tunnel and not via WAN. IMPORTANT: Then you must, in the ASUS RT-N66U make a port forward inwards from TUN to the phone client. I would suggest

I have a Linksys 2102 ATA here that does call forwarding internally with the *72 code, however our Asterisk 1.6.2.17 server doesn't forward the call properly. This is what shows up in the console when an incoming call is made while the ATA is call-forwarded: -- Called Username -- Got SIP response 302 "Moved Temporarily" back from XX.XXX.XX.XXX -- Now forwarding DAHDI/1-1

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

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:

Hi all, I am trying to perform the following outgoing call: exten => _49.,1,Log(NOTICE,Dialing German number: ${EXTEN}) same => n,Set(route=DAHDI/g1/00${EXTEN}) same => n,Dial(${route}) exten => _0049.,1,Goto(${EXTEN:2},1) exten => _01149.,1,Goto(${EXTEN:3},1) exten => _+49.,1,Goto(${EXTEN:1},1) But this is not working. I have also tried changing the

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

I'm attempting to convert from ZAP to DAHDI with 1.6.0. I was using 1.6.0-beta9. I followed the directions I could find. I moved /etc/zapata to /etc/dahdi/system.conf I moved /etc/asterisk/zapata.conf to /etc/asterisk/chan_dahdi.conf I don't undestand how to deal with extensions.conf? I replaced Dial (ZAP/ ...) with Dial (DAHDI/ ... ) All my inbound calls from DAHDI work the same as

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI> module load chan_dahdi.so receive the following: signalling must be specified before any channels are. CLI> Warning [4663]: chan_dahdi.c:11627 process_dahdi: Ignoring signalling Error[4663]: chan_dahdi.c:10946 build_channels: Unable to reconfigure channel '1' Error[4663]:

Hi all, I am having trouble setting the signalling method for the B410P using DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or 'bri_net' - but it doesn't mind having 'pri_cpe' etc. ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown signalling method 'bri_net' Dahdi - dahdi-linux-complete-2.1.0.4+2.1.0.2 Asterisk - 1.4.23.1 Libpri - 1.4.9

Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no answer, the extension hangs up, but the dial status is busy instead of no answer. How do I get this to work -- do I need to update dahdi? The card is an X400p using its FXS module. Thanks in advance for any ideas on this. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?

So, tweaking configs, rebuilding this and that... restarting, twiddling, it works (yeah!), but fails on re-boot to work at all. Consistently, though. I believe it comes down to this: I can call out only *after* I've received a call. So, cold boot. Then: modprobe dahdi modprobe wctc4xxp modprobe wcfxo dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.3