similar to: Realtime rtignoreexpire bugged ??

All, we're running realtime and ast 1.2.7.1 stable. The problem I'm having is when you register a SIP device on ServerA, works fine, sip show peer works fine. When you dial the SIP device from ServerB however, it tries to dial, even does a MySQL query like it should but comes back saying no route to dest. When I do a 'sip show peer ..' on ServerB, the peer comes back fine, it even

Having problems getting realtime working, I'm trying to use odbc for all of this. I've got Fedora 3 and have been fighting with odbc for a day now. I think I got it working correctly, however I can't seem to get the realtime portion working. In asterisk 'odbc show' shows it connected, I see it on my (odbc) mysql server connected and all, it connects and just idles. So, without

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

Can anyone shed some light on realtime caching? My desired behavior is that MWI works with realtime voicemail/sip/extensions AND updates to the database take place on the next call to the extensions. Right now I have rtcachefriends=yes, and MWI works, but updates to the database for a cached user seem to still require a reload. It is my understating that removing rtcachefriends will

Dear All, I am currently working on asterisk cvs-head version in order to use realtime with mysql, 2 asterisk servers with duplicate mysql databases, one asterisk server is serving the sip phones and the data is logged to the database and replicated to the other asterisk database, when the first server fails though it has the sip phones data in it's database the sip phones need to re-register

I don't see anything with sip show users and sip show peers, however it works! Is there a trick? I have installed realtime (sipbuddies) on one machine and see sip show peers/users and on my new installed system I don't. Have I forgotten something? bye Ronald

Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura "support". Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-)

Any ideas? I edited the Makefile as instructed, ty. Now compiling .... sccp_channel.c 279 lines sccp_channel.c: In function `sccp_channel_send_callinfo': sccp_channel.c:48: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple

We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer

Has something changed in the recent modifications to Asterisk that would break dialing of the IAX peer? We're getting these authority failures everywhere. Everything is configured just the way it was half a year ago, this is the message we're getting on the most recent vers of asterisk. Wiki says nothing, nor does the ast-dev list.. -lost Mar 18 12:55:23 NOTICE[3479]: chan_iax2.c:6545

All, I'm not sure how to word this question but we're noticing a lot of our asterisk boxes no longer have multiple asterisk child processes. i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used to seeing 8+ .. There is no rhyme or reason to it, and we're using the safe_asterisk script which has always worked in the past. Ast 1.2.4, zap 1.2.4, naturally..

Ok, here's an odd one. I would have opened a bug on this but last time I tried that I got flamed.. :) Problem: When proxy requests digest challenge (SIP) Asterisk responds normally with the exception that for some reason it changes the FROM: (Also changes Contact: )to what's in the original TO: line. Why on earth is it doing this?! It must be a bug, I've gone over my extensions.conf

Hello, I'm having a problem when trying to create realtime stuff for asterisk 13 in mysql DB. Following https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic, I've successfully created the main tables using "config" schema. > Update |script_location|to the schema to update. Asterisk currently > supports two sets of schemas: > > 1.

On Mon, Sep 12, 2016 at 3:01 PM, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 9/12/16 3:39 PM, George Joseph wrote: > > > > On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > >> >> >> On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> >> wrote: >> >>> Has

ala cisco 7960 -----Original Message----- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck.

On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> > wrote: > >> Has anyone successfully used Mysql realtime PJSIP with Asterisk >> 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the >> following error now: >>

Hi, The asterisk wiki page says: "Sorcery.conf allows you to try to configure other PJSIP objects such as transport using realtime and it currently won't stop you from doing so. However, some of these object types should not be used with realtime and this can lead to errant behavior." Which objects and is this still true in 1.13.1 ? Thanks, Antonio. PS:

Hi, I'm having 2 problems: 1) MOH in realtime is not working, I have configured it but never go to look at the database, no warning or error found and I can do a query using realtime and the family from the cli. 2) I have SIP phones via realtime, if I register one of them and a call to a queue comes the call is never delivered to the phone, I have to make a call from the

Is the template capability in sip.conf compatible with realtime sip.conf entries such as users in a database? I notice that contrib/realtime/mysql/sippeers.sql and the wiki page don't mention a template column: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure while some third-party examples do suggest that a column named "template" is permitted: