similar to: auto load SIP peers on startup

I have read and followed as much as I can find but I am missing something. What I want to do is get as much as I can running from mysql and keep the *.conf files for static things. So I have setup a SIP users/peers table in a mysql database and I have populated it with a few peers. I have configured asterisk addons and from the asterisk CLI I am able to search the sip users / peers tables using

Hi! I've read in the asterisk docs (AstARA.html) that realtime users/peers can't be notified (MWI with SIP NOTIFY) when they have new voicemail messages, because their object are not persistent in memory?! But that's what we really need!!! Is their any work in progress to get these things working? Or are their any known workarounds? I'm using asterisk version 1.2.1 with OBDC

I've been going in circles for a few weeks now with Realtime SIP. My extconfig.conf has: sipusers => mysql,dbname,ast_sip_users sippeers => mysql,dbname,ast_sip_users When I do a 'sip show peers' I see all my phones. When I do a 'sip show users' I only see a few of them. I can't work out why this is the case. They are also coming up with NAT as

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

Greetings, I'm having some trouble with the realtime engines. When asterisk loads, everything looks fine, there don't seem to be any problems via notices or anything. Furthermore, cdr_odbc is working, and actively logging my failed call attempts to db through ODBC using the same DSN. unixODBC and the mysql drivers are installed from source. Here are the relevant parts of the config:

Im glad to let you know that finally I invested some time to make work Unicall in Asterisk 1.4, I must say not much testing could be done since I have no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

Hi im running several gentoo servers with Asterisk, only using IAX2 and SIP. Recently we decided to implement h323. All the necessary dependences for oh323-0.7.3 were installed by portage (package manager of Gentoo distro), including openh323, pwlib etc. The module is successfully loaded (load chan_oh323.so) but when asterisk is stopped (stop now) or the oh323 module is unloaded (unload

Hi Folks, I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format, are there some open source aplication for Asterisk that be easier for use. Pls anything suggestion will be very appreciate. Thanks Rgds. -- Omar E.P.T ----------------- Certified Networking Professionals make better Connections! http://omarept.blogspot.com/

Hi, I'm testing Asterisk with MySql and I would want to insert sip users in a table "sip_users". After I modified extconfig.conf with "sipusers => odbc,asterisk" and I create the table sipusers, which changes must I make to sip.conf? Thank's Maury P.S.: C'? qualche utente italiano nella mailing list? -------------- next part -------------- An HTML attachment

Does anybody have a working Asterisk server with Unicall using MFCR2 in Brazil? Were having problems. It seems SPANDSP never detect the tones from the telco. Im using brazil protocol variant. Im having lots of problems to find out why spandsp seems to not detect the MF tones. We send the first digit, the telco says they receive it, and respond with the proper signal to ask for the next digit, we

Good day all We installed a box a long time ago and they bought g729a licenses Now we want to upgrade and reinstall,whats going to happen with the codec,if I give the box the same ip as always will it work? Please Help

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 =

Hi, Anyone there could figure me out on how to install my unicall. I followed the instruction below in the stated site at; http://soft-switch.org/unicall/installing-mfcr2.html. Questions: 1. On what directory/folder should I copy the chan_unicall.c, channels_makefile.patch? 2. On what directory/folder should I command patch < channels_makefile.patch? 3. On what directory/folder should I

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header

Hi all, I have a trouble when I try to configure asterisk to make calls between IAX and SIP. IAX I'm using to connect between asterisks a on SIP I have phones. The calls come from higher asterisk to my on IAX, SIP phone is ringing and when I hang up then dial command ends and connection is loss. When I'll make connection between asterisks on SIP then all work fine. Does anybody has any

Hello all, Has anyone tried 8xE1 in one box using Asterisk and Digium boards ? What is the CPU needed for sustained performance in this capacity ? Is this affected if G.729 codec is used ? Regards, Jorge A.

Dear Moises Silva, Chris Thompson and others, I've finally managed to find the problem with my PHP AGI script! Apparently, my machine has 2 versions of PHP installed. One for CGI and the other for CLI. Without knowing, to hand test my script I was calling the CLI version (/usr/local/bin/php), whereas Asterisk was calling the CGI version (/usr/bin/php). It turns out that my script

Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Uys, Steve Underwood I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I get the far and local end unblocked but as soon as I try to make a call I get dialing and then protocol failure.. Do you guys know if there are any issues with sangoma and unicall? Anybody has an a101 card working with unicall and r2mfc? Are you out there Steve? :)