similar to: [IAX] Everyone is busy/congested at this time (1:0/0/1)

I am trying to find where the authoritative sql definitions for Asterisk Realtime Architecture ARA are located. I have found many locations but each and everyone seems to be different. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example Files included with the distribution:

Hi, I think I encountered a bug in the qualify timers for IAX on asterisk 1.8 but I'd like to check if I'm not messing up in my config somewhere before reporting a bug. In my IAX peer configuration I have this: [remote-host] type=friend host=172.16.6.45 username=remote-host secret=test notransfer=yes qualify=16000 qualifyfreqnotok=30000 disallow=all allow=alaw allow=ulaw allow=ilbc

Hi all I have asked this question before but have not got any helping input. I'm really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand Brands is not used, Cards just makes the cards. Routed in the dialplan and pricelist, Trunks is for ASTCC to

After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip Feb 17 15:20:03 WARNING[15317]: config.c:823 read_config_maps: The

This has come up before on the list and archives but I don't seem to find a solution for this. On just a few nodes we have this situation where we see the IP disappear from the CLI iax2 show peers list but the status shows OK: 3012/3012 (Unspecified) (D) 255.255.255.255 0 OK (89 ms) How can the status be OK a few milliseconds ago and have no IP ?? The strange thing is

Good morning everybody, Can someone explain to me the interconnection between these four things: indications.conf, SetLanguage(), zaptel.conf and ring-back ? If there is any !! :- ) I am having this case where some users cannot hear ring back from a DeadAGI script and it seems to be interconnected to these items. These users are from the iaxfriends table, they _can_ hear ring-back from a

Qualify=yes will send a SIP OPTIONS periodically and keep the NAT open, if you use 1 to 1 NAT (versus PAT where it is "many to one NAT") it will work because port 5060 on the private address will still be port 5060 on the public address. With PAT the port could be anything over 1024, but usually much higher, and the originator will send to port 5060, which your NAT router will drop.

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

Hello I was wandering If I let 2 asterisk boxes (let's name them ast01 and ast02) connect to one SQL realtime iaxfriends/sipfriends database What happens if I register my client to ast01, The ast01 box will update the client's record in the iaxfriends database (ipaddr/port/regseconds) Let's say there is an incoming call then for this client but this call arrives on ast02 (the box

dear all, does anyone have a clue why in the event messages it show that Unregistered '1000' (AUTHENTICATED) if i'm using iaxfriends ? if using iax.conf text file configuration ... the status showed Registered '1000' (AUTHENTICATED) i'm using asterisk 1.0.3 and iaxcomm-linux (pre CVS 28 Feb 2004) regards, __________________________________ Do you Yahoo!? Yahoo!

Hi everyone, I'm a newbie to Asterisk. Will Asterisk run on RH9, easily or does it have to run on FreeBSD? Will the drivers for the Digium cards work on RH9? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040919/7d981e5e/attachment.htm

Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip

Can someone shed some light on this? It sounds like exactly what I am looking for. Does it handle extensions.conf or just sip/iax/voicemail? (not that to say that _just_ those things would be cool) I have googled for some more information, but so far the only thing I can find is in the bug tracker and perhaps I'm missing something, but I don't get a full explanation. Any insight would be

Hello everybody! We at the Max Planck Society of Molecular Plant Physiologie want to know, if there is R version available for a 64 bit computer. If yes for which processor an amd or for a 64 bit Mac Thanks a lot Daniel ____________________________________________________ Daniel Weicht Max Planck Institute of Molecular Plant Physiology Am Mühlenberg 1 14476 Golm,

Greetings list, I have a scenario where half a dozen phones at a site appear to be dropping offline for a few seconds every few hours, but the connection between them and the asterisk server remains up. It's been suggested to me that the problem might be to do with qualify - which is enabled in this case. However, I don't really want to disable it if at all possible - it's a very

I know this has been moved to contrib, but is anybody using it successfully? We are looking at using Asterisk for the fine IVR features it has, tied in with another platform. Calls are getting routed to it, but the following is happening: (I've redacted phone#s and passwords) asterisk log: Sep 17 11:56:24 WARNING[17423]: No entry in voicemail config file for '+13609XX2000' Sep 17

I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12

Hello list, what is the difference between mohsuggest & mohinterpret when defining a SIP peer ?! If a certain SIP peer puts another channel on hold, what field then determines the moh class that Asterisk will choose to play to that channel ? If I take the test and call from peer A to peer B, and peer A puts peer B in hold, then the class of peer B is taken... that's not what I want.

I'm wondering if moh is still broken in Asterisk 1.4 beta3. In Asterisk 1.2, when a callee put a caller on hold, the musiconhold class that was played was not the one the callee wanted the caller to hear, but something else. Even after using mohsuggest in Asterisk 1.4, it still appears that this is not working correctly. Here's the results of a simple test: CASE CALLER CALLEE