similar to: Realtime & sip.conf

Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=><did>:<secret>@<domain>/<did context> and use realtime realtime (funny name!) for peers and friends: [myprovider]

I am upgrading to Asterisk-Realtime and stumbled upon a problem converting my existing sip.conf register command to the RealTime format. It seems that sip_friends table setup doesn't allow for such thing to happen. So far the only way I see to do this is dumping the sip_friends table setup in favor of Asterisk RealTime Static (

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

Hi, I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0. Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail. There's been no changes to our configuration I bet I'm expereiencing a Linux problem rather than an Asterisk problem, but because I know only as much Linux as required to get

Hi, I'm having trouble compiling zaptel 1.4.4 on SUSE 10.1. I'm really only interested in getting ztdummy to work because this is a dev machine with no zaptel h/w. SUSE 10.1 is a 2.6 kernel: asterisk-dev:/home/hugh # uname -r 2.6.16.13-4-default It seems that my problem is related to autoconf.h - I cannot find that file: asterisk-dev:/home/hugh # find / -name 'autoconf.h'

I have an Asterisk serving 15 people using the X-Lite soft-phone. Currently they all register to the internal IP address of Asterisk (192.168.1.110). I only use VoIP internally. External calls go PSTN. I'd like to arrange it so that they register to our external WAN address (port forwarded to Asterisk) so that they can go mobile and still have Asterisk service. Is it possible to arrange it

On Tue, Sep 16, 2008 at 2:21 AM, Michiel van Baak <michiel at vanbaak.info> wrote: > On 22:46, Mon 15 Sep 08, hugolivude wrote: >> I have two Asterisk servers running on the same LAN. One starts fine, >> but when I start the other I get: >> >> pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: >> Address already in use >> >>

Hi all, I have a problem with using remote MySQL database server with two Asterisk (1.4.17) servers. PhoneA registers with Asterisk#1 using realtime into MySQL on remote server and everything is working fine and when I call Phone A from Phone B (also registered with Asterisk#1) call is established. Problem is when I call PhoneA (which is registered with Asterisk#1) from PhoneC (which is

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

I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets "clipped" to static if they speak too loudly. Many of these users have migrated to Skype ? much

Hi there asterisk goes to 90% cpu usage when trying to authenticate a sip friend using realtime mysql, no other message does appear at cli and asterisk hungs; here some info: *CLI> realtime load sipfriends name 104 Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:109 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipfriends WHERE name = '104' Jan 13 11:52:21 DEBUG[8928]:

In the following example exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider) exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)}) exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider) exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue) exten => _1NXXNXXXXXX,n(denied),NoOp(There are too many calls up) exten =>

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

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

I am currently in the process of testing out realtime support for sip.conf. I have followed all of the directions that are listed in the Wiki, but for some reason this does not work. When utilizing a flat file, I am able to register endpoints without any problems, and calls can proceed. One interesting side effect that I have noticed is that when I am using realtime for sip, I am unable to see

Quick question re. NAT traversal. I understand how sitting behind a NAT could cause problems for a SIP UA. The SIP UA would create SIP mesages using IP addresses from inside the network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses are of course unnavigable for the recipient. What I don't get is why don't web browsers suffer the same problem? A web brower behind a NAT sends an

Curious whether the ChanIsAvail command is thread safe. By that I mean, if I use ChanIsAvail to determine which channel to use, can I be sure that it will still be available when I go to Dial it on the next line? It occurs to me that there's a possibility the channel could get used by a competing thread AFTER my thread has determined it is available and BEFORE my thread gets a chance to

hello i m using asterisk-1.0.9. i want to connect to db through odbc. isql is working. but asterisk is not getting user information from this table. can any one pls check this /etc/asterisk/extconfig.conf [settings] sipusers => odbc,mysql1,sip_buddies sippeers => odbc,mysql1,sip_buddies sip.conf => odbc,mysql1,sip_buddies sipfriends => odbc,mysql1,sip_buddies

Asterisk 1.2.7 RedHat 9.0 Hi, I've run into some voice degradation problems with IAX2: I frequently have calls come in on a DiD provided by an ITSP. I often have to redirect these calls back out to the PSTN (i.e. to a cell phone). When this happens, I don't want my server in the media path, I want to hand it off to my ITSP instead and let them handle both ends of the call. I've

I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). Before I implemented Asterisk, some users were using Bell services to set-up 5 way calling: The user would set up a three way call on one line, switch to the second line, set up another 3 way call and then link the two lines together with the Flash key, thus establishing a 5 way call (the user, 2 others on line 1, another 2 on line 2). How