similar to: Ser 0.9.0 adding a user?

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 =

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'm assuming asterisk does not have a SIP jitter buffer in place? Any ideas on how to help with this going over a data T1 where VoIP is shared with regular traffic? Problem is when people are downloading the voice is jittery, even lossy. Matt

I have several Asterisk servers installed and one SER server which will act as a gateway to PSTN, en redirect server. I was thinking to implement it the following way: - Register all the * servers at SER (is this neccessary?) -> this works via register=>asterisk:password@serbox in sip.conf - Setup aliases in SER for the telephonenumbers to the appropiate * server: serctl alias add

Hi, I'm looking for a tutorial or installation guide for SER to be used with asterisk to solve the remote SIP agent problem. All the documents available are for large scale installation. Any help is highly appreciated. Regards. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

Try using the dlls from windows 95 cd, I'm not talking about the dcom95 dlls, but the dlls win95 installs in c:windows/system. > > > puoti@inwind.it wrote: > > Did you use only win95 dlls? If you used dlls from any other version of > > win, htere could be problems with dcom 95. > I used dll's only from dcom95 which are listed at appdb.winehq.org > > Without

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

Hi Guys, My name is Marcias, and I am setting up for the first time an Asterisk PBX, I am learning as I go along. I have been able to download and install Asterisk, Libpri and I have been able to get Asterisk up and running. I have several questions: 1 .I can call the Asterisk server from my Xten phone and it picks up. I have 3 computers (one of them being the asterisk server) I can seem to call

2015-03-02 23:41 GMT+03:00 Brian Rak <brak@gameservers.com>: > In IRC, I was directed to this patch: > https://www.redhat.com/archives/libvir-list/2015-February/msg01212.html ... > which does exactly what I was looking for. It doesn't build cleanly in that > state, but it's pretty trivial fix (needs actualType added to the function > definition for

On Jul 27, 2007, at 7:04 PM, Gabriel McArthur wrote: > I offered to do the web site a couple of weeks ago. Life got in > the way temporarily, and it took me a long time to make it look the > way that I wanted and to make it look the same in IE6. Anyway, I > wanted to get some feedback on what I had done. Note to self/dumba**: include web site next time.

All, I'm having a heck of a time getting hdlc to work on kernel 2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the kernel (note into, and not 'modules'). System comes up, I configured zaptel.conf span=1,0,0,esf,b8zs nethdlc=1-24 modprobe wct4xxp ztcfg sethdlc hdlc0 cisco ifconfig hdlc0 up All of this works fine, believe it or not. I have a T1 cross plugged

Is it possible to enable automatic forwarding of Xprint server displays? The -X/-Y switches just forward the video Xserver display but it seems there is no way to get the Xprint server displays forwarded, too. Thanks... Felix Schulte -- _ Felix Schulte _|_|_ mailto:felix.schulte at gmail.com (0 0) ooO--(_)--Ooo

Kevin wrote: > There is a reboot script posted on the wiki to reboot Polycom > telephones. When I execute this script, I get the following messages. > I am concerned as this is causing issues with asterisk and the PRI. > Does anyone have any ideas why this would be happening? > > > > asterisk console: > > -- Remote UNIX connection > -- Remote UNIX

Dear R-people, I want to use an idea from LaTeX in the work flow with R. It is possible in LaTeX to have a main file from which other files are called (ie included). So for example if you have book, the main index file would call each chapter separately. Is there something comparable in R that follows the above 'include' idea from LaTeX? A main file that calls parts of an analysis or

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 :-)

Bueller? Is this a lib of some kind? Google and lists bring up nada, this is from ast cvs head latest on Fedora Core 3. /usr/bin/ld: cannot find -lidn collect2: ld returned 1 exit status make[1]: *** [app_curl.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 [root@zoot asterisk]# uname -a Linux zoot 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004

Hi All, I''ve downloaded a Puppet module and I''m trying to add a parameter to it by editing lib/puppet/type/<resource>.rb. I simply added: newproperty(:pcfree) do desc "My description here" end In the corresponding file in lib/puppet/provider, I do something with :pcfree. Whenever I call the resource from Puppet with my new parameter, it keeps

Hello, I have problem with interpretation of CDR entries. What happened? ------------- There was: 1. at 09:00:26 we received call from unknown caller (no callerid) to secretary with extension 17 2. secretary answered and after some conversation called to extension 18 to check if she could transfer customer call 3. at 09:02:55 extension 18 answered and call was forwarded 4. call was very long

I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one

I have a gateway, which I call from my mobile phone (free of charge, since it is the same phone company). This gateway gives me a dial tone. I can than dial to any extension number or even other gateways, .... It is getting more a trouble to remember all the numbers, or to key in all the long phone numbers when you got the dialtone. I was thinking of using for this Sphinx2. How can I