similar to: FW: OH323 with Asterisk@home - seems incomplete

I installed oh323 and everything seemed to go smoothly (compile & everything upto calling through using oh323). I must admit, there is some behavior that's doesn't seem right but generally, I'm able to dial-out of any oh323 device whether to an extension or to a trunk. Audio is sometimes muted when dialing out until the extension or dialed number answers. Sound quality is good

I'm and Asterisk@home user - been so now for almost a year. Lately, I've upgraded to the latest & greatest.. (which is built on 1.2.5) and am unable to install oh323. I've already asked over at the (A@H) Sourceforge forum but no one seems to think it worth answering. The error I get is pretty obvious but I don't know where to go from here. More importantly, I need to have

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Ahhh this could be my problem! I just checked which wires on the RJ11 cable had a voltage across them and it was the yellow and green (3 & 4?). From what someone posted the other day it's supposed to be Bumble Bee and Christmas Tree. I did have to get a technician out to fix my line when it was first installed because it was dead. Maybe he wired it up incorrectly or maybe they just do it

I know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an <mailto:Asterisk@home> Asterisk@home installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack

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Same thing here. I had this problem awhile ago and made this workaround. Going to another trunk does not work because they are answering and not sending a error code. If you are using AAH code then this waits 10 seconds on your Voip then times out and goes to PSTN. You can modify for your needs The pertinent line is 14 in macro-dialout-trunk I am going to clean it up and repost under my

I want to say thanks to everyone for the help so far. I figured out a way to modify some AAH code that worked for me (well sort of). The line I modified is s,14 in macro-dialout-trunk. Then I just added a variable and passed it from 9_outside. I just have one last problem. This waits for an answer not ringing. So if the called party has a long ring to voice mail the call is dropped and goes

Hello, I've got an Asterisk system with 3 SIP trunks configured. Each SIP trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound call routing (via AMP 1.10.007a) uses the 3 trunks in descending order, all set with max channels to 4. Unfortunately, when the first trunk reports a "480 Service Unavailable" (all ports in use), Asterisk reports congestion without

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi there, I am looking to install R on Mandriva-linux. But on the UK mirror site I found only Debian, Redhat, Suse, and Ubuntu, but not Mandriva. Any help would be greatly appreciated Thanks -- ------------------------------------ Abdel Hannachi Reading, UK

Hi, I have an XML file which contains among other nodes : ===myXMLfile.xml=== (?) <nbRelations>2</nbRelations> <nbActors>2</nbActors> (...) <nbRuns>5</nbRuns> <nbStep>2000</nbStep> (?) ===End file=== I need to extract those values and to make them R variables such as: nbRelations = 2 nbActors = 2 nbRuns = 5 nbSteps = 2000 I read the help and have

According to the wiki at voip-info.org, the dial structure for using oh323 without a gatekeeper is: OH323/<exten>@<host>:<port> or OH323/<exten> The second option is valid only in the case where a gatekeeper is used. NOTE: OpenH323 library v1.12.0 has a bug in the parsing of the destination host. When this version is used then the above syntax should be:

On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi <shinacalypse at gmail.com>wrote: > Hi! > I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide > a conference bridge for an existing Avaya PBX. I have no control over the > Avaya system, but I am able to speak with the admin in charge when I need > stuff done. I am running all this in a VirtualBox

> From: Gamal Abdel-Azim [mailto:gamal at crinet.com] > > While trying to expand the memory/object size in R, I noticed > that R might be using > only heap memory. Is this true? Are all objects in R created > in the heap not > allocated? It's not logical that this is the case!! Otherwise > the whole R project would > be a total waste of time and resources.

Hi, erverybody The Asterisk is running well in the linux system. Now I would like to add oh323 in Asterisk. I have download pwlib(version is 1.6.6) and openh323(version is 1.13.5). And I sucessfully maked and installed these two packages. But I got the following errors when compling the asterisk-oh323 0.6.3a: for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]:

Hello everybody, I have a strange comportment with oh323 and asterisk, I'start testing asterisk but with this I can't understant plesae help me ! Thanks Eltorio ---------------------------------------------------------- 1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a Modem[i4l] line ---------------------------------------------------------- Nothing happens

Michael, Thanks a bunch, I downloaded from inaccessnetworks.com thinking that it is the latest :). Ok I will upgrade it. just for the record, following worked. exten => _87.,1,Dial(OH323/H323:${EXTEN:1}@16.52.153.206) Cheers Sathya Date: Fri, 28 Nov 2003 11:28:59 +0200 From: Michael Manousos <manousos@inaccessnetworks.com> Organization: inAccess Networks To:

Hi guys, I've been trying to update my chan_oh323 from 6.1 to 6.3b. I built asterisk from cvs-head on the date Micheal said he made it compatible, pwlib-1.6.6 and openh323-1.13.5 (both with nothing more than the ./configure, make, well aplied patch on openh323) When I start * with my normal config I get this: [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing