similar to: How to call an extension number from ohphone to astersisk

I've been banging my head on this one for a few days and am quite stuck. I've got a gatekeeper running and everything works there. Netmeeting works calling other netmeeting clients. Netmeeting calling asterisk connects, but netmeeting can't generate the signals to make the demo do anything other than talk. But connection from ohphone always disconnects straight away. I can't seem

Hi, all Here is a something I found on the web: http://www.voipbuster.com And it works OK too. Now, I want to use it via asterisk, so I ccan use my normal phones instead of PC application. Did anyone try to connect astersisk and VoipBuster? Thanks, Rudolf

Hi all! I am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it

Hello I'm using H323 channel and client used ohPhone-1.4.1 (with gatekeeper). when at client side dial to asterisk server (dial 7777, test mode). ohPhone don't hear any thing sounds (no audio). i dial between ohphone (with gatekeeper). sounds are good. my current setting. Asterisk-1.1.x, GNUGK 2.2, PWLIB-1.8.3, OpenH323-1.15.2, ohPhone (for windows) 1.4.1 Please help me. Thanks

Hai Friends, I was using wine happily with ubuntu 8.04. Now I have ubuntu 9.04 clean install on a new machine. I could install wine but when I open wine-program-accessories-notepad nothing happens. Also configure wine does n't funtion. Also when I try to install any exe file this message comes: here is one example [/home/illusions/Desktop/TeamViewer_Setup.exe] End-of-central-directory

I'v got the following scenario: Two clients (ohphone) are calling (one at a time) the host with asterisk, which then connects to the SIP client. One of these hosts let's asterisk crash with a segmentation fault (i can provide the core file, if needed) in the second, the SIP client accepts the call. However .. if that client get's to the voicemail instead, because the SIP client is

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

Hello all, This email is intended rather informative than questioning. While developing some script-generated dial plan, we figured out that there are differences between Snom 190's and BudgeTone 101's relating to transfers. It appeared that the 190's will have their own 'Caller ID' set as the 'CALLERID' variable in astersisk when transfering a call, while the

Dear All, I' I have a problem in installing asterisk 1.4.0. how can i compile res_config_mysql.c in astersisk-addons dir. I've downloaded asterisk-addons-1.4.0 compiling and installing it. But i can't found shared object of res_config_mysql.so. My system is : Debian Linux 3.1 Kernel 2.6.8-11 asterisk-1.4.0 zaptel-1.4.0 asterisk-addons-1.4.0 libmysqlclient using apt-get webserver :

Joost Witteveen (joost@iliana.nl) wrote: > > So, each UDP package with 20 bytes speex-data, we send: > > 20 bytes speex > 12 bytes ogg headers (and others?) > 28 bytes UDP/IP headers (2 IP numerbers, 2 portnumbers, checksum, etc, etc) > > and, if it goes over the phone, each package has a few ppp headers. > > Am I overlooking something, or does this fixed frame

I've a problem with my X100P card. I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I call an PSTN phone number, some digits are duplicated, so I'm unable to call the right person. Not very clear ? I'll try to do better (sorry, I'm french...) example : I use ohphone (with quicknet hardware), I call asterisk (*192*168*1*204#), asterisk answers, I choose

Hi! When I try to make a call with ohphone, that is the message I get: Call to "06302" aborted, insufficient bandwidth Can anybody tell me a solution or a reason why this messages appears? Thanks a lot! Regards, Mireia

Hi Duncan, >>are you trying to get the name "@_ZTIi" or "@__cxa_throw"? yes! i want get the name @_ZTi or @__cxa_throw, the latter @__cxa_throw i can get it throw value->getName(), but the @_ZTi it did n't has name! zhangzw thanks 2009/3/26 Duncan Sands <baldrick at free.fr>: > Hi zhangzw, > >> invoke void @__cxa_throw(i8* %7, i8*

hi I am struck with a problem in igraph package of R. My problem is as follows I want to plot a power law fit for my data (in .net format --- pajek format) syntax for that in R is g <- read.graph("filename.net", "pajek") d <- degree (g, mode="in") power.law.fit (d+1, 2) it gives me desired out put if my if input a single file but I want to use a variable

Reading archives I did found letter related to my question: Stephen C. Tweedie: > > On Fri, Jul 20, 2001 at 08:27:44PM +0300, Nikolaos Kefalas wrote: >> I want only to ask , if it is necessary to run tun2fs -J the kernel to have >> support for ext3 . >> I want to boot , from another linux to convert my partions since the partition >> must not be >> mounted

Hi zhangzw, >> invoke void @__cxa_throw(i8* %7, i8* bitcast >> (%struct.__fundamental_type_info_pseudo* @_ZTIi to i8*), void (i8*)* >> null) >> noreturn to label %invcont unwind label %lpad > > >>are you trying to get the name "@_ZTIi" or "@__cxa_throw"? > > yes! i want get the name @_ZTi or @__cxa_throw, > the latter

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

Hi, I'm using asterisk with chan_h323 together with gnugk. chan_h323 and gnugk were recently compiled with pwlib-1.5.2 and openh323-1.12.2 as advised. When connecting asterisk directly by ohphone (without gatekeeper), everthing is fine. When using gnugk for usage control in routed mode, I find a funny situation in asterisk's H.323 debug: == New H.323 Connection created. --

Hi to all, I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper. I've tested h323 using ohphone and I can talk between them, then I've tested SIP with Twinkle softphones and function very well. Now I have to perform call from h323 to sip and viceversa. How can I do it ???? I receive h323 call from a Cisco Voice GW to my Asterisk and this call have to go to a SIP phone:

> Framesize always refers to the decoded data frame size in samples. > Framesize is dependent on the encoding mode > Narrowband (8kHz): framesize = 160 samples = 320 bytes of PCM > The size of the encoded data depends on the quality setting, so if you > know for instance that you are using quality 3 on narrowband, that is > 119 bits of encoded data per frame which is rounded to