search for: ohphone

...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 to enable any kind tracing on the asterisk end: Asterisk Server: openh323 v1.12.2 pwlib v1.5.2 asterisk v1.0 asterisk-oh323 v0.5.10 inAccess Networks OpenH323 Wrapper OhPhone: (attempt one) openh323 v1.12.2 pwlib v1.5.2 ohphone v1.4...

Hi friends Can you please say me "How to send an extension number from ohphone to astersisk". For eg I have an extension 5454 at the asterisk. How can I make a call to that extension from ohphone. I tried with the command ohphone 5454@IPAddressOfAsterisk. But I could n't call that number. I want to do it without using any gatekeeper. Can you please suggest me...

...elephone 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 unless I can use the onbord G723.1 codecs on the quicknet cards. Ohphone allows this through h323 I think but I want an asterisk solution. If not a fullblown asterisk install on my brothers machine, maybe set it up as a h323 client to mine. I am currently working on setting up one of my lan machines with ohphone to connect to my asterisk box to call FWD and such. Is...

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...

...P 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 size mean that > the "8kbps" in reality means at least 24kbps? > > Or is there something ohphone/gnomemeeting/etc can/should do to > put more than one fram inside each UDP package? Er, why would you be sending ogg headers in a VoIP scenario? That's needless overhead. Check out the Speex RTP format document at http://www.speex.org/drafts/draft-herlein-speex-rtp-profile-00.txt. Figu...

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 wind...

...39;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 "9" (to do an extern call, see my extensions.conf below), so I've a dial tone. Now I call "0684357917" with my OH323 client but asterisk calls something like "06884335779117" I...

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

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 voic...

...s not adhere to any signalling standards at all, and not being currently maintained. LinPhone (www.linphone.org) using the OpenSIP stack has used Speex from day one, and it's in wide use. Very real world. Speex is *already* supported in the OpenH323 stack (www.openh323.org) and in use in the ohphone, openphone, Gnomemeeting, etc. VoIP implementations that use that library stack. Very real world. :) Greg <p><p><p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message...

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

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 name: [500] -- Ca...

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. -- Received SETUP message -- Setting up Call -- Call token: [ip$213.30.206.5:32957/4316]...

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: - PSTN ==> Cisco...

...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. <p>Am I overlooking something, or does this fixed frame size mean that the "8kbps" in reality means at least 24kbps? Or is there something ohphone/gnomemeeting/etc can/should do to put more than one fram inside each UDP package? Thanks, joostje --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' contai...

Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.

...ice 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 unless I can use the onbord G723.1 codecs on the quicknet >>cards. >>Ohphone allows this through h323 I think but I want an asterisk >>solution. If not a fullblown asterisk install on my brothers machine, >>maybe set it up as a h323 client to mine. >> >>I am currently working on setting up one of my lan machines with >>ohphone to connect to...

...y.com/products/tkphone/). I don't know whether it's compilant with the IETF Speex draft. Also with the fixed-point port that's progressing, we may be closer to a hardware phone. > 3) Any suggested SIP soft-phones that support Speex? There's OpenH323 and all the derived phones (ohphone, gnomemeeting, ...). There's also a SIP phone called Linphone (http://www.linphone.org) which is at least compliant with the first draft of the RTP profile (don't know at the second one). I'd guess there are probably others. If you're working with SIP, I suggest you test for interop...

...risk Slackware 9.0 IAX protocol GSM libraries I have everything installed on my Slackware box (actually I have two of them and I want to set them up so that they can call each other) with an analog phone connected to the PhoneJack card. How exactly do I go about making a call? I have downloaded OhPhone, as I would like to not have to rely on a graphical interface, but I can't figure out how to use it. Any help will be greatly appreciated. Thank you. _________________________________________________________________ Tired of spam? Get advanced junk mail protection with MSN 8. http://join....

hello, I would like to realize a VoIP Gateway, with some extra-features. The aim is to get the phone number of the caller, to make research in our database, and to put him automatically through the good employee. The company is equipped with a VoIP network : software : - PSTNgw - Ohphone - OpenGatekeeper hardware : - Quicknet phonejack - gateway : Voicetroniw OpenLine4 Is it possible to do it with Asterisk ? I don't know if Voicetronix cards are well supported, but it's ok to buy a new one if needed. Thanks, Fabrice Tereszkiewicz