similar to: help: H323 and SIP

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

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.

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

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

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

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

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

I am attempting a fresh install of ASTCC on Ubuntu. Getting install invalid user as bellow. Has any one seen this? Can some one help out? /usr/src/astcc# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Detected dry run! ./astcc-admin.cgi >/dev/null DBI

> http://www.speakfreely.org/ > > I think this would be one of the best real-world tests of the speex codec. > This software doesnt use ACM or directsound api's but uses straight C code. > I was thinking the speexenc/speexdec should be easy enough to add. The last time I looked at this it was still very much old news - mostly half duplex audio, does not adhere to any

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https://bugzilla.samba.org/show_bug.cgi?id=12920 Bug ID: 12920 Summary: Invalid path from sender Product: rsync Version: 3.1.2 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: core Assignee: wayned at samba.org Reporter: crazyjim_68 at

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 everybody. Is there a way to set up liveice so it reconnects to Icecast automatically when the connection is lost?. I tried to automate this process by using a crontab and a perl script, but when liveice is run with the -@ 2 option It seams It doesn't realize that the connection has been broken and continues "running". If I do ps ax, I can see liveice is "running",

> 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

Hi all, After having tested the SIP part (successful :-)) we are now testing the H323 part of Asterisk. The H323 channel is up and running (using NuFone Network's Open H.323 Channel Driver)) However when dialing to * using ohphone the call can not be set up / established /H323 debug enabled *CLI> == New H.323 Connection created. -- Received SETUP message...

Hello All, I have an off topic question about rsync and was wondering if i can get some kind person help with it. I have two servers with each server have three same directories on them /dir1/ /dir2/ /dir3/ . How would i achieve this by using rsync? I have tried rsync -avrt --delete server_ip:/dir1/ /dir2/ /dir3/ /dir1/ /dir2/ /dir3/ this does not do anything except give errors. Someone on IRC