similar to: chan ooh323 choppy sound

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ?

Hi all. Trying to setup H.323 via Asterisk between a PLANET H.323 box and my SIP phones. When calling from the SIP phones, it connects but quickly disconnects citing the following error message: **** --- build_peer +++ build_peer +++ reload_config +++ ooh323_do_reload -- Executing Dial("SIP/yyy-2965", "OOH323/203@xxx") in new stack --- ooh323_request - data

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip-> asterisk server -sip-> quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip->

Dears; I am having same problem, that when I place a call from the H323 end point (even if it is not added in the ooh323.conf), then asterisk handle the call and play the wave file in the default context. Also I added endpoint to the ooh323.conf and same thing, it keep goes for default context whatever the context placed. My Asterisk vesion is 1.4.25 My Asterisk add-on version is: 1.4.8 What I

Hi, I recently changed my version of asterisk to 11.XX, and I see a waning with h323, with version 1.8 did not have these warning I have connected one avaya ip office 500 h323 with asterisk and the 1.8 version did not have these messages Oct 23 17:20:35] WARNING[7593][C-000000aa]: chan_ooh323.c:1413 ooh323_indicate: Don't know how to indicate condition 33 on ooh323c_60 [Oct 23 17:20:35]

Hi all, I just configured a quescom 400 to route all gsm incoming calls to asterisk, now i would route all outgoing asterisk calls to gsm port of the quescom. Anyone has any idea how implement it? I did a configuration but i always get this error -- Got SIP response 503 "Service Unavailable" back from <ip_add_quescom400> Thanks in advance. Giordano

Hi, I am using asterisk 1.4. I have confgured the musiconhold.conf file. However, when i make a call and then hold the call it does nothing. in the CLI i do not see the starting/stopping musiconhold messages. i am making calls from sip to h323 using asterisk assip/h323 gateway (with gnugk and ooh323). i get the following messages when putting the call on hold: -- Executing [204 at default:1]

I've this setup : CiscoAta186 -> asterisk with oh323 chan -> gsmgateway dtmf doesn't work, tryed inband, with g711a and g729 codecs CiscoAta186 -> gsmgateway works, even with g729, so it seems the problem is in * oh323.conf has inBandDTMF=yes, what else may I need to tweak ?

I've got Asterisk 1.6 bridging to an Avaya using H323. The Avaya is autoanswering calls to music (as expected) and audio seems fine, but I see this error on bridging: WARNING[8833]: chan_ooh323.c:1054 ooh323_indicate: Don't know how to indicate condition 20 on ooh323c_o_2 Is this a warning I should be concerned about? What does condition 20 mean? Thanks! Michelle -------------- next

Hi there, I'd like to know if some of you knew Quescom products (http://www.quescom.com) and if it is possible to make run Asterisk with a Quescom400. Thanks. -- Asega Adrole

I've been asked to purchase a gsm gateway for use with our asterisk server (for our use, not reselling) I have a spare ISDN port on the server, so I have use either a PRI or VOIP gsm gateway. What would people recommend ? Has anyone used the QuesCom 400 ? I would also love to know a rough idea of cost ;) Once I've gotten the info, I'll post a message on the biz list for a

Hi all, My scenario is this one: LandLine------------------E1---------------|-------------| |-------------------| |OLDPBX|-------E1-----------|Asterisk1.2.5|-----VoIPusers GSMGateway---------Analogue------ |-------------| |-------------------| What is happening: 1- SipUserAgent "A" Dials a call to a Local Extension "B" in the OldPbx 2- "B" , the called party

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

I'm considering switching my incoming phones lines from standard analog to a T-1 service from XO communications. They propose to bring in an "IAD" which has 12 lines of voice and 768k of internet bandwidth as part of a package deal. Since I want to keep the voice traffic in the digital domain the equipment they're proposing is a "Lucent Digital Vina Integrator" IAD

Hi, For MGCP users. Is there any success stories with any MGCP IAD vendor. I?m trying to find an IAD which works with Asterisk. I?ve tried the Cisco IAD 2430 without success; but SIP on this IAD works but it?s limited (no authentication, no notify messages, etc) and with higher density IAD (16 or more ports) it?s nice to control using MGCP. Any information will be apreciated ! Thanks. --

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come

let's suppose I have this dialplan : exten => _X.,1,Playtones(ring) exten => _X.,2,Dial(CAPI/contr1/${EXTEN},,g) exten => _X.,3,AGI(update) where "update" updates some db tables we have based on the type of extension Now, from the wiki : If the /g/ option is specified, and the called party hangs up before the calling party, then Dial exits with a return code of 0 to

I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other