similar to: MWI for voicemail - H323

I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. Firstly, despite the warnings in h323/README, I decided to try using the distro-specific versions of openh323 and pwlib. Of course, the Makefiles in channels and channels/h323 assume that openh323 and pwlib have been specially compiled in $HOME, so I modified the Makefiles to look for headers and libraries in

Like some other people on here, I am trying to integrate Asterisk for VM with CCM version 3.x. I've got gnugk and Asterisk running, I've got CCM registering with the GK, I've got the voicemail pilot and profiles setup. A call comes into a CCM phone, it rings, rolls to the correct VM on ASterisk and asterisk emails the voicemail and I can check the voicemail, but I cannot get MWI

Hi , I am new to Asterisk community. I have some queries. Please guide me on the following : 1)I want to configure H.323 softphones, How do I do that ? I am using the Asterisk windows versio 0.60.There is no chan_h.323.so file .Also there are no help files or documents for configuring h.323 softphones. Can someone guide me inthis regard ? 2)And I want to know how the supplementary

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the 480i

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the

Hi all, I have this installation: Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0. I have a problem that is, when a call comes from H323 and goes to a Sip phone the asterisk sends two rtp streams to the sip. I checked this with tcpdump, save the payload (voice is in G711u), one is the ringing indication and the other is the voice coming from the user in h323 side. And

Hello all, I just installed Asterisk with H323 support (chan_h323 from Jeremy McNamara). But experience problem while connecting OpenPhone to Asterisk Here is h.323 trace: 5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720, handle=27 5:37.444 H225 Answer:9cc1250 transports.cxx(564) H225

Hi all, I'm using following software and equipment and I have very strange behavior: Asterisk CVS-NHEAD-05/30/05-16:42:41 H323 gatekeeper - GnuGK 2.2.2 IAX2 client - Firefly 1.9.8 build 3945 H323 client - SJPhone Build 1.50.271d H323 gateway - Welltech Wellgate 3504A When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected. When I dial from SJPhone (H323) ->

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

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the

Here's a link to it: http://forums.digium.com/viewtopic.php?t=4363&highlight= Regards, Scott -----Original Message----- From: Scott Keagy Sent: Monday, December 04, 2006 5:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. A while back I posted a fully functional though somewhat elaborate

Dear List, I'm having a blinking MWI light on the snom 320 even when there's no message waiting in Asterisk. We've managed to make the voicemail button work using fromdomain=192.168.0.1 in sip.conf vmexten=2500 (our VoicemailMain application extension in extensions.conf). We also added notifymimetype=application/simple-message-summary also in sip.conf to allow SIP simple MWI

I am using Nu Fone Network's h323 drivers. I can place H323 calls using following in extensions.conf file, exten => _1732.,1,Dial(H323/${EXTEN}@192.168.1.2) If I need to use h323.conf to do the same I cannot configure h323 to do the same. I get everyone is busy message and I do not see IP packets being generated by * trying to communicate to 192.168.1.2. Can someone point out what I

Hi, I have about one year of experience with Asterisk, working with ZAP (digium, junghanns) ZAPHFC, SIP and IAX. These technologies are quite clear to me, the problem is that I have no experience with H323, but now, I need to use this also. The problem that I have is very trivial, so I think that this should be a very easy question for you guys whom know how it works. All I want to do,

Hi - We've got multiple offices with their own asterisk boxes (CVS HEAD 11/03/04-14:59:37) connecting to each other using IAX forwards. All users are on SIP phones. Voicemail is centralized to one location. Everything is hunky dory except that the users in the remote offices don't get MWI on their phones. I've seen the other posts to this list regarding this, and

Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the

Dear All. I have installed Asterisk-1.0.6 on FreeBSD-5.4 RELEASE via port, so the chan_h323.so modules already included. When i try the SIP channel, asterisk works fine. In this case, i use the XLite softphone as client, i can hear the voice transfered through clearly. Asterisk is good!! :D The problem is when i try the H323 channel, the voice cannot be transferred through. I have

Hello all! I have a problem with ringing indication when calling from h323 (oh323+open phone client) to sip users. The phone rings and users can talk to each other with no problems but the calling h323 user hear silence unless sip user picks up the phone. Calling to pstn no problems. Calling from sip to that open phone client also no problems. I tried latest ooh323 and oh323... no difference

Hi, I didn't get any response to this posting so thought I'd post again in the hope that anyone who missed the posting the first time may be able to offer some ideas. This problem isn't specific to the particular model of phone - I can see from sip debug that the local extension itself is not being sent any Messages-Waiting headers. Any alternative strategy would also be welcomed

I try to get H323 to run, but have so far only partial success: There is a Gatekeeper GK, where asterisk connects to. The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper. From the Network on the GK, asterisk is reachable via the number 070333333. I have an extension on asterisk 6002, which is reachable. I try to call a number attached to the gatekeeper (070168177) with the