similar to: H.323 call problemm (no sound)

* compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin H323 debug enabled --

My configuration is: [ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] -- [CISCO ip phone 12SP+/Skinny] When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN everything working ok (RTP is ok). But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone IP phone party can hear ISDN party, but ISDN (incoming) party canNOT hear IP phone party

Try sending 5350 config and oh323.conf, versions, etc... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj. Sent: Wednesday, December 22, 2004 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party

Hi, I'm using D-Link DG-104SH (H323 4 port FXS gateway) with analig phone connected to it and X-Lite softphone as endpoints with * When I calling from X-Lite to analog phone it's ok When I dilaling X-Lite from analog phone, X-Lite si ringind but whei I picked up X-Lite connection drops IP of DG-104SH is 192.168.1.3, H323 ID is GW1 X-Lite number is 233 Here is * output: -- 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

Hello, I'm running Asterisk 1.0.1 (the same was with 0.9, 1.0). When it receives inbound H.323 call it makes connection and uses local 127.0.0.1 address to send audio stream: remoteIpAddress: 127.0.0.1 When making outbound calls from Asterisk it makes correct connection to send audio stream. Is it a bug in h.323? Is there some more settings to make in .conf files? See detailed debug below:

Hi all, I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2). So far I have been using the H.323 channel included in the tarball (Nufone ?). I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box : =====> here is the H.323 configuration for the incoming calls (192.168.1.50 is the IP of the

Hello... I'm having problems with H323/G729 setup. Below is the output of h.323 debug when making a call. I use a SIP phone connected to an * box in the same LAN. The * connects to a h323/g729 PSTN terminator through internet. Calls rings and are answered in the other side, but I get no sound at all nor the other side does (complete silence in both sides). I thought this would just happen

Hello... I'm having problems with H323/G729 setup. Below is the output of h.323 debug when making a call. I use a SIP phone connected to an * box in the same LAN. The * connects to a h323/g729 PSTN terminator through internet. Calls rings and are answered in the other side, but I get no sound at all nor the other side does (complete silence in both sides). I thought this would just happen

Hi all: I've got some core dumps when I use chan_h323. I dial an extension using h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes * hangs, sometimes not. The client used for test es SjPhone (http://www.sjlabs.com/). This is the data for one core dump: (gdb) bt #0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790 #1 0x41f8879c in create_connection

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Hi, I have Slackware 8 Linux Box with Samba-2.2.5 and HP LJ 1200 printer shared by samba (with LPRng). The problemm is: when printing from W2K clients users cannot change print options (like portrait/landscape page orientation, number of copies etc). When printing from Win98 clients all is ok. Could someone help vt with this problemm? -- Sincerely, Elman Efendiyev elman@megacom.com.ua

Hi, I need to send and receive faxes over VoIP in realtime. I mean: user ? calls from VoIP network to fax machine on PSTN, but starts voice conversation with user B on that fax machine. Then users agree to send a fax (any direction), pressed "start", completed fax transmission and then continue a voice conversation. This is one of generic ways to use analog fax machine. As I understand

Hi All, I need to replace old analog PBX with Asteriskl and X-Lise SIP SoftPhones as client phones. First: I have problems with implementation of PBX functions. I need and unsuccesfully tried theese functions (took info at http://voip-info.org/wiki-Asterisk+PBX+functions) Call Pickup: Supported in the standard installation (*8 - defined in res_parking.c +54) - Just don't understand how to

Hi I have successfully deployed winbindd in redhatlinux 7.1 os. It worked fine for a couple of days and boom the server locked up. i restarted it and winbindd crashed (i think). i restarted winbindd again and it's never been the same again. My users connect to the drive now but at a very slow speed. Also when I try to move large file form snap server to the linux server i get this error

Hello, As I understand * don't supports T.38 in Zap channels (please correct me if I'm wrong, BTW is there plans for such support?) I believe it's should support T.38 in "pass-thru" mode. I mean setup like this: Hardware gate with T.38 ------ Asterisk ------ Hardware gate with T.38 But I had troubles with this setup (no faxing) while two gates conneted directly with same

I've got a problem connecting to Cisco call manager. I dial a numder and hear only ringing h.323 debug shows this Allowed Codecs: Table: G.711-ALaw-64k{sw} <1> Set: 0: 0: G.711-ALaw-64k{sw} <1> -- Making call to 3200@10.1.105.3. == New H.323 Connection created. -- 6129 is calling host 3200@10.1.105.3 -- Call token is

Hi all, I am using chan_h323 from Asterisk CVS to interconnect with GNUGK v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on Asterisk. However, I only heard ringing when the call was answered on SIP side. Below is the debug from chan_h323. Any help is welcome. Thanks. *CLI> == New H.323 Connection created. -- Setting up Call -- Call token:

Could you help me with this problem? When I call H323 gateway there is no sound in both ways. Here is h323 debug: ----- begin ------------------------ -- Executing Dial("SIP/msn-6297", "H323/73952389512@peer:1720") in new stack Allowed Codecs: Table: G.729A{sw} <1> G.729{sw} <2> G.711-uLaw-64k <3> G.711-ALaw-64k <4>