similar to: MGCP problem

* 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 --

Hello everybody, I?ve been pulling my hair for a week now over a problem, and I really don?t know where to look anymore. Here?s my setup: There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I can use it to send and receive calls from physical phones attached to it. I have setup Asterisk 1.0.3, with H323 and openH323, and on the same server I also setup GnuGK (10.253.30.1). I

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

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 there, What are you using for text search engine in your rails applications ? I have been using simple search for simple applications, but then I need to search text in model instances inside collections such as user.documents and the like. I believe that simple search does not support any sort of ranged queries, and sql like might be overkill for tables with lots of records. Bests, Sergi

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

Hello, I''m trying to get a freebsd virtual machine running following : http://www.yuanjue.net/xen/howto.html But I get : # xm create freebsd_xen_INSTALL -c Using config file "./freebsd_xen_INSTALL". Error: (14, ''Bad address'') Am I missing something? I am using xen 3.0.4. Thanks and regards, Sergi .... log output from xend.log: [2007-02-27 12:46:59

Hello, I have tried to create a package, and I have got it. I checked the files and I built the package. Nevertheless, I want a .pdf file with the package's documentation. Anyone know what I have to do? Thanks in advance, Sergi Martínez [[alternative HTML version deleted]]

Hi guys, I'm working on a package, and I want to create a new version file pdf. On R 2.6.2 it ran ok with the code: R CMD Rd2dvi.sh --pdf pkg. But it doesn't run on R 2.8.0. What I'm doing wrong? These are my components: ActivePerl-5.8.8.822-MSWin32-x86-280952 basic-miktex-2.7.2960 htmlhelp MinGW-3.2.0-rc-3 Rtools28 Thanks in advance, Sergi M.Garrido

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context

Hello there, I have a windows box (spanish locale, charset cp1252) which is backup to a linux server via rsync. Until now I've had problems with file names containing non us-ascii characters. Since the new stable version of rsync with support for iconv I've upgraded rsync on my linux (Debian) to 3.0.0 and also on my windows (cygwin, compiled from source). It works quite right, getting

Hi, My configuration is SipPhone<-->*1<--->*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best

I am getting the following compile error on h323. Its an old redhat 7.3 system with asterisk 1.2.14, zaptel 1.2.12 pwlib 1.5.2 and openh323 1.12.2 I have pwlib compiled and installed. I have openh323 compiled and installed. I went in the channels/h323 directory and did "make opt" What shall I do? Jerry ---------------------------------------- ../../include/asterisk/utils.h: In

hi all im trying to compile h323 i have got the pwlib and openh323 working that is simph323 is running properly but when i try to compile h323 in the channels directory it gives me the following error can anybody please help me with [root@test src]# cd /usr/src/asterisk/channels/h323/ [root@test h323]# make opt g++ -DNDEBUG -I../../include -Wmissing-prototypes -fPIC -DP_LINUX=2.6.5-1.358

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:

Hi there, I know this is the classic RTFM list question but... I've really tried hard on this and no result! This is what I'm receving from logcheck: System Events =-=-=-=-=-=-= Apr 3 06:55:13 bsg sshd[32246]: pam_unix(sshd:auth): authentication failure; logname= uid=0 euid=0 tty=ssh ruser= rhost=123.233.245.226 user=root Apr 3 06:55:19 bsg sshd[32248]: pam_unix(sshd:auth):

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

Okay, it's 15 minutes late, but it's out, thanks very much to all the people who worked so hard this weekend to make this possible! Mark p.s. there was no 0.6.0 release.

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement and