similar to: OpenH323 channel driver, Q931 Calling party number

Displaying 20 results from an estimated 90 matches similar to: "OpenH323 channel driver, Q931 Calling party number"

2009 May 29
1
Call telco transfer q931
Hello Please help me, i need transfer a call in asterisk to other telco number and free the channel. Can i do with any q931 function?. Thanks a lot Aris... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090528/2bcd93ae/attachment.htm
2004 Jan 13
1
E100P without q931?
Hi, does anyone know if its feasible to run asterisk with a PRI card but not run any q931 signalling.. basically push calls down the PRI and tell asterisk in some other way to pickup a particular Zap channel? Steve
2004 Jun 10
0
oh323 0.6.2 q931 messages
- I've just installed 0.6.2, & I would like to see the q931 messages going back & forth. I turned on debugging with "h323 debug toggle", which the README says is "very verbose", but I don't see much. Is there a way for me to see more debugging information, like the "debug isdn q931" of IOS? Or am I missing something? Thanks, Glen IAS
2005 Oct 12
1
send Q931 information element keypadfacility ?!
Hi all, I'm looking for a way with any asterisk-version with TE410P (cpe EuroISDN, Q931) for sending an INFORMATION ELEMENT KeypadFacility, eg. *87, during a connected call to the PSTN switch. Are there existing functions in asterisk to generate & send such IE ? If not what existing modules would be best to derive from? TIA, Bruno -------------- next part -------------- A non-text
2010 May 15
1
q931.c modifications for CLID Presentation
Hi Guys, We have a problem with Caller ID not being displayed. I want to test everything to see where the problem is with the incoming Caller ID and why it's not displaying. I am tracking this down to "Presentation prohibited of network provided number" even though the Caller doesn't use *67 and even though they haven't asked their provider to block their CLID for outbound.
2010 May 15
1
Re-compiling q931.c
Hi Guys, Can q931.c be re-compiled using gcc or something else without the need to re-do the whole libpri? Some changes were made to q931.c and I want those to be reflected in .a .o .so .lo files as I think those are the files read by Asterisk vs the .c file. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Aug 23
2
[Asterisk-Dev] q931 dial errors
Skipped content of type multipart/alternative-------------- next part -------------- _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
2011 Aug 11
0
Asterisk 1.8.5 - Ubuntu Pkg from diguim Repo - OPENH323 error
Hi Paul, Maybe you can give some help here: I'm trying to compile and build the debian source file of asterisk_1.8.5.0.orig.tar.gz and asterisk_1.8.5.0-1digium1~natty.debian.tar.gz. Howerver every time I'm trying to compile it, using ./configure of dpkg-buildpackage -rfakeroot -us -uc I get errors like this: checking for mandatory modules: CAP GSM OPENH323 IMAP_TK PWLIB... fail
2003 Jun 10
0
chan_h323 + openh323 CVS = no go? (fwd)
---------- Forwarded message ---------- Date: Wed, 11 Jun 2003 01:10:16 +0200 (CEST) From: Siggi Langauf <langausd@fachschaft.informatik.uni-stuttgart.de> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go? On Tue, 10 Jun 2003, Jeremy McNamara wrote: > If you would have followed the build instructions laid out by the Open > H.323 folks
2003 Aug 19
1
Speex & openh323
hi, I'm currently trying to use Speex with Asterisk from my OpenH.323 client. It seems to mismatch the codecs, below is my log from Asterisk. My Openh323 client crashes in responding to a Speex request for bits per frame. I'm guessing it either isn't running the codec correctly or doesn't support the same subset of speex codecs as openh323. (I'm using speex-1.0.1 with
2003 Sep 23
0
Cisco Callmanager 3.3 Asterisk OpenH323
Hi, i'm searching and trying, but can't get it working. I'm trying to send calls from Cisco Callmanager to Asterisk with oh323 channel driver. Therefor the asterisk is defined as a H323 Gateway in the Cisco Callmanager. The Call comes from CCM to Asterisk and it works but i didn't get the called number. This is needed because i want to make Voicemailboxes. If i connect via
2004 Sep 10
0
Re: Problem with Openh323 channel driver
Date: Fri, 10 Sep 2004 16:37:33 +0300 > From: Michael Manousos <manousos@inaccessnetworks.com> > Subject: Re: [Asterisk-Users] Problems with 0penh323 Channel Driver > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <4141AE1D.3020403@inaccessnetworks.com> > Content-Type: text/plain; charset=us-ascii;
2004 Sep 19
1
openh323 compile for Asterisk
HI, I have the latest RC2 of Asterisk on a RH 9 non-modified-load box. I have an Avaya IP phone that uses h323, so I am trying to compile h323 into Asterisk. Now, I downloaded pwlib and openh323 tar files and I have compiled this according to the instructions: pwlib: ./configure make opt openh323: ./configure make opt cd asterisk/channels/h323 make cd asterisk make clean make install I am
2005 Jan 28
0
two OpenH323 vulnerabilities
www.sans.org has two vulnerabilities in OpenH323, one as 'high', one as 'other'. Jason Sjobeck ICQ 5579183
2006 Jan 12
0
latest openh323...still compile error
ver 1.17.2 [root@71 openh323_v1_17_2]# make opt /usr/src/openh323_v1_17_2/openh323u.mak:192: usr/src/pwlib_v1_9_1/make/ptlib.mak: No such file or directory make: *** No rule to make target `usr/src/pwlib_v1_9_1/make/ptlib.mak'. Stop. [root@71 openh323_v1_17_2]# thx in advance _________________________________________________________________ Express yourself instantly with MSN Messenger!
2008 Dec 04
1
OT - Is sourceforge OpenH323 active ?
Hi, A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 Aug 06
0
[OpenH323]BUG NOT CORRECTED
Hi. I've found a large bug in Speex Codec implementation in Openh323 code. I've signalled this bug in the Openh323 site, in the section "Report bugs". THE BUG NUMBER IS 143! Everybody can verify that the report is arrived monday 7 July!!!!! But no one has analyze this bug....!!!! So the result is: Openh323 implements Speex Codec that runs with wrong bit rates and that
2005 Sep 09
0
woomera doesn't work (same OpenH323 problem as with chan_h323)
Banging my head against a brick wall trying to get a working H.323 implementation for CVS-HEAD. (The ONLY H.323 I have had working is OH323 v0.6.5 with CVS-STABLE - see my other post regarding compile problems on OH323 for HEAD) So, I thought, lets try this wonderful chan_woomera (dubbed "H.323 for Asterisk that works!"). I get exactly the same kind of problem as I have previously had
2008 Jul 17
1
OpenH323 and ptlib version for asterisk 1.4.21.1
Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser
2004 Aug 06
1
openh323 conflict; can speex_encode take short?
openh323 speexcodec.cxx has: BOOL SpeexCodec::EncodeFrame(BYTE * buffer, unsigned & length) { // convert PCM to float float floatData[SAMPLES_PER_FRAME]; PINDEX i; for (i = 0; i < SAMPLES_PER_FRAME; i++) floatData[i] = sampleBuffer[i]; // encode PCM data in sampleBuffer to buffer speex_bits_reset(bits); speex_encode(coder_state, floatData, bits);