similar to: woomera doesn't work (same OpenH323 problem as with chan_h323)

Hi, when I receive a call from ISDN BRI (with a Sangoma A500) and I try to playback something I get the following error: **[WOOMERA]** HW DTMF supported s1c1- -- Executing [number at from-pstn:1] Answer("WOOMERA/g1/1-7b29", "") in new stack **[WOOMERA]** +++ANSWER WOOMERA/g1/1-7b29 -- Executing [number at from-pstn:2] Playback("WOOMERA/g1/1-7b29",

Hello, there has been a post to this list somewhere arount april which said that it is possible to use a Sangoma BRI A500 card with zaptel and asterisk bristuff. That is, without sangoma_brid and sangoma_mgd daemons and without woomera channels. Could anybody give me a short hint how to configure this? I tried wanpipe-driver + zaptel + asterisk-bristuffed, but I couldn't get zaptel to

Hi, folks! Sangoma has an informal user survey up on their home page (at http://www.sangoma.com) asking if people would use their A500 card for North American BRI, if it were supported. I encourage anyone with an interest in voice BRI in North America to vote; this information will be used for deciding whether to make the investment in developing a National ISDN driver layer for it. Benefits to

Through the generosity of our Premier sponsor, Sangoma Technologies we are proud to welcome Craig Southeren all the way from Australia. Mr. Southeren.s work has pioneered the development of open source telephony applications with his ground-breaking OpenH323 protocol stack that stood alone as the only open source VOIP software for quite some time. Today, Craig continues to raise the bar

I am trying oh323(version 0.67) , make call from sip UA to h323 gateway, can't get Call-id pass from sip UA to h323 gateway, h323 always gets call-ID sent from Asterisk as *. are there any configure to pass the correct call-id from sip UA to h323 gateway? or this is a bug in oh323 0.67? how about oh323 0.73 ? Mario On 9/29/05, Kanishka Somaratne <kani@technoportal.biz>

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

Hello! I've got a question regarding the Q.931 Setup-field Calling Party Number. It contains five things: Type of number, Number Plan, Presentation and Screening indicators and the actual number. Our provider uses some of those to decide if the numer should be presented or not to the outside world. I've done a crude hack in our GnuGK to always change those so that our numbers are

---------- 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

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

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

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;

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

www.sans.org has two vulnerabilities in OpenH323, one as 'high', one as 'other'. Jason Sjobeck ICQ 5579183

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!

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:

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

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

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);

Thanks Vamsi I have not been able to locate pwlib - 1.6.6 or openh323 1_13_5. I found the latest versions through sourceforge and I found some older versions on another site, but not these versions. This has been quite frustrating. Anyway, I think by using the asterisk-oh323 branch under channels in the asterisk source tree I will have more luck. At present it seems to compile successfully, but

Hi there I have Asterisk running beautifully on our test server. Over the past few days I have been tearing my hair out trying to compile various versions of asterisk-oh323 on various versions of pwlib and openh323. pwlib is now up to 1.8.3 and openh323 is now 1.15.2 stable. asterisk-oh323 is currently 0.7.1 I have tried these three with many errors. I have tried 0.7.1 with pwlib 1.5.2 and