similar to: No callerid on outgoing call over chan_h323

Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -vvvvcd). In debug log I can see the

On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> wrote: > Hello, > > After I have re-read the "PJSIP Advanced Codec negotiation" document, it > occurred to me that the desired behavior should actually happen > automatically, just due to the codec negotiation logic, but it looks like > asterisk doesn't actually follow the described

Oh, that's great. It wasn't clear from that page, at least not for me. :-( Having it clearly stated on the document would save me (and probably others) lots of time. Thanks for clarifying it. Any idea on the timeframe of implementation? *Michael Ulitskiy* Ace Innovative Networks, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-1203 https://www.aceinnovative.com On 7/6/23 12:47, Joshua

Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can "hear" me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external

Hello, I've started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do:

Hi, I'm wondering if there's a way within a dialplan or AGI to find out if an extension (SIP client) is already in use and the person is already on the phone? By default the channel is assumed available and callwaiting tone is transmitted to the called extension. AFAIK there's no way to turn off callwaiting from within the dialplan. I need to avoid the callwaiting behavior in some

Hi all, I try to ask helps again about chan_h323 extensions. I define this in h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 allow=gsm allow=ulaw gatekeeper = DISABLE context=default [gm1] type=friend host=192.168.1.20 context=default [gm2] type=friend host=192.168.1.25 context=default and I have in extensions.conf : [demo]

Hello, After I have re-read the "PJSIP Advanced Codec negotiation" document, it occurred to me that the desired behavior should actually happen automatically, just due to the codec negotiation logic, but it looks like asterisk doesn't actually follow the described logic which is likely a bug. Can you please follow with me through a simple sip call and see if I'm missing

hi .. I have finally managed to get Chan_H323 & G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib & oH323 with versions taken from nufone's site

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

I have installed Asterisk, pwlib and openh323. How do install chan_h323 and test to make sure it works. Any how to chan_h323 Linux distro: Redhat 9.0 Regards Patrick __________________________________ Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo. http://search.yahoo.com

Hi all, i have two *. One makes call to another over chan_H323. At the called side i have defined a user (friend) like --> [thaeger] type=friend host=172.20.23.100 context=incomingh323 At the calling side i have following entry in extensions.conf exten => _X.,1,Dial,H323/thaeger@172.20.23.10 Now i have two questions: 1. how will be transfered the callerid (normaly

Hi, I'm wondering if there are any plans on adding secondary gatekeeper support to asterisk h323 channel drivers. Also I've noticed that chan_h323 is crashing asterisk at startup if primary gatekeeper is not available. Wouldn't it be a more correct behavior if it doesn't crashing but continue registration attempts in the background? Didn't test it with chan_oh323. Thank you.

I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled, and chan_h323 module does not load with "undefined symbol _ZTI19H323AudioCapability". What could be the problem? Peter

Hi folks, I?m using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info won?t work with Asterisk?s voicemail system. I?m using the g.729 codec for h323 and Asterisk. I?m told dtmfmode=inband won?t work with g.729. Is it possible to use

Hello all, Encouraged by the successful "demo", I'am getting on with Asterisk CVS. I added 2 H.323 extensions in extensions.conf [default] include => demo exten => 701,1,Dial(H323/gm2@192.168.1.20/s) exten => 702,1,Dial(H323/gm2@192.168.1.25/s) With: - [demo] is defined by default in sample.extensions.conf - Asterisk server is running on host 192.168.1.20, on the

Hi all Asterisk Gurus. I am really badly in need of help. Asterisk is very lovely software, but has one big disadvantage.. lack of documentation.But let's get to the point. 1. Is it normal that I get such a crappy quality with iax, some drops and clicks? Could anyone with some similar setup check my quality and say if this is what the people are so excited about? ( I used to work as a speech

hi just after talking to Jeremy, I downloaded the source he's using and compiled. core dump! This was without gatekeeper: (gdb) bt #0 0x40141118 in mallopt () from /lib/libc.so.6 #1 0x401400fc in malloc () from /lib/libc.so.6 #2 0x405e981b in PAbstractArray::SetSize(int) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x405ed006 in PString::SetSize(int) () from

I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk. Has anyone gotten a Webswitch running? When I try to connect asterisk thinks everything works fine, while the webswitch just rings. I belive chan_h323 is picking the wrong port to talk at the webswitch on, however I'm not sure, nor am I sure how to fix it. Any clues/hints? A tcpdump is attached to show the session.

I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [asterisk@jonux h323]# make clean rm -f *.o *.so core.* [root@jonux h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN