similar to: h323 channel compile error

Heya, I changed the Makefile of the h323-channel-makefile (I downloaded cvs of a couple of hours ago) so that I don't have to rebuild pwlib and openh323, but can use the precompiled versions. I'm using pwlib 1.8.3 and openh323 1.15.2. There aren't many changes. I replaced OPENH323DIR with OPENH323INC ,which points to /usr/include/openh323 for me and OPENH323LIB, which points to

i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/

Hi, I am trying to install Gatekeeper. I have installed pwlib and trying to install h323 plus. I have set the path as PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/open323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH I also configured /etc/ld.so.conf include /root/pwlib/lib include /root/h323plus/lib Then I tried running ldconfig. But it

Hi, erverybody The Asterisk is running well in the linux system. Now I would like to add oh323 in Asterisk. I have download pwlib(version is 1.6.6) and openh323(version is 1.13.5). And I sucessfully maked and installed these two packages. But I got the following errors when compling the asterisk-oh323 0.6.3a: for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]:

Hi, I have Asterisk SVN-branch-1.4-r47845 installed in a Ubuntu Dapper. Its works as sip server and I am trying to get h323 support. I installed these packages: libpt-1.10.0 libopenh323-1.18.0 And I set the next global variables: PWLIBDIR=/usr/share/pwlib OPENH323DIR=/usr/share/openh323 Then, when I execute the configure script (before installation) and finishes with this message: #

I am trying to build PWLIB to get OH323 up and running. I am not an expert in linux but can someone help telling me how I can do the following: How can I add a directory to LD_LIBRARY_PATH?! Thanks in advance ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- ---------------------- For

Hi All; I am trying now to compile h323 to be able to use it, I did the pwlib and openh323 successfully and I exported the PWLIBDIR=/usr/src/pwlib_v1_10_0 and the OPENH323DIR=/usr/src/openh323_v1_18_0, then I was need to compile h323 as following: cd /usr/src/asterisk-1.4/channels/h323 When I type make, it gives me: make: Nothing to be done for 'degault' And when I type make opt, it

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, I'm on a RH 9 box, fresh install. When I try to compile channel h323, I get multiple compile errors. Can someone help? asterisk, ptlib, openh323 all are fresh from CVS. Thanks! Here's what I've done so far: All the source is in /usr/src/asterisk/ [root@localhost openh323]# ls -la /usr/src/asterisk/ total 164 drwxr-xr-x 8 root root 4096 Sep 25 00:59 .

FC2 Asterisk 1.0 When I dial a H323 dialup to an existing OKI Voip Router (BV1250), I get an EndedByRefusal yet the OKI Gateway is setup with the corrent reply ip addresses etc etc, unfortunately its an existing multiple voip router setup with g723.1 and g729a, so changing the codec on the router maybe an issue. I have compile in the h323 as per the channels/h323 setup with the listed libraries.

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

Hello, Users. I loaded module chan_h323.so, chan_vpb.so. I have met a message : "No one is available to answer at this time". I don?t know what I do.. My 'h.323 trace 5' result is : == vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR] -- Executing Dial("vpb/1-8", "h323/192.168.1.107") in new stack 1:21:34.936 ThreadID=0x06f2bbb0

Hi all! This is my VoIP network scheme H323EndPoint ----- --- GW H323/SIP-IN -- -- SIP Phone | | (Sipquest) | | | | | |

Hi List I have one question about MGCP in asterisk. I have a media gateway, and I want to have asterisk to work with the media gateway. As I was told that the media gateway can communicate with the switch via standard interface MGCP/ICGP. Question is if the asterisk MGCP supports such MGCP message ??? Thanks. George Lin

Hello all, I just installed Asterisk with H323 support (chan_h323 from Jeremy McNamara). But experience problem while connecting OpenPhone to Asterisk Here is h.323 trace: 5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720, handle=27 5:37.444 H225 Answer:9cc1250 transports.cxx(564) H225

Hello ! First time I have instaled Asterisk without problem and working with a SIP clinet (X-Lite). Then I try to make the H323 with came with Asterisk. So, I DL pwlib v.1.5.2 in /root (./configure ; make) no errors DL openh323 in /root (./configure ; make opt): ------------ /root/openh323/src/h248.cxx:6178: internal error: Segmentation fault Please submit a full bug report, with preprocessed

Greetings All- I have an * box with the NuFone H.323 channel driver installed. I also have an Altigen VoIP system with a PRI to the PSTN. I can sucessfully make a call from a SIP extension (snom190) to an H.323 extension (altigen phone) The thing I can't seem to make work is a call from a H.323 phone to a SIP extension. Here's the layout:

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

hi, i had asterisk and gnugk running on fedora core 2. it worked quite well. then, i needed to change to red hat 9, and i'm experiencing troubles with h.323 :-( making a call from a h.323 phone (innovaphone) does not work, and dial-in also doesn't. below is an excerpt of what happens, when i try to dial-in my extension (126). it takes about 10(!) seconds, until the 'Called 126'