similar to: chan_h323.c compile error

Hi can anyone help me with this g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I../../include -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include

New 5.3 install. Saw some stuff around about problems with pwlib but those were in 0.9 and have long since been fixed and I haven't found anything else out there to explain this stuff. Anyone have any ideas? gmake[2]: Entering directory `/usr/ports/net/asterisk/work/asterisk-1.0.2/channels/h323' c++ -O -pipe -c -fno-rtti -o ast_h323.o -O -pipe -Wall -Wstrict-prototypes

Setup: Asterisk with chan_h323 (chan_iax was connecting the two clients directly, dropping asterisk out of the picture) Clients are two pentium class computers on the same network with ohphone installed. The idea is simply to have one client call into asterisk (a client calling from outside) to be answered by an operator then transferred to a different extension. Extensions.conf: [default]

Hi, I am trying to install the h323 channel on the latest Asterisk CVS. I have re-compiled openh323 and pwlib, and now make in /usr/src/asterisk/channels/h323 works g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I../../include -I/usr/src/pwlib/include

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

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

Today my pbx stopped responding to my sip phones.. looking into the log, here what I got: Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc): Unable to create RTP session: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 4655

Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was released in November, but I have yet to get my hands on it. The Polycom site has way more docs online, but the link to the firmware only brings up the release notes. -sb

I am having a problem when a SIP registration fails. I get the following messages in the log: Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874 (sip_reg_timeout): Registration for '<user>@fwd.pulver.com@65.39.205.114' timed out, trying again Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 5119 (handle_request): Registration from

Hi, I am having trouble installing the Open H.323 Channel driver in Asterisk. I have installed PWLib and OpenH323 V.1.11.7 This is the output l get before it bails out. asterisk:/usr/src/asterisk/channels/h323# make g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX

I'm trying to make the chan_h323 in /usr/src/asterisk/channels/h323 But I'm getting all kinds of errors about PWLIB... I built using the newest PWLIB and OpenH323 from CVS Error log from make below make g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -DP_HAS_SEMAPHORES

I'm trying to make the chan_h323 in /usr/src/asterisk/channels/h323 But I'm getting all kinds of errors about PWLIB... I built using the newest PWLIB and OpenH323 from CVS Error log from make below make g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -DP_HAS_SEMAPHORES

Caller*ID used to work as some point, but I can't seem to get it going these days. The card is a x101p. I've tried going up and down the rxgain scale. Can the txgain effect it at all? When I plug in a phone into the line with a splitter it can decode caller id with no problems. Reading through the mailing list archives hasn't given me any move clues. Any ideas?

In my setup, which is FreeBSD, using pjsip 2.5.5 as sip backend I am observing a regression when testing the latest Release Candidate. Any calls get refused and the following error shown on console: [Nov 22 10:49:26] WARNING[101105]: res_rtp_asterisk.c:2400 int create_new_socket(const char *, int): Unable to allocate RTP socket: Protocol not supported [Nov 22 10:49:26] WARNING[101105]:

Hi, Can anyone help? I get the following error when trying to complie the h323 channel under the source installation directory asterisk/channels/h323 i have read the readme file and kept to the recomended versions; h.323 v1.12.2 and PWLIB v1.5.2 Thanks in advance [root@gw01 h323]# make g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDE BUG -DDO_CRASH

Hello.... Asterisk is compiled and running perfectly... But when i try to compile Channel_h323... that's another story :-|. I'm compiling using RH9, OpenH323_1.12.2, pwlib_1.5.2, i compiled and installed them myself... GnuGk is running smoothly using them. Also my intention is to terminate calls from a SIP client through Asterisk and into my GnuGK... has anyone accomplished this ?

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Hi, I am taking over an asterisk system from another person and having an issue where a sip trunk is restricting the outgoing codecs to just g.729 I am dialing in from a Cisco 7960. The Invite from the Cisco has the folowing M line: m=audio 17022 RTP/AVP 18 0 8 101. So it is allowing g.729, ulaw and alaw. Asterisk is tandeming the call out over a SIP trunk sip.conf tandem trunk config:

The following occurs with code from yesterday's cvs (asterisk) and current OpenH323 code: [root@raid-2 h323]# make clean install rm -f *.o *.so core.* 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_ENDIA N -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING