RE: [Asterisk-Users] oh323 compile error.
I have similar problem on FreeBSD. Gcc and pwlib upgrade solve my problem... Regards, Primoz -Original Message- From: Kim Daeyong [mailto:[EMAIL PROTECTED] Sent: 4. maj 2005 9:11 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] oh323 compile error. Hi. I downloaded pwlib_1.18.1 and openh323_1.15.1 to install Asterisk CVS HEAD version. I tried to install asterisk-oh323-0.7.1. I patched openh323 as typing 'patch -p1 < /usr/src/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch' in openh323 directory. Then I compiled pwlib, openh323 and installed Asterisk. After that, I edited 'Makefile' in asterisk-oh323-0.7.1 directory. I typed 'make', and there is an error. [EMAIL PROTECTED] make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper' ./check_ver /root/work/pwlib pwlib ./check_ver /root/work/openh323 openh323 g++ -DP_USE_PRAGMA -D_REENTRANT -Wall -fPIC -I/root/work/pwlib/include -DPTRACING -I/usr/local//include/openh323 -DHAS_OSS -DHAS_VPB -Wall -x c ++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.8.1\" -DOPENH323VERSION=\"1.15.1\" -I/root/work/pwlib/include/ptlib/unix -I/root/work/pwlib/include -I/root/work/openh323/include -I/root/work/openh323/include/openh323 -I../asterisk-driver -c wrapper_misc.cxx -o wrapper_misc.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -D_REENTRANT -Wall -fPIC -I/root/work/pwlib/include -DPTRACING -I/usr/local//include/openh323 -DHAS_OSS -DHAS_VPB -Wall -x c ++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.8.1\" -DOPENH323VERSION=\"1.15.1\" -I/root/work/pwlib/include/ptlib/unix -I/root/work/pwlib/include -I/root/work/openh323/include -I/root/work/openh323/include/openh323 -I../asterisk-driver -c asteriskaudio.cxx -o asteriskaudio.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -D_REENTRANT -Wall -fPIC -I/root/work/pwlib/include -DPTRACING -I/usr/local//include/openh323 -DHAS_OSS -DHAS_VPB -Wall -x c ++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.8.1\" -DOPENH323VERSION=\"1.15.1\" -I/root/work/pwlib/include/ptlib/unix -I/root/work/pwlib/include -I/root/work/openh323/include -I/root/work/openh323/include/openh323 -I../asterisk-driver -c wrapconnection.cxx -o wrapconnection.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -D_REENTRANT -Wall -fPIC -I/root/work/pwlib/include -DPTRACING -I/usr/local//include/openh323 -DHAS_OSS -DHAS_VPB -Wall -x c ++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.8.1\" -DOPENH323VERSION=\"1.15.1\" -I/root/work/pwlib/include/ptlib/unix -I/root/work/pwlib/include -I/root/work/openh323/include -I/root/work/openh323/include/openh323 -I../asterisk-driver -c wrapendpoint.cxx -o wrapendpoint.o wrapendpoint.cxx: In member function `virtual BOOL WrapH323EndPoint::OpenAudioChannel(H323Connection&, int, unsigned int, H323AudioCodec&)': wrapendpoint.cxx:915: error: `IsDescendant' undeclared (first use this function) wrapendpoint.cxx:915: error: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [wrapendpoint.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper' make: *** [subdirs_build] Error 1 [EMAIL PROTECTED] Please let me know to solve that problem. Thanks for reading. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP problems
Hi, thanks a lot. After adding username, password and changing extensions.conf my very basic * setup works excellent. So, next milestone is establishing connection from my office to my home server where my * resides... Thanks and Regards, Primoz -Original Message- From: Bellows, Jared [mailto:[EMAIL PROTECTED] Sent: 2. maj 2005 22:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP problems It looks like you haven't defined a "tina" extension. You have the "tina" SIP account set to be extension "1000". If you want to dial extension "tina" change "1000" to "tina". -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Primoz Kragelj Sent: Monday, May 02, 2005 12:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] SIP problems Hi all, I'm newbie to VOIP/SIP/asterisk... and I having problems with SIP on local network. I have Freebsd server 5.3 running asterisk and two x-lite clients. I added following lines to sip.conf [tina] type=friend host=dynamic dtmfmode=inband context=sip [primozz] type=friend host=dynamic dtmfmode=inband context=sip And following to extensions.conf [sip] exten => 1000,1,Dial,SIP/tina exten => 2000,1,Dial,SIP/primozz *CLI> sip show users Username Secret Accountcode Def.Context ACL NAT primozz sip No RFC35 tina sip No RFC35 I have X-Lite clinet on Win XP and while trying to make call to "tina" I got 404 error - not found. Same for vice versa...Both users are local. >From debug below following line: To: ;tag=as1283188b is very strange to me. Instead od 192.168.1.3 there should be 192.168.1.1. Do I need to put some ware static IP for each client ? And following is debug from asterisk: Peer audio RTP is at port 192.168.1.3:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for tina in sip list_route: hop: Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz ;tag=1716760483 To: ;tag=as1283188b Call-ID: [EMAIL PROTECTED] CSeq: 22324 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.1.3:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz ;tag=1716760483 To: ;tag=as1283188b Contact: Call-ID: [EMAIL PROTECTED] CSeq: 22324 ACK Max-Forwards: 70 Content-Length: 0 Thanks for help ! Regards, Primoz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Debuging SIP
sip debug Hope this will help.. Regrads, Primoz -Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: 2. maj 2005 20:44 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Debuging SIP Guys. Im having NAT problems. Any good tips on how to debug remote SIPS, how to see which ports are been sent and received, etc? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP problems
Hi all, I'm newbie to VOIP/SIP/asterisk... and I having problems with SIP on local network. I have Freebsd server 5.3 running asterisk and two x-lite clients. I added following lines to sip.conf [tina] type=friend host=dynamic dtmfmode=inband context=sip [primozz] type=friend host=dynamic dtmfmode=inband context=sip And following to extensions.conf [sip] exten => 1000,1,Dial,SIP/tina exten => 2000,1,Dial,SIP/primozz *CLI> sip show users Username Secret Accountcode Def.Context ACL NAT primozz sip No RFC35 tina sip No RFC35 I have X-Lite clinet on Win XP and while trying to make call to "tina" I got 404 error - not found. Same for vice versa...Both users are local. >From debug below following line: To: ;tag=as1283188b is very strange to me. Instead od 192.168.1.3 there should be 192.168.1.1. Do I need to put some ware static IP for each client ? And following is debug from asterisk: Peer audio RTP is at port 192.168.1.3:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for tina in sip list_route: hop: Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz ;tag=1716760483 To: ;tag=as1283188b Call-ID: [EMAIL PROTECTED] CSeq: 22324 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.1.3:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bKBD00F479A53B4C97B6FF319C9D6B06CA From: Primoz ;tag=1716760483 To: ;tag=as1283188b Contact: Call-ID: [EMAIL PROTECTED] CSeq: 22324 ACK Max-Forwards: 70 Content-Length: 0 Thanks for help ! Regards, Primoz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
