[Asterisk-Users] Disconnecting after 1 min while Communicating Clarent class 5 call manager
Hi List I installed asterisk server and tried to transfer calls from asterisk to Clarent class 5 call manager. The calls are passing through with out any problem but after 60 seconds the call get disconnected automatically. Please help me to sort out this problem. Attaching here with my sip configuration file Sip.conf. [general] rtpholdtimeout=300 rtptimeout=300 defaultexpirey=20 context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes [clarent] register => test:[EMAIL PROTECTED]/123456 type=friend secret=1234 username=testuser host=192.168.10.150 fromuser=test1234 insecure=very fromdomain=192.168.10.150 canreinvite=no nat=no disallow=all allow=g723 context=default Thanks in advance. Anil ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 registration Failure
Hi List, I am very new to asterisk. I downloaded asterisk from CVS head yesterday and compiled it in Redhat linux 9. I created a sip account for testing and configured it in the Firefly. While Firefly try to connect to the asterisk server i am getting an error as below and failing the registration. Oct 1 08:00:57 NOTICE[23415]: chan_sip.c:10646 handle_request_register: Registration from '"200" ' failed for '192.168.10.200' - Not a local SIP domain My sip configuration is as below, [general] context=default ; Default context for incoming calls bindport=5060 bindaddr=0.0.0.0 [test] type=peer secret=200 username=200 host=dynamic nat=no #disallow=all allow=all context=default Please help me to find out the problem in my configuration. Thanks in advance Rgds Anil ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Meetme2 compilation problem
Finally i installed the asterisk home , meetme2 is working perfectly in it. Thanks a lot anil On Tue, 22 Mar 2005 20:34:41 +1100, PHP Mechanic <[EMAIL PROTECTED]> wrote: > ï > > User=guest, password=restricted. > This account wil be open util friday. > > Nope: > 220 Welcome to the Vink Consultancy FTP server. Please login... > Name (ftp.vinkconsult.com:brianc): guest > 331 Password required for guest. > Password: > 530 Login incorrect. > Login failed. > > Yep: > $ ftp ftp://guest:[EMAIL PROTECTED]/ > Connected to services.vinkconsult.com. > 220 Welcome to the Vink Consultancy FTP server. Please login... > Remote system type is UNIX. > Using binary mode to transfer files. > 331 Password required for guest. > 230 User guest logged in. > 200 Type set to I > > ___ > 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] Meetme2 compilation problem
I did the patch also . That didnt help me. I am using CVS head of 17th March . Googling didnt give me much info other than this patch. Thanks On Fri, 18 Mar 2005 10:18:26 -0500, Giovanni Powell <[EMAIL PROTECTED]> wrote: > I'm sure there was a patch for meetme2 regarding compilation... google > for meetme2 + patch. It worked for me. > ___ 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] Meetme2 compilation Err
Hi , While compiling meetme2 i am getting the following error. Please guide me to make it work. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:317: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:317: for each function it appears in.) make: *** [app_dial.o] Error 1 Thanks in advance Anil ___ 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] Meetme2 compilation problem
Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:317: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:317: for each function it appears in.) make: *** [app_dial.o] Error 1 Thanks in advance Anil ___ 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
