search for: nsthi

...communicator.. it mean i can able to make a call and receive.. but i can't hear the other person voice. but my voice he can able to hear... some times i can't able to make (Between 2 sip comm.)call also... I'm using asterisk 1.4 versoin... could u tell me any suggestions.. Regards, nsthi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061206/cb463407/attachment.htm

...ference to `ast_copy_string' ast_expr2f.o(.text+0x1198): In function `ast_expr': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ast_expr2.fl:248: undefined reference to `ast_copy_string' collect2: ld returned 1 exit status make[1]: *** [aelparse] Error... plz could answer this issue. -nsthi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061213/596dc3d1/attachment.htm

Hello ppl, Reading all over the net. Learnt quite a lot, but that has left me confused-a-lot as well. Need answers to a few questions. Before that, I have an ISP(fax gateway) which will help me send/recv faxes using the T.38 protocol. I am using Asterisk 1.2.12.1. Now to the few questions I had: 1) Do I need any additional hardware on the Asterisk box?? I did download the spandsp and

...l steps ..in order to check and compile i 've to do recompile the whole asterisk module and it consuke to much time... please anyone couls you tell me, how can i modify it , compile and test the I/O in asterisk applications in a easy way... plz do reply .. Thanks for ur commands, Regards, nsthi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061204/40c82d82/attachment.htm

...communicator.. it mean i can able to make a call and receive.. but i can't hear the other person voice. but my voice he can able to hear... some times i can't able to make (Between 2 sip comm.)call also... I'm using asterisk 1.4 versoin... could u tell me any suggestions.. Regards, nsthi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061206/e1b91cca/attachment.htm

...communicator.. it mean i can able to make a call and receive.. but i can't hear the other person voice. but my voice he can able to hear... some times i can't able to make (Between 2 sip comm.)call also... I'm using asterisk 1.4 versoin... could u tell me any suggestions.. Regards, nsthi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061208/f3fdd22d/attachment.htm

...communicator.. it mean i can able to make a call and receive.. but i can't hear the other person voice. but my voice he can able to hear... some times i can't able to make (Between 2 sip comm.)call also... I'm using asterisk 1.4 versoin... could u tell me any suggestions.. Regards, nsthi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061211/3aa61435/attachment.htm

Hi moises, Hi i need to done some modify/changes in one of the asterisk c source code ,,eg: app_meetme.c How can i compile and debug it without compile the whole module of asterisk... and also let me know which editor suitable it ,I'm using suse linux.. Plz help me reg. this .. -nsthi, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070104/8c9c00d3/attachment.htm