Thanks for the input. Yes, I did call out many times, but the recording doesnt happen even after the call is bridged and there is two way audio. I also took out the b option and so it should recording the ringing right (even before call is bridged) ..
Thank you so much that solved my database issue. Now how asterisk will
forward call ?
Or I need to specify gotoif statment in my stdexten to check database
key and take ..
All; I need to configure the SIP account so if first IP address failed then to send for the second IP address. How to do this? While configuring the sip account, at the host parameter, can I give two IP addresses separated by comma? Or what shoul..
does anybody know a good tutorial on how to setup asterisk to receive faxes
(no need to send them) ? ive tried using app_fax.so with T38 but i keep
getting Transmission ..
I have a couple of questions about asterisk 1.6:
Can you change codecs mid-call upon re-invite?
Can you handle the SDP offer-answer in the 200-ACK instead of the usual
Thanks in advance for any insight.
Thanks, looks really helpful for managing connected users (half my problem). On the web-interface question, how do I create a website with a [call-in] button? Im using Drupal, so will make it a members only page. Basically they click the [call-in] butt..
I am running Asterisk 22.214.171.124, dahdi 2.4.0 and libpri 126.96.36.199 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card.All PCI Express cards. Co-workers are hearing hissing sou..
I see a lot of these messages in the debug log : /[May3 15:47:09] DEBUG audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May3 15:47:09] DEBUG audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [M..
All i have a asterisk install from rpm (digium yum) and i wish to install res spandsp i comipled from source using the same asterisk version and copyied the file to the module directory but when trying to load i get this WARNING loader.c: Mod..