i am seriously stuck with an asterisk and sip problem.
the following sip.conf works with respect to some_peer:
[general] bindaddr = x.y.z.w nat = no
[some_peer] type=peer host=somehost secret=somesecret some other unrelated options
here x.y.z.w is the ip address of the interface pointing to the network containing somehost. more precisely its the address of tun0 and route -n prints Destination Gateway Genmask Flags Metric Ref Use Iface [..] x.y.z.0 0.0.0.0 255.255.255.0 U 0 0 0 tun0 [..]
here 'it works' implies that i have to change and reload sip.conf after ifup tun0, or anything that forces tun0 to go down, like my dsl provider.…
I am planning on building a testing module which would spawn about 500 calls in order to test the performance of the network by transferring audio/speech files to end points at that juncture.Is it possible to spawn as many concurrent calls (or nearly concurrent calls) using just call files.Is there a limit as to the maximum number that could be spawned.? I tried doing this for about 20 calls and found that there is autofallthrough after a point of time.Is this a problem with my dialplan or is it because of the call files (i also get a warning which…
I have an Asterisk server configured to run as voicemail with a T1 and SMDI. It has 188.8.131.52 (dahdi 184.108.40.206) and CentOS 5.6 and has worked great for a few years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on
The problem I am having appears to be related to DTMF detection. When the test phone number is called (2704083000) Asterisk only receives a portion of the dialed number. It varies as to what numbers are detected. Sometimes it sees a single digit, sometimes 3 or 4 of the digits of the dialed number.
When I compare…