Sip.conf And Binaddr Issue


Hi there.

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 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.…

Asterisk Users 3.3 years ago 3 Answers

Maximum Concurrent Calls Using Call Files


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…

Asterisk Users 3.3 years ago 2 Answers



I have an Asterisk server configured to run as voicemail with a T1 and SMDI. It has (dahdi 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 CentOS 5.8

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…

Asterisk Users 3.3 years ago 10 Answers

Asterisk Trying To Call A Queue With No Members



I am trying to configure some static queues in asterisk, it's almost working, the problem is that asterisk is not verifying if the queue has logged members. For example, if I create queue called test, which has no members logged in, and try to place a call using Queue(test) I get into the queue, even if all phones are turned off, I tried to verify it with the QUEUE_MEMBER function, using the "ready" parameter, it shows me that all members are logged in. Here is my queues.conf:

[general] persistentmembers = yes monitor-type = MixMonitor

[default-queue](!) musicclass

Asterisk Users 3.3 years ago 5 Answers

Can I Install Asterisk Normally And Then Installing The GUI "asterisk Now"



Is it possible if I have already asterisk installed on Fedora machine to install the GUI "asterisk now" without doing a fresh installation using the Asterisk Now CD?

Which version of the GUI that should be selected to work with the asterisk version? For example, if I have asterisk 1.8 then which GUI version to select? "I am talking about compatibility".

Can I say that Freepbx is Asterisk + Asterisk Now?

Regards Bilal

Asterisk Users 3.3 years ago 1 Answer

Call File And NFS Server



I have 3 server, 2 running with asterisk and another one generate call files say some directory callfile/serverA and callfile/serverB (NFS Sharing) and mounted this directory to respectively on Server A (Asterisk) and Server B(Asterisk) on /var/spool/asterisk/outgoing.

Server A has Asterisk 1.8.0-rc2 and Server B has asterisk version, and both asterisk compile ./configure --without-inotify

Callfile will execute call successfully on both machine, but got the following problem

*[Jul 6 16:15:04] WARNING[26921]: pbx_spool.c:278 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/ Operation not permitted * I have set the folder (callfile/Server{A/B}) permission to 777 as well as call file permission to 777.

-- Regards,

Chandrakant Solanki

Asterisk Users 3.3 years ago 4 Answers