* You are viewing the archive for July 3rd, 2012


I have been doing some testing with queues. I have been experiencing some strange behavior and I wanted to see if anyone else sees this. I am using 1.8.11-cert2.

It is my understanding that I cannot directly tell from the dial plan that a member is dynamic or static unless I check RQMSTATUS from RemoveQueueMember for NOTDYNAMIC. Is this true? (If I try something like using AST_CONFIG then I can only get back the first static member for a queue) The problem that I am seeing is that if I try to remove a queue member that is static, the values from then on that I see from ${QUEUE_MEMBER(,free) ${QUEUE_MEMBER(,ready) are incorrect. They are always one less than the actual value for that queue. Can anyone else confirm this? Thank you.

Chet Stevens

Outbound Asterisk Calls Default Directmedia Specifications

I am using call files to make calls to a remote machine but can’t seem to quite understand the directmedia options that are set by default in
Asterisk.Is there any way i can specify the directmedia options using call files?

Free PBX: Hangup Even If Did Not Dial # In The Voicemail


What is the setting to be done on freepbx to let the voicemail go for hangup after while (or after leaving the message) even if the caller did not dial #. It is very important for me to be sure of the hangup status.


IAX Trunking Stopped Working

I administer a group of Asterisk servers running a mix of 10.3, 10.4, and (mostly 10.4). One of those servers is a call concentrator/relay for E911 service. All of the other servers make an IAX connection to the relay server, which then hands off to a SIP trunk to my E911 provider. It all worked as recently as 2 weeks ago, but I discovered that sometime between then and now it stopped working without any explanation. Last modified time on the config files is over 2 months ago.

The setup is as follows:

On the call relay (IAX “receiver”)
type=user host=dynamic username=my-remote-username encryption=yes secret=my-remote-secret context=my-call-context deny=

On the VoIP servers (IAX “sender”)

- One of the servers is set to register: register => my-remote-username:my-remote-secret@call.relay.server.ip

- Another is set to just use the peer definition as below without trying to register
type=peer host=call.relay.server.ip username=my-remote-username secret=my-remote-secret qualify=no

Dialplan on the VoIP servers:
exten => 911,1,Verbose()
same => n,Dial(IAX2/my-remote-server/911)

Dialplan on the relay server:
exten => 911,1,Verbose()
same => n,Dial(Relay to E911)

The issue I’m seeing is this:

- On the servers that are set to register, the relay server is rejecting the registration (I’ve confirmed the username/peername/secrets are an exact match on both sides, and nothing has changed from when they were working). IAX debug on the relay server shows the auths come in and the relay server send REGREJ – Registration Refused, Cause Code 29. IAX debug on the server attempting to register shows sending the REGAUTH packets and receiving the REGREJ packets. The IP address shown in the IAX debug packets matches the IP address in the permit rule for each peer that’s supposed to register.

- On the server that is set to just send the calls, an attempt to dial 911 just hangs for 60 seconds and eventually times out without sending the call. IAX debug on the relay server shows the call start frame get RX’d, shows the relay server try to TX a CTOKEN frame, and nothing further (other than retransmissions of the call start frame). IAX debug on the server trying to send the call to the relay server shows the TX for the call start frame, but no RX for the CTOKEN frames.

Ultimately, this has gone from working to totally broken without any apparent change to my configuration. I need help to try to troubleshoot it further, I’ve tried everything I can think of (including transferring the backed-up working config files to a brand new clean-load server, upgrading Asterisk, and recreating the configurations by hand), and nothing seems to be helping.

Thank you,

How To Play Different Different Hold Music.

Dear All,

I have two server ‘A‘ and ‘B‘ . In Server ‘A‘, five different ivr (Sevices) is playing and call is *forwarding *into Server ‘B‘. Server ‘B‘ basically use for agent login(Extension). I want to play different hold music(Server ‘B‘) bases on the corresponding services which is running into server ‘A‘.

A single agent takes the call from different different services but hold music is play astrisk own by default. Is there any way to play different hold music bases on services which run into server A.

I have some changes into musiconhold.conf (server B) but problem is no solve.

please help me.


AMR – Segmentation Fault

Hi All,

OS : Cent OS 5 64Bit
Asterisk : 1.8.0-rc2

AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/

When I tried to call or start asterisk, I found “Segmentation Fault”. Below
I paste same for AMR

Loaded symbols for /usr/lib/asterisk/modules/app_db.so
Core was generated by `asterisk -qg’.
Program terminated with signal 11, Segmentation fault.
#0 D_plsf_3 (st=, mode=,
bfi=, indice=,
at sp_dec.c:567
567 tmp = ( ( cos_table[ind+1]-cos_table[ind] )*offset ) << 1;
(gdb) br
Breakpoint 1 at 0x2aaab57093f1: file sp_dec.c, line 567.
(gdb) bt
#0 D_plsf_3 (st=, mode=,
bfi=, indice=,
at sp_dec.c:567
#1 0x00002aaab570df95 in Decoder_amr (st=02aaad6147d00, mode=MR515,
parm=07fff11d06a40, frame_type=,
A_t=07fff11d06730) at sp_dec.c:4717
#2 0x00002aaab5712e6a in Speech_Decode_Frame (st=02aaad613e200, mode