Hello,

I am having a problem with SendDTMF – it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.

I use software phone to test it… when I dialed 501, I cant hear anything
for about 10 seconds (this is because of SendDTMF) and then I can hear
the operator saying to enter the numbers but SendDTMF already did it?!

Asterisk server are connected to voip.ms provider.

I have spent many hours trying to get to work, how to fix this issue?

See the configuration and debug log below:

extensions.conf
================
[test]
exten => 501,1,Set(CALLERID(num)=004471XXXXXXX)
exten => 501,n,Dial(SIP/+44797XXXXXX@voipms,30,M(sendnumber)t)
exten => 501,n,Hangup()

[macro-sendnumber]
exten => s,1,Wait(3)
exten => s,n,SendDTMF(www0w7w8w8wXwXwXwXwXwX)

sip.conf
==========
[general]
context=default
tcpbindaddr=0.0.0.0
dtmfmode = rfc2833
register => xxxxx:vxxxxx@london.voip.ms:5060

[test]
type=peer
secret=2xxx
host=dynamic
context=test

[voipms]
canreinvite=no
host=london.voip.ms
secret=xxxxxx
type=peer
username=135xxx ;your account
disallow=all
allow=gsm
; allow=g729 ; Uncomment if you support G729
fromuser=135xxx
insecure=invite
trustrpid=yes
sendrpid=yes
nat=yes
dtmfmode=rfc2833

debug:
=====
== Using SIP RTP CoS mark 5

I have been looking online for a definitive how-to on using Asterisk as a
SIP trunk termination point . I am seeing conflicting messages and
methodologies for doing this.

I am not going to use a commercial vendor for this trunk, it will be used in
testing out various customer scenarios and am looking at Asterisk as one
alternative.

I see two ways to do from all my research
1.use two asterisk in VMs(or on bare metal) to originate and terminate a
peer trunk.
- this would be good if one doesn’t have an IP PBX or control over the
remote end.
2.use asterisk to terminate one end of the trunk by using it to log in to a
sip trunk and then define the peering.

I know the first one is easy, but the second may be the way to go.
What does asterisk need to supply the remote end of the sip trunk? I realize
that it is based on the remote end, I just haven’t seen any example for the
most popular IP PBXs, like Cisco UC or Avaya CM/SES.

Ids this is all trust, I could see the SIP trunk peer entry and trusted and
maybe bypassing the peer username/password in the definition.

Has anyone done this in a proof of concept lab or in production?

Jonas Kellens wrote:
> Will ODBC become the default then ?

As far as I’m aware, yes.

You need to make sure that unix-odbc development libraries are installed.

Under Mandriva, Mageia:

urpmi unixodbc-devel

Ubuntu/Debian:

apt-get install unixodbc-dev

And here is a good link to get you started:

http://nerdvittles.com/?p=604

Doug

Hi,
Because I had an incoming caller ID issue, I plan to upgrade Dahdi to the
latest version, hopefully this will solve the caller ID problem.

I’m running Elastix 2.2 with Dahdi tool 2.4.1.2

It it “safe” to upgrade to the latest version of dahdi? What is the best
way to do it? through yum?

Many thanks,
Anam.

Hello,

notice in the console output beneath that there is a resultid 6 but it
can not be cleared :

[May 5 11:46:27] — Executing [s@sub:3] MYSQL(“SIP/vart-00000336″,
“Connect connid localhost dialplan host Asterisk”) in new stack
[May 5 11:46:27] — Executing [s@sub:4] MYSQL(“SIP/vart-00000336″,
“Query resultid 4 DELETE FROM pickuptbl WHERE pickmark LIKE
“%SIP/vart2-00000336%”") in new stack
[May 5 11:46:27] — Executing [s@sub:5] MYSQL(“SIP/vart-00000336″,
“Clear 6″) in new stack
[May 5 11:46:27] WARNING[17803]: app_mysql.c:194 find_identifier:
Identifier 6, identifier_type 2 not found in identifier list
[May 5 11:46:27] WARNING[17803]: app_mysql.c:510 aMYSQL_clear: Invalid
result identifier 6 passed in aMYSQL_clear
[May 5 11:46:27] — Executing [s@sub:6] MYSQL(“SIP/vart-00000336″,
“Disconnect 4″) in new stack
[May 5 11:46:27] — Executing [s@sub:7] Return(“SIP/vart-00000336″,
“”) in new stack

How come ??