[asterisk-ss7] IAM AND SAM ISSUE;

Goke Aruna goksie at gmail.com
Thu Jul 6 05:52:02 MST 2006

Below is the copy from the archive on the SAM and IAM issue.

Though, I have not join the community then but I need somebody to get this
clearer inorder to allow overlap dial.

"So I added a timer that waits for a SAM after an IAM and starts  again if a
SAM is received"
how can I add a timer or has this been catered for?

However, each time my ss7 provider dialed (the PSTN) dialed 009 + their
number ... it gets only the first 7 digits and I need to receive the remain

How can I achieve that.

I am using chan_ss7-0.8.4 with asterisk-1.2.8,

Kindly assist please.


My experiences with chan_ss7, some questions and a solution for the ringback
tone ________________________________

From: Kai Militzer <km (at) westend.com>
Date: Wed, 15 Mar 2006 16:07:30 +0100   ________________________________
   Hello comunity,

I tought I could share my experience with chan_ss7 with you and maybe  get
some answers/opinions from the rest of you. How wants to know the  solution
for the ringback tone will have to read til the end of this  mail. ;) What
is most important to know for the most, is I guess, that chan_ss7  works
with an Alcatel S12 switch. I have it (in a lab config) running  relativly
stable since late december 2005 (starting with chan_ss7-0.2)  with one E1
(30 Channels). Three weeks ago (befor I went on vacation ;)  ) I added
another E1 and this also seems to work (say: I was still able  to make calls
after my return today). The version I am currently running is modified
version 0.8. The  modifications were neccesarry because I use chan_ss7 to
"convert" calls  from ss7 to SIP and vice versa without terminating them on
this asterisk  instance. The SIP part of the call is simply forwarded to a
SIP Server  that then terminates the call. The problem I had was, that I
cannot tell  on the asterisk with chan_ss7 if the dialed number is complete
and  equiped and so I have to match everything with _X. This approach did
not  work with overlap dialing, because it would match directly after
the  IAM. So I added a timer that waits for a SAM after an IAM and
starts  again if a SAM is received. In my opinion this is the only way to
use  chan_ss7 as a gateway without knowledge of the numberingplan on
the  final destination. Sifira didn't see it this way and wouldn't take
my  patch into the main chan_ss7 ;( , maybe some of you could convince
them  to do so. ;) Another problem I had was with the handling of the
hangupcause which  weren't translated correctly from SS7 to SIP and other
way round. In my  opinion the error was in ast_softhangup_nolock in
asterisk, but seems  not to be the case (see
http://bugs.digium.com/view.php?id=6550). (at) sifira, if you are reading
this: would it be possible to fix this in  chan_ss7? Now my question to the
comunity: Is anyone running asterisk with  chan_ss7 as PSTN-to-SIP Gateway
anf if yes what are your experiences?  Does it work reliable, what call
volumes do you handle with it? And last but not least, I also had the
problem that no ringback tones  were generated by asterisk. The following
two lines in the dialplan  inserted before the Dial statement do the trick:
exten => _X.,n,SetLanguage(de)
exten => _X.,n,Playtones(ring)

I hope that helps. ;)

Best regards,

Kai Militzer                 WESTEND GmbH  |  Internet-Business-Provider
Technik                      CISCO Systems Partner - Authorized Reseller
                             Lütticher Straße 10      Tel 0241/701333-14
km (at) westend.com               D-52064 Aachen              Fax
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