Below is the copy from the archive on the SAM and IAM issue.<br><br>Though, I have not join the community then but I need somebody to get this clearer inorder to allow overlap dial.<br><br>"So I added a timer that waits for a SAM after an IAM and starts again if a SAM is received"
<br><span style="font-weight: bold;">how can I add a timer or has this been catered for?</span><br style="font-weight: bold;"><br>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 SAM.
<br><br>How can I achieve that.<br><br>I am using chan_ss7-0.8.4 with asterisk-1.2.8,<br>asterisk-oh323-0.7.3.<br><br>Kindly assist please.<br><br>goksie<br><br><span style="font-weight: bold;">My experiences with chan_ss7, some questions and a solution for the ringback tone
</span> ________________________________<br> <br>From: Kai Militzer <km (at) <a href="http://westend.com">westend.com</a>><br>Date: Wed, 15 Mar 2006 16:07:30 +0100 ________________________________<br> Hello comunity,
<br><br><br>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.
<span style="font-weight: bold;">So I added a timer that waits for a SAM after an IAM and starts again if a SAM is received</span>. 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
<a href="http://bugs.digium.com/view.php?id=6550">http://bugs.digium.com/view.php?id=6550</a>). (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)
<br>exten => _X.,n,Playtones(ring)<br><br>I hope that helps. ;)<br><br>Best regards,<br>Kai<br><br>--<br>Kai Militzer WESTEND GmbH | Internet-Business-Provider<br>Technik CISCO Systems Partner - Authorized Reseller
<br> Lütticher Straße 10 Tel 0241/701333-14<br>km (at) <a href="http://westend.com">westend.com</a> D-52064 Aachen Fax 0241/911879<br>