Kai, <br>
<br>
could you please share your patch with us and describe more in detail what does it do?<br><br><div><span class="gmail_quote">2006/3/15, Kai Militzer <<a href="mailto:km@westend.com">km@westend.com</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello comunity,<br><br>I tought I could share my experience with chan_ss7 with you and maybe<br>get some answers/opinions from the rest of you. How wants to know the<br>solution for the ringback tone will have to read til the end of this
<br>mail. ;)<br><br>What is most important to know for the most, is I guess, that chan_ss7<br>works with an Alcatel S12 switch. I have it (in a lab config) running<br>relativly stable since late december 2005 (starting with chan_ss7-
0.2)<br>with one E1 (30 Channels). Three weeks ago (befor I went on vacation ;)<br>) I added another E1 and this also seems to work (say: I was still able<br>to make calls after my return today).<br><br>The version I am currently running is modified version
0.8. The<br>modifications were neccesarry because I use chan_ss7 to "convert" calls<br>from ss7 to SIP and vice versa without terminating them on this asterisk<br>instance. The SIP part of the call is simply forwarded to a SIP Server
<br>that then terminates the call. The problem I had was, that I cannot tell<br>on the asterisk with chan_ss7 if the dialed number is complete and<br>equiped and so I have to match everything with _X. This approach did not
<br>work with overlap dialing, because it would match directly after the<br>IAM. So I added a timer that waits for a SAM after an IAM and starts<br>again if a SAM is received. In my opinion this is the only way to use<br>
chan_ss7 as a gateway without knowledge of the numberingplan on the<br>final destination. Sifira didn't see it this way and wouldn't take my<br>patch into the main chan_ss7 ;( , maybe some of you could convince them<br>to do so. ;)
<br><br>Another problem I had was with the handling of the hangupcause which<br>weren't translated correctly from SS7 to SIP and other way round. In my<br>opinion the error was in ast_softhangup_nolock in asterisk, but seems
<br>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>).<br>@sifira, if you are reading this: would it be possible to fix this in<br>chan_ss7?<br><br>Now my question to the comunity: Is anyone running asterisk with
<br>chan_ss7 as PSTN-to-SIP Gateway anf if yes what are your experiences?<br>Does it work reliable, what call volumes do you handle with it?<br><br>And last but not least, I also had the problem that no ringback tones<br>
were generated by asterisk. The following two lines in the dialplan<br>inserted before the Dial statement do the trick:<br><br>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><a href="mailto:km@westend.com">km@westend.com</a>
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Aachen Fax
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