[asterisk-users] requirecalltoken and Realtime
Hi, I've just had to enable the requirecalltoken=no option in iax.conf for one of my IAX2 trunks, and I don't think it works properly in the realtime version. I've created the requirecalltoken field in my (Postgres via ODBC) database, type text, and have variously tried it with 'yes', 'no' and 'auto' in the field. But the setting never seems to be used and thus calls fail down the trunk. If I try the same thing using iax.conf flat file, the requirecalltoken parameter works fine, so I was wondering if anyone else has seen this and wonder if I've tripped over a bug? All this was tested using 1.6.1 SVN, r216266. Gary H -- Gary Hawkins MBCS gary.hawk...@garyhawkins.me.uk OpenPGP Key ID: 0x9A1037BB Web: http://www.garyhawkins.me.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] requirecalltoken and Realtime
Tilghman Lesher wrote: On Friday 04 September 2009 12:08:26 Gary Hawkins wrote: I've just had to enable the requirecalltoken=no option in iax.conf for one of my IAX2 trunks, and I don't think it works properly in the realtime version. [snip] Please try the attached patch. I've just tried the patch - but it doesn't seem to have made any difference - iax.conf entries still work though exactly as before. Gary H -- Gary Hawkins MBCS gary.hawk...@garyhawkins.me.uk OpenPGP Key ID: 0x9A1037BB Web: http://www.garyhawkins.me.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gosubs broken since r160626 (1.6.0 SVN) ?
Mark Michelson wrote: In a fit of wild curiosity, I decided to double-check to be sure that the problem was an AEL parser issue and not one of my own. I actually discovered a bug introduced by my changes. I have fixed this bug in revision 161494 of the 1.6.0 branch. I suspect this will fix the problem you were seeing, too. I've just tested with this revision and all seems to be well again. Thanks for finding and fixing the bug! Gary H ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gosubs broken since r160626 (1.6.0 SVN) ?
Hi all,
I've just upgraded to latest 1.6.0 SVN from a few days ago and my Gosubs
have stopped working.
This is from the verbose logs:
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/aaisp-3802,
1?5:7) in new stack
-- Goto (incoming-aaisp,0407271,5)
-- Executing [EMAIL PROTECTED]:5] Gosub(IAX2/aaisp-3802,
macro-announcement,s,1(anonymous_call_rejection,22)) in new stack
== Spawn extension (incoming-aaisp, 0407271, 6) exited non-zero on
'IAX2/aaisp-3802'
-- Hungup 'IAX2/aaisp-3802'
This was the original AEL2 code:
0407271 = {
Verbose(We got here);
AGI(caller_id_rewriter/caller_id_rewriter.py);
Set(CALLERID(name)=1 ${CALLERID(name)});
if (${WITHHELD} = yes) {
macro-announcement(anonymous_call_rejection,22);
Hangup(22);
}
Dial(${ALLPHONES},20);
if (${DIALSTATUS} = BUSY) {
VoiceMail(201,b);
}
else
{
VoiceMail(201,u);
}
Hangup(${HANGUPCAUSE});
}
This was working on 1.6.0 SVN before r160626 and I have not changed any
of the code. The Gosubs were generated by the AEL parser. In the AEL2
dialplan I am calling
macro-announcement(anonymous_call_rejection,22);
Has anyone seen similar problems to this?
Thanks
Gary H
--
Gary Hawkins [EMAIL PROTECTED]
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[asterisk-users] chan_misdn and Asterisk 1.4.1 - Early B3 not working any more
Hi, I recently upgraded to Asterisk 1.4.1 from Asterisk 1.4.0 and suddenly my early B3 seems to have stopped working. When I call a number that is out of service from my SIP phones, instead of hearing the announcement from the telco, I immediately receive a SIP Busy message and the call is disconnected. hear no audio. This was working fine in Asterisk 1.4.0 with exactly the same configuration, early_bconnect options is enabled in misdn.conf, and all other mISDN features are working perfectly, so I am not sure exactly what has gone wrong here. The same problem also is exhibited on today's stable branch 1.4 SVN (revision 58669). Telco is BT, using an ISDN2e line. Debug output from my system is available here: http://www.pastebin.ca/389195 Does anyone have any ideas on what has changed? Thanks Gary Hawkins -- Gary Hawkins MBCS [EMAIL PROTECTED] PGP/GnuPG: 0x4D82B019 (expires 31 Dec 2007) Web: http://www.garyhawkins.me.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.11 and ${SIPDOMAIN} variable
Hi,
I've just upgraded from Asterisk 1.2.10 to 1.2.11 and I've noticed that the
${SIPDOMAIN} variable now contains a different (and to my mind, incorrect)
value than what it used to. Instead of (say) example.com, it now contains
the string example.com;user=phone instead which causes calls to fail if you
then try and use the Dial app to call [EMAIL PROTECTED] or try to do a
match on a particular domain. I just wanted to find out if anyone else has
noticed this so I can get some evidence to report this as a bug...
Thanks
Gary H
--
Gary Hawkins MBCS [EMAIL PROTECTED]
PGP: 0x6D4E5C77 (expires 31 Dec 2006)
Web: http://www.garyhawkins.me.uk
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Re: [asterisk-users] Outgoing MSNs and chan_misdn
Marco Mouta wrote: in your init-misdn.conf (or misdn.conf, not sure now...) you can choose the MSNs for your incoming Ports or Outgoing ports, msns=3223242,3223243,3223244 for example. Then in your calls, just set the outgoing callerid for your trunk, to one of them. Be aware that as far as i know you must own the MSN you r going to set otherwise you are spoofing MSN Please give some feedback. I've got it working now -- thank you! I notice that you can also use msns=* as well as setting the individual numbers. Once I'd entered that into misdn.conf, and used a command of the form Set(CALLERID(num)=234567) in the dialplan, it now works as I want it to. Gary H -- Gary Hawkins MBCS [EMAIL PROTECTED] PGP: 0x6D4E5C77 (expires 31 Dec 2006) Web: http://www.garyhawkins.me.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing MSNs and chan_misdn
Hi, Does anybody know if it is possible to set the outgoing MSN to a different value than the default set in misdn.conf for a single call only via chan_misdn 0.3.x, and if so, how to do it? I can't find any info on how to do this via Google, and I've tried a few things myself, none of which seem to work. This is definitely possible via chan_capi... Thanks Gary H ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi [0.4.0|-cm-0.5.4] and Asterisk 1.2.0-beta1 - early B3 not early enough sometimes
Hi, I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded chan_capi and compiled it in and run it. (For comparison purposes, I've tried this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and chan_capi-cm latest CVS). Whilst most things are fine, it seems that if I specify the 'b' parameter in the dial string before the number, sometimes the early B3 isn't early enough or not there at all. From the limited tests that I did last night, it would appear that it seems to depend on which carrier I use to make the call on my BRI line (I am based in the UK, and have a BT ISDN2e line). If I use my CPS provider to make the call, I get full early B3 including the ringing tone passed through from the exchange. If I route the call through BT by using the 1280 prefix, I do not get ringing tone at all and only get the sound through when either (a) a recorded anouncement is played or (b) the call is answered. What is more strange is that early B3 has been flawless whilst using Asterisk 1.0.x (currently 1.0.9) and chan_capi-0.3.5 (that is, it does the right thing in all cases.) So I'm making the assumption here that it probably isn't the fault of the telephone companies. Has anyone else come across this? Is it a bug in chan_capi? TIA Gary H -- Gary Hawkins MBCS [EMAIL PROTECTED] PGP: 0x6355BF46 (expires 31 Dec 2005) Web: http://www.garyhawkins.me.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
