Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] --> channel:0 mode:TE cause:0 ocause:0 rad: cad: The request from the telephone company's switch seems correct, a SETUP message (if 08 is Q.931, 05 is SETUP). 02 ff 03 08 01 04 05 a1 04 03 80 90 a3 18 01 80 6c 0b 01 83 39 31 36 33 39 31 37 34 32 70 03 c1 38 34 I've changed misdn.conf so it sends a release cause as 17 (user busy), but I get the same behaviour - cause:0 ocause:0. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo Santos misdn.conf: http://pastebin.com/FmgECqkU misdn debug: http://pastebin.com/Tg6wPKBD [1] http://www.mail-archive.com/asterisk-users at lists.digium.com/msg244330.html
I'm resending this email to the list, apparently the first one didn't go through. If it did, I apologize for the re-post. Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] --> channel:0 mode:TE cause:0 ocause:0 rad: cad: The request from the telephone company's switch seems correct, a SETUP message (if 08 is Q.931, 05 is SETUP). 02 ff 03 08 01 04 05 a1 04 03 80 90 a3 18 01 80 6c 0b 01 83 39 31 36 33 39 31 37 34 32 70 03 c1 38 34 I've changed misdn.conf so it sends a release cause as 17 (user busy), but I get the same behaviour - cause:0 ocause:0. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo Santos misdn.conf: http://pastebin.com/FmgECqkU misdn debug: http://pastebin.com/Tg6wPKBD [1] http://www.mail-archive.com/asterisk-users at lists.digium.com/msg244330.html
Still I am also facing the call disconnection when there is a third call. I am using Netmod BRI router and the output of the BRI router lines are connected to FXO ports in Asterisk. Where in Asterisk I am facing the call disconnection when there is a third call.. On Tue, Sep 28, 2010 at 4:22 PM, Paulo Santos <paulo.r.santos at sapo.pt>wrote:> Hello, > > Following my first mail about this issue [1], I think I know now what > the problem is. > > When I have both lines being used and a third call comes in, the person > calling doesn't get a busy tone, he gets something like line unavailable. > > I've been debugging mISDN and I think the reason is because asterisk is > sending the release cause as 0. > > P[ 3] --> channel:0 mode:TE cause:0 ocause:0 rad: cad: > > The request from the telephone company's switch seems correct, a SETUP > message (if 08 is Q.931, 05 is SETUP). > > 02 ff 03 08 01 04 05 a1 04 03 80 90 > a3 18 01 80 6c 0b 01 83 39 31 36 33 > 39 31 37 34 32 70 03 c1 38 34 > > I've changed misdn.conf so it sends a release cause as 17 (user busy), > but I get the same behaviour - cause:0 ocause:0. > > Anyone knows how can I force asterisk to send cause 16 or 17 in this > situation? > > Thanks in advance. > > Best regards, > Paulo Santos > > misdn.conf: http://pastebin.com/FmgECqkU > misdn debug: http://pastebin.com/Tg6wPKBD > > [1] > http://www.mail-archive.com/asterisk-users at lists.digium.com/msg244330.html > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Thank you with regards, Gopalakrishnan A.N, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101005/92bd2adc/attachment.htm
Paulo Santos wrote:> Hello, > > Following my first mail about this issue [1], I think I know now what > the problem is. > > When I have both lines being used and a third call comes in, the person > calling doesn't get a busy tone, he gets something like line unavailable. > > I've been debugging mISDN and I think the reason is because asterisk is > sending the release cause as 0. > > P[ 3] --> channel:0 mode:TE cause:0 ocause:0 rad: cad: > > The request from the telephone company's switch seems correct, a SETUP > message (if 08 is Q.931, 05 is SETUP). > > 02 ff 03 08 01 04 05 a1 04 03 80 90 > a3 18 01 80 6c 0b 01 83 39 31 36 33 > 39 31 37 34 32 70 03 c1 38 34 > > I've changed misdn.conf so it sends a release cause as 17 (user busy), > but I get the same behaviour - cause:0 ocause:0. > > Anyone knows how can I force asterisk to send cause 16 or 17 in this > situation? > > Thanks in advance. > > Best regards, > Paulo Santos > > misdn.conf: http://pastebin.com/FmgECqkU > misdn debug: http://pastebin.com/Tg6wPKBD > > [1] > http://www.mail-archive.com/asterisk-users at lists.digium.com/msg244330.html >Ok, I've encountered a similar issue on a different installation but instead of being PTP it's PTMP. Plus, it's a setup with 2 BRI lines with call forwarding between them - main number of BRI1 forwards to secondary number of BRI2 when busy/unavailable and vice-versa. I've called the phone company and confirmed that call waiting is disabled, yet I get a message in misdn debug saying: P[ 2] --> Call Waiting on PMP sending RELEASE_COMPLETE I don't know if this is the actual call waiting feature or if it is just an information of some kind. In the misdn debug I get this: http://pastebin.com/D7wv0qqm The P[ 2] is the port of the BRI line I called in the first place, then it is forwarded to P[ 1] where I get an error: P[ 1] Decoding FACILITY failed! (-1) And the same issue I said in the previews email: P[ 1] --> channel:0 mode:TE cause:0 ocause:0 rad: cad: I changed isdn_lib.c and now I'm sending ocause:17 (user busy). I've done this in the PTP line mentioned in the previews email as well. For the PTP line it appears to have worked, I have the regular busy signal. It worked only after the first time I tried to place a 3rd call. Now the 3rd call doesn't even reach Asterisk, which was what I wanted from the phone company in the first place. On the PTMP line it didn't work, I still don't get the busy signal. Maybe cause 17 isn't the right one? And what can be that "FACILITY" mentioned in the debug? Thanks in advance. Best regards, Paulo Santos
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