currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet. here si the sip debug msg, we got a Message type: DISCONNECT (69) and unable to dial any numbers. Jul 12 12:56:26 WARNING[1523]: chan_zap.c:1931 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! -- Making new call for cr 32771> Protocol Discriminator: Q.931 (8) len=47 > Call Ref: len= 2 (reference 3/0x3) (Originator) > Message type: SETUP (5) > [04 03 80 90 a3] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) > Ext: 1 User information layer 1: A-Law (35) > [18 04 e9 81 83 81] > Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 > ChanSel: Reserved > Ext: 1 DS1 Identifier: 1 > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 1 ] > [28 08 4a 69 61 6e 20 4c 69 75] > Display (len= 8) [ Jian Liu ] > [6c 04 21 81 31 30] > Calling Number (len= 6) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) > Presentation: Presentation permitted, user number passed network screening (1) '10' ] > [70 0d a1 30 31 33 39 30 31 30 33 35 34 33 36] > Called Number (len=15) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '013901035436' ]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending> Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 3/0x3) (Originator) > Message type: DISCONNECT (69) > [08 02 81 90] > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) > Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request, peerstate Disconnect Indication -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050712/96cbdbd2/attachment.htm
On Tue, Jul 12, 2005 at 10:59:42PM +0800, matt001 wrote:> currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet. > > here si the sip debug msg, we got a Message type: DISCONNECT (69) and unable to dial any numbers. > > Jul 12 12:56:26 WARNING[1523]: chan_zap.c:1931 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! > -- Making new call for cr 32771 > > Protocol Discriminator: Q.931 (8) len=47 > > Call Ref: len= 2 (reference 3/0x3) (Originator) > > Message type: SETUP (5) > > [04 03 80 90 a3] > > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) > > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) > > Ext: 1 User information layer 1: A-Law (35) > > [18 04 e9 81 83 81] > > Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 > > ChanSel: Reserved > > Ext: 1 DS1 Identifier: 1 > > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > > Ext: 1 Channel: 1 ] > > [28 08 4a 69 61 6e 20 4c 69 75] > > Display (len= 8) [ Jian Liu ] > > [6c 04 21 81 31 30] > > Calling Number (len= 6) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) > > Presentation: Presentation permitted, user number passed network screening (1) '10' ] > > [70 0d a1 30 31 33 39 30 31 30 33 35 34 33 36] > > Called Number (len=15) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '013901035436' ] > NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending > > Protocol Discriminator: Q.931 (8) len=9 > > Call Ref: len= 2 (reference 3/0x3) (Originator) > > Message type: DISCONNECT (69) > > [08 02 81 90] > > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) > > Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] > NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request, peerstate Disconnect IndicationHave you played with the overlapdial settings any to see if that helps? -- Matthew Fredrickson
We have an Asterisk server with a single Digium E1. Everzthign works as it should except for one minor issue. When we place a call to a number that is busy, Asterisk does not seem to properly send the busy signal back to the SIP phones. There is no indication on the phone of anything at all, just silence, like the call did not go through. As you might imagine, this can be quite frustrating. The only indication is that we see a 403 Forbidden SIP message on softphones. I would appreciate any ideas of how to solve this issue. I have yet to do extensive PRI debugging to see what the Telecom provider sends back, so I am assuming that it correct signaling. Regards, Joe
Looks like you got a configuration issue, you should test for the
${DIALSTATUS} variable and set the signalling to the phones based on
that.
You can do:
exten => _X.,1,Dial(Zap/g1/${EXTEN})
exten => _X.,2,Goto,s-${DIALSTATUS},1)
exten => s-CANCEL,1,Playtones(congestion)
exten => s-CANCEL,2,Congestion
exten => s-NOANSWER,1,Goto(s-CANCEL,1)
exten => s-BUSY,1,Playtones(busy)
exten => s-BUSY,2,Busy
exten => s-CONGESTION,1,Goto(s-CANCEL,1)
exten => s-CHANUNAVAIL,1,Goto(s-CANCEL,1)
Check this:
http://www.voip-info.org/wiki-asterisk+cmd+dial
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+DIALSTATUS
On 1/6/06, Joseph Rothstein <jrothstein@comcentrixs.com>
wrote:> We have an Asterisk server with a single Digium E1. Everzthign works as it
> should except for one minor issue.
>
> When we place a call to a number that is busy, Asterisk does not seem to
> properly send the busy signal back to the SIP phones. There is no
indication
> on the phone of anything at all, just silence, like the call did not go
> through. As you might imagine, this can be quite frustrating. The only
> indication is that we see a 403 Forbidden SIP message on softphones.
>
> I would appreciate any ideas of how to solve this issue. I have yet to do
> extensive PRI debugging to see what the Telecom provider sends back, so I
am
> assuming that it correct signaling.
>
> Regards,
> Joe
>
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Thanks for the suggestion, but I can't seem to get this to work for some
reason.
When I dial my zap channel it does not seem to go beyond the first priority.
I have setup the following just as a test, but never see the output of Noop:
exten => _0.,1,Dial(Zap/g1/${EXTEN:2},,f)
exten => _0.,2,Noop(${DIALSTATUS})
I'd appreciate any suggestions along these lines.
Regards,
Joe
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On Sunday 08 January 2006 05:48, Joseph Rothstein wrote:> When I dial my zap channel it does not seem to go beyond the first > priority.> exten => _0.,1,Dial(Zap/g1/${EXTEN:2},,f) > exten => _0.,2,Noop(${DIALSTATUS})No application continues upon hangup unless there are special conditions which permit this. The Dial() option "g" does this. -A.
If you mean the call keeps ringing, then the reason why is you have no
timeout in your dial statement. When there is no timeout the system
will never give up on line one so it can continue to the second
priority.
Thanks,
Steve
_____
From: Joseph Rothstein [mailto:jrothstein@comcentrixs.com]
Sent: Sunday, January 08, 2006 5:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] PRI problem
Thanks for the suggestion, but I can't seem to get this to work for some
reason.
When I dial my zap channel it does not seem to go beyond the first
priority.
I have setup the following just as a test, but never see the output of
Noop:
exten => _0.,1,Dial(Zap/g1/${EXTEN:2},,f)
exten => _0.,2,Noop(${DIALSTATUS})
I'd appreciate any suggestions along these lines.
Regards,
Joe