RE: [Asterisk-Users] Problems with PRI with T410 messages
This is a problem I pointed out to Digium a while back, but I am not sure Markster understood the issue and I didn't really have the time to follow it up. It does need fixing though, as it is a major drawback in the current architecture. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aimable Sent: 17 June 2004 10:29 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problems with PRI with T410 messages Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Phone Services TERRACOM Broadband [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 10:56 AM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Soekris Engineering net4801 (Senad Jordanovic) 2. Accepting SIP calls from unregistered gateways (Axel) 3. Re: pri with TE410P not working (Austria) (Peter Svensson) 4. Re: ZAPHFC - only for * 0.7.2? (Holger Schurig) 5. Calling the firefly network? (Martijn van Oosterhout) 6. RE: IAX2 no compatible codecs (Jason Penton) 7. Re: pri with TE410P not working (Austria) (Wolfgang Pichler) 8. Re: embedded Asterisk (Klaus-Peter Junghanns) 9. Re: pri with TE410P not working (Austria) (Michael Bielicki) 10. RE: Cost of IP Phones, or Isn't It Just Software? (Andy Powell) 11. Re: pri with TE410P not working (Austria) (Peter Svensson) --__--__-- Message: 1 From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Soekris Engineering net4801 Date: Thu, 17 Jun 2004 08:34:01 +0100 Reply-To: [EMAIL PROTECTED] John Bittner wrote: Hi, I have it working great. I have debian running on it with music on hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with calls on all 10 phones at the same time through voicepulse with no issues. I ran top with all the phones running and I was only up to 45% cpu. Seems to run ok but I am still in the testing phase. Great... Have you tried to connect a X100P or TDM400P to it? --__--__-- Message: 2 From: Axel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 17 Jun 2004 03:43:12 -0400 Subject: [Asterisk-Users] Accepting SIP calls from unregistered gateways Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable Hi, Is there a way to accept SIP calls from unregistered gateways? autocreatpeer=3Dyes seems to disable checking credentials but the = originating gateway is still required to register itself with a username = and password (which can be anything since it won't check it). I like to be able to receive the call from any gateway without them = having to register even, just like a Cisco gateway that you can = terminate a call from clients who are not registered. Is such thing = possible with Asterisk? Best regards, Axel --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 4.0 Transitional//EN HTMLHEAD META http-equiv=3DContent-Type content=3Dtext/html; = charset=3Diso-8859-1 META content=3DMSHTML 6.00.2800.1400 name=3DGENERATOR STYLE/STYLE /HEAD BODY bgColor=3D#ff DIVFONT face=3DArial size=3D2Hi,/FONT/DIV DIVFONT face=3DArial size=3D2Is there a way to accept SIP calls from = unregistered gateways?/FONT/DIV DIVFONT face=3DArial size=3D2autocreatpeer=3Dyes seems to disable = checking=20 credentials but the originating gateway is still required to register = itself=20 with a username and password (which can be anything since it won't check = it)./FONT/DIV DIVFONT face=3DArial size=3D2I like to be able to
Re: [Asterisk-Users] Problems with PRI with T410 messages
Send traces. - Original Message - From: Aimable [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 17, 2004 6:28 AM Subject: [Asterisk-Users] Problems with PRI with T410 messages Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Phone Services TERRACOM Broadband [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 10:56 AM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Soekris Engineering net4801 (Senad Jordanovic) 2. Accepting SIP calls from unregistered gateways (Axel) 3. Re: pri with TE410P not working (Austria) (Peter Svensson) 4. Re: ZAPHFC - only for * 0.7.2? (Holger Schurig) 5. Calling the firefly network? (Martijn van Oosterhout) 6. RE: IAX2 no compatible codecs (Jason Penton) 7. Re: pri with TE410P not working (Austria) (Wolfgang Pichler) 8. Re: embedded Asterisk (Klaus-Peter Junghanns) 9. Re: pri with TE410P not working (Austria) (Michael Bielicki) 10. RE: Cost of IP Phones, or Isn't It Just Software? (Andy Powell) 11. Re: pri with TE410P not working (Austria) (Peter Svensson) --__--__-- Message: 1 From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Soekris Engineering net4801 Date: Thu, 17 Jun 2004 08:34:01 +0100 Reply-To: [EMAIL PROTECTED] John Bittner wrote: Hi, I have it working great. I have debian running on it with music on hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with calls on all 10 phones at the same time through voicepulse with no issues. I ran top with all the phones running and I was only up to 45% cpu. Seems to run ok but I am still in the testing phase. Great... Have you tried to connect a X100P or TDM400P to it? --__--__-- Message: 2 From: Axel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 17 Jun 2004 03:43:12 -0400 Subject: [Asterisk-Users] Accepting SIP calls from unregistered gateways Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable Hi, Is there a way to accept SIP calls from unregistered gateways? autocreatpeer=3Dyes seems to disable checking credentials but the = originating gateway is still required to register itself with a username = and password (which can be anything since it won't check it). I like to be able to receive the call from any gateway without them = having to register even, just like a Cisco gateway that you can = terminate a call from clients who are not registered. Is such thing = possible with Asterisk? Best regards, Axel --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 4.0 Transitional//EN HTMLHEAD META http-equiv=3DContent-Type content=3Dtext/html; = charset=3Diso-8859-1 META content=3DMSHTML 6.00.2800.1400 name=3DGENERATOR STYLE/STYLE /HEAD BODY bgColor=3D#ff DIVFONT face=3DArial size=3D2Hi,/FONT/DIV DIVFONT face=3DArial size=3D2Is there a way to accept SIP calls from = unregistered gateways?/FONT/DIV DIVFONT face=3DArial size=3D2autocreatpeer=3Dyes seems to disable = checking=20 credentials but the originating gateway is still required to register = itself=20 with a username and password (which can be anything since it won't check = it)./FONT/DIV DIVFONT face=3DArial size=3D2I like to be able to receive the call = from any=20 gateway without them having to register even, just like a Cisco gateway = that you=20 can terminate a call from clients who are not registered.nbsp; Is such = thing=20 possible with Asterisk?/FONT/DIV
Re: [Asterisk-Users] Problems with PRI with T410 messages
This is a problem I pointed out to Digium a while back, but I am not sure Markster understood the issue and I didn't really have the time to follow it up. It does need fixing though, as it is a major drawback in the current architecture. Rgds Tim Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Call Proceeding can be sent only by transit network, not by the local switch or pbx. AFAIK, * behavior for this scenario is like as local switch. Certainly, this is not a normal behavior. Regards, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with PRI with T410 messages
I do not believe you are correct. We see CALL PROCEEDING in both directions as part of the normal ISDN call setup process. See trace below. Asterisk sends 'CALL PROCEEDING' followed immediately by 'ALERTING'. CALL PROCEEDING is normally an acknowledgement to a SETUP. See Q931 below: 3.1.2 CALL PROCEEDING This message is sent by the called user to the network or by the network to the calling user to indicate that requested call establishment has been initiated and no more call establishment information will be accepted. See Table 3-3. ALERTING has a very specific meaning: 3.2.1 ALERTING This message is sent by the called user to the network to indicate that called user alerting has been initiated. See Table 3 23. i.e. the channel to the called party has been established, and the phone at the other end is physically ringing or making some other indication that an incoming call is there to be answered. It is 'ALERTING' that is being sent in the wrong place, as Asterisk sends 'ALERTING' before the remote party (be it a SIP or IAX channel) is actually 'ringing'. Receipt of 'ALERTING' from the called party is the trigger for the calling party to be presented with 'ringback tone'. So to send a 'RELEASE' message with 'busy' after the caller has been told the phone is ringing is not a logical thing to do, and causes a lot of problems here. It needs fixing Rgds Tim Connected to Asterisk CVS-D2004.05.25.23.00.00-06/14/04-12:46:31 currently running on localhost (pid = 4875) mote UNIX connection Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 1 (reference 1/0x1) (Originator) Message type: SETUP (5) Sending Complete (len= 4) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10) Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] Calling Number (len=18) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Called Number (len= 5) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '14' ] -- Making new call for cr 1 -- Processing Q.931 Call Setup -- Processing IE 33 (Sending Complete) -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 30 (Progress Indicator) -- Processing IE 108 (Calling Party Number) -- Processing IE 112 (Called Party Number) Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 129/0x81) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 129/0x81) (Terminator) Message type: ALERTING (1) Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] -- Executing Wait(Zap/1-1, 2) in new stack -- Accepting call from '0044125679' to '14' on channel 1, span 1 -- Executing Goto(Zap/1-1, default|8714|1) in new stack -- Goto (default,8714,1) -- Executing SetMusicOnHold(Zap/1-1, default) in new stack -- Executing Answer(Zap/1-1, ) in new stack Protocol Discriminator: Q.931 (8) len=11 Call Ref: len= 1 (reference 129/0x81) (Terminator) Message type: CONNECT (7) Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing SayDigits(Zap/1-1, 0044125679) in new stack -- Playing 'digits/0' (language 'en') Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 1/0x1) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CW_ASN Sent: 17 June 2004 12:13 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with PRI with T410 messages This is a problem I pointed out to Digium a while back, but I am not sure Markster understood the issue and I didn't really have the time to follow it up. It does need fixing though, as it is a major drawback in the current architecture. Rgds Tim Hi all, I
RE: [Asterisk-Users] Problems with PRI with T410 messages
I do not believe you are correct. We see CALL PROCEEDING in both directions as part of the normal ISDN call setup process. See trace below. Asterisk sends 'CALL PROCEEDING' followed immediately by 'ALERTING'. CALL PROCEEDING is normally an acknowledgement to a SETUP. See Q931 below: 3.1.2 CALL PROCEEDING This message is sent by the called user to the network or by the network to the calling user to indicate that requested call establishment has been initiated and no more call establishment information will be accepted. See Table 3-3. ALERTING has a very specific meaning: 3.2.1 ALERTING This message is sent by the called user to the network to indicate that called user alerting has been initiated. See Table 3 23. i.e. the channel to the called party has been established, and the phone at the other end is physically ringing or making some other indication that an incoming call is there to be answered. It is 'ALERTING' that is being sent in the wrong place, as Asterisk sends 'ALERTING' before the remote party (be it a SIP or IAX channel) is actually 'ringing'. Receipt of 'ALERTING' from the called party is the trigger for the calling party to be presented with 'ringback tone'. So to send a 'RELEASE' message with 'busy' after the caller has been told the phone is ringing is not a logical thing to do, and causes a lot of problems here. It needs fixing Rgds Tim Tim: Call proceeding is not mandatory in local termination (at least in EuroISDN). Alerting is mandatory (obviously). Some class 5 switches sends Call Proceeding only when the received SETUP will be routed thru CCS or CAS routes, and only when a timer (I can't remember the timer number) expires. The Call Proceeding must be retransmitted to A side. Call Proceeding message is used mostly in transit environments. Obviously, Ringing can't be used when unallocated or busy conditions are detected. The correct procedure for successful call with Call Proceeding and Setup Acknowledge: 1) A-Setup 2) Setup acknowledge -B 3) Call Proceeding -B 4) Ringing -B 5) Answer -B Or 5) Release A-B (by expiration time) The correct procedure for unsuccessful (1 or 17 cause) call without Call Proceeding, with Setup Acknowledge: 1) A-Setup 2) Setup acknowledge -B 3) Release -B (ITU-T release cause i.e.: 1 or 17) As you said, it needs to be fixed. Regards, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
