RE: [Asterisk-Users] Problems with PRI with T410 messages

2004-06-17 Thread CW_ASN
> 
> 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



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RE: [Asterisk-Users] Problems with PRI with T410 messages

2004-06-17 Thread Robinson Tim-W10277
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]

Re: [Asterisk-Users] Problems with PRI with T410 messages

2004-06-17 Thread CW_ASN
>
> 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



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Re: [Asterisk-Users] Problems with PRI with T410 messages

2004-06-17 Thread CW_ASN
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
>
> 
> 
>  charset=3Diso-8859-1">
> 
> 
> 
> 
> Hi,
> Is there a way to accept SIP calls from =
>
> unregistered gateways?
> autocreatpeer=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).
> I 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.  Is such =
> thing=20
> possible with Asterisk?
>  
> Best regards,
>  
> Axel
>
> --=_NextPart_000_0351_01C4541D.36B45830--
>
>
>
> --__--__--
>
> Message: 3
> Date: Thu, 17 Jun 2004 09:43:37 +0200 (CEST)
> From: Peter Svensson <[EMAIL P

RE: [Asterisk-Users] Problems with PRI with T410 messages

2004-06-17 Thread Robinson Tim-W10277

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

   Hi, Is there a way to accept SIP calls from =

unregistered gateways?
autocreatpeer=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).
I 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.  Is such = thing=20 
possible with Asterisk?   Best 
regards,   Axel

--=_NextPart_000_0351_01C4541D.36B45830--



--__--__--

Message: 3
Date: Thu, 17 Jun 2004 09:43:37 +0200 (C