Re: [asterisk-users] Overlap SIP dialing
On 09/07/2011 11:06 AM, Daniel Tryba wrote:
The aim of the quest for overlap dialing is to let the user enter a
number at their own pace but immediatly dial when all digits are
received (just like plain old ISDN does). My trunk is a bunch of E1 PRIs
in overlap mode. The following just works for any SIP client (without
overlap dialing):
exten = _X.,1,Answer()
exten = _X.,n,Dial(${TRUNK})
Unless I'm mis-remembering, this was the point of adding the '!'
dialplan match character. If you use _X!, and you have your SIP
endpoints configured to send an INVITE as soon as the user has entered
two digits (and you have no other patterns in the context that could
match), then the dialplan will match against that and initiate a Dial()
on your ISDN PRI. Since the number is not yet complete, the SETUP
message on the PRI won't result in the call proceeding, and as the user
of the phone presses additional digits they'll be sent to Asterisk as
DTMF, bridged over to chan_dahdi, and it will send them as INFORMATION
messages rather than as DTMF digits, because it knows the outbound call
is still in 'dialing' state.
However, this is still going to 'mess with CDRs' as you put it, because
the only switch in the network that knows the complete number that was
dialed is the PSTN switch that your PRI is connected to. It seems
possible that chan_dahdi could 'update' the EXTEN on the current channel
as the additional digits are dialed so that the CDR contains the
complete number, but I have no idea whether it does or not.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] Overlap SIP dialing
On Thu, Sep 8, 2011 at 9:38 AM, Kevin P. Fleming kpflem...@digium.com wrote:
On 09/07/2011 11:06 AM, Daniel Tryba wrote:
The aim of the quest for overlap dialing is to let the user enter a
number at their own pace but immediatly dial when all digits are
received (just like plain old ISDN does). My trunk is a bunch of E1 PRIs
in overlap mode. The following just works for any SIP client (without
overlap dialing):
exten = _X.,1,Answer()
exten = _X.,n,Dial(${TRUNK})
Unless I'm mis-remembering, this was the point of adding the '!' dialplan
match character. If you use _X!, and you have your SIP endpoints configured
to send an INVITE as soon as the user has entered two digits (and you have
no other patterns in the context that could match), then the dialplan will
match against that and initiate a Dial() on your ISDN PRI. Since the number
is not yet complete, the SETUP message on the PRI won't result in the call
proceeding, and as the user of the phone presses additional digits they'll
be sent to Asterisk as DTMF, bridged over to chan_dahdi, and it will send
them as INFORMATION messages rather than as DTMF digits, because it knows
the outbound call is still in 'dialing' state.
However, this is still going to 'mess with CDRs' as you put it, because the
only switch in the network that knows the complete number that was dialed is
the PSTN switch that your PRI is connected to. It seems possible that
chan_dahdi could 'update' the EXTEN on the current channel as the additional
digits are dialed so that the CDR contains the complete number, but I have
no idea whether it does or not.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
Exactly Kevin. I remember now that I was using it for my
http://etel.wiki.oreilly.com/wiki/index.php/Asterisk_Man_in_the_Middle
in some setup/testing.
--
~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~
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Re: [asterisk-users] Overlap SIP dialing
On Thu, Sep 08, 2011 at 08:38:39AM -0500, Kevin P. Fleming wrote:
The following just works for any SIP client (without
overlap dialing):
exten = _X.,1,Answer()
exten = _X.,n,Dial(${TRUNK})
Unless I'm mis-remembering, this was the point of adding the '!'
dialplan match character. If you use _X!, and you have your SIP
endpoints configured to send an INVITE as soon as the user has entered
two digits (and you have no other patterns in the context that could
match), then the dialplan will match against that and initiate a Dial()
on your ISDN PRI. Since the number is not yet complete, the SETUP
message on the PRI won't result in the call proceeding, and as the user
of the phone presses additional digits they'll be sent to Asterisk as
DTMF, bridged over to chan_dahdi, and it will send them as INFORMATION
messages rather than as DTMF digits, because it knows the outbound call
is still in 'dialing' state.
That is what I read, but I can't find any working examples. With the
following context:
[overlap]
exten = _X!,1,Dial(${TRUNK)
This happens:
Caller hits a number, which is immediatly send with an INVITE. SIP
endpoint receives a TRYING. A DAHDI channel is opened.
SIP endpoint locks up, no further input iws accepted.
Nothing happens for 10s.
SIP endpoint received a TRYING.
Nothing happens for 40s.
DAHDI fails, SIP endpoint received a 503 Service Unavailable:
X-Asterisk-HangupCause: Mandatory information element is missing
X-Asterisk-HangupCauseCode: 96
SIP = SIP/SDP Request: INVITE sip:0...@ouzo.pocos.nl, with session description
SIP = SIP Status: 100 Trying
DAHDI= Message Type: SETUP (5)
DAHDI= Calling Number
DAHDI= Presentation: Presentation permitted, user number not screened (0)
'' ]
DAHDI= Called Number
DAHDI= Message Type: STATUS (125)
DAHDI= Cause
DAHDI= Ext: 1 Cause: Invalid information element contents (100), class =
Protocol Error (e.g. unknown message) (6) ]
DAHDI= Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call
state: Call Initiated (1)
DAHDI= Message Type: SETUP ACKNOWLEDGE (13)
DAHDI= Channel ID
DAHDI= ChanSel: As indicated in following octets
DAHDI= Ext: 1 Coding: 0 Number Specified Channel Type: 3
DAHDI= Ext: 1 Channel: 1 Type: CPE]
DAHDI= Progress Indicator
DAHDI= Ext: 1 Progress Description: Inband information or appropriate
pattern now available. (8) ]
q931.c:7194 post_handle_q931_message: Call 32806 enters state 2 (Overlap
Sending). Hold state: Idle
DAHDI= Message Type: CALL PROCEEDING (2)
SIP = SIP Status: 100 Trying
DAHDI= Message Type: DISCONNECT (69)
DAHDI= Message Type: RELEASE (77)
DAHDI= Cause
DAHDI= Ext: 1 Cause: Mandatory information element is missing (96), class =
Protocol Error (e.g. unknown message) (6) ]
DAHDI= Message Type: RELEASE COMPLETE (90)
DAHDI= Cause
DAHDI= Ext: 1 Cause: Mandatory information element is missing (96), class =
Protocol Error (e.g. unknown message) (6) ]
SIP = SIP Status: 503 Service Unavailable
SIP = SIP Request: ACK sip:0...@ouzo.pocos.nl
Notice the Call 32806 enters state 2 (Overlap Sending). This is not
propagated to the SIP endpoint.
However, this is still going to 'mess with CDRs' as you put it, because
the only switch in the network that knows the complete number that was
dialed is the PSTN switch that your PRI is connected to. It seems
possible that chan_dahdi could 'update' the EXTEN on the current channel
as the additional digits are dialed so that the CDR contains the
complete number, but I have no idea whether it does or not.
Didn't think about this CDR problem (was only concerned about billsec).
As far as I can see this doesn't happen in 1.6.2.x. The lack of
destination in CDR makes overlap dialing useless since I can't bill my
customers.
--
Daniel Tryba
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Re: [asterisk-users] Overlap SIP dialing
On 09/08/2011 10:04 AM, Daniel Tryba wrote:
On Thu, Sep 08, 2011 at 08:38:39AM -0500, Kevin P. Fleming wrote:
The following just works for any SIP client (without
overlap dialing):
exten = _X.,1,Answer()
exten = _X.,n,Dial(${TRUNK})
Unless I'm mis-remembering, this was the point of adding the '!'
dialplan match character. If you use _X!, and you have your SIP
endpoints configured to send an INVITE as soon as the user has entered
two digits (and you have no other patterns in the context that could
match), then the dialplan will match against that and initiate a Dial()
on your ISDN PRI. Since the number is not yet complete, the SETUP
message on the PRI won't result in the call proceeding, and as the user
of the phone presses additional digits they'll be sent to Asterisk as
DTMF, bridged over to chan_dahdi, and it will send them as INFORMATION
messages rather than as DTMF digits, because it knows the outbound call
is still in 'dialing' state.
That is what I read, but I can't find any working examples. With the
following context:
[overlap]
exten = _X!,1,Dial(${TRUNK)
This happens:
Caller hits a number, which is immediatly send with an INVITE. SIP
endpoint receives a TRYING. A DAHDI channel is opened.
SIP endpoint locks up, no further input iws accepted.
Nothing happens for 10s.
SIP endpoint received a TRYING.
Nothing happens for 40s.
DAHDI fails, SIP endpoint received a 503 Service Unavailable:
X-Asterisk-HangupCause: Mandatory information element is missing
X-Asterisk-HangupCauseCode: 96
SIP = SIP/SDP Request: INVITE sip:0...@ouzo.pocos.nl, with session
description
SIP= SIP Status: 100 Trying
DAHDI= Message Type: SETUP (5)
DAHDI= Calling Number
DAHDI=Presentation: Presentation permitted, user number not screened (0)
'' ]
DAHDI= Called Number
DAHDI= Message Type: STATUS (125)
DAHDI= Cause
DAHDI= Ext: 1 Cause: Invalid information element contents (100), class =
Protocol Error (e.g. unknown message) (6) ]
DAHDI= Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call
state: Call Initiated (1)
DAHDI= Message Type: SETUP ACKNOWLEDGE (13)
DAHDI= Channel ID
DAHDI= ChanSel: As indicated in following octets
DAHDI= Ext: 1 Coding: 0 Number Specified Channel Type: 3
DAHDI= Ext: 1 Channel: 1 Type: CPE]
DAHDI= Progress Indicator
DAHDI= Ext: 1 Progress Description: Inband information or appropriate
pattern now available. (8) ]
q931.c:7194 post_handle_q931_message: Call 32806 enters state 2 (Overlap
Sending). Hold state: Idle
DAHDI= Message Type: CALL PROCEEDING (2)
SIP= SIP Status: 100 Trying
DAHDI= Message Type: DISCONNECT (69)
DAHDI= Message Type: RELEASE (77)
DAHDI= Cause
DAHDI= Ext: 1 Cause: Mandatory information element is missing (96), class =
Protocol Error (e.g. unknown message) (6) ]
DAHDI= Message Type: RELEASE COMPLETE (90)
DAHDI= Cause
DAHDI=Ext: 1 Cause: Mandatory information element is missing (96), class
= Protocol Error (e.g. unknown message) (6) ]
SIP= SIP Status: 503 Service Unavailable
SIP = SIP Request: ACK sip:0...@ouzo.pocos.nl
Notice the Call 32806 enters state 2 (Overlap Sending). This is not
propagated to the SIP endpoint.
OK, yes, I can see how this would occur. Explicit Answer() (or even
Progress()) before Dial() would resolve that problem, but makes the CDR
situation even worse.
Honestly, I'm not really sure that there is a practical solution here.
ISDN overlap dialing was intended for 'dumb' phones, and SIP phones
aren't 'dumb' :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
--
_
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Re: [asterisk-users] Overlap SIP dialing
8 sep 2011 kl. 17:17 skrev Kevin P. Fleming: Honestly, I'm not really sure that there is a practical solution here. ISDN overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb' :-) That's a quote that goes to my quote storage layer. /O ;-) -- _ -- 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
Re: [asterisk-users] Overlap SIP dialing
On Thu, Sep 8, 2011 at 11:21 AM, Olle E. Johansson o...@edvina.net wrote: 8 sep 2011 kl. 17:17 skrev Kevin P. Fleming: Honestly, I'm not really sure that there is a practical solution here. ISDN overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb' :-) That's a quote that goes to my quote storage layer. /O ;-) -- I want a t-shirt SIP phones aren't 'dumb' :-) Overlap dialing has very limited use, however I found it helpful when testing integration with other PBX/VM/PSTN connections. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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
Re: [asterisk-users] Overlap SIP dialing
8 sep 2011 kl. 17:26 skrev Andrew Latham: On Thu, Sep 8, 2011 at 11:21 AM, Olle E. Johansson o...@edvina.net wrote: 8 sep 2011 kl. 17:17 skrev Kevin P. Fleming: Honestly, I'm not really sure that there is a practical solution here. ISDN overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb' :-) That's a quote that goes to my quote storage layer. /O ;-) -- I want a t-shirt SIP phones aren't 'dumb' :-) Overlap dialing has very limited use, however I found it helpful when testing integration with other PBX/VM/PSTN connections. Yes, but the solution is not 484, but as Kevin stated to answer the darn call on the SIP side and provide a dial tone from the other side. And yes, we do have dial tones on ISDN PRI trunks... /O -- _ -- 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
Re: [asterisk-users] Overlap SIP dialing
On Thu, Sep 08, 2011 at 11:26:45AM -0400, Andrew Latham wrote: I want a t-shirt SIP phones aren't 'dumb' :-) Overlap dialing has very limited use, however I found it helpful when testing integration with other PBX/VM/PSTN connections. My quest for overlap dialing started with the request to create a transparant SIP connection to a ISDN BRI PBX without adding unnecessary waiting for timeouts on the side of the SIP gateway to get all digits to send them en-block to Asterisk. My feedback for this request to the customer will be: -just use en-block dialing or -terminate the dialed number with # when not using en-block dialing And make sure to strip # from en-block numbers :) But I'm willing to give up now I see that this will not work with billing anyway. Thanks for the feedback and hopefully other people wanting to do overlap dialing will find this thread and may take it into account. -- Daniel Tryba -- _ -- 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
[asterisk-users] Overlap SIP dialing
Looking at the history of the list I don't expect any answer but lets try anyway: Does anybody use overlap dialing from SIP devices to asterisk? Does anybody have a working example? -- Daniel Tryba -- _ -- 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
Re: [asterisk-users] Overlap SIP dialing
7 sep 2011 kl. 15:59 skrev Daniel Tryba: Looking at the history of the list I don't expect any answer but lets try anyway: Does anybody use overlap dialing from SIP devices to asterisk? Does anybody have a working example? To add to your question: Does anyone have a phone that supports this properly? /O -- _ -- 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
Re: [asterisk-users] Overlap SIP dialing
On Wednesday, September 7, 2011, Olle E. Johansson wrote: 7 sep 2011 kl. 15:59 skrev Daniel Tryba: Looking at the history of the list I don't expect any answer but lets try anyway: Does anybody use overlap dialing from SIP devices to asterisk? Does anybody have a working example? To add to your question: Does anyone have a phone that supports this properly? /O Yup, I have a few... http://wiki.snom.com/Settings/overlap_dialing -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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
Re: [asterisk-users] Overlap SIP dialing
7 sep 2011 kl. 16:20 skrev Andrew Latham: On Wednesday, September 7, 2011, Olle E. Johansson wrote: 7 sep 2011 kl. 15:59 skrev Daniel Tryba: Looking at the history of the list I don't expect any answer but lets try anyway: Does anybody use overlap dialing from SIP devices to asterisk? Does anybody have a working example? To add to your question: Does anyone have a phone that supports this properly? /O Yup, I have a few... http://wiki.snom.com/Settings/overlap_dialing Great. Haven't seen this - thank you. The whole concept is interesting. Suppose the call forks and one UA answers with 484, another with 486 and another with 180 ringing. What are you supposed to do? I think there's a problem with the RFC 3261 here and don't know if it's been clarified. Now - in the case of Asterisk if we call out to two devices from the dialplan and one responds with 484 and another with 180 ringing - what happens in Asterisk? /O -- _ -- 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
Re: [asterisk-users] Overlap SIP dialing
On Wed, Sep 7, 2011 at 10:28 AM, Olle E. Johansson o...@edvina.net wrote: 7 sep 2011 kl. 16:20 skrev Andrew Latham: On Wednesday, September 7, 2011, Olle E. Johansson wrote: 7 sep 2011 kl. 15:59 skrev Daniel Tryba: Looking at the history of the list I don't expect any answer but lets try anyway: Does anybody use overlap dialing from SIP devices to asterisk? Does anybody have a working example? To add to your question: Does anyone have a phone that supports this properly? /O Yup, I have a few... http://wiki.snom.com/Settings/overlap_dialing Great. Haven't seen this - thank you. The whole concept is interesting. Suppose the call forks and one UA answers with 484, another with 486 and another with 180 ringing. What are you supposed to do? I think there's a problem with the RFC 3261 here and don't know if it's been clarified. Now - in the case of Asterisk if we call out to two devices from the dialplan and one responds with 484 and another with 180 ringing - what happens in Asterisk? /O In the past (2004/2005) I have dealt with this and hoot and holler* systems... * http://en.wikipedia.org/wiki/Hoot-n-holler -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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
Re: [asterisk-users] Overlap SIP dialing
On Wed, Sep 07, 2011 at 10:20:40AM -0400, Andrew Latham wrote:
To add to your question: Does anyone have a phone that supports this
properly?
Yup, I have a few... http://wiki.snom.com/Settings/overlap_dialing
I'm using a Grandstream (GXP2000) to test.
What I got so far:
Overlap works IF you know *ALL* patterns available. Which is ofcourse
impossible to get into a dialplan.
Example (partial Dutch dialplan):
exten = _11X,1,Dial(${TRUNK}/${EXTEN})
exten = _1[46]XXX,1,Dial(${TRUNK}/${EXTEN})
exten = _1[46],1,Dial(${TRUNK}/${EXTEN})
exten = _0[1-8],1,Dial(${TRUNK}/${EXTEN})
exten = _0[89]0X,1,Dial(${TRUNK}/${EXTEN})
exten = _0[89]0,1,Dial(${TRUNK}/${EXTEN})
Dialing a normal 10 digit number works with overlap. When the 10th
digit is touched the destination rings immediate.
But calling service numbers is impossible, these numbers exists in 8 and
11 digit lengths, eg. 08001234 08001234567. Calling the 11 digit variant
is impossible since as soon as the 8th digit is entered the Asterisk
dials the 8 digit number.
= SIP Request: ACK sip:080...@ouzo.pocos.nl
= SIP/SDP Request: INVITE sip:0800...@ouzo.pocos.nl, with session description
= SIP Status: 484 Address Incomplete
= SIP Request: ACK sip:0800...@ouzo.pocos.nl
= SIP/SDP Request: INVITE sip:08000...@ouzo.pocos.nl, with session description
= SIP Status: 100 Trying
= SIP Status: 100 Trying
= SIP Status: 180 Ringing
= SIP/SDP Status: 200 OK, with session description
For internation calls this gets worse. For example Germany uses variable
length domestic numbers.
The possible (but hopefully avoidable) workarounds are:
-DISA
-Answering the call, Dial(${TRUNK}) and let the user enter the destination
by DTMF
The aim of the quest for overlap dialing is to let the user enter a
number at their own pace but immediatly dial when all digits are
received (just like plain old ISDN does). My trunk is a bunch of E1 PRIs
in overlap mode. The following just works for any SIP client (without
overlap dialing):
exten = _X.,1,Answer()
exten = _X.,n,Dial(${TRUNK})
But this affects CDRs.
--
Daniel Tryba
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