Hello Freeswitch Developers, ISDN in germany (maybe also somewhere else) has a feature, which colides with the design of a dialplan: it sends the numbers the user types into his phone LIVE (!) through the ISDN network. If the number is complete, the ISDN network tells it to the caller; Only now the call get established (means the dialplan gets invoked). So you have to do an own step (live and interactive determination of the number) BEFORE the dialplan comes in line. Because this is object not supported by freeswith, in the following situation User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <-> Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation you have to do "overlap dialing". Means, freeswitch waits some seconds until the user has entered the last number and then it goes directly to the dialplan. -> This is not what I need! I want that it works as if there is no FreeSwitch in between. Means, freeswitch should relay each number the user types into his phone to the ISDN on the remote side, and make the dialplan stuff AFTER the number has been dicovered for completeness. How do you think this should be implemented? If nobody has an object, I would code this the following way into FreeTDM: If I get a call with a starting number I know that it must go to the external ISDN (means: a second dialplan), I first relay the typed numbers and determine the full number. Only now, when I got the full number, I would give this call to the higher layers. What you think about this? Regards, Markus Mueller
hi Markus, how do you determine if it's a complete number? Do you expect the outgoing ISDN channel to tell you that? I also wonder what happens if you attach an ISDN-SIP gateway, like Patton. Will you have a new SIP message (which?) on every dialed digit? ________________________________ From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Saturday, September 7, 2013 1:52 PM Subject: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN) Hello Freeswitch Developers, ISDN in germany (maybe also somewhere else) has a feature, which colides with the design of a dialplan: it sends the numbers the user types into his phone LIVE (!) through the ISDN network. If the number is complete, the ISDN network tells it to the caller; Only now the call get established (means the dialplan gets invoked). So you have to do an own step (live and interactive determination of the number) BEFORE the dialplan comes in line. Because this is object not supported by freeswith, in the following situation User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <-> Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation you have to do "overlap dialing". Means, freeswitch waits some seconds until the user has entered the last number and then it goes directly to the dialplan. -> This is not what I need! I want that it works as if there is no FreeSwitch in between. Means, freeswitch should relay each number the user types into his phone to the ISDN on the remote side, and make the dialplan stuff AFTER the number has been dicovered for completeness. How do you think this should be implemented? If nobody has an object, I would code this the following way into FreeTDM: If I get a call with a starting number I know that it must go to the external ISDN (means: a second dialplan), I first relay the typed numbers and determine the full number. Only now, when I got the full number, I would give this call to the higher layers. What you think about this? Regards, Markus Mueller _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Hi all,
hi Markus, how do you determine if it's a complete number? Do you expect the outgoing ISDN channel to tell you that? Sorry that I don't have all details; I am not as familary as I want to with the ISDN protocols. But I know that each number goes one by one to the outgoing ISDN and then outgoint ISDN tells you back when you have reached a destination. If needed I can make a log when I have my Siemens Hipath placed behind my Freeswitch in NT mode; there you can see the numbers comming one by one.
I also wonder what happens if you attach an ISDN-SIP gateway, like Patton. Will you have a new SIP message (which?) on every dialed digit? Normaly all the products do the "overlap dialing" crap, as mentioned in my first email. I don't know for sure but I think there is no alternative, especially if you make only SIP and not ISDN.
No comments about my design suggestion? Regards, Markus
------------------------------------------------------------------------ *From:* Markus Müller <freeswitchdev@priv.de> *To:* freeswitch-dev@lists.freeswitch.org *Sent:* Saturday, September 7, 2013 1:52 PM *Subject:* [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hello Freeswitch Developers,
ISDN in germany (maybe also somewhere else) has a feature, which colides with the design of a dialplan: it sends the numbers the user types into his phone LIVE (!) through the ISDN network. If the number is complete, the ISDN network tells it to the caller; Only now the call get established (means the dialplan gets invoked). So you have to do an own step (live and interactive determination of the number) BEFORE the dialplan comes in line.
Because this is object not supported by freeswith, in the following situation
User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <-> Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation
you have to do "overlap dialing". Means, freeswitch waits some seconds until the user has entered the last number and then it goes directly to the dialplan.
-> This is not what I need!
I want that it works as if there is no FreeSwitch in between. Means, freeswitch should relay each number the user types into his phone to the ISDN on the remote side, and make the dialplan stuff AFTER the number has been dicovered for completeness.
How do you think this should be implemented? If nobody has an object, I would code this the following way into FreeTDM: If I get a call with a starting number I know that it must go to the external ISDN (means: a second dialplan), I first relay the typed numbers and determine the full number. Only now, when I got the full number, I would give this call to the higher layers.
What you think about this?
Regards, Markus Mueller
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org <mailto:FreeSWITCH-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
I guess there's no such ting as "complete number", but a simple interdigit timeout. If you stop dialing and don't enter digits for 10 seconds, it automatically places the call. So, you need a gateway which supports such an interdigit timeout. In theory, you can implement that with an external script in FreeSWITCH. On practice, needs testing and a proof of concept. ________________________________ From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Saturday, September 7, 2013 10:08 PM Subject: Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN) Hi all, hi Markus,
how do you determine if it's a complete number? Do you expect the outgoing ISDN channel to tell you that?
Sorry that I don't have all details; I am not as familary as I want to with the ISDN protocols. But I know that each number goes one by one to the outgoing ISDN and then outgoint ISDN tells you back when you have reached a destination. If needed I can make a log when I have my Siemens Hipath placed behind my Freeswitch in NT mode; there you can see the numbers comming one by one.
I also wonder what happens if you attach an ISDN-SIP gateway, like Patton. Will you have a new SIP message (which?) on every dialed digit?
Normaly all the products do the "overlap dialing" crap, as mentioned in my first email. I don't know for sure but I think there is no alternative, especially if you make only SIP and not ISDN. No comments about my design suggestion? Regards, Markus
________________________________ From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Saturday, September 7, 2013 1:52 PM Subject: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hello Freeswitch Developers,
ISDN in germany (maybe also somewhere else) has a feature,
which colides
with the design of a dialplan: it sends the numbers the user types into his phone LIVE (!) through the ISDN network. If the number is complete, the ISDN network tells it to the caller; Only now the call get established (means the dialplan gets invoked). So you have to do an own step (live and interactive determination of the number) BEFORE the dialplan comes in line.
Because this is object not supported by freeswith, in the following situation
User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <-> Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation
you have to do "overlap dialing". Means, freeswitch waits some seconds until the user has entered the last number and then it goes directly to the dialplan.
-> This is not what I need!
I want that it works as if there is no FreeSwitch in between. Means, freeswitch should relay each number the user types into his phone to the ISDN on the remote side, and make the dialplan stuff AFTER the number has been dicovered for completeness.
How do you think this should be implemented? If nobody has an object, I would code this the following way into FreeTDM: If I get a call with a starting number I know that it must go to the external ISDN (means: a second dialplan), I first relay the typed numbers and determine the full number. Only now, when I got the full number, I would give this call to the higher layers.
What you think about this?
Regards, Markus Mueller
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Hi Stanislav Sinyagin, what you mean is the "overlap dialing" feature, this is already implemented and working in freetdm. It waits some seconds after the last number and then dials. ====>> This is not what I am searching for ! This is also not what I am talking about ! Please accept this and make sure how the reality is before saying that I am telling wrong things. I am searching what I worte, if you believe or not. ISDN in germany currently works this way, and yes, there is a "completeness" of numbers which is interactively determined by sending each key through the ISDN network and getting back when a destination has been found. Regards, Markus Mueller
I guess there's no such ting as "complete number", but a simple interdigit timeout. If you stop dialing and don't enter digits for 10 seconds, it automatically places the call.
So, you need a gateway which supports such an interdigit timeout. In theory, you can implement that with an external script in FreeSWITCH. On practice, needs testing and a proof of concept.
------------------------------------------------------------------------ *From:* Markus Müller <freeswitchdev@priv.de> *To:* freeswitch-dev@lists.freeswitch.org *Sent:* Saturday, September 7, 2013 10:08 PM *Subject:* Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hi all,
hi Markus, how do you determine if it's a complete number? Do you expect the outgoing ISDN channel to tell you that? Sorry that I don't have all details; I am not as familary as I want to with the ISDN protocols. But I know that each number goes one by one to the outgoing ISDN and then outgoint ISDN tells you back when you have reached a destination. If needed I can make a log when I have my Siemens Hipath placed behind my Freeswitch in NT mode; there you can see the numbers comming one by one.
I also wonder what happens if you attach an ISDN-SIP gateway, like Patton. Will you have a new SIP message (which?) on every dialed digit? Normaly all the products do the "overlap dialing" crap, as mentioned in my first email. I don't know for sure but I think there is no alternative, especially if you make only SIP and not ISDN.
No comments about my design suggestion?
Regards, Markus
------------------------------------------------------------------------ *From:* Markus Müller <freeswitchdev@priv.de> <mailto:freeswitchdev@priv.de> *To:* freeswitch-dev@lists.freeswitch.org <mailto:freeswitch-dev@lists.freeswitch.org> *Sent:* Saturday, September 7, 2013 1:52 PM *Subject:* [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hello Freeswitch Developers,
ISDN in germany (maybe also somewhere else) has a feature, which colides with the design of a dialplan: it sends the numbers the user types into his phone LIVE (!) through the ISDN network. If the number is complete, the ISDN network tells it to the caller; Only now the call get established (means the dialplan gets invoked). So you have to do an own step (live and interactive determination of the number) BEFORE the dialplan comes in line.
Because this is object not supported by freeswith, in the following situation
User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <-> Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation
you have to do "overlap dialing". Means, freeswitch waits some seconds until the user has entered the last number and then it goes directly to the dialplan.
-> This is not what I need!
I want that it works as if there is no FreeSwitch in between. Means, freeswitch should relay each number the user types into his phone to the ISDN on the remote side, and make the dialplan stuff AFTER the number has been dicovered for completeness.
How do you think this should be implemented? If nobody has an object, I would code this the following way into FreeTDM: If I get a call with a starting number I know that it must go to the external ISDN (means: a second dialplan), I first relay the typed numbers and determine the full number. Only now, when I got the full number, I would give this call to the higher layers.
What you think about this?
Regards, Markus Mueller
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org <mailto:FreeSWITCH-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org <mailto:FreeSWITCH-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Markus, how "the PSTN" would know the length of the destination (German) phone number? I think it's just impossible, especially that there are multiple independent providers in Germany, and the phone numbers differ in length even within the same end customers. just a few examples: http://www.de-cix.net/contact/contact/ https://www1.pbxes.com/iptel_impressum.html So, I still think somewhere along the chain there's a simple interdigit timeout. Probably it's in your outgoing ISDN channel at the provider side. How else would the system know when to place the call? So, instead of relying on someone else's interdigit timeout, you can handle your own, and place the outbound call when you have enough digits. ________________________________ From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Sunday, September 8, 2013 2:41 AM Subject: Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN) Hi Stanislav Sinyagin, what you mean is the "overlap dialing" feature, this is already implemented and working in freetdm. It waits some seconds after the last number and then dials. ====>> This is not what I am searching for ! This is also not what I am talking about ! Please accept this and make sure how the reality is before saying that I am telling wrong things. I am searching what I worte, if you believe or not. ISDN in germany currently works this way, and yes, there is a "completeness" of numbers which is interactively determined by sending each key through the ISDN network and getting back when a destination has been found. Regards, Markus Mueller I guess there's no such ting as "complete number", but a simple interdigit timeout. If you stop dialing and don't enter digits for 10 seconds, it automatically places the call.
So, you need a gateway which supports such an interdigit
timeout. In theory, you can implement that with an external script in FreeSWITCH. On practice, needs testing and a proof of concept.
________________________________ From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Saturday, September 7, 2013 10:08 PM Subject: Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hi all,
hi Markus,
how do you determine if it's a complete number? Do
you expect the outgoing ISDN channel to tell you that?
Sorry that I don't have all details; I am not as familary as I want to with the ISDN protocols. But I know that each number goes one by one to the outgoing ISDN and then outgoint ISDN tells you back when you have reached a destination. If needed I can make a log when I have my Siemens Hipath placed behind my Freeswitch in NT mode; there you can see the numbers comming one by one.
I also wonder what happens if you attach an ISDN-SIP gateway, like Patton. Will you have a new SIP message (which?) on every dialed digit?
Normaly all the products do the "overlap dialing" crap, as mentioned in my first email. I don't know for sure but I think there is no alternative, especially if you make only SIP and not ISDN.
No comments about my design suggestion?
Regards, Markus
________________________________ From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Saturday, September 7, 2013 1:52 PM Subject: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hello Freeswitch Developers,
ISDN in germany (maybe also somewhere else)
has a feature, which colides
with the design of a dialplan: it sends the numbers the user types into his phone LIVE (!) through the ISDN network. If the number is complete, the ISDN network tells it to the caller; Only now the call get established (means the dialplan gets invoked). So you have to do an own step (live and interactive determination of the number) BEFORE the dialplan comes in line.
Because this is object not supported by freeswith, in the following situation
User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <-> Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation
you have to do "overlap dialing". Means, freeswitch waits some seconds until the user has entered the last number and then it goes directly to the dialplan.
-> This is not what I need!
I want that it works as if there is no FreeSwitch in between. Means, freeswitch should relay each number the user types into his phone to the ISDN on the remote side, and make the dialplan stuff AFTER the number has been dicovered for completeness.
How do you think this should be implemented? If nobody has an object, I would code this the following way into FreeTDM: If I get a call with a starting number I know that it must go to the external ISDN (means: a second dialplan), I first relay the typed numbers and determine the full number. Only now, when I got the full number, I would give this call to the higher layers.
What you think about this?
Regards, Markus Mueller
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
What your searching for IS overlap dialing. And it works by sending the setup and receiving the error code for incomplete. This should pass across a bridge, you can also handle this yourself in the dialplan if you like. If this is not passing across a tdm to tdm bridge, that is a bug and should get filed in jira. Mike On Sep 7, 2013, at 8:41 PM, Markus Müller <freeswitchdev@priv.de> wrote:
Hi Stanislav Sinyagin,
what you mean is the "overlap dialing" feature, this is already implemented and working in freetdm. It waits some seconds after the last number and then dials.
====>> This is not what I am searching for ! This is also not what I am talking about ! Please accept this and make sure how the reality is before saying that I am telling wrong things.
I am searching what I worte, if you believe or not. ISDN in germany currently works this way, and yes, there is a "completeness" of numbers which is interactively determined by sending each key through the ISDN network and getting back when a destination has been found.
Regards, Markus Mueller
I guess there's no such ting as "complete number", but a simple interdigit timeout. If you stop dialing and don't enter digits for 10 seconds, it automatically places the call.
So, you need a gateway which supports such an interdigit timeout. In theory, you can implement that with an external script in FreeSWITCH. On practice, needs testing and a proof of concept.
From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Saturday, September 7, 2013 10:08 PM Subject: Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hi all,
hi Markus, how do you determine if it's a complete number? Do you expect the outgoing ISDN channel to tell you that? Sorry that I don't have all details; I am not as familary as I want to with the ISDN protocols. But I know that each number goes one by one to the outgoing ISDN and then outgoint ISDN tells you back when you have reached a destination. If needed I can make a log when I have my Siemens Hipath placed behind my Freeswitch in NT mode; there you can see the numbers comming one by one.
I also wonder what happens if you attach an ISDN-SIP gateway, like Patton. Will you have a new SIP message (which?) on every dialed digit? Normaly all the products do the "overlap dialing" crap, as mentioned in my first email. I don't know for sure but I think there is no alternative, especially if you make only SIP and not ISDN.
No comments about my design suggestion?
Regards, Markus
From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Saturday, September 7, 2013 1:52 PM Subject: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hello Freeswitch Developers,
ISDN in germany (maybe also somewhere else) has a feature, which colides with the design of a dialplan: it sends the numbers the user types into his phone LIVE (!) through the ISDN network. If the number is complete, the ISDN network tells it to the caller; Only now the call get established (means the dialplan gets invoked). So you have to do an own step (live and interactive determination of the number) BEFORE the dialplan comes in line.
Because this is object not supported by freeswith, in the following situation
User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <-> Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation
you have to do "overlap dialing". Means, freeswitch waits some seconds until the user has entered the last number and then it goes directly to the dialplan.
-> This is not what I need!
I want that it works as if there is no FreeSwitch in between. Means, freeswitch should relay each number the user types into his phone to the ISDN on the remote side, and make the dialplan stuff AFTER the number has been dicovered for completeness.
How do you think this should be implemented? If nobody has an object, I would code this the following way into FreeTDM: If I get a call with a starting number I know that it must go to the external ISDN (means: a second dialplan), I first relay the typed numbers and determine the full number. Only now, when I got the full number, I would give this call to the higher layers.
What you think about this?
Regards, Markus Mueller
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Hi Mike, thanks for helping against the unteachable ignoramus here... I am very bugged about this SHIT! Currently in default this works as you described; I type the "0" of "089xxxxxx". This goes to the SIP gateway (=Freeswitch), it send it out on the external ISDN, and this answeres something. It might be "incomplete"... don't know. In any case: The calling is aborted by the SIP gateway (=Freeswitch) and there is currently only the possibility to enable this crap "overlap dialing" feature in freeswitch (it just waits some seconds if another digit comes and then dials after the last). If it is just wrong implemented: Fine! Lets fix it. But... Don't you think it makes it necessary to change the dialplan behaviour at all? Currently it is fully based on complete numbers; it just cannot handle incomplete numbers, is it? In any case: I'll do an bug report with log files and so on. BUT PLEASE: I cannot find anything anymore about creating pcap file. Does anybody have a hint how to do this currently with FreeTDM/libpri? Regards, Markus Mueller
What your searching for IS overlap dialing. And it works by sending the setup and receiving the error code for incomplete. This should pass across a bridge, you can also handle this yourself in the dialplan if you like. If this is not passing across a tdm to tdm bridge, that is a bug and should get filed in jira.
Mike
On Sep 7, 2013, at 8:41 PM, Markus Müller <freeswitchdev@priv.de <mailto:freeswitchdev@priv.de>> wrote:
Hi Stanislav Sinyagin,
what you mean is the "overlap dialing" feature, this is already implemented and working in freetdm. It waits some seconds after the last number and then dials.
====>> This is not what I am searching for ! This is also not what I am talking about ! Please accept this and make sure how the reality is before saying that I am telling wrong things.
I am searching what I worte, if you believe or not. ISDN in germany currently works this way, and yes, there is a "completeness" of numbers which is interactively determined by sending each key through the ISDN network and getting back when a destination has been found.
Regards, Markus Mueller
I guess there's no such ting as "complete number", but a simple interdigit timeout. If you stop dialing and don't enter digits for 10 seconds, it automatically places the call.
So, you need a gateway which supports such an interdigit timeout. In theory, you can implement that with an external script in FreeSWITCH. On practice, needs testing and a proof of concept.
------------------------------------------------------------------------ *From:* Markus Müller <freeswitchdev@priv.de> *To:* freeswitch-dev@lists.freeswitch.org *Sent:* Saturday, September 7, 2013 10:08 PM *Subject:* Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hi all,
hi Markus, how do you determine if it's a complete number? Do you expect the outgoing ISDN channel to tell you that? Sorry that I don't have all details; I am not as familary as I want to with the ISDN protocols. But I know that each number goes one by one to the outgoing ISDN and then outgoint ISDN tells you back when you have reached a destination. If needed I can make a log when I have my Siemens Hipath placed behind my Freeswitch in NT mode; there you can see the numbers comming one by one.
I also wonder what happens if you attach an ISDN-SIP gateway, like Patton. Will you have a new SIP message (which?) on every dialed digit? Normaly all the products do the "overlap dialing" crap, as mentioned in my first email. I don't know for sure but I think there is no alternative, especially if you make only SIP and not ISDN.
No comments about my design suggestion?
Regards, Markus
------------------------------------------------------------------------ *From:* Markus Müller <freeswitchdev@priv.de> <mailto:freeswitchdev@priv.de> *To:* freeswitch-dev@lists.freeswitch.org <mailto:freeswitch-dev@lists.freeswitch.org> *Sent:* Saturday, September 7, 2013 1:52 PM *Subject:* [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hello Freeswitch Developers,
ISDN in germany (maybe also somewhere else) has a feature, which colides with the design of a dialplan: it sends the numbers the user types into his phone LIVE (!) through the ISDN network. If the number is complete, the ISDN network tells it to the caller; Only now the call get established (means the dialplan gets invoked). So you have to do an own step (live and interactive determination of the number) BEFORE the dialplan comes in line.
Because this is object not supported by freeswith, in the following situation
User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <-> Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation
you have to do "overlap dialing". Means, freeswitch waits some seconds until the user has entered the last number and then it goes directly to the dialplan.
-> This is not what I need!
I want that it works as if there is no FreeSwitch in between. Means, freeswitch should relay each number the user types into his phone to the ISDN on the remote side, and make the dialplan stuff AFTER the number has been dicovered for completeness.
How do you think this should be implemented? If nobody has an object, I would code this the following way into FreeTDM: If I get a call with a starting number I know that it must go to the external ISDN (means: a second dialplan), I first relay the typed numbers and determine the full number. Only now, when I got the full number, I would give this call to the higher layers.
What you think about this?
Regards, Markus Mueller
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org <mailto:FreeSWITCH-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org <mailto:FreeSWITCH-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org <mailto:FreeSWITCH-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
You can configure this all in freeswitch dialplan if freeswitch is answering, simply don't answer first, you can return incomplete, you can just bridge to something else that will return incomplete and that will pass across the bridge. Mike On Sep 9, 2013, at 10:31 AM, Markus Müller <freeswitchdev@priv.de> wrote:
Hi Mike,
thanks for helping against the unteachable ignoramus here... I am very bugged about this SHIT!
Currently in default this works as you described; I type the "0" of "089xxxxxx". This goes to the SIP gateway (=Freeswitch), it send it out on the external ISDN, and this answeres something. It might be "incomplete"... don't know. In any case: The calling is aborted by the SIP gateway (=Freeswitch) and there is currently only the possibility to enable this crap "overlap dialing" feature in freeswitch (it just waits some seconds if another digit comes and then dials after the last). If it is just wrong implemented: Fine! Lets fix it. But... Don't you think it makes it necessary to change the dialplan behaviour at all? Currently it is fully based on complete numbers; it just cannot handle incomplete numbers, is it?
In any case: I'll do an bug report with log files and so on. BUT PLEASE: I cannot find anything anymore about creating pcap file. Does anybody have a hint how to do this currently with FreeTDM/libpri?
Regards, Markus Mueller
What your searching for IS overlap dialing. And it works by sending the setup and receiving the error code for incomplete. This should pass across a bridge, you can also handle this yourself in the dialplan if you like. If this is not passing across a tdm to tdm bridge, that is a bug and should get filed in jira.
Mike
On Sep 7, 2013, at 8:41 PM, Markus Müller <freeswitchdev@priv.de> wrote:
Hi Stanislav Sinyagin,
what you mean is the "overlap dialing" feature, this is already implemented and working in freetdm. It waits some seconds after the last number and then dials.
====>> This is not what I am searching for ! This is also not what I am talking about ! Please accept this and make sure how the reality is before saying that I am telling wrong things.
I am searching what I worte, if you believe or not. ISDN in germany currently works this way, and yes, there is a "completeness" of numbers which is interactively determined by sending each key through the ISDN network and getting back when a destination has been found.
Regards, Markus Mueller
I guess there's no such ting as "complete number", but a simple interdigit timeout. If you stop dialing and don't enter digits for 10 seconds, it automatically places the call.
So, you need a gateway which supports such an interdigit timeout. In theory, you can implement that with an external script in FreeSWITCH. On practice, needs testing and a proof of concept.
From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Saturday, September 7, 2013 10:08 PM Subject: Re: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hi all,
hi Markus, how do you determine if it's a complete number? Do you expect the outgoing ISDN channel to tell you that? Sorry that I don't have all details; I am not as familary as I want to with the ISDN protocols. But I know that each number goes one by one to the outgoing ISDN and then outgoint ISDN tells you back when you have reached a destination. If needed I can make a log when I have my Siemens Hipath placed behind my Freeswitch in NT mode; there you can see the numbers comming one by one.
I also wonder what happens if you attach an ISDN-SIP gateway, like Patton. Will you have a new SIP message (which?) on every dialed digit? Normaly all the products do the "overlap dialing" crap, as mentioned in my first email. I don't know for sure but I think there is no alternative, especially if you make only SIP and not ISDN.
No comments about my design suggestion?
Regards, Markus
From: Markus Müller <freeswitchdev@priv.de> To: freeswitch-dev@lists.freeswitch.org Sent: Saturday, September 7, 2013 1:52 PM Subject: [Freeswitch-dev] Call relaying (ISDN <> Freeswitch <> ISDN)
Hello Freeswitch Developers,
ISDN in germany (maybe also somewhere else) has a feature, which colides with the design of a dialplan: it sends the numbers the user types into his phone LIVE (!) through the ISDN network. If the number is complete, the ISDN network tells it to the caller; Only now the call get established (means the dialplan gets invoked). So you have to do an own step (live and interactive determination of the number) BEFORE the dialplan comes in line.
Because this is object not supported by freeswith, in the following situation
User <-> Analog Phone <-> Siemens Hipath <-> ISDN <-> [FreeTDM <-> Freeswitch <-> FreeTDM] <-> ISDN <-> World <-> ISDN Destiation
you have to do "overlap dialing". Means, freeswitch waits some seconds until the user has entered the last number and then it goes directly to the dialplan.
-> This is not what I need!
I want that it works as if there is no FreeSwitch in between. Means, freeswitch should relay each number the user types into his phone to the ISDN on the remote side, and make the dialplan stuff AFTER the number has been dicovered for completeness.
How do you think this should be implemented? If nobody has an object, I would code this the following way into FreeTDM: If I get a call with a starting number I know that it must go to the external ISDN (means: a second dialplan), I first relay the typed numbers and determine the full number. Only now, when I got the full number, I would give this call to the higher layers.
What you think about this?
Regards, Markus Mueller
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
On Sat, Sep 7, 2013 at 4:08 PM, Markus Müller <freeswitchdev@priv.de> wrote:
Hi all,
hi Markus, how do you determine if it's a complete number? Do you expect the outgoing ISDN channel to tell you that?
Sorry that I don't have all details; I am not as familary as I want to with the ISDN protocols. But I know that each number goes one by one to the outgoing ISDN and then outgoint ISDN tells you back when you have reached a destination. If needed I can make a log when I have my Siemens Hipath placed behind my Freeswitch in NT mode; there you can see the numbers comming one by one.
I also wonder what happens if you attach an ISDN-SIP gateway, like Patton. Will you have a new SIP message (which?) on every dialed digit?
Normaly all the products do the "overlap dialing" crap, as mentioned in my first email. I don't know for sure but I think there is no alternative, especially if you make only SIP and not ISDN.
No comments about my design suggestion?
Your design suggestion is pretty thin on details, so is hard to comment on that. However, overall it does not sound like a good idea to me to try to bridge a call without going through the FreeSWITCH core. I believe your best shot at getting this done and integrated would be to make it work with the existing bridge() application. This is a rough description of how to do it: 1. Send the call up from the signaling stack to FreeSWITCH immediately so the call is sent to the routing engine and you can call the bridge() application on an outbound FreeTDM channel. 2. The outbound channel sends the first call message (ie SETUP) 3. Relay the following digits as subsequent signaling messages from the inbound leg to the outbound leg. On the FreeTDM side using FTDM_SIGEVENT_COLLECTED_DIGIT, then enqueuing a freeswitch message such as SWITCH_MESSAGE_INDICATE_INFO, on the outgoing leg you would receive the indication and send the digits to the upstream provider. Note you described some sort of pass-thru overlap functionality, if we wanted to implement overlap dialing where FreeSWITCH takes the routing decision based on digits as received some extra effort is required to have XML re-evaluated for each digit received. Cheers, Moy *Moises Silva **Manager, Software Engineering*** msilva@sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 **<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`| | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
participants (4)
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Markus Müller
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Michael Jerris
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Moises Silva
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Stanislav Sinyagin