[Freeswitch-dev] FreeSwitch, application or module!?
Hi Friends! I wish to do something like: - Authorize all sip clients in external DB, for example postgresql. - Control each call session duration, just like prepaid calling card. - LCR, if one voip provider can't connect call into PSTN, try next PSTN supplier. Please help me, i can't understand what is docs i might to read to know how to do it. Could you show me some links and terms. explain it? Thank you so much! DM
Dmitry, FreeSWITCH can definitely help you with all of this, HOWEVER, there's a lot in this scenario that isn't specifically FS. Let me ask you this question: do you have a programmer that can handle the interfacing necessary with FS? I believe you will need to read up on a few things: Auth all SIP clients from external db - http://wiki.freeswitch.org/wiki/Mod_xml_curl Call control - http://wiki.freeswitch.org/wiki/Event_Socket LCR - http://wiki.freeswitch.org/wiki/Mod_lcr I can tell you that there isn't already a whole package with all of this, but rather just some of the individual components that will need to be put together. Will you also need a billing system? If so you'll need to make sure that you can handle CDRs. You have multiple choices there as well. FS does not support logging directly to a backend database, but it does support generating CDR records that are "INSERT INTO..." commands that can easily be run to autopopulate your CDR table. There's also XML CDRs which allow for extremely specific parsing of call history. The catch is that you need a good XML parser and some business logic to make sure that this information is all useful to you. Hope this helps! -MC On Sun, Dec 21, 2008 at 11:08 PM, Dmitry Mordovin <d.mordovin@dwide.com>wrote:
Hi Friends!
I wish to do something like: - Authorize all sip clients in external DB, for example postgresql. - Control each call session duration, just like prepaid calling card. - LCR, if one voip provider can't connect call into PSTN, try next PSTN supplier.
Please help me, i can't understand what is docs i might to read to know how to do it. Could you show me some links and terms. explain it?
Thank you so much!
DM
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Michael Collins wrote:
Dmitry,
FreeSWITCH can definitely help you with all of this, HOWEVER, there's a lot in this scenario that isn't specifically FS. Let me ask you this question: do you have a programmer that can handle the interfacing necessary with FS? I believe you will need to read up on a few things:
Auth all SIP clients from external db - http://wiki.freeswitch.org/wiki/Mod_xml_curl Call control - http://wiki.freeswitch.org/wiki/Event_Socket LCR - http://wiki.freeswitch.org/wiki/Mod_lcr
I can tell you that there isn't already a whole package with all of this, but rather just some of the individual components that will need to be put together. Will you also need a billing system? If so you'll need to make sure that you can handle CDRs. You have multiple choices there as well. FS does not support logging directly to a backend database, but it does support generating CDR records that are "INSERT INTO..." commands that can easily be run to autopopulate your CDR table. There's also XML CDRs which allow for extremely specific parsing of call history. The catch is that you need a good XML parser and some business logic to make sure that this information is all useful to you.
Hope this helps! -MC
On Sun, Dec 21, 2008 at 11:08 PM, Dmitry Mordovin <d.mordovin@dwide.com <mailto:d.mordovin@dwide.com>> wrote:
Hi Friends!
I wish to do something like: - Authorize all sip clients in external DB, for example postgresql. - Control each call session duration, just like prepaid calling card. - LCR, if one voip provider can't connect call into PSTN, try next PSTN supplier.
Please help me, i can't understand what is docs i might to read to know how to do it. Could you show me some links and terms. explain it?
Thank you so much!
DM
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Hello Michael! Thank you so much! Now i have another problem, 1000@example.com (kphone) calling -> FS -> 1001@example.com (X-Lite). Sometimes ago 1000 and 1001 connected and talking. When 1001 user hungup, FS recv BYE from 1001, i see (in log) FS doing some HUNGUPs function, writes log " Sending BYE to sofia/internal/1000@example.com". But 1000@example.com Don't receive any packets since receive 200 OK on connect. I tried UDP and TCP protocol. my dialplan: <extension name="PBX Extension"> <condition field="destination_number" expression="^(1[0-9]{3})$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/default/$1%$${domain}"/> </condition> </extension> I found discussion about it and bug fix, but now its happens again, or its my mistake in configuration? Thank you all! DM
This sounds like a nat issue where we are sending a BYE but its not getting to the other end. Check the sip trace and see if that is the case. Mike On Dec 23, 2008, at 5:09 AM, Dmitry Mordovin wrote:
Michael Collins wrote:
Dmitry,
FreeSWITCH can definitely help you with all of this, HOWEVER, there's a lot in this scenario that isn't specifically FS. Let me ask you this question: do you have a programmer that can handle the interfacing necessary with FS? I believe you will need to read up on a few things:
Auth all SIP clients from external db - http://wiki.freeswitch.org/wiki/Mod_xml_curl Call control - http://wiki.freeswitch.org/wiki/Event_Socket LCR - http://wiki.freeswitch.org/wiki/Mod_lcr
I can tell you that there isn't already a whole package with all of this, but rather just some of the individual components that will need to be put together. Will you also need a billing system? If so you'll need to make sure that you can handle CDRs. You have multiple choices there as well. FS does not support logging directly to a backend database, but it does support generating CDR records that are "INSERT INTO..." commands that can easily be run to autopopulate your CDR table. There's also XML CDRs which allow for extremely specific parsing of call history. The catch is that you need a good XML parser and some business logic to make sure that this information is all useful to you.
Hope this helps! -MC
On Sun, Dec 21, 2008 at 11:08 PM, Dmitry Mordovin <d.mordovin@dwide.com
wrote: Hi Friends!
I wish to do something like: - Authorize all sip clients in external DB, for example postgresql. - Control each call session duration, just like prepaid calling card. - LCR, if one voip provider can't connect call into PSTN, try next PSTN supplier.
Please help me, i can't understand what is docs i might to read to know how to do it. Could you show me some links and terms. explain it?
Thank you so much!
DM
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Hello Michael!
Thank you so much!
Now i have another problem,
1000@example.com (kphone) calling -> FS -> 1001@example.com (X-Lite).
Sometimes ago 1000 and 1001 connected and talking. When 1001 user hungup, FS recv BYE from 1001, i see (in log) FS doing some HUNGUPs function, writes log " Sending BYE to sofia/internal/1000@example.com ".
But 1000@example.com Don't receive any packets since receive 200 OK on connect. I tried UDP and TCP protocol.
my dialplan:
<extension name="PBX Extension"> <condition field="destination_number" expression="^(1[0-9]{3})$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/default/$1%$ ${domain}"/> </condition> </extension>
I found discussion about it and bug fix, but now its happens again, or its my mistake in configuration?
Thank you all!
DM _______________________________________________ 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
Michael Jerris wrote:
This sounds like a nat issue where we are sending a BYE but its not getting to the other end. Check the sip trace and see if that is the case.
Mike
On Dec 23, 2008, at 5:09 AM, Dmitry Mordovin wrote:
Michael Collins wrote:
Dmitry,
FreeSWITCH can definitely help you with all of this, HOWEVER, there's a lot in this scenario that isn't specifically FS. Let me ask you this question: do you have a programmer that can handle the interfacing necessary with FS? I believe you will need to read up on a few things:
Auth all SIP clients from external db - http://wiki.freeswitch.org/wiki/Mod_xml_curl Call control - http://wiki.freeswitch.org/wiki/Event_Socket LCR - http://wiki.freeswitch.org/wiki/Mod_lcr
I can tell you that there isn't already a whole package with all of this, but rather just some of the individual components that will need to be put together. Will you also need a billing system? If so you'll need to make sure that you can handle CDRs. You have multiple choices there as well. FS does not support logging directly to a backend database, but it does support generating CDR records that are "INSERT INTO..." commands that can easily be run to autopopulate your CDR table. There's also XML CDRs which allow for extremely specific parsing of call history. The catch is that you need a good XML parser and some business logic to make sure that this information is all useful to you.
Hope this helps! -MC
On Sun, Dec 21, 2008 at 11:08 PM, Dmitry Mordovin <d.mordovin@dwide.com <mailto:d.mordovin@dwide.com>> wrote:
Hi Friends!
I wish to do something like: - Authorize all sip clients in external DB, for example postgresql. - Control each call session duration, just like prepaid calling card. - LCR, if one voip provider can't connect call into PSTN, try next PSTN supplier.
Please help me, i can't understand what is docs i might to read to know how to do it. Could you show me some links and terms. explain it?
Thank you so much!
DM
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Hello Michael!
Thank you so much!
Now i have another problem,
1000@example.com (kphone) calling -> FS -> 1001@example.com (X-Lite).
Sometimes ago 1000 and 1001 connected and talking. When 1001 user hungup, FS recv BYE from 1001, i see (in log) FS doing some HUNGUPs function, writes log " Sending BYE to sofia/internal/1000@example.com".
But 1000@example.com Don't receive any packets since receive 200 OK on connect. I tried UDP and TCP protocol.
my dialplan:
<extension name="PBX Extension"> <condition field="destination_number" expression="^(1[0-9]{3})$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/default/$1%$${domain}"/> </condition> </extension>
I found discussion about it and bug fix, but now its happens again, or its my mistake in configuration?
Thank you all!
DM _______________________________________________ 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
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Thanks a lot Michael! Checked, its a problem with NAT. Will try to find solution. DM
participants (3)
-
Dmitry Mordovin
-
Michael Collins
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Michael Jerris