NPDI tel URI parameters appearing in SIP URI
Hello, SIP call comes on leg A as following: INVITE sip:6135555555*;npdi=yes*@1.2.3.4:5060 SIP/2.0 ${destination_number} var becomes "6135555555;npdi=yes" On leg B I send the call to gateway using unaltered ${destination_number}: <action application="bridge" data="sofia/gateway/test_gw/${destination_number}"/> I see the following on leg B: INVITE sip:6135555555@5.6.7.8:5060*;npdi=yes* SIP/2.0 So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554 Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway. Thank you, Victor
You'll have to use a regex and capture it manually. See: http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Porta... -Avi On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy < victor.chukalovskiy@gmail.com> wrote:
Hello,
SIP call comes on leg A as following: INVITE sip:6135555555*;npdi=yes*@1.2.3.4:5060 SIP/2.0
${destination_number} var becomes "6135555555;npdi=yes"
On leg B I send the call to gateway using unaltered ${destination_number}: <action application="bridge" data="sofia/gateway/test_gw/${destination_number}"/>
I see the following on leg B: INVITE sip:6135555555@5.6.7.8:5060*;npdi=yes* SIP/2.0
So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554
Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway.
Thank you, Victor
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If its actually moving the ndpi to the host side thats a bug open a jira and attach debug logs with sip trace on Sent from my iPhone On Feb 19, 2013, at 6:25 PM, Avi Marcus <avi@avimarcus.net> wrote:
You'll have to use a regex and capture it manually. See: http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Porta...
-Avi
On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com> wrote:
Hello,
SIP call comes on leg A as following: INVITE sip:6135555555;npdi=yes@1.2.3.4:5060 SIP/2.0
${destination_number} var becomes "6135555555;npdi=yes"
On leg B I send the call to gateway using unaltered ${destination_number}: <action application="bridge" data="sofia/gateway/test_gw/${destination_number}"/>
I see the following on leg B: INVITE sip:6135555555@5.6.7.8:5060;npdi=yes SIP/2.0
So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554
Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Yes, it moves it to the right of the host. Thank you for confirming, looks like I'll have to open a jira. On 02/19/2013 07:57 PM, Ken Rice wrote:
If its actually moving the ndpi to the host side thats a bug open a jira and attach debug logs with sip trace on
Sent from my iPhone
On Feb 19, 2013, at 6:25 PM, Avi Marcus <avi@avimarcus.net <mailto:avi@avimarcus.net>> wrote:
You'll have to use a regex and capture it manually. See: http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Porta...
-Avi
On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com <mailto:victor.chukalovskiy@gmail.com>> wrote:
Hello,
SIP call comes on leg A as following: INVITE sip:6135555555 <tel:6135555555>*;npdi=yes*@1.2.3.4:5060 <http://1.2.3.4:5060> SIP/2.0
${destination_number} var becomes "6135555555 <tel:6135555555>;npdi=yes"
On leg B I send the call to gateway using unaltered ${destination_number}: <action application="bridge" data="sofia/gateway/test_gw/${destination_number}"/>
I see the following on leg B: INVITE sip:6135555555@5.6.7.8:5060 <mailto:sip:6135555555@5.6.7.8:5060>*;npdi=yes* SIP/2.0
So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554
Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
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Avi, I know this part of the WiKi....thank you but I don't think it helps here. -Victor On 02/19/2013 07:25 PM, Avi Marcus wrote:
You'll have to use a regex and capture it manually. See: http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Porta...
-Avi
On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com <mailto:victor.chukalovskiy@gmail.com>> wrote:
Hello,
SIP call comes on leg A as following: INVITE sip:6135555555 <tel:6135555555>*;npdi=yes*@1.2.3.4:5060 <http://1.2.3.4:5060> SIP/2.0
${destination_number} var becomes "6135555555 <tel:6135555555>;npdi=yes"
On leg B I send the call to gateway using unaltered ${destination_number}: <action application="bridge" data="sofia/gateway/test_gw/${destination_number}"/>
I see the following on leg B: INVITE sip:6135555555@5.6.7.8:5060 <mailto:sip:6135555555@5.6.7.8:5060>*;npdi=yes* SIP/2.0
So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554
Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway.
Thank you, Victor
_________________________________________________________________________ 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-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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
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Ah sorry, I should have read your question more carefully! -Avi On Wed, Feb 20, 2013 at 6:43 AM, Victor Chukalovskiy < victor.chukalovskiy@gmail.com> wrote:
Avi,
I know this part of the WiKi....thank you but I don't think it helps here.
-Victor
On 02/19/2013 07:25 PM, Avi Marcus wrote:
You'll have to use a regex and capture it manually. See: http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Porta...
-Avi
On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy < victor.chukalovskiy@gmail.com> wrote:
Hello,
SIP call comes on leg A as following: INVITE sip:6135555555*;npdi=yes*@1.2.3.4:5060 SIP/2.0
${destination_number} var becomes "6135555555;npdi=yes"
On leg B I send the call to gateway using unaltered ${destination_number}: <action application="bridge" data="sofia/gateway/test_gw/${destination_number}"/>
I see the following on leg B: INVITE sip:6135555555@5.6.7.8:5060*;npdi=yes* SIP/2.0
So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554
Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway.
Thank you, Victor
_________________________________________________________________________ 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
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Did a bit more testing today. <action application="bridge" data="sofia/test_profile/*sip:6135555555;npdi=yes*@24.114.15.50:5060"/> gives what I need as per WiKi: INVITE sip:6135555555;npdi=yes@5.6.7.8:5060 SIP/2.0 But here is a caveat: this is not gateway! I do need to use gateway in the bridge app (because of OPTIONS PING that determines if gateway is up or down). So, I try: <action application="bridge" data="sofia/gateway/test_gateway/6135555555;npdi=yes"/> This is no good, ;npdi=yes moved in to the domain part: INVITE sip:6135555555@5.6.7.8:5060*;npdi=yes* SIP/2.0 Also, tried this: <action application="bridge" data="sofia/gateway/test_gateway/sip:6135555555;npdi=yes"/> But this is no good either. Looks like sip: syntax does not work with gateways: INVITE *sip:sip:*6135555555@5.6.7.8:5060*;npdi=yes* SIP/2.0 ---------------- So, my immediate question is, is there a way to use sip: syntax when bridging to a gateway? ---------------- And on a broader scale, something along these lines: http://freeswitch-users.2379917.n2.nabble.com/generating-RFC-3966-and-RFC-46... Was there much work done since then on RFC3966 and RFC4694 support in FS? Ideally, that would be something that's parsed as channel variable on incoming call and can be manipulated as a channel variable on the outgoing call. FS would take care of parsing tel URI on incoming and putting it together on the outgoing. -Victor On 02/20/2013 03:32 AM, Avi Marcus wrote:
Ah sorry, I should have read your question more carefully! -Avi
On Wed, Feb 20, 2013 at 6:43 AM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com <mailto:victor.chukalovskiy@gmail.com>> wrote:
Avi,
I know this part of the WiKi....thank you but I don't think it helps here.
-Victor
On 02/19/2013 07:25 PM, Avi Marcus wrote:
You'll have to use a regex and capture it manually. See: http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Porta...
-Avi
On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com <mailto:victor.chukalovskiy@gmail.com>> wrote:
Hello,
SIP call comes on leg A as following: INVITE sip:6135555555 <tel:6135555555>*;npdi=yes*@1.2.3.4:5060 <http://1.2.3.4:5060> SIP/2.0
${destination_number} var becomes "6135555555 <tel:6135555555>;npdi=yes"
On leg B I send the call to gateway using unaltered ${destination_number}: <action application="bridge" data="sofia/gateway/test_gw/${destination_number}"/>
I see the following on leg B: INVITE sip:6135555555@5.6.7.8:5060 <mailto:sip:6135555555@5.6.7.8:5060>*;npdi=yes* SIP/2.0
So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554
Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway.
Thank you, Victor
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
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Hello, I really appreciate if Brian or Anthony could have a look here. Looking for some help and advice to understand if I got it right and how to move forward: Is there a way to pass tel URI parameters (;npdi= ;rn=) correctly when bridging SIP call to a gateway? Or is it not possible without opening a Jira bug to fix? What is the current support of RFC3966 and RFC4694 in FS? Does it require a feature request with perhaps a bounty? Many thanks, Victor On 13-02-20 02:26 PM, Victor Chukalovskiy wrote:
Did a bit more testing today.
<action application="bridge" data="sofia/test_profile/*sip:6135555555;npdi=yes*@24.114.15.50:5060"/> gives what I need as per WiKi: INVITE sip:6135555555;npdi=yes@5.6.7.8:5060 SIP/2.0
But here is a caveat: this is not gateway! I do need to use gateway in the bridge app (because of OPTIONS PING that determines if gateway is up or down). So, I try:
<action application="bridge" data="sofia/gateway/test_gateway/6135555555;npdi=yes"/> This is no good, ;npdi=yes moved in to the domain part: INVITE sip:6135555555@5.6.7.8:5060*;npdi=yes* SIP/2.0
Also, tried this: <action application="bridge" data="sofia/gateway/test_gateway/sip:6135555555;npdi=yes"/> But this is no good either. Looks like sip: syntax does not work with gateways: INVITE *sip:sip:*6135555555@5.6.7.8:5060*;npdi=yes* SIP/2.0
---------------- So, my immediate question is, is there a way to use sip: syntax when bridging to a gateway? ---------------- And on a broader scale, something along these lines: http://freeswitch-users.2379917.n2.nabble.com/generating-RFC-3966-and-RFC-46... Was there much work done since then on RFC3966 and RFC4694 support in FS? Ideally, that would be something that's parsed as channel variable on incoming call and can be manipulated as a channel variable on the outgoing call. FS would take care of parsing tel URI on incoming and putting it together on the outgoing.
-Victor
On 02/20/2013 03:32 AM, Avi Marcus wrote:
Ah sorry, I should have read your question more carefully! -Avi
On Wed, Feb 20, 2013 at 6:43 AM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com <mailto:victor.chukalovskiy@gmail.com>> wrote:
Avi,
I know this part of the WiKi....thank you but I don't think it helps here.
-Victor
On 02/19/2013 07:25 PM, Avi Marcus wrote:
You'll have to use a regex and capture it manually. See: http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Porta...
-Avi
On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com <mailto:victor.chukalovskiy@gmail.com>> wrote:
Hello,
SIP call comes on leg A as following: INVITE sip:6135555555 <tel:6135555555>*;npdi=yes*@1.2.3.4:5060 <http://1.2.3.4:5060> SIP/2.0
${destination_number} var becomes "6135555555 <tel:6135555555>;npdi=yes"
On leg B I send the call to gateway using unaltered ${destination_number}: <action application="bridge" data="sofia/gateway/test_gw/${destination_number}"/>
I see the following on leg B: INVITE sip:6135555555@5.6.7.8:5060 <mailto:sip:6135555555@5.6.7.8:5060>*;npdi=yes* SIP/2.0
So, FS moved ";npdi=yes" which is a tel URI parameter into the very end as if it was a SIP URI parameter. Such behavior does not seem correct. Is this a bug? Found the following implementation discussion: http://comments.gmane.org/gmane.ietf.sip-implementors/21554
Please help to make FS send tel URI parameters unaltered using bridge application and a gateway. WiKi for bridge dial-plan tool has some examples but it does not make use of gateway and I need to use gateway.
Thank you, Victor
_________________________________________________________________________ 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-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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
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Did you ever open the jira that Ken requested you open on this? All of the information you found about this issue should be on a jira or its not going to get the attention you desire. Mike On Feb 21, 2013, at 1:10 PM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com> wrote:
Hello,
I really appreciate if Brian or Anthony could have a look here. Looking for some help and advice to understand if I got it right and how to move forward:
Is there a way to pass tel URI parameters (;npdi= ;rn=) correctly when bridging SIP call to a gateway? Or is it not possible without opening a Jira bug to fix?
What is the current support of RFC3966 and RFC4694 in FS? Does it require a feature request with perhaps a bounty?
Many thanks, Victor
Mike, Thanks I did now FS-5118. Sorry I thought better to double-check it here before opening a Jira (I'm not sure at this point weather it's a bug or a feature request). Thank you, Victor On 13-02-21 01:35 PM, Michael Jerris wrote:
Did you ever open the jira that Ken requested you open on this? All of the information you found about this issue should be on a jira or its not going to get the attention you desire.
Mike
On Feb 21, 2013, at 1:10 PM, Victor Chukalovskiy <victor.chukalovskiy@gmail.com> wrote:
Hello,
I really appreciate if Brian or Anthony could have a look here. Looking for some help and advice to understand if I got it right and how to move forward:
Is there a way to pass tel URI parameters (;npdi= ;rn=) correctly when bridging SIP call to a gateway? Or is it not possible without opening a Jira bug to fix?
What is the current support of RFC3966 and RFC4694 in FS? Does it require a feature request with perhaps a bounty?
Many thanks, Victor
_________________________________________________________________________ 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-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On Thu, Feb 21, 2013 at 2:02 PM, Victor Chukalovskiy < victor.chukalovskiy@gmail.com> wrote:
Mike,
Thanks I did now FS-5118.
Sorry I thought better to double-check it here before opening a Jira (I'm not sure at this point weather it's a bug or a feature request).
You never have to ask on the ML whether to open a Jira. Just open it and the interested parties will comment there. -MC
participants (5)
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Avi Marcus
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Ken Rice
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Michael Collins
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Michael Jerris
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Victor Chukalovskiy