[Freeswitch-users] NPDI tel URI parameters appearing in SIP URI
Victor Chukalovskiy
victor.chukalovskiy at gmail.com
Thu Feb 21 21:10:58 MSK 2013
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 at 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 at 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 at 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-4694-calls-td2544806.html
> 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 at gmail.com
>> <mailto:victor.chukalovskiy at 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_Portability.29
>>>
>>> -Avi
>>>
>>> On Wed, Feb 20, 2013 at 12:25 AM, Victor Chukalovskiy
>>> <victor.chukalovskiy at gmail.com
>>> <mailto:victor.chukalovskiy at 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 at 5.6.7.8:5060
>>> <mailto:sip:6135555555 at 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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>>
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