[cisco-voip] CFA/SNR fails with 503 Request Timeout at Proxy...

[Jonathan Charles]

Just an update; the solution was to create a new CSS for Cal Forward All
and Single Number Reach that prefixes 999 on them; we then match a dial
peer using different SIP profile that adds the Diversion (the normal SIP
profile does not add a diversion); we also run it thru a translation to
remove the 999....

Just in case for future reference:


voice class sip-profiles 1
 request INVITE sip-header Diversion modify "<sip:(.*)@(.*>)" "<
sip:8475551212 at 10.10.10.10:5060>"
 request INVITE sip-header Contact modify "<sip:(.*)@(.*>)" "<
sip:8475551212 at 10.10.10.10:5060>"
!
voice class sip-profiles 2
 request INVITE sip-header Diversion modify "<sip:(.*)@(.*>)" "<
sip:8475551212 at 10.10.10.10:5060>"
 request INVITE sip-header Contact modify "<sip:(.*)@(.*>)" "<
sip:8475551212 at 10.10.10.10:5060>"
 request INVITE sip-header Diversion add "Diversion: \"8475551212\" <
sip:8475551212 at 10.10.10.10>;reason=unconditional;privacy=off;screen=yes"
!


voice translation-rule 12
 rule 1 /^999\(1..........\)/ /\1/
!

voice translation-profile ANIOUT
 translate called 12
!

dial-peer voice 200 voip
 description OUTBOUND Voice SIP calls to VzB
 translation-profile outgoing ANIOUT
 max-conn 200
 destination-pattern 1[2-9]..[2-9]......$
 session protocol sipv2
 session target sip-server
 incoming called-number .
 voice-class sip dtmf-relay force rtp-nte
 voice-class sip early-offer forced
 voice-class sip profiles 1
 no voice-class sip pass-thru content sdp
 dtmf-relay rtp-nte
 codec g711ulaw
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco
 no vad
!


!
dial-peer voice 200000 voip
 description OUTBOUND CFA-SNR Voice SIP calls to VzB
 translation-profile outgoing ANIOUT
 max-conn 200
 destination-pattern 9991[2-9]..[2-9]......$
 session protocol sipv2
 session target sip-server
 incoming called-number .
 voice-class sip dtmf-relay force rtp-nte
 voice-class sip early-offer forced
 voice-class sip profiles 2
 no voice-class sip pass-thru content sdp
 dtmf-relay rtp-nte
 codec g711ulaw
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco
 no vad
!




Jonathan


On Tue, Dec 16, 2014 at 3:37 PM, Jonathan Charles <jonvoip at gmail.com> wrote:

> OK, added the Diversion, and now on some calls, we are getting a 606 Not
> Acceptable...
>
> voice class sip-profiles 1
> request INVITE sip-header Diversion add "Diversion: \"2125551212\" <
> sip:2125551212 at 1.1.1.1>;reason=unconditional;privacy=off;screen=yes"
>
> IP/2.0 606 Not acceptable
> Via: SIP/2.0/UDP 157.130.97.178:5060;branch=z9hG4bKAF9861B3B
> From: "John Smith" <sip:7085551212 at scn10001.company.globalipcom.com
> >;tag=389FB18-DAE
> To: <sip:12125551212 at scn10001.company.globalipcom.com
> >;tag=1255691601-1418765447466
> Call-ID: 9B1AA540-84A111E4-B96AFDC8-B7F757A2 at 157.130.97.178
> CSeq: 101 INVITE
> Timestamp: 1418765435
> Content-Length: 0
>
> Any ideas?
>
>
>
>
> Jonathan
>
>
>
> On Sat, Dec 13, 2014 at 4:23 PM, Jonathan Charles <jonvoip at gmail.com>
> wrote:
>>
>> Cool, let me try it.. I didn't know I could add as well (that seems a tad
>> dangerous, but cool)...
>>
>>
>> Jonathan
>>
>> On Sat, Dec 13, 2014 at 4:03 PM, NateCCIE <nateccie at gmail.com> wrote:
>>>
>>> You just say add instead of modify.
>>>
>>> Now if you add when there is already one then there will be two.
>>>
>>> So I delete then add so there is always the one I want.
>>>
>>> Sent from my iPhone
>>> +1 801 718 2308
>>>
>>> On Dec 13, 2014, at 2:47 PM, Jonathan Charles <jonvoip at gmail.com> wrote:
>>>
>>> The problem from Verizon is that there is no diversion header in the
>>> message, how do we add one, I am modifying diversion headers but we don't
>>> even see a diversion header on these calls... Are they addable?
>>>
>>>
>>> Jonathan
>>>
>>> On Fri, Dec 12, 2014 at 1:42 AM, Amit Kumar <amit3.kum at gmail.com> wrote:
>>>>
>>>> Hi,
>>>>
>>>> Do we see this message inbound on cube??,  then it would be worth
>>>> asking vz if they need something specific on redirected calls.
>>>> On 12-Dec-2014 10:33 am, "Jonathan Charles" <jonvoip at gmail.com> wrote:
>>>>
>>>>> CCM - H323 - CUBE - SIP TRUNK - Verizon SIP
>>>>>
>>>>> We are seeing the following SIP errors on CFA/SNR calls:
>>>>>
>>>>> SIP/2.0 503 Request Timeout At Proxy
>>>>>
>>>>> Via: SIP/2.0/UDP 157.130.97.178:5060;branch=z9hG4bKACD87F86
>>>>>
>>>>> From: "PLACE" <sip:ORIGINAL OUTSIDE CALLING
>>>>> PARTY at scn10001.company.globalipcom.com>;tag=EB500DAC-20BF
>>>>>
>>>>> To: <sip:SNR DESTINATION @scn10001.company.globalipcom.com
>>>>> >;tag=aprqngfrt-9h004400000a6
>>>>>
>>>>> Call-ID: 41B1BE63-80EF11E4-B597FDC8-B7F757A2 at 157.130.97.178
>>>>>
>>>>> CSeq: 101 INVITE
>>>>>
>>>>> Timestamp: 1418358981
>>>>>
>>>>> Content-Length: 0
>>>>>
>>>>>
>>>>>
>>>>> We have multiple office codes on our SIP trunk, all of them work
>>>>> except this one; errors are consistent.
>>>>>
>>>>>
>>>>> There is no DIVERSION field in any of the Invites.
>>>>>
>>>>>
>>>>> Any ideas?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Jonathan
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>  _______________________________________________
>>> cisco-voip mailing list
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>>>
>>>
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