[cisco-voip] Unity xfer Getting 603 from carrier

[Anthony Holloway]

Most of the time it' the verbose nature of the systems to perform a
handoff.  I.e., They are chatty!

There are two methods to reduce the chatter that I know of:

1) On the CUBE the command mid-call signlaing passthru media-change
(spellcheck?)
2) On the CUCM a service param called Duplex Streaming Enabled set to True

Maybe others know of more.

Anyway, if neither of these commands reduce the delay, you would need to
look at a ladder diagram of the end-to-end (which is why sip end-to-end is
nice) signaling to see where the delay comes from.

On Fri, Aug 14, 2020 at 8:16 AM fred at browardcommunications.com <
fred at browardcommunications.com> wrote:

> Hello, thanks again for the help, a combination of Diversion and PAI is
> finally what let the call go through with ATT, so that is working now.
> Issue now is, there is a very long delay between the time the caller
> selects the option, and the call actually gets routed, any thoughts on what
> would cause this delay?
> Thank you.
>
>
>
>
> Sent from my iPhone
>
> On Jul 30, 2020, at 4:41 PM, Anthony Holloway <
> avholloway+cisco-voip at gmail.com> wrote:
>
> Midcall signaling wont help the calling number issue. The link I posted
> has examples.
>
> On Thu, Jul 30, 2020 at 3:29 PM fred at browardcommunications.com <
> fred at browardcommunications.com> wrote:
>
>> Thank you.
>> So, it looks like I have 3 options:
>> - Set the midcall signaling, which I am trying tonight
>>
>> - setting the sip trunk on calling party selection to “last redirect
>> number”
>> And reset
>>
>> - then if all else fails I will try the sip profile.
>>
>> Do you have an example sip profile / dial-peer config handy?
>>
>> Thank you much!
>>
>>
>>
>> Sent from my iPhone
>>
>> On Jul 30, 2020, at 3:46 PM, Anthony Holloway <
>> avholloway+cisco-voip at gmail.com> wrote:
>>
>> If your assessment is correct about the calling number, the carrier will
>> see the original or outside caller's phone number in the From field and
>> reject your call.  I see this being solved most of the time with a SIP
>> Profile on the outgoing dial-peer, which adds either a Diversion header or
>> a P-Asserted-Identity header.
>>
>> E.g.,
>>
>> https://community.cisco.com/t5/collaboration-voice-and-video/configure-and-troubleshoot-call-forward-to-the-pstn-using-sip/ta-p/3118287
>>
>> On Wed, Jul 29, 2020 at 6:45 AM fred at browardcommunications.com <
>> fred at browardcommunications.com> wrote:
>>
>>>
>>> Greetings all, this might be simple fix, I just haven’t dealt with this
>>> in a while.
>>>
>>> We have a unity AA that, when external callers call, select menu
>>> options, etc. Unity will send the call back to. CtiRp, which then CFA to an
>>> external number. You hear the Unity transfer message, 1 second of MoH, then
>>> 10 seconds of nothing, then the call drops.
>>>
>>> Internal calls to the CFA ctirp works
>>>
>>> Internal calls to the unity ctirp then back to the CFA ctirp works.
>>>
>>> I tried having unity call the pstn number directly with same results
>>>
>>> I think the issue is with the calling number is why the carrier is
>>> sending the 603, but it is next to impossible to get them to tell us that.
>>>
>>> Call flow:
>>> Pstn>sipt>cucm >unity>cucm>ctirp-CFA-pstn
>>>
>>> Where would I change the calling number being CFA’ed from Unity to the
>>> PSTN?
>>>
>>> Any ideas?
>>>
>>> Thank you.
>>>
>>> /FW
>>>
>>> Sent from my iPhone
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
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