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Today's Topics:
1. Re: AS5400 working as CUBE (Peter Slow)
2. Re: AS5400 working as CUBE (Maciej Bylica)
3. Re: Using DTMF # as input for Company Directory w/ CVP and
ICM (Anthony Holloway)
4. TRAI regulation for call treatment over WAN (Abdul Salam .)
5. Re: Strange Issue CP-3905 (Jean Carlos Coelho)
6. IRC (Jason Faraone)
7. Re: Jabber status for phone forwarded? (Ryan Ratliff (rratliff))
8. fast busy when phone CFA via CUBE (Erick Wellnitz)
9. Re: fast busy when phone CFA via CUBE (Kenneth Hayes)
10. Re: fast busy when phone CFA via CUBE (Kenneth Hayes)
----------------------------------------------------------------------
Message: 1
Date: Wed, 31 Jul 2013 11:05:38 -0400
From: Peter Slow <[email protected]>
To: [email protected]
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] AS5400 working as CUBE
Message-ID:
<CAMa5Jw6OF3bBiy=w9l+the3zcbx9w6km2gxyn9ok+wffh9o...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
I am fairly certain that at one point or another this platform allowed
you to use the dspfarm config commands, but i dont rememeber what the
specifics were.
It would be helpful to know which DSPs you've got. you've also got a
couple different options - even if you can't do all of the CUBE things
you need to do on the 5400, you might (should) be able to get it to
register as a DSP farm to some other device, either a CUBE or a CUCM.
I haven't ever actually gotten one of those to register as a dspfarm
before, but i'm willing to take a look with you if you're interested
=)
-Pete
On Wed, Jul 31, 2013 at 9:12 AM, Mark Holloway <[email protected]> wrote:
> I believe it was IOS 15.0(1)M which added SIP to SIP transcoding with CUBE.
> You should be able to support this function on the AS5400 as long as you're
> on a the right version. IOS 12.x releases do not support SIP to SIP
> transcoding (documented here).
>
> On Jul 30, 2013, at 4:08 PM, Joel Perez <[email protected]> wrote:
>
> That was one of the areas we had trouble with. We couldn't get trans-coding
> to work locally on the CUBE router. If i remember correctly we couldn't
> register the resources locally or with CUCM. It has been a while but that is
> what I remember off the top of my head.
>
> Joel P.
>
>
> On Tue, Jul 30, 2013 at 4:04 PM, Maciej Bylica <[email protected]> wrote:
>>
>> Hi Joel,
>>
>> Thanks for prompt answer.
>> Have you tried to use transcoding in your case?
>>
>> Thanks,
>> Maciej.
>>
>>
>> 2013/7/30 Joel Perez <[email protected]>
>>>
>>> Hi Mac,
>>>
>>> I haven't used an AS5400 for CUBE but I have used an AS5350. They are
>>> somewhat similar so i would venture to say that you could.
>>> In my case I used it for sip-sip connections only also, never had any
>>> E1's configured.
>>>
>>> Joel P
>>>
>>>
>>> On Tue, Jul 30, 2013 at 3:47 PM, Maciej Bylica <[email protected]> wrote:
>>>>
>>>> Hello,
>>>>
>>>> I want to know if i could use AS5400 for sip-to-sip connections only.
>>>> If possitive, could i just use incomig and outgoing voip dialpeers for
>>>> this purpose or i need to proceed with back-to-back E1 configuration?
>>>>
>>>> Of course i need to proxy RTP traffic as well.
>>>>
>>>> Thanks in advance for information.
>>>> Mac.
>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> [email protected]
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>
>>
>
> _______________________________________________
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> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> _______________________________________________
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>
------------------------------
Message: 2
Date: Wed, 31 Jul 2013 20:06:20 +0200
From: Maciej Bylica <[email protected]>
To: Peter Slow <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] AS5400 working as CUBE
Message-ID:
<CACynCY64=_rVhCLQF9Kw9+Y3EXUp05zWLFbL1V75yVr=uvc...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Thanks Mark and Peter for the hints.
I will try to get 15.x ios and DSP spec.
Mac.
2013/7/31 Peter Slow <[email protected]>
> I am fairly certain that at one point or another this platform allowed
> you to use the dspfarm config commands, but i dont rememeber what the
> specifics were.
>
> It would be helpful to know which DSPs you've got. you've also got a
> couple different options - even if you can't do all of the CUBE things
> you need to do on the 5400, you might (should) be able to get it to
> register as a DSP farm to some other device, either a CUBE or a CUCM.
>
> I haven't ever actually gotten one of those to register as a dspfarm
> before, but i'm willing to take a look with you if you're interested
> =)
>
> -Pete
>
> On Wed, Jul 31, 2013 at 9:12 AM, Mark Holloway <[email protected]>
> wrote:
> > I believe it was IOS 15.0(1)M which added SIP to SIP transcoding with
> CUBE.
> > You should be able to support this function on the AS5400 as long as
> you're
> > on a the right version. IOS 12.x releases do not support SIP to SIP
> > transcoding (documented here).
> >
> > On Jul 30, 2013, at 4:08 PM, Joel Perez <[email protected]> wrote:
> >
> > That was one of the areas we had trouble with. We couldn't get
> trans-coding
> > to work locally on the CUBE router. If i remember correctly we couldn't
> > register the resources locally or with CUCM. It has been a while but
> that is
> > what I remember off the top of my head.
> >
> > Joel P.
> >
> >
> > On Tue, Jul 30, 2013 at 4:04 PM, Maciej Bylica <[email protected]> wrote:
> >>
> >> Hi Joel,
> >>
> >> Thanks for prompt answer.
> >> Have you tried to use transcoding in your case?
> >>
> >> Thanks,
> >> Maciej.
> >>
> >>
> >> 2013/7/30 Joel Perez <[email protected]>
> >>>
> >>> Hi Mac,
> >>>
> >>> I haven't used an AS5400 for CUBE but I have used an AS5350. They are
> >>> somewhat similar so i would venture to say that you could.
> >>> In my case I used it for sip-sip connections only also, never had any
> >>> E1's configured.
> >>>
> >>> Joel P
> >>>
> >>>
> >>> On Tue, Jul 30, 2013 at 3:47 PM, Maciej Bylica <[email protected]>
> wrote:
> >>>>
> >>>> Hello,
> >>>>
> >>>> I want to know if i could use AS5400 for sip-to-sip connections only.
> >>>> If possitive, could i just use incomig and outgoing voip dialpeers for
> >>>> this purpose or i need to proceed with back-to-back E1 configuration?
> >>>>
> >>>> Of course i need to proxy RTP traffic as well.
> >>>>
> >>>> Thanks in advance for information.
> >>>> Mac.
> >>>>
> >>>> _______________________________________________
> >>>> cisco-voip mailing list
> >>>> [email protected]
> >>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>>
> >>>
> >>
> >
> > _______________________________________________
> > cisco-voip mailing list
> > [email protected]
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > [email protected]
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
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Message: 3
Date: Wed, 31 Jul 2013 14:25:41 -0500
From: Anthony Holloway <[email protected]>
To: Ryan Burtch <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Using DTMF # as input for Company Directory
w/ CVP and ICM
Message-ID:
<cacrcjoim5owjp2c9e-g+4zxu0_gfyx5cgjg_e1mwobx5mxk...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
I'm no UCCE person, but would it require the value to be enclosed in quotes?
Example: Call.user.microapp.caller_input == "#"
How does it look when you want to check equality for a numeric value?
Example: Call.user.microapp.caller_input == 1 OR
Call.user.microapp.caller_input == "1"
On Tue, Jul 30, 2013 at 10:39 AM, Ryan Burtch <[email protected]> wrote:
> All:
>
> I'm trying to use DTMF # as an input in a CVP Menu to direct callers to a
> company directory. I have already changed the VXML termchar value to A, in
> CVP Studio, but when I try to add Call.user.microapp.caller_input==# to my
> if statement in ICM, it gives me an error (see below).
>
> Does anyone know the proper config in ICM to get it to access the CED of #
> when returning from CVP?
>
>
>
> Sincerely,
>
> Ryan Burtch
>
> [image: Inline image 1]
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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------------------------------
Message: 4
Date: Thu, 1 Aug 2013 06:02:16 +0530
From: "Abdul Salam ." <[email protected]>
To: "cisco-voip ([email protected])"
<[email protected]>
Subject: [cisco-voip] TRAI regulation for call treatment over WAN
Message-ID:
<CAKav0XRnoySY0uiLn0wHpcizJJeTMgz9ZG2==20n=lhtj5j...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Is it ok to extend India PSTN calls over WAN for call treatment or VM?
*---AS*
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Message: 5
Date: Thu, 1 Aug 2013 12:46:55 +0000
From: Jean Carlos Coelho <[email protected]>
To: Jean Carlos Coelho <[email protected]>,
"[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Strange Issue CP-3905
Message-ID: <ce1fde53.225%[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi, just to keep the records..
I upgraded the firmware of the problematic phone, the results was fine,
then I applied this new firmware to all the phones.
Next.. :)
On 24/07/13 16:00, "Jean Carlos Coelho" <[email protected]>
wrote:
>i have an strange issue on cp-3905, when i do some calls to external
>numbers , it works ok, but when the call is from or to DN in the same
>network, the call is answered i can hear the dial tone in the
>conversation, but only one side can listen the dial tone at the
>conversation, only the side that make the call, someone got this problem
>too?
>
>Thank You
>
>
>
>
>
>
>_______________________________________________
>cisco-voip mailing list
>[email protected]
>https://puck.nether.net/mailman/listinfo/cisco-voip
>
------------------------------
Message: 6
Date: Thu, 1 Aug 2013 13:56:01 +0000
From: Jason Faraone <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] IRC
Message-ID:
<fb0beb07ca11e54a83b331ad1c263ed665d8f...@chq-mdf-mbx-01.paulo.com>
Content-Type: text/plain; charset="us-ascii"
Is there a voice specific IRC channel? #cisco on freenode has a few voice
admins but I was hoping to find a dedicated channel. If there isn't one, I'll
be idling in #voip on freenode.
------------------------------
Message: 7
Date: Thu, 1 Aug 2013 14:05:06 +0000
From: "Ryan Ratliff (rratliff)" <[email protected]>
To: Scott Voll <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Jabber status for phone forwarded?
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="iso-8859-1"
I'm not sure how you forward a user, but Jabber Mac doesn't show anything at
all in my status when I use it to set CFA.
-Ryan
On Jul 30, 2013, at 12:32 PM, Scott Voll <[email protected]> wrote:
What does Jabber show the status of a user, when the user is forwarded?
TIA
Scott
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cisco-voip mailing list
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------------------------------
Message: 8
Date: Thu, 1 Aug 2013 09:35:03 -0500
From: Erick Wellnitz <[email protected]>
To: cisco-voip <[email protected]>
Subject: [cisco-voip] fast busy when phone CFA via CUBE
Message-ID:
<CAK0wOsD=lya1wu+rdw+4wtcts6u+8gdnqshthtlnrm0mjon...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I have an issue which is probably very easy to fix but I think I'm making
it more complex than I need to.
I have a Nortel that we are migrating to CUCM 9.1 from. In the interim, we
have interated the two using SIP via CUBE. PSTN gateway is H323. I
wouldn't normally do this but decisions get made and we have to deal with
it.
Call from Nortel to IP Phone works fine. Call forward that same IP Phone
to a PSTN number and the call connects to the external number as if nothing
was wrong but the Nortel phone gets dropped with a 'fast busy'.
Traces and TAC (626731957) that my colleague opened seem to show a codec
/capabilities mismatch but I can't find a codec mismatch in the system at
all and our reporting/monitoring tool shows the call trying to connect as
G711u on both sides.
Is this a situation where an MTP is gong to be necessary or is there a way
to avoid mandatory MTP?
Thanks in advance!
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Message: 9
Date: Thu, 1 Aug 2013 09:48:26 -0500
From: Kenneth Hayes <[email protected]>
To: Erick Wellnitz <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] fast busy when phone CFA via CUBE
Message-ID: <3088492081936765650@unknownmsgid>
Content-Type: text/plain; charset=ISO-8859-1
CSS or MTP issue on the trunk?
Sent from my iPhone
On Aug 1, 2013, at 9:36 AM, Erick Wellnitz <[email protected]> wrote:
> I have an issue which is probably very easy to fix but I think I'm making it
> more complex than I need to.
>
> I have a Nortel that we are migrating to CUCM 9.1 from. In the interim, we
> have interated the two using SIP via CUBE. PSTN gateway is H323. I wouldn't
> normally do this but decisions get made and we have to deal with it.
>
> Call from Nortel to IP Phone works fine. Call forward that same IP Phone to
> a PSTN number and the call connects to the external number as if nothing was
> wrong but the Nortel phone gets dropped with a 'fast busy'.
>
> Traces and TAC (626731957) that my colleague opened seem to show a codec
> /capabilities mismatch but I can't find a codec mismatch in the system at all
> and our reporting/monitoring tool shows the call trying to connect as G711u
> on both sides.
>
> Is this a situation where an MTP is gong to be necessary or is there a way to
> avoid mandatory MTP?
>
> Thanks in advance!
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
Message: 10
Date: Thu, 1 Aug 2013 09:49:01 -0500
From: Kenneth Hayes <[email protected]>
To: Erick Wellnitz <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] fast busy when phone CFA via CUBE
Message-ID: <4413912057972518445@unknownmsgid>
Content-Type: text/plain; charset=ISO-8859-1
Do you have MTP configured on your IOS router?
Sent from my iPhone
On Aug 1, 2013, at 9:36 AM, Erick Wellnitz <[email protected]> wrote:
> I have an issue which is probably very easy to fix but I think I'm making it
> more complex than I need to.
>
> I have a Nortel that we are migrating to CUCM 9.1 from. In the interim, we
> have interated the two using SIP via CUBE. PSTN gateway is H323. I wouldn't
> normally do this but decisions get made and we have to deal with it.
>
> Call from Nortel to IP Phone works fine. Call forward that same IP Phone to
> a PSTN number and the call connects to the external number as if nothing was
> wrong but the Nortel phone gets dropped with a 'fast busy'.
>
> Traces and TAC (626731957) that my colleague opened seem to show a codec
> /capabilities mismatch but I can't find a codec mismatch in the system at all
> and our reporting/monitoring tool shows the call trying to connect as G711u
> on both sides.
>
> Is this a situation where an MTP is gong to be necessary or is there a way to
> avoid mandatory MTP?
>
> Thanks in advance!
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
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