Re: IVR

I would love to join.  What are conditions for joining this project
and how do I join?

Thanks

On Sun, Jan 29, 2012 at 10:42 PM, Joegen Baclor <jbaclor <at> ezuce.com> wrote:
> Roman,
>
> Join us in sipXcolab.  We can collaborate in person what you have in mind.
>  I will be working on a scripting engine for sipx in the hackfest.  I hope
> you can join.
>
> Joegen
>
>
> On 01/30/2012 08:37 AM, Roman Gelfand wrote:
>>
>> Actually, I have used speech server 2007.  I found it pretty decent.
>> However, I have reservations about writing an application which is
>> IDE, even version, dependent and environment is bloated and
>> inefficient.
>>
>> Thanks
>> On Sun, Jan 29, 2012 at 1:27 PM, Chris Rawlings<cm.rawlings <at> gmail.com>
>>  wrote:
>>>
>>> i would suggest using an external SIP Based IVR System that your forward
>>> calls to through SipXecs... register the IVR as an extension on the
>>> system... the system you create needs to be able to send calls back to
>>> SipXecs similar to a phone with blind transfers.

Re: IVR

(no subject)

No cdr or cse record for forwarded calls

Re: No cdr or cse record for forwarded calls

On Wed, Feb 1, 2012 at 10:17 AM, Luciano Berardi
<luciano.berardi <at> sip2ser.it> wrote:
>> Hi to everybody,
>>
>> let me begin saying that I'm new in sipXecs so I'm sorry if something is
>> not clear in my question or scenario-explanation.
>> My current objective is to retrieve cdr's information about different call
>> types (e.g. transferred, forwarded ..) and perform  some manipulation on
>> these (working on sipXecs release-4.4).
>>
>> Analyzing trace of forwarded calls I noticed that if the forwarding target
>> is external, there is no record in cse logging this forwarding handle and
>> showing the external called.
>> Furthermore when the call is forwarded, a CANCEL is sent to the user to be
>> forwarded and the call passes trough sipXbridge.
>> At this point a new call starts from sipXbridge to target external number;
>> the sipXbridge seems to talk with internal calling number using the old
>> internal as contact (the one CANCELLED) and talks with external number using
>> the original calling number.
>>
>> My questions are:
>> - Is this a correct behaviour?
>
No this is not correct behavior and I have also noticed it. (i.e.
forwarded calls do not show in CDR). I think a JIRA should be opened
in this case but only after checking the tracker to see if an issue
already exists.
>
>> - why in the cse (and cdr) there is no trace of the external called
>> contact?
>>
>> Thank you in advance to all.
>>
>> _______________________________________________
>> sipx-users mailing list
>> sipx-users <at> list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
>
> _______________________________________________
> sipx-users mailing list
> sipx-users <at> list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/

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Any gotchas with T.38 and Cisco SPA3102?

Re: Any gotchas with T.38 and Cisco SPA3102?

Re: Any gotchas with T.38 and Cisco SPA3102?

Tony,

That's what I thought. The SPA3102 fully supports T.38, although it 
looks like the provisioning profile in SipX does not include the 
options, so manual configuration is likely. Or at least partial manual.

Not too worried, as long as it will negotiate T.38 with the 
termination/origination provider (that properly supports T.38) and play 
nicely, then we're good.

Now the question is -- how can I have the ATA add a dialing prefix 
automatically so I can create a custom Fax dialplan in SipX... :D

-- Robert

On 2/1/2012 9:50 AM, Tony Graziano wrote:
> sipxbridge does not block the codec.
>
> at the same time the question is really a half question:
>
> if the ATA is configured to handle t.38, you can connect a fax machine 
> to it (and even register it to sipx if it is capable) and send a fax 
> from the plain paper fax machine to a unified messaging fax account on 
> sipx.
>
> At the same time, if the DID (PSTN call) points to the ATA registered 
> device and the call is from a sip trunk provider who properly handles 
> t.38 (i.e., it works to a unified messaging account on sipx) the fax 
> machine will also handle the inbound fax. It also would not matter if 
> the PSTN gateway was a POTS line or PRI, as long as the PSTN facing 
> gateway handles t.38 properly.
>
> I cannot attest to the model you are using. We do this with Patton 
> gateways/PRI and FXS devices and it just always works.
>
> The ATA WILL ABSOLUTLEY INTERFERE if it is not t.38 capable or 
> configured properly.  Also, since the inbound or outbound call has to 
> negotiate a codec, the UA must support t.38. Sipx DOES NOT transcode 
> this. It has to be natively available and offered on both ends of the 
> call.

MOH and Call Park (4.4 latest)

I have uploaded music on hold to a system for both MOH and Park. I
have set the user to use "Use System Configuration".

When a call is transferred they hear the MOH. When the user presses
the HOLD button, no MOH is heard. When a call is parked they hear the
correct MOH, but on timeout and during the "transfer back" process
they hear the system (shipping) default MOH which is not what I think
they should hear (I think they should hear the chosen MOH system file
that was uploaded, and would normally hear during a transfer).

Can anyone else confirm this behavior?

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Re: Any gotchas with T.38 and Cisco SPA3102?

We do this with the Patton's via a dialplan entry but I don't mess
with the Cisco/Linksys stuff so I can't say how flexible that is. They
do have manuals and forums for this kind of thing and you will
probably find your answer pretty quickly in their forums.

Another route I've tried with mixed results... put the gateway and the
ATA/User in a different branch and try to get the branch to handle the
gateway permissions. It looks like this is "broken" or not finished
yet though, and there are JIRA's on it.

On Wed, Feb 1, 2012 at 10:58 AM, Robert B <devo <at> spudland.com> wrote:
> Tony,
>
> That's what I thought. The SPA3102 fully supports T.38, although it looks
> like the provisioning profile in SipX does not include the options, so
> manual configuration is likely. Or at least partial manual.
>
> Not too worried, as long as it will negotiate T.38 with the
> termination/origination provider (that properly supports T.38) and play
> nicely, then we're good.
>
> Now the question is -- how can I have the ATA add a dialing prefix
> automatically so I can create a custom Fax dialplan in SipX... :D
>
> -- Robert
>
>
>
>
> On 2/1/2012 9:50 AM, Tony Graziano wrote:
>>
>> sipxbridge does not block the codec.
>>
>> at the same time the question is really a half question:
>>
>> if the ATA is configured to handle t.38, you can connect a fax machine to
>> it (and even register it to sipx if it is capable) and send a fax from the
>> plain paper fax machine to a unified messaging fax account on sipx.
>>
>> At the same time, if the DID (PSTN call) points to the ATA registered
>> device and the call is from a sip trunk provider who properly handles t.38
>> (i.e., it works to a unified messaging account on sipx) the fax machine will
>> also handle the inbound fax. It also would not matter if the PSTN gateway
>> was a POTS line or PRI, as long as the PSTN facing gateway handles t.38
>> properly.
>>
>> I cannot attest to the model you are using. We do this with Patton
>> gateways/PRI and FXS devices and it just always works.
>>
>> The ATA WILL ABSOLUTLEY INTERFERE if it is not t.38 capable or configured
>> properly.  Also, since the inbound or outbound call has to negotiate a
>> codec, the UA must support t.38. Sipx DOES NOT transcode this. It has to be
>> natively available and offered on both ends of the call.
>
>

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Telephone: 434.984.8430
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~~~~~~~~~~~~~~~~~~
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