[cisco-voip] Call Preserve SIP

[Justin Steinberg]

I'm not sure whether a MTP on the local device providing SRST would keep the
call up.   I tend to think it won't but that's just my opinion.  If it was
me dealing with this, I would upgrade CM to a version that has a fix for
that defect and then remove the MTP from the call path.

On Mon, Dec 20, 2010 at 8:05 AM, Bill Riley <bill at hitechconnection.net>wrote:

> Anyone else?  Anyway to keep the call up when switching to SRST mode?
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Bill Riley
> *Sent:* Friday, December 17, 2010 3:16 PM
> *To:* 'Buchanan, James'
>
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Call Preserve SIP
>
>
>
> Yes same bug different issue.  The bug causes me to need to use an MTP for
> transfers to work correctly even though I shouldn’t need one. My current
> problem is that when the WAN drops I lose my call to the SIP provider so I
> wonder if it is because I am using the software MTP on Call manager.
>
>
>
> *From:* Buchanan, James [mailto:jbuchanan at presidio.com]
> *Sent:* Friday, December 17, 2010 2:56 PM
> *To:* Bill Riley
> *Subject:* RE: [cisco-voip] Call Preserve SIP
>
>
>
> Isn’t that the one you hit a while back?
>
>
>
> *James Buchanan *|* Technology Manager, UC *| *South Region *|* Presidio
> Networked Solutions
> 12 Cadillac Dr, Suite 130, Brentwood, TN 37027 *|* **
> jbuchanan at presidio.com <jbuchanan at ctiusa.com>
> **D: 615-866-5729* | *F:* *615-866-5781* | *www.presidio.com*
>
> *CCIE #25863, Voice*
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Bill Riley
> *Sent:* Friday, December 17, 2010 2:41 PM
> *To:* 'Mike Lydick'
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Call Preserve SIP
>
>
>
> Call Manager bug
>
>
>
>
>
> CSCtb01167
>
>
>
>
>
> *CUCM fails to ACK the 200 OK after blind transfer over SIP trunk *
>
> *Symptom:*
>
> After blind transferring a call the transfer destination gets one-way voice
> and eventually the call disconnects
>
>
>
>
>
>
>
>
>
> *From:* Mike Lydick [mailto:mike.lydick at gmail.com]
> *Sent:* Friday, December 17, 2010 11:12 AM
> *To:* Bill Riley
> *Subject:* Re: [cisco-voip] Call Preserve SIP
>
>
>
> What is requiring the MTP? Is there an early offer or DTMF codec/media
> codec mismatch?
>
>
> Best Regards,
>
> Mike Lydick
>
> On Fri, Dec 17, 2010 at 11:24 AM, Bill Riley <bill at hitechconnection.net>
> wrote:
>
> main up during the transition to SRST?  I do have MTP required o
>
>
>
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>
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