[cisco-voip] Call Preserve SIP

[Jason Aarons (US)]

I tested this the other week with 8.0.3a and IOS 15.1.2T with a PRI and SIP Trunk back to CallManager.  I recall with MTP Required checked (always a bad idea) and a local IOS Software MTP begin the only registered MTP in MRG/MRGL the call would not stay up upon invoking SRST by simulating wan failure.  A good design should work to avoid/mitiigate MTP/Transcoding (assuming bandwidth is inexpensive).

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin Steinberg
Sent: Monday, December 20, 2010 6:03 PM
To: Bill Riley
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Preserve SIP

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<mailto: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> [mailto: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<mailto: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<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<mailto:jbuchanan at ctiusa.com>
D: 615-866-5729 | F: 615-866-5781 | www.presidio.com<http://www.presidio.com/>
CCIE #25863, Voice

From: cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net> [mailto: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<mailto: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<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<mailto:bill at hitechconnection.net>> wrote:
main up during the transition to SRST?  I do have MTP required o


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