[cisco-voip] [cisco-VoIP] SIP trunk one way audio

Buchanan, James jbuchanan at presidio.com
Thu Nov 11 15:31:11 EST 2010


The dial peers, everything under voice service, and the voice classes.

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: Bill Riley [mailto:bill at hitechconnection.net]
Sent: Thursday, November 11, 2010 2:25 PM
To: Buchanan, James; 'Mark Holloway'
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] [cisco-VoIP] SIP trunk one way audio

I could send parts. What do you want to see.

From: Buchanan, James [mailto:jbuchanan at presidio.com]
Sent: Thursday, November 11, 2010 2:20 PM
To: Bill Riley; 'Mark Holloway'
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] [cisco-VoIP] SIP trunk one way audio

Can you send us an IOS config?

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: Bill Riley [mailto:bill at hitechconnection.net]
Sent: Thursday, November 11, 2010 2:15 PM
To: 'Mark Holloway'
Cc: Buchanan, James; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] [cisco-VoIP] SIP trunk one way audio

Another internal extension.

From: Mark Holloway [mailto:mh at markholloway.com]
Sent: Thursday, November 11, 2010 2:08 PM
To: Bill Riley
Cc: 'Buchanan, James'; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] [cisco-VoIP] SIP trunk one way audio

Are you forwarding back to the PSTN or to another internal extension?


On Nov 11, 2010, at 11:48 AM, Bill Riley wrote:

Anyone else?

From: 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: Thursday, November 11, 2010 12:14 PM
To: 'Buchanan, James'
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] [cisco-VoIP] SIP trunk one way audio

Whatever the default setting. When I go into Voice-services-voip to configure it globally all I have is media forking.

From: Buchanan, James [mailto:jbuchanan at presidio.com]
Sent: Thursday, November 11, 2010 12:09 PM
To: Bill Riley
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] SIP trunk one way audio

Are you using media flow-around or flow-through?

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: Bill Riley [mailto:bill at hitechconnection.net]
Sent: Thursday, November 11, 2010 12:06 PM
To: Buchanan, James
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] SIP trunk one way audio

Sip

From: Buchanan, James [mailto:jbuchanan at presidio.com]
Sent: Thursday, November 11, 2010 12:05 PM
To: Bill Riley
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] SIP trunk one way audio

Is your trunk SIP or H323 into UCM?

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: Bill Riley [mailto:bill at hitechconnection.net]
Sent: Thursday, November 11, 2010 11:52 AM
To: Buchanan, James
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] SIP trunk one way audio

Well according to TAC if the call is sip to sip I shouldn't need the MTP at all.

I thought MTP required check box allocated the MTP if it needed one or not and leaving it unchecked  allocated the MTP if it needed to use one.

From: Buchanan, James [mailto:jbuchanan at presidio.com]
Sent: Thursday, November 11, 2010 11:49 AM
To: Bill Riley
Subject: RE: [cisco-voip] SIP trunk one way audio

Because it won't invoke the MTP without it.

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: Bill Riley [mailto:bill at hitechconnection.net]
Sent: Thursday, November 11, 2010 11:19 AM
To: Buchanan, James
Subject: RE: [cisco-voip] SIP trunk one way audio

No I don't have "MTP Required" Checked. The MTP's are set to only use G.711. I don't see why I need  to have MTP Required on this trunk.

From: Buchanan, James [mailto:jbuchanan at presidio.com]
Sent: Thursday, November 11, 2010 11:12 AM
To: Bill Riley
Subject: RE: [cisco-voip] SIP trunk one way audio

Dumb question-I assume you have "MTP Required" checked, are set to use only G711 to the MTP, and reset the SIP trunk after setting it? If you require G729, you will need to setup an IOS MTP. These do not use DSPs, but will allow other codecs besides G711.

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] On Behalf Of Bill Riley
Sent: Thursday, November 11, 2010 10:15 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] SIP trunk one way audio

I have a new SIP trunk terminating in a 2921 CUBE bundle. When I call in from the Trunk directly to an IP phone it works correctly. If I call from the trunk to IP phone and the IP phone transfers the call without waiting for the remote party to answer I get one way audio. From reading this looks like and MTP issue but I have an MTP set in the MRGL for the trunk.
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