Re: [OSL | CCIE_Voice] Calculation QOS sanity check

2008-12-15 Thread James Key
I would agree to round up also, but I would also suggest you clarify with the Proctor just to make sure. James From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shadab Abbasi (moabbasi) Sent: Sunday, December 14, 2008 8:38 PM To: Ryan

Re: [OSL | CCIE_Voice] ccm -siptrunk-cme-call cannt be forwarded to cue

2008-12-15 Thread saralilin2006
fyi debug ccsip info gives an error response code of 3456XX *Dec 16 02:19:12.607: //-1//SIP/Info/sipTcpQueueSendData: Data queued length: 369 *Dec 16 02:19:22.607: //35/C36D36D5802F/SIP/Info/ccsip_call_forward: Call forward target num *Dec 16 02:19:22.607:

[OSL | CCIE_Voice] ccm -siptrunk-cme-call cannt be forwarded to cue

2008-12-15 Thread saralilin2006
sip trunk question again. i have sip trunk between ccm and cme ccm ext 1001 unity subscriber cme ext 3001 cue subscriber -ccm call cue pilot is working -cme call ccm call forward to unity is working but ccm calling cue trunk is not working. the moment the call hit cue dial-peer call get

Re: [OSL | CCIE_Voice] SIP Analog phone fast busy

2008-12-15 Thread rob
Hi Ryan, Do you see a codec specified in SDP content of the INVITE sent to CM when calling from FXS to CM? Rob On Mon, Dec 15, 2008 at 12:45 AM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: SIP analog phone that hangs off an FXS port. I created a SIP trunk (MTP required) in CM. It

Re: [OSL | CCIE_Voice] Alias command not working in SRST

2008-12-15 Thread James Key
I don't believe Alias with work if you have dialplan-pattern defined. James -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of anil batra Sent: Monday, December 15, 2008 1:13 AM To: Ryan Trauernicht Cc:

Re: [OSL | CCIE_Voice] ccm -siptrunk-cme-call cannt be forwarded to cue

2008-12-15 Thread Vik Malhi
I don¹t think this call is supported in THIS release of IOS. -- Vik Malhi ­ CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities:

Re: [OSL | CCIE_Voice] SIP Analog phone fast busy

2008-12-15 Thread Vik Malhi
Bind SIP control and media to the IP defined in the SIP trunk: Voice service voip sip bind all Ensue that the FXS port is not registered to GK: Dial-peer voice xx port no e164 reg Ensure that the MTP is correctly configured (in a Device Pool which communicates using g711u). -- Vik

Re: [OSL | CCIE_Voice] ccm -siptrunk-cme-call cannt be forwarded to cue

2008-12-15 Thread saralilin2006
thanks vik for your reply. what is the exact feature not supported? is there any document from cisco? Sara Vik Malhi vma...@ipexpert.com wrote: I don’t think this call is supported in THIS release of IOS. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc.

[OSL | CCIE_Voice] Services gives blank screen

2008-12-15 Thread Chris Parker
Anyone ever run into the issue where you press services on the phone and all you get back is a blank screen? No http errors ... just blank. Chris

Re: [OSL | CCIE_Voice] Alias command not working in SRST

2008-12-15 Thread Jose Gregorio Linero (jlinero)
Hi: Alias commando will not work with non registered phones, you have to find another way. Regards, Jose -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of James Key Sent: Lunes, 15 de Diciembre de 2008 08:58

Re: [OSL | CCIE_Voice] Alias command not working in SRST

2008-12-15 Thread Shadab Abbasi (moabbasi)
It will work, provided it has a higher preference value than what defined under max-dn. Regards, Shadab -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jose Gregorio Linero (jlinero) Sent: Tuesday, December 16,

[OSL | CCIE_Voice] Site B to HQ Sip Trunk

2008-12-15 Thread Ryan Trauernicht
Trying to get Site B phone to call HQ Sip trunk that goes to an analog phone. I know the SIP trunk must support 711only, but I have it set to HQ device pool right now. 711 within HQ and 729 to Site B. I have a HW Xcoder at the HQ side to be invoked to transcode the call so it can be 729 across

Re: [OSL | CCIE_Voice] Alias command not working in SRST

2008-12-15 Thread anil batra
Hey Vik, High Five for your detailed kind responce on this. Appreciate it. Regards, Anil --- On Tue, 12/16/08, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] Alias command not working in SRST To: James Key j...@jackhenry.com,