Re: [cisco-voip] how codec transparent works?

2015-05-10 Thread s m
hello Ryan,

thank you for your reply. without codec transparent, my phone rings but
when i answer i have no voice and it hangs up after 5 seconds. asterisk
says no answer. this is so strange for me. i think media negotiation
failed, right? is there any hint to have h323 trunk to asterisk with
specific codec (not transparent one)???

On Sun, May 10, 2015 at 4:32 PM, Ryan Huff ryanh...@outlook.com wrote:

 Codec transparent just passes sdp through to the other call leg without
 trying to do media negotiations.

 So without codec transparent,  what happens?
 Thanks,

 Ryan


  Original Message 
 From: s m sam.gh1...@gmail.com
 Sent: Sunday, May 10, 2015 01:19 AM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] how codec transparent works?

 hello everybody,

 anybody knows how codec transparent works?

 i have a strange problem. i want to set h323 trunk between asterisk and
 cisco 2800. it only works when i set codec transparent in dial-peer nodes.
 show commands in cisco shows that i have a call with g711alaw but if i set
 codec g711alaw in dial-peers, i do not have any success call. i know it is
 codec compatibility problem. is there any difference between g711 codecs
 which cisco and asterisk utilize? what happened when codec is set to
 transparent? dose anyone know anything about it?

 thanks is advance
 SAM

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Re: [cisco-voip] RTMT??? or some other way to monitor a phone

2015-05-10 Thread Justin Steinberg
RTMT has a device search that let's you filter by device pool or subnet.
On May 10, 2015 7:38 PM, Abhiram Kramadhati (akramadh) akram...@cisco.com
wrote:

  This might be helpful: https://www.singlewire.com/free-xml-apps.html


   From: Scott Voll svoll.v...@gmail.com
 Date: Saturday, 9 May 2015 12:24 am
 To: Brian Meade bmead...@vt.edu
 Cc: cisco-voip@puck.nether.net cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] RTMT??? or some other way to monitor a phone

   I was really hoping not to pull SDL traces (more work) and I was hoping
 just to look at a small segment of phones not everything.

  Guess that answers my question.

  Scott



 On Thu, May 7, 2015 at 5:19 PM, Brian Meade bmead...@vt.edu wrote:

 Using the Syslog Viewer in RTMT, open the AlternateSyslog for the primary
 subscriber the phones are registered to.  That will show all phone
 registrations/unregistrations.

  On Thu, May 7, 2015 at 5:42 PM, Scott Voll svoll.v...@gmail.com wrote:

  we have 8861 phones that are just below an AP that are registering and
 unregistering (but working well enough that we have not moved them back to
 wired) so we are trying to monitor how much it's happening.  Already have a
 TAC case open on the wireless side.

  Scott


 On Thu, May 7, 2015 at 2:38 PM, Ryan Huff ryanh...@outlook.com wrote:

 Might be easier to start the other way around.

 What is actually wrong?

 RTMT does work with registered phones, and presumably,  your wireless
 phones are registered?

 Are the wireless phones not registering to call manager?

 Thanks,

 Ryan


  Original Message 
 From: Scott Voll svoll.v...@gmail.com
 Sent: Thursday, May 7, 2015 05:34 PM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] RTMT??? or some other way to monitor a phone

 We are troubleshooting a wireless deployment of 8861 phones.

  is there any way to monitor just a phone / phone model / device pool
 / etc so I can just see what I'm looking for?

  Looks like RTMT is just all registered phones.

  TIA

  Scott



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Re: [cisco-voip] how codec transparent works?

2015-05-10 Thread Ryan Huff
Codec transparent just passes sdp through to the other call leg without trying 
to do media negotiations. 

So without codec transparent,  what happens?
Thanks,

Ryan

 Original Message 
From: s m sam.gh1...@gmail.com
Sent: Sunday, May 10, 2015 01:19 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] how codec transparent works?

hello everybody,

anybody knows how codec transparent works?

i have a strange problem. i want to set h323 trunk between asterisk and
cisco 2800. it only works when i set codec transparent in dial-peer nodes.
show commands in cisco shows that i have a call with g711alaw but if i set
codec g711alaw in dial-peers, i do not have any success call. i know it is
codec compatibility problem. is there any difference between g711 codecs
which cisco and asterisk utilize? what happened when codec is set to
transparent? dose anyone know anything about it?

thanks is advance
SAM

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