[cisco-voip] CCM 5.11 vs. CME 4.1 h323 RTP stream fails - seems intermittent
Ryan Ratliff
rratliff at cisco.com
Tue Apr 10 14:39:17 EDT 2007
Are you using an MTP with the CME? With it added as an ICT you
should have the "MTP Required" set but if using it as an H.323
gateway an MTP is not required.
one-way or no-way audio is a routing issue. If the signaling was
incorrect then the call would fail at media negotiation. Check the
StartMediaTransmission sent to each phone to see what IP address it
is trying to send RTP to. Then make sure that address is routable
from that phone's subnet.
If your firewall is doing NAT then you should have H.323 fixup
enabled. If no NAT then make sure fixup is disabled.
-Ryan
On Apr 10, 2007, at 1:52 PM, Kristján Ólafur Eðvarðsson wrote:
Has anyone experienced RTP streaming issues or perhaps Call Control
issued between CME 4.x and CCM 5.1x ?
My issue has the background that initially 2 CME where connected via
h.323 without any such problems. But when changing one out for a CCM
5.1 platform
and configuring the CME as a h323 gateway (even tried what some
recommend to configure the CME as a Intercluster-trunk (none-
gatekeeper controlled) on the CCM 5.1 site. But The problem where
similar.
The problem description is that when a CCM 5.1 user (phone) calls CME
phone via h.323 gateway (CME on other end) there is ringing tones
e.t.c but
when offhook the voice is not heard in both directions. This may work
when the CME user dials back, sometimes it doesn't. I have seen logs
in the
ASA firewall at one of the location that rtp session is being
generated and a connection is started.
Any service parameters on CCM 5 on the h323 that need changing or are
there any suggestions ?
regards. Kristjan Edvardsson
Sensa ehf.
Cisco Silver Partner - Iceland
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of cisco-voip-request at puck.nether.net
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Today's Topics:
1. IPCC - Can Users Change Prompt (DUNCAN, W B.)
2. PIN-Assignment for RJ21-Connector on VG224 (J?rg Wesely)
3. Re: FXO disconnect problem (Ahmed Elnagar)
4. Re: FXO disconnect problem Correction (Ahmed Elnagar)
----------------------------------------------------------------------
Message: 1
Date: Tue, 10 Apr 2007 10:47:08 -0500
From: "DUNCAN, W B." <duncanw at otc.edu>
Subject: [cisco-voip] IPCC - Can Users Change Prompt
To: <cisco-voip at puck.nether.net>
Message-ID: <06D1B6D4926222458F803D0D3EDCCB7E01BA565C at EXM1.otc.edu>
Content-Type: text/plain; charset="us-ascii"
I am relatively new to IPCC 4.0, and have created a script for use by
our help desk. It is a basic queue script which starts by playing a
file which says, "Thank you for calling the OTC help desk..." It then
either transfers callers to an available agent or puts them in the
queue. This would be the normal greeting, but during outages, it would
be nice if the helpdesk could re-record this to say, "Thank you for
calling the help desk, we are currently experiencing trouble with system
X...", then when the crisis is over easily change it back (sort of like
alternate greetings in Unity). I can't figure out have to do this, does
anyone have any suggestions?
Thanks,
W. Brian Duncan
Coordinator of Telecommunications
Ozarks Technical Community College
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Message: 2
Date: Tue, 10 Apr 2007 18:47:23 +0200
From: J?rg Wesely <joerg.wesely at intact-is.com>
Subject: [cisco-voip] PIN-Assignment for RJ21-Connector on VG224
To: cisco-voip at puck.nether.net
Message-ID: <evgf2v$2k0$1 at sea.gmane.org>
Content-Type: text/plain; charset=ISO-8859-15; format=flowed
Hello,
we've got here a VG224. Does anyone know the PIN-Assignment of the RJ-21
connector?
thanks in advance
J?rg
------------------------------
Message: 3
Date: Tue, 10 Apr 2007 19:00:00 +0200
From: "Ahmed Elnagar" <aelnagar at ACT-EG.COM>
Subject: Re: [cisco-voip] FXO disconnect problem
To: "Steve G" <smgustafson at gmail.com>, "ciscovoip"
<cisco-voip at puck.nether.net>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED889FD2 at mail.ACT-EG.COM>
Content-Type: text/plain; charset="iso-8859-1"
Groud start
Thanks and Best Regards
Ahmed A. Elnagar
Network Field Engineer
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
Postal Code:12411 Cairo Egypt
Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com
________________________________
From: Steve G [mailto:smgustafson at gmail.com]
Sent: Tue 10-Apr-07 5:41 PM
To: Ahmed Elnagar; ciscovoip
Subject: Re: [cisco-voip] FXO disconnect problem
What signalling method is the FXO port using? Loop-start? Ground-
start?
On 4/10/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
Hello all;
I have a 2811 Router with FXO ports (VIC2-4FXO) and I have the
famous disconnect problem. I have tried all methods for disconnect
but with no help seems that I will go for the voice class
configuration but the problem is that I don't know who to measure the
frequency and the cadence of the disconnect tone that is sent from
the Provider, anyone knows a tool to do this. For more details see
the Problem description below:
When calling from PSTN through FXO port to inside IP Phones and
without the IP Phone answerers the call, I disconnect the call from
the PSTN side the IP Phone still rings until the timeout for Ringing
finishes and it stops ringing and the port on the gateway returns to
idle state.
When calling from PSTN through FXO port to inside IP Phones and with
the IP Phone answerers the call, I disconnect the call from the PSTN
side I hear the disconnect call playing on the IP Phone (that is why
I am sure that this the method used for disconnect, and I tried at a
different site that is not having this problem and I heard no
disconnect call) then when the disconnect call plays for a couple of
seconds the call still up for no limit.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 4
Date: Tue, 10 Apr 2007 19:22:11 +0200
From: "Ahmed Elnagar" <aelnagar at ACT-EG.COM>
Subject: Re: [cisco-voip] FXO disconnect problem Correction
To: "Ahmed Elnagar" <aelnagar at ACT-EG.COM>, "Steve G"
<smgustafson at gmail.com>, "ciscovoip" <cisco-voip at puck.nether.net>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED889FD3 at mail.ACT-EG.COM>
Content-Type: text/plain; charset="iso-8859-1"
Sorry Loop start
Thanks and Best Regards
Ahmed A. Elnagar
Network Field Engineer
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
Postal Code:12411 Cairo Egypt
Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com
________________________________
From: cisco-voip-bounces at puck.nether.net on behalf of Ahmed Elnagar
Sent: Tue 10-Apr-07 7:00 PM
To: Steve G; ciscovoip
Subject: Re: [cisco-voip] FXO disconnect problem
Groud start
Thanks and Best Regards
Ahmed A. Elnagar
Network Field Engineer
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
Postal Code:12411 Cairo Egypt
Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com
________________________________
From: Steve G [mailto:smgustafson at gmail.com]
Sent: Tue 10-Apr-07 5:41 PM
To: Ahmed Elnagar; ciscovoip
Subject: Re: [cisco-voip] FXO disconnect problem
What signalling method is the FXO port using? Loop-start? Ground-
start?
On 4/10/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
Hello all;
I have a 2811 Router with FXO ports (VIC2-4FXO) and I have the
famous disconnect problem. I have tried all methods for disconnect
but with no help seems that I will go for the voice class
configuration but the problem is that I don't know who to measure the
frequency and the cadence of the disconnect tone that is sent from
the Provider, anyone knows a tool to do this. For more details see
the Problem description below:
When calling from PSTN through FXO port to inside IP Phones and
without the IP Phone answerers the call, I disconnect the call from
the PSTN side the IP Phone still rings until the timeout for Ringing
finishes and it stops ringing and the port on the gateway returns to
idle state.
When calling from PSTN through FXO port to inside IP Phones and with
the IP Phone answerers the call, I disconnect the call from the PSTN
side I hear the disconnect call playing on the IP Phone (that is why
I am sure that this the method used for disconnect, and I tried at a
different site that is not having this problem and I heard no
disconnect call) then when the disconnect call plays for a couple of
seconds the call still up for no limit.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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