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Today's Topics:
1. error message IVR-3-APP_ERR:**** CVP HANDOFF.TCL
(FRANKLYN GONZALEZ)
2. AUTO: Oluwatosin Odubanjo is out of the office (returning
07/12/2013) ([email protected])
3. Re: error message IVR-3-APP_ERR:**** CVP HANDOFF.TCL (Peter Slow)
4. FXS Port one way audio - no dtmf (Aaron Blair)
5. Re: FXS Port one way audio - no dtmf (Aaron Blair)
6. Re: error message IVR-3-APP_ERR:**** CVP HANDOFF.TCL (Pavan K)
7. Re: FXS Port one way audio - no dtmf (Aaron Blair)
8. Polycom Analog or IP (David Zhars)
9. Re: Polycom Analog or IP (Jason Faraone)
10. Re: Polycom Analog or IP (Ryan Ratliff)
11. Re: Polycom Analog or IP (Erick Wellnitz)
12. Re: SIP between CUCM clusters (Russell Chaseling)
13. Re: SIP between CUCM clusters (Erick Wellnitz)
----------------------------------------------------------------------
Message: 1
Date: Thu, 11 Jul 2013 18:00:17 +0000
From: FRANKLYN GONZALEZ <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] error message IVR-3-APP_ERR:**** CVP HANDOFF.TCL
Message-ID: <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"
I have a Gateway which presents the following message in log I have no
affectation of service but I wonder if they know that this message occurs
*Jul 11 17:46:52.765: %IVR-3-APP_ERR:**** CVP HANDOFF.TCL:
AF8028B2.E98811E2.B9318497.5F57BD47 abnormally disconnected with code 38. ****
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Message: 2
Date: Thu, 11 Jul 2013 19:27:37 +0100
From: [email protected]
To: [email protected]
Subject: [cisco-voip] AUTO: Oluwatosin Odubanjo is out of the office
(returning 07/12/2013)
Message-ID:
<OFB4DBA0AF.0F9FEE44-ON41257BA5.00656804-41257BA5.00656806@LocalDomain>
Content-Type: text/plain; charset="us-ascii"
I will be out of the office starting 07/11/2013 and will not return until
07/12/2013
Dear All,
I will be away from office starting from today 11th July to 12th July. I
will resume on Monday 15th July. In my absence, Obiajulu Francis will be
responsible for blackberry/cisco ip phones/tablets/Glo credit issues. You
can reach him on 1010.
Note: This is an automated response to your message "cisco-voip Digest,
Vol 117, Issue 9" sent on 7/11/2013 5:00:11 PM.
This is the only notification you will receive while this person is
away._____________________Disclaimer________________________
The information transmitted is intended only for the person or entity
to which it is addressed and may contain confidential and/or
privileged material. Any review, retransmission, dissemination or
other use of, or taking of any action in reliance upon, this information
by persons or entities other than the intended recipient is prohibited.
If you received this in error, please contact the sender and delete
the material from any computer.
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Message: 3
Date: Thu, 11 Jul 2013 14:58:08 -0500
From: Peter Slow <[email protected]>
To: FRANKLYN GONZALEZ <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] error message IVR-3-APP_ERR:**** CVP
HANDOFF.TCL
Message-ID:
<CAMa5Jw4T2kYN_585Zvg_gU9xe9mdJRigU=tys=ndcaus6ac...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
you most likely do have an affectation of service, that pretty much
means you dropped a call, sir.
you are gonna need some stuff from CVP... You are in for some fun!
-Pete
On Thu, Jul 11, 2013 at 1:00 PM, FRANKLYN GONZALEZ
<[email protected]> wrote:
>
> I have a Gateway which presents the following message in log I have no
> affectation of service but I wonder if they know that this message occurs
>
>
> *Jul 11 17:46:52.765: %IVR-3-APP_ERR:**** CVP HANDOFF.TCL:
> AF8028B2.E98811E2.B9318497.5F57BD47 abnormally disconnected with code 38.
> ****
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
------------------------------
Message: 4
Date: Thu, 11 Jul 2013 22:40:42 +0000
From: Aaron Blair <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] FXS Port one way audio - no dtmf
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi all,
I'm having some issues with a basic FXS port setup.. We've tried two
phones/brands (Uniden + NEC (wasn't really expecting NEC to work)) but still
seeing the same thing.
voice-port 0/1/0
disconnect-ack
disc_pi_off
cptone AU
timeouts initial 12
timeouts interdigit 8
timeouts call-disconnect 40
description Uniden Test
station-id name Test Name
station-id number 1234567
caller-id enable
Voice port config as above. Router is 2921 with 1x VWIC3-1MFT-T1/E1, 2x
VIC3-2FXS/DID, 1x PVDM3-64, 1x ISM-SRE-300-K9.
It's a CME setup, with voip phones all registered and talking just fine.
What we're seeing is the phone on the FXS port is receiving dialtone, but any
attempts to dial digits do not break the dial-tone, and the router does not see
any digits being dialled.
When calling the FXS port from a VOIP phone, the FXS port rings, and when
picked up there is one way audio from the voip phone to the FXS port, no audio
from the FXS port to the voip phone. The voip phone appears to be receiving RTP
just fine from the gateway.
Phone we're currently testing is a Uniden DECT2035, but have tried NEC phones
as well.
Another possibly unrelated issue is that we're seeing one way audio on the E1.
Outbound voice is working fine (VOIP to isdn), but inbound is not (ISDN to
voip). Also, when dialing out from voip phone, there is no ringing on the voip
phone, it just connects when picked up.
Any help would be much appreciated.
Cheers,
Aaron Blair
Network Engineer
Bridge Point Communications
Direct: +61 7 3231 5458<tel:+61732315458>
Office: +61 7 3231 5444<+61732315444>
Support: +61 7 3231 5422<tel:+61732315422>
Fax: +61 7 3231 5411
IM: [email protected]<xmpp:[email protected]>
Your Information and Data Security Specialists
ISO27001 Certified & PCI Qualified Security Assessor (QSA)
[Description: Description: Description: Description: Description: Description:
Description: Description: Description: cid:[email protected]]
This email and any attachments are subject to copyright. They may also contain
confidential information. This email and any attachments may not be
distributed, reproduced, copied, stored or transmitted in any form or by any
means, without the prior written consent of Bridge Point Communications Pty Ltd
ABN 29 083 424 668. Any personal information in this email must be handled in
accordance with the Privacy Act 1988 (Cth). Emails may be interfered with, may
contain computer viruses or other defects and may not be successfully
replicated on other systems. Bridge Point Communications Pty Ltd gives no
warranties in relation to these matters. If you have any doubts about the
authenticity of an email purportedly sent by us, please contact Bridge Point
Communications Pty Ltd immediately.
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------------------------------
Message: 5
Date: Thu, 11 Jul 2013 22:42:27 +0000
From: Aaron Blair <[email protected]>
To: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] FXS Port one way audio - no dtmf
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Also, code version on the router is 15.2(4)M2
Aaron Blair
Network Engineer
Bridge Point Communications
Direct: +61 7 3231 5458<tel:+61732315458>
Office: +61 7 3231 5444<+61732315444>
Support: +61 7 3231 5422<tel:+61732315422>
Fax: +61 7 3231 5411
IM: [email protected]<xmpp:[email protected]>
Your Information and Data Security Specialists
ISO27001 Certified & PCI Qualified Security Assessor (QSA)
[Description: Description: Description: Description: Description: Description:
Description: Description: Description: cid:[email protected]]
This email and any attachments are subject to copyright. They may also contain
confidential information. This email and any attachments may not be
distributed, reproduced, copied, stored or transmitted in any form or by any
means, without the prior written consent of Bridge Point Communications Pty Ltd
ABN 29 083 424 668. Any personal information in this email must be handled in
accordance with the Privacy Act 1988 (Cth). Emails may be interfered with, may
contain computer viruses or other defects and may not be successfully
replicated on other systems. Bridge Point Communications Pty Ltd gives no
warranties in relation to these matters. If you have any doubts about the
authenticity of an email purportedly sent by us, please contact Bridge Point
Communications Pty Ltd immediately.
From: cisco-voip [mailto:[email protected]] On Behalf Of Aaron
Blair
Sent: Friday, 12 July 2013 8:41 AM
To: [email protected]
Subject: [cisco-voip] FXS Port one way audio - no dtmf
Hi all,
I'm having some issues with a basic FXS port setup.. We've tried two
phones/brands (Uniden + NEC (wasn't really expecting NEC to work)) but still
seeing the same thing.
voice-port 0/1/0
disconnect-ack
disc_pi_off
cptone AU
timeouts initial 12
timeouts interdigit 8
timeouts call-disconnect 40
description Uniden Test
station-id name Test Name
station-id number 1234567
caller-id enable
Voice port config as above. Router is 2921 with 1x VWIC3-1MFT-T1/E1, 2x
VIC3-2FXS/DID, 1x PVDM3-64, 1x ISM-SRE-300-K9.
It's a CME setup, with voip phones all registered and talking just fine.
What we're seeing is the phone on the FXS port is receiving dialtone, but any
attempts to dial digits do not break the dial-tone, and the router does not see
any digits being dialled.
When calling the FXS port from a VOIP phone, the FXS port rings, and when
picked up there is one way audio from the voip phone to the FXS port, no audio
from the FXS port to the voip phone. The voip phone appears to be receiving RTP
just fine from the gateway.
Phone we're currently testing is a Uniden DECT2035, but have tried NEC phones
as well.
Another possibly unrelated issue is that we're seeing one way audio on the E1.
Outbound voice is working fine (VOIP to isdn), but inbound is not (ISDN to
voip). Also, when dialing out from voip phone, there is no ringing on the voip
phone, it just connects when picked up.
Any help would be much appreciated.
Cheers,
Aaron Blair
Network Engineer
Bridge Point Communications
Direct: +61 7 3231 5458<tel:+61732315458>
Office: +61 7 3231 5444<+61732315444>
Support: +61 7 3231 5422<tel:+61732315422>
Fax: +61 7 3231 5411
IM: [email protected]<xmpp:[email protected]>
Your Information and Data Security Specialists
ISO27001 Certified & PCI Qualified Security Assessor (QSA)
[Description: Description: Description: Description: Description: Description:
Description: Description: Description: cid:[email protected]]
This email and any attachments are subject to copyright. They may also contain
confidential information. This email and any attachments may not be
distributed, reproduced, copied, stored or transmitted in any form or by any
means, without the prior written consent of Bridge Point Communications Pty Ltd
ABN 29 083 424 668. Any personal information in this email must be handled in
accordance with the Privacy Act 1988 (Cth). Emails may be interfered with, may
contain computer viruses or other defects and may not be successfully
replicated on other systems. Bridge Point Communications Pty Ltd gives no
warranties in relation to these matters. If you have any doubts about the
authenticity of an email purportedly sent by us, please contact Bridge Point
Communications Pty Ltd immediately.
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------------------------------
Message: 6
Date: Thu, 11 Jul 2013 18:58:12 -0500
From: Pavan K <[email protected]>
To: FRANKLYN GONZALEZ <[email protected]>
Cc: [email protected]
Subject: Re: [cisco-voip] error message IVR-3-APP_ERR:**** CVP
HANDOFF.TCL
Message-ID:
<cajdpbuu8roqyc5ckbfz_deg45fdpyn+vsfvt06jcw717tzc...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
If you are using CVP I would probably pull logs from your call server to
see what the heck happened to the call.
If you are not using cvp, time to cleanup your vxml apps (bootstrap and
others) on the gateway config
On Jul 11, 2013 1:03 PM, "FRANKLYN GONZALEZ" <[email protected]> wrote:
>
> I have a Gateway which presents the following message in log I have no
> affectation of service but I wonder if they know that this message occurs
>
>
> *Jul 11 17:46:52.765: %IVR-3-APP_ERR:**** CVP HANDOFF.TCL:
> AF8028B2.E98811E2.B9318497.5F57BD47 abnormally disconnected with code 38.
> ****
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 7
Date: Fri, 12 Jul 2013 06:25:38 +0000
From: Aaron Blair <[email protected]>
To: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] FXS Port one way audio - no dtmf
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
I fixed all the listed issues by relocating the PVDM from slot 0 to slot 1.
Cheers,
Aaron Blair
Network Engineer
Bridge Point Communications
Direct: +61 7 3231 5458<tel:+61732315458>
Office: +61 7 3231 5444<+61732315444>
Support: +61 7 3231 5422<tel:+61732315422>
Fax: +61 7 3231 5411
IM: [email protected]<xmpp:[email protected]>
Your Information and Data Security Specialists
ISO27001 Certified & PCI Qualified Security Assessor (QSA)
[Description: Description: Description: Description: Description: Description:
Description: Description: Description: cid:[email protected]]
This email and any attachments are subject to copyright. They may also contain
confidential information. This email and any attachments may not be
distributed, reproduced, copied, stored or transmitted in any form or by any
means, without the prior written consent of Bridge Point Communications Pty Ltd
ABN 29 083 424 668. Any personal information in this email must be handled in
accordance with the Privacy Act 1988 (Cth). Emails may be interfered with, may
contain computer viruses or other defects and may not be successfully
replicated on other systems. Bridge Point Communications Pty Ltd gives no
warranties in relation to these matters. If you have any doubts about the
authenticity of an email purportedly sent by us, please contact Bridge Point
Communications Pty Ltd immediately.
From: cisco-voip [mailto:[email protected]] On Behalf Of Aaron
Blair
Sent: Friday, 12 July 2013 8:42 AM
To: [email protected]
Subject: Re: [cisco-voip] FXS Port one way audio - no dtmf
Also, code version on the router is 15.2(4)M2
Aaron Blair
Network Engineer
Bridge Point Communications
Direct: +61 7 3231 5458<tel:+61732315458>
Office: +61 7 3231 5444<+61732315444>
Support: +61 7 3231 5422<tel:+61732315422>
Fax: +61 7 3231 5411
IM: [email protected]<xmpp:[email protected]>
Your Information and Data Security Specialists
ISO27001 Certified & PCI Qualified Security Assessor (QSA)
[Description: Description: Description: Description: Description: Description:
Description: Description: Description: cid:[email protected]]
This email and any attachments are subject to copyright. They may also contain
confidential information. This email and any attachments may not be
distributed, reproduced, copied, stored or transmitted in any form or by any
means, without the prior written consent of Bridge Point Communications Pty Ltd
ABN 29 083 424 668. Any personal information in this email must be handled in
accordance with the Privacy Act 1988 (Cth). Emails may be interfered with, may
contain computer viruses or other defects and may not be successfully
replicated on other systems. Bridge Point Communications Pty Ltd gives no
warranties in relation to these matters. If you have any doubts about the
authenticity of an email purportedly sent by us, please contact Bridge Point
Communications Pty Ltd immediately.
From: cisco-voip [mailto:[email protected]] On Behalf Of Aaron
Blair
Sent: Friday, 12 July 2013 8:41 AM
To: [email protected]<mailto:[email protected]>
Subject: [cisco-voip] FXS Port one way audio - no dtmf
Hi all,
I'm having some issues with a basic FXS port setup.. We've tried two
phones/brands (Uniden + NEC (wasn't really expecting NEC to work)) but still
seeing the same thing.
voice-port 0/1/0
disconnect-ack
disc_pi_off
cptone AU
timeouts initial 12
timeouts interdigit 8
timeouts call-disconnect 40
description Uniden Test
station-id name Test Name
station-id number 1234567
caller-id enable
Voice port config as above. Router is 2921 with 1x VWIC3-1MFT-T1/E1, 2x
VIC3-2FXS/DID, 1x PVDM3-64, 1x ISM-SRE-300-K9.
It's a CME setup, with voip phones all registered and talking just fine.
What we're seeing is the phone on the FXS port is receiving dialtone, but any
attempts to dial digits do not break the dial-tone, and the router does not see
any digits being dialled.
When calling the FXS port from a VOIP phone, the FXS port rings, and when
picked up there is one way audio from the voip phone to the FXS port, no audio
from the FXS port to the voip phone. The voip phone appears to be receiving RTP
just fine from the gateway.
Phone we're currently testing is a Uniden DECT2035, but have tried NEC phones
as well.
Another possibly unrelated issue is that we're seeing one way audio on the E1.
Outbound voice is working fine (VOIP to isdn), but inbound is not (ISDN to
voip). Also, when dialing out from voip phone, there is no ringing on the voip
phone, it just connects when picked up.
Any help would be much appreciated.
Cheers,
Aaron Blair
Network Engineer
Bridge Point Communications
Direct: +61 7 3231 5458<tel:+61732315458>
Office: +61 7 3231 5444<+61732315444>
Support: +61 7 3231 5422<tel:+61732315422>
Fax: +61 7 3231 5411
IM: [email protected]<xmpp:[email protected]>
Your Information and Data Security Specialists
ISO27001 Certified & PCI Qualified Security Assessor (QSA)
[Description: Description: Description: Description: Description: Description:
Description: Description: Description: cid:[email protected]]
This email and any attachments are subject to copyright. They may also contain
confidential information. This email and any attachments may not be
distributed, reproduced, copied, stored or transmitted in any form or by any
means, without the prior written consent of Bridge Point Communications Pty Ltd
ABN 29 083 424 668. Any personal information in this email must be handled in
accordance with the Privacy Act 1988 (Cth). Emails may be interfered with, may
contain computer viruses or other defects and may not be successfully
replicated on other systems. Bridge Point Communications Pty Ltd gives no
warranties in relation to these matters. If you have any doubts about the
authenticity of an email purportedly sent by us, please contact Bridge Point
Communications Pty Ltd immediately.
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------------------------------
Message: 8
Date: Fri, 12 Jul 2013 09:05:40 -0400
From: David Zhars <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Polycom Analog or IP
Message-ID:
<CADe=jTEiZ=azoexal3lszdepfjyalrnb0s4kfqg2hgppx8c...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
My boss is pushing hard for a Polycom for a conference room....ok whatever,
let's make him happy!
I have an ATA-187 still in the box. So I could get an Analog version,
though it looks like the 187 doesn't handle POE, so I'll need to have it
near a power source.
Is it better to just get the IP versions and set it up for SIP? Anyone
have any experiences either way and did it bring happiness? ;-)
Thanks.
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Message: 9
Date: Fri, 12 Jul 2013 13:26:30 +0000
From: Jason Faraone <[email protected]>
To: "'David Zhars'" <[email protected]>, "[email protected]"
<[email protected]>
Subject: Re: [cisco-voip] Polycom Analog or IP
Message-ID:
<fb0beb07ca11e54a83b331ad1c263ed665d2a...@chq-mdf-mbx-02.paulo.com>
Content-Type: text/plain; charset="us-ascii"
I stopped deploying analog Polycoms in favor of Cisco 7937s and have been much
happier. Having a softkey template is handy for joining conferences, etc. To
take it a step further, I deployed one of these over a year ago and it's been
performing like a champ. It is configured as a "basic" SIP device so you don't
get any advanced features, but it's wireless and comes with a couple satellite
mics.
http://revolabs.com/Products/Product-Line/FLX-Wireless-Conference-Phone.aspx
From: cisco-voip [mailto:[email protected]] On Behalf Of David
Zhars
Sent: Friday, July 12, 2013 8:06 AM
To: [email protected]
Subject: [cisco-voip] Polycom Analog or IP
My boss is pushing hard for a Polycom for a conference room....ok whatever,
let's make him happy!
I have an ATA-187 still in the box. So I could get an Analog version, though
it looks like the 187 doesn't handle POE, so I'll need to have it near a power
source.
Is it better to just get the IP versions and set it up for SIP? Anyone have
any experiences either way and did it bring happiness? ;-)
Thanks.
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Message: 10
Date: Fri, 12 Jul 2013 09:40:40 -0400
From: Ryan Ratliff <[email protected]>
To: Jason Faraone <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Polycom Analog or IP
Message-ID: <[email protected]>
Content-Type: text/plain; charset="windows-1252"
On a somewhat-related topic has anyone tried the new 8831 yet?
I'd love to hear feedback if so.
-Ryan
On Jul 12, 2013, at 9:26 AM, Jason Faraone <[email protected]> wrote:
I stopped deploying analog Polycoms in favor of Cisco 7937s and have been much
happier. Having a softkey template is handy for joining conferences, etc. To
take it a step further, I deployed one of these over a year ago and it?s been
performing like a champ. It is configured as a ?basic? SIP device so you don?t
get any advanced features, but it?s wireless and comes with a couple satellite
mics.
http://revolabs.com/Products/Product-Line/FLX-Wireless-Conference-Phone.aspx
From: cisco-voip [mailto:[email protected]] On Behalf Of David
Zhars
Sent: Friday, July 12, 2013 8:06 AM
To: [email protected]
Subject: [cisco-voip] Polycom Analog or IP
My boss is pushing hard for a Polycom for a conference room....ok whatever,
let's make him happy!
I have an ATA-187 still in the box. So I could get an Analog version, though
it looks like the 187 doesn't handle POE, so I'll need to have it near a power
source.
Is it better to just get the IP versions and set it up for SIP? Anyone have
any experiences either way and did it bring happiness? ;-)
Thanks.
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 11
Date: Fri, 12 Jul 2013 09:04:32 -0500
From: Erick Wellnitz <[email protected]>
To: David Zhars <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Polycom Analog or IP
Message-ID:
<cak0wosanzehqakelnthrr4jwnlyldet4c6drp7cmnydkyfd...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I deployed 2 of the soundstation IP versions on CME. I think they were
7000 series.
Once I figured out all of the Polycom settings they registered and worked
like a champ. We went with them because we could set them up in a large
conference room with mics and external speakers.
On Fri, Jul 12, 2013 at 8:05 AM, David Zhars <[email protected]> wrote:
> My boss is pushing hard for a Polycom for a conference room....ok
> whatever, let's make him happy!
>
> I have an ATA-187 still in the box. So I could get an Analog version,
> though it looks like the 187 doesn't handle POE, so I'll need to have it
> near a power source.
>
> Is it better to just get the IP versions and set it up for SIP? Anyone
> have any experiences either way and did it bring happiness? ;-)
>
> Thanks.
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 12
Date: Fri, 12 Jul 2013 14:08:11 +0100
From: Russell Chaseling <[email protected]>
To: Ryan Ratliff <[email protected]>, Carlo Calabrese
<[email protected]>
Cc: 'Cisco-voip' <[email protected]>
Subject: Re: [cisco-voip] SIP between CUCM clusters
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
I'm also looking at deploying SIP trunks between two CUCM 8.6 clusters for IP
calling for the simple reason that I believe that presence status can be shared
between the two ie IP Phone BLF and CUEAC BLF (particularly CUEAC BLF)
1) First questions is have I heard this right? Will BLF work between 2 x
CUCM clusters?
2) Has anyone deployed SIP trunks between clusters in production? If so is
it worth the pain?
Cheers
Russell
From: cisco-voip [mailto:[email protected]] On Behalf Of Ryan
Ratliff
Sent: 05 June 2013 15:24
To: Carlo Calabrese
Cc: 'Cisco-voip'
Subject: Re: [cisco-voip] SIP between CUCM clusters
You can use SIP trunks between CUCM clusters in lieu of ICTs. The 8.x SRND
specifically states that QSIG-tunneled SIP trunks have feature parity with
H.323 ICTs.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1122456
The SIP trunk features available in the current release of Unified CM make SIP
the preferred choice for new and existing trunk connections. The QSIG over SIP
feature provides parity with H.323 intercluster trunks and can also be used to
provide QSIG over SIP trunk connections to Cisco IOS gateways (and on to
QSIG-based TDM PBXs). The ability to run on all Unified CM nodes and to handle
up to 16 destination IP addresses improves outbound call distribution from
Unified CM clusters and reduces the number of SIP trunks required between
clusters and devices. SIP OPTIONS ping provides dynamic reachability detection
for SIP trunk destinations, rather than per-call reachability determination.
SIP Early Offer support for voice and video calls (insert MTP if needed) can
reduce or eliminate the need to use MTPs and allows voice, video, and encrypted
calls to be made over SIP Early Offer trunks.
In general anything you can do to reduce the complexity of your call flows
makes your life that much easier both in configuration and troubleshooting.
-Ryan
On Jun 5, 2013, at 8:32 AM, Carlo Calabrese
<[email protected]<mailto:[email protected]>> wrote:
This is the problem I am having. Intercluster turnks are H323 based not sip. I
have them built now, but it breaks all sorts of stuff.
multicast, not building rtp streams correctly. Dropped calls DTMF.
From: Kenneth Hayes [mailto:[email protected]<http://gmail.com/>]
Sent: Wednesday, June 05, 2013 3:26 AM
To: Carlo Calabrese
Subject: Re: [cisco-voip] SIP between CUCM clusters
That would be call a Intercluster trunk. Build it like you would a regular SIP
trunk but instead of selecting SIP trunk go to Trunk->Add New->Trunk type
should be InterCluster Trunk Non-Gatekeeper Controlled" put in the required
information and I'm not sure if MTP is needed or not it's been awhile since
I've done a ICT but that should get you going. Also you will have to make sure
you have the correct PT's and CSS on both ends and dial-plan modifications.
On Wed, Jun 5, 2013 at 12:19 AM, Carlo Calabrese
<[email protected]<mailto:[email protected]>> wrote:
Has anyone done a sip trunk between CUCM clusters? I am running 8.6 in
production and 8.0 in the lab.
I can make it work with intercluster trunks ok. But I keep running into odd
bugs with calls going from sip to h323 back to sip
Or can you point me in the right direction.
Thanks.
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
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------------------------------
Message: 13
Date: Fri, 12 Jul 2013 10:07:47 -0500
From: Erick Wellnitz <[email protected]>
To: Russell Chaseling <[email protected]>
Cc: Cisco-voip <[email protected]>
Subject: Re: [cisco-voip] SIP between CUCM clusters
Message-ID:
<CAK0wOsCa=KWwbE2uMHYkOJzGBLBo7tT56+-HkMni9=pk7pe...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
I have deployed SIP between a 6.1 cluster and an 8.6 cluster for migration
purposes.
It worked perfect once I configured it correctly.
Make sure each side only points to CUCM nodes in the CM group of the
trunk's device pool...if that makes sense. Or, I believe you could check
the 'run on all nodes' box in the trunk configuration for both/all clusters.
On Fri, Jul 12, 2013 at 8:08 AM, Russell Chaseling
<[email protected]>wrote:
> I?m also looking at deploying SIP trunks between two CUCM 8.6 clusters for
> IP calling for the simple reason that I believe that presence status can be
> shared between the two ie IP Phone BLF and CUEAC BLF (particularly CUEAC
> BLF)****
>
> ** **
>
> **1) **First questions is have I heard this right? Will BLF work
> between 2 x CUCM clusters? ****
>
> **2) **Has anyone deployed SIP trunks between clusters in
> production? If so is it worth the pain?****
>
> ** **
>
> Cheers****
>
> Russell****
>
> ** **
>
> *From:* cisco-voip [mailto:[email protected]] *On Behalf
> Of *Ryan Ratliff
> *Sent:* 05 June 2013 15:24
> *To:* Carlo Calabrese
> *Cc:* 'Cisco-voip'
> *Subject:* Re: [cisco-voip] SIP between CUCM clusters****
>
> ** **
>
> You can use SIP trunks between CUCM clusters in lieu of ICTs. The 8.x
> SRND specifically states that QSIG-tunneled SIP trunks have feature parity
> with H.323 ICTs.****
>
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1122456
> ****
>
> ** **
>
> The SIP trunk features available in the current release of Unified CM make
> SIP the preferred choice for new and existing trunk connections. The QSIG
> over SIP feature provides parity with H.323 intercluster trunks and can
> also be used to provide QSIG over SIP trunk connections to Cisco IOS
> gateways (and on to QSIG-based TDM PBXs). The ability to run on all
> Unified CM nodes and to handle up to 16 destination IP addresses improves
> outbound call distribution from Unified CM clusters and reduces the number
> of SIP trunks required between clusters and devices. SIP OPTIONS ping
> provides dynamic reachability detection for SIP trunk destinations, rather
> than per-call reachability determination. SIP Early Offer support for voice
> and video calls (insert MTP if needed) can reduce or eliminate the need to
> use MTPs and allows voice, video, and encrypted calls to be made over SIP
> Early Offer trunks.****
>
> ** **
>
> In general anything you can do to reduce the complexity of your call flows
> makes your life that much easier both in configuration and troubleshooting.
> ****
>
> ** **
>
> -Ryan ****
>
> ** **
>
> On Jun 5, 2013, at 8:32 AM, Carlo Calabrese <[email protected]>
> wrote:****
>
> ** **
>
> This is the problem I am having. Intercluster turnks are H323 based not
> sip. I have them built now, but it breaks all sorts of stuff.****
>
> multicast, not building rtp streams correctly. Dropped calls DTMF.****
>
> ****
>
> *From:* Kenneth Hayes [mailto:[email protected]]
> *Sent:* Wednesday, June 05, 2013 3:26 AM
> *To:* Carlo Calabrese
> *Subject:* Re: [cisco-voip] SIP between CUCM clusters****
>
> ****
>
> That would be call a Intercluster trunk. Build it like you would a regular
> SIP trunk but instead of selecting SIP trunk go to Trunk->Add New->Trunk
> type should be InterCluster Trunk Non-Gatekeeper Controlled" put in the
> required information and I'm not sure if MTP is needed or not it's been
> awhile since I've done a ICT but that should get you going. Also you will
> have to make sure you have the correct PT's and CSS on both ends and
> dial-plan modifications.****
>
> ****
>
> On Wed, Jun 5, 2013 at 12:19 AM, Carlo Calabrese <
> [email protected]> wrote:****
>
> Has anyone done a sip trunk between CUCM clusters? I am running 8.6 in
> production and 8.0 in the lab.****
>
> I can make it work with intercluster trunks ok. But I keep running into
> odd bugs with calls going from sip to h323 back to sip****
>
> Or can you point me in the right direction.****
>
> ****
>
> Thanks.****
>
> ****
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ****
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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