Re: [cisco-voip] h323 trunk between cisco and asterisk

[s m]

thank you Brian, yes i have set bind address. when i enable h245
debugging,  all messages have no ip address like this:
value OpenLogicalChannel ::=
{
  forwardLogicalChannelNumber 1001
  forwardLogicalChannelParameters
  {
dataType nullData : NULL
multiplexParameters none : NULL
  }
  reverseLogicalChannelParameters
  {
dataType audioData : g711Ulaw64k : 20
multiplexParameters h2250LogicalChannelParameters :
{
  sessionID 1
  mediaChannel unicastAddress : iPAddress :
  {
network 'C0A80047'H
tsapIdentifier 17680
  }
  mediaControlChannel unicastAddress : iPAddress :
  {
network 'C0A80047'H
tsapIdentifier 17681
  }
}
  }
}



Apr 29 05:09:23.499: H245 FS OLC INCOMING ENCODE BUFFER::=
0003E90C6013800A04000100C0A800474511
Apr 29 05:09:23.499:
Apr 29 05:09:23.499: H245 FS OLC INCOMING PDU ::=

value OpenLogicalChannel ::=
{
  forwardLogicalChannelNumber 1002
  forwardLogicalChannelParameters
  {
dataType audioData : g711Ulaw64k : 20
multiplexParameters h2250LogicalChannelParameters :
{
  sessionID 1
  mediaControlChannel unicastAddress : iPAddress :
  {
network 'C0A80047'H
tsapIdentifier 17681
  }
}
  }
}

i think it is problem. cisco does not know where should send rtp packets.
am i right??? do you have any hint about it???

thank you for your attention.
SAM

On Tue, Apr 28, 2015 at 8:04 PM, Brian Meade  wrote:

> Do you have "h323-gateway voip bind srcaddr x.x.x.x" configured on an
> interface?
>
> You'll want to run "debug h245 asn1" to see if media negotiations as well.
>
> On Tue, Apr 28, 2015 at 3:55 AM, s m  wrote:
>
>> hello guys,
>>
>> i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with
>> ooh323 module. i configured both side and have successful call from cisco
>> to asterisk. but when call comes from asterisk to cisco, my phone rings but
>> no audio is heard and call is disconnected after 5 second. i enable "debug
>> voice rtp" in cisco and see the source address for receiving rtp packets is
>> 0.0.0.0
>>
>>  Apr 28 07:46:34.765: RTP(50493): ps rx s=0.0.0.0(0),
>> d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9
>>
>> any body knows how should i fix it?
>>
>> this is my cisco config:
>>
>> voice service voip
>>  allow-connections h323 to sip
>>  allow-connections sip to h323
>>  allow-connections sip to sip
>>  sip
>> !
>> !
>> !
>> voice class codec 1
>>  codec preference 1 g711ulaw
>>  codec preference 2 g711alaw
>>  codec preference 3 g729r8
>> !
>> dial-peer voice 1 voip
>>  destination-pattern 2.+
>>  voice-class codec 1
>>  session protocol sipv2
>>  session target ipv4:192.168.0.240
>> !
>> dial-peer voice 2 voip
>>  destination-pattern 1.+
>>  voice-class codec 1
>>  session target ipv4:192.168.0.71:1720
>>
>> any comments or hints are really appreciated.
>> SAM
>>
>>
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>>
>>
>
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Re: [cisco-voip] Phone Restart on Subscriber add

[Adam Frankel (afrankel)]

Brian is right.

Because TVS is not an activated service, it’s automatically added to the ITL 
during installation, forcing a phone reset.  There is not currently a way 
around this – other than potentially  enabling the rollback parameter prior to 
node installation (which in itself causes a phone reset).

--
Adam

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: Monday, April 27, 2015 4:07 PM
To: Tommy Schlotterer; Ryan Huff; bmead...@vt.edu
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Phone Restart on Subscriber add

Despite the phone resets,  the customer’s outsourced security group panicked 
when they saw a couple thousand packets at the TFTP servers when a sub got 
added to 10.5 cluster. Guess they thought a DDOS attack was going down or 
something.  Who knows what a 3rd party security company with IDS sensors would 
think….needless to say it created a small ruckus.

From: Tommy Schlotterer [mailto:tschlotte...@netechcorp.com]
Sent: Monday, April 27, 2015 4:02 PM
To: Ryan Huff; Jason Aarons (AM); bmead...@vt.edu
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Phone Restart on Subscriber add

It’s not a full reboot, looks just like an apply config, just enough for users 
to complain.

Tommy

Tommy Schlotterer | Systems Engineer
CCNA, CCNA Voice
48325 Alpha Dr. Ste. 150
Wixom, MI 48393
p 248.468.0710
e tschlotte...@netechcorp.com
w netechcorp.com
 [cid:DE00F175-A6C9-45A6-B3AC-D658551F1586]
[cid:62EB95BF-20B4-4A1B-98FE-9E042594730A]
 [cid:47C21B6C-578D-4D72-BFF4-8A482CE7A978] 
  
[cid:A362BD4D-9EC8-47CC-96A8-8A17AF38C15C]   
[cid:F52A69B8-DA49-4CD7-91E9-C057917D90C2] 


From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Monday, April 27, 2015 3:59 PM
To: Jason Aarons, (AM); bmead...@vt.edu; Tommy 
Schlotterer
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Phone Restart on Subscriber add


I'll have to lab this up and see what I am missing; last week I added a 9.1.x 
cluster node and I didnt get any phone bounces during the install.

Strange.

Thanks,

Ryan


 Original Message 
From: "Jason Aarons (AM)" 
mailto:jason.aar...@dimensiondata.com>>
Sent: Monday, April 27, 2015 02:54 PM
To: Brian Meade mailto:bmead...@vt.edu>>,Tommy Schlotterer 
mailto:tschlotte...@netechcorp.com>>
Subject: Re: [cisco-voip] Phone Restart on Subscriber add
CC: cisco-voip@puck.nether.net
So building the subscriber resets all phones in cluster to download updated 
Trust List?

It has nothing to do with adding subscriber to the CallManager group the phone 
is using?

I’ve heard of this issue but wasn’t clear (or forgot) about the specifics.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Brian 
Meade
Sent: Monday, April 27, 2015 3:50 PM
To: Tommy Schlotterer
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Phone Restart on Subscriber add


There's an option during the install to just check connectivity and then wait 
for the full install.  That's as far as you can go without resetting all the 
phones unfortunately.

On Mon, Apr 27, 2015 at 3:31 PM, Tommy Schlotterer 
mailto:tschlotte...@netechcorp.com>> wrote:
Yes, I do understand that. I just need to know when in the install, The 
customer doesn’t want a 4 hour maintenance window to install a new node.

Tommy

Tommy Schlotterer | Systems Engineer
CCNA, CCNA Voice
48325 Alpha Dr. Ste. 150
Wixom, MI 48393
p 248.468.0710
e tschlotte...@netechcorp.com
w netechcorp.com
 [cid:DE00F175-A6C9-45A6-B3AC-D658551F1586]
[cid:62EB95BF-20B4-4A1B-98FE-9E042594730A]
 [cid:47C21B6C-578D-4D72-BFF4-8A482CE7A978] 
  
[cid:A362BD4D-9EC8-47CC-96A8-8A17AF38C15C]   
[cid:F52A69B8-DA49-4CD7-91E9-C057917D90C2] 


From: bmead...@gmail.com 
[mailto:bmead...@gmail.com] On Behalf Of Brian Meade
Sent: Monday, April 27, 2015 3:28 PM
To: Tommy Schlotterer
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Phone Restart on Subscriber add

It happens when the cluster becomes aware of the new nodes certificates.  It 
adds a new TVS entry to the ITL of each node and resets the phones to ensure 
they download the new ITL before any other ITL changes are made.  This is to 
prevent too many change

Re: [cisco-voip] 3rd party conference phones

[Walenta, Philip]

Third party devices (SIP advanced or basic) are not allowed to use CM 
conference resources.  You need to mix on box or redirect somewhere else.

Sent from my iPhone

> On Apr 28, 2015, at 4:23 PM, Nick Thompson  wrote:
> 
> I just checked with a customer who has a bunch of the FLX2’s deployed and he 
> is seeing the same behavior.  He also tested with the standard SIP device and 
> advanced SIP device with no change.  He was going to call Revo and see if 
> there is something he is missing or if it is just a limitation of the device 
> firmware.
> 
> Nick
> 
>> On Apr 28, 2015, at 10:09 AM, Ed Leatherman  wrote:
>> 
>> Circling around on this thread,
>> 
>> So we're trying out revolabs FLX2. 
>> 
>> They are pretty easy to use and have some nice features. The unit reminds 
>> you to put everything back on the charging station after a call is over, 
>> which is an interesting touch. 
>> 
>> Main concern for me right now is that you cannot invoke an ad hoc conference 
>> that uses a cm controlled conference bridge (that I can tell) from the unit. 
>> The conference function can merely join the two line appearances that the 
>> unit has. 
>> 
>> We're still going to try it out in one of our rooms where the executive 
>> absolutely had to have "wireless" but I don't see widely deploying it 
>> elsewhere; people use the ad hoc bridges an aweful lot.
>> 
>> Ed
>> 
>> 
>> 
>> On Thu, Apr 9, 2015 at 10:05 AM, Nick Thompson  wrote:
>> 
>> Revolabs FLX2 - http://www.revolabs.com/flx2 
>> 
>> Revolabs is the OEM for the 8831 conference phone as well.
>> 
>> 
>>> On Apr 9, 2015, at 7:41 AM, Ed Leatherman  wrote:
>>> 
>>> Hello
>>> 
>>> I've had a few requests for "Wireless" conference phones lately for some 
>>> medium sized rooms. In these cases the executive wants to get any cables 
>>> etc off the table, so a wired base station w/ cordless module for the table 
>>> fits the bill as much as a WiFi device would. And because "Wireless"!
>>> 
>>> I'm curious what other people might be using, I see that polycom has the 
>>> SoundStation2W that uses a POTS line. I've also had some folks using a 
>>> Jabra Speak510 paired with some of the bluetooth capable Cisco Phones, but 
>>> this only works well in small rooms.
>>> 
>>> Thanks!
>>> 
>>> 
>>> 
>>> -- 
>>> Ed Leatherman
>>> ___
>>> cisco-voip mailing list
>>> cisco-voip@puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> 
>> 
>> 
>> 
>> -- 
>> Ed Leatherman
> 
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Re: [cisco-voip] 3rd party conference phones

[Roger Wiklund]

We are using a couple of SNOM conference phones, they do the job just
fine. Not wireless but PoE.

https://www.snom.com/en/products/snom-complementary-line/snom-meetingpoint-sip-conference-phone/

On Tue, Apr 28, 2015 at 11:22 PM, Nick Thompson  wrote:
> I just checked with a customer who has a bunch of the FLX2’s deployed and he 
> is seeing the same behavior.  He also tested with the standard SIP device and 
> advanced SIP device with no change.  He was going to call Revo and see if 
> there is something he is missing or if it is just a limitation of the device 
> firmware.
>
> Nick
>
>> On Apr 28, 2015, at 10:09 AM, Ed Leatherman  wrote:
>>
>> Circling around on this thread,
>>
>> So we're trying out revolabs FLX2.
>>
>> They are pretty easy to use and have some nice features. The unit reminds 
>> you to put everything back on the charging station after a call is over, 
>> which is an interesting touch.
>>
>> Main concern for me right now is that you cannot invoke an ad hoc conference 
>> that uses a cm controlled conference bridge (that I can tell) from the unit. 
>> The conference function can merely join the two line appearances that the 
>> unit has.
>>
>> We're still going to try it out in one of our rooms where the executive 
>> absolutely had to have "wireless" but I don't see widely deploying it 
>> elsewhere; people use the ad hoc bridges an aweful lot.
>>
>> Ed
>>
>>
>>
>> On Thu, Apr 9, 2015 at 10:05 AM, Nick Thompson  wrote:
>>
>> Revolabs FLX2 - http://www.revolabs.com/flx2
>>
>> Revolabs is the OEM for the 8831 conference phone as well.
>>
>>
>>> On Apr 9, 2015, at 7:41 AM, Ed Leatherman  wrote:
>>>
>>> Hello
>>>
>>> I've had a few requests for "Wireless" conference phones lately for some 
>>> medium sized rooms. In these cases the executive wants to get any cables 
>>> etc off the table, so a wired base station w/ cordless module for the table 
>>> fits the bill as much as a WiFi device would. And because "Wireless"!
>>>
>>> I'm curious what other people might be using, I see that polycom has the 
>>> SoundStation2W that uses a POTS line. I've also had some folks using a 
>>> Jabra Speak510 paired with some of the bluetooth capable Cisco Phones, but 
>>> this only works well in small rooms.
>>>
>>> Thanks!
>>>
>>>
>>>
>>> --
>>> Ed Leatherman
>>> ___
>>> cisco-voip mailing list
>>> cisco-voip@puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>>
>> --
>> Ed Leatherman
>
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Re: [cisco-voip] 3rd party conference phones

[Nick Thompson]

I just checked with a customer who has a bunch of the FLX2’s deployed and he is 
seeing the same behavior.  He also tested with the standard SIP device and 
advanced SIP device with no change.  He was going to call Revo and see if there 
is something he is missing or if it is just a limitation of the device firmware.

Nick

> On Apr 28, 2015, at 10:09 AM, Ed Leatherman  wrote:
> 
> Circling around on this thread,
> 
> So we're trying out revolabs FLX2. 
> 
> They are pretty easy to use and have some nice features. The unit reminds you 
> to put everything back on the charging station after a call is over, which is 
> an interesting touch. 
> 
> Main concern for me right now is that you cannot invoke an ad hoc conference 
> that uses a cm controlled conference bridge (that I can tell) from the unit. 
> The conference function can merely join the two line appearances that the 
> unit has. 
> 
> We're still going to try it out in one of our rooms where the executive 
> absolutely had to have "wireless" but I don't see widely deploying it 
> elsewhere; people use the ad hoc bridges an aweful lot.
> 
> Ed
> 
> 
> 
> On Thu, Apr 9, 2015 at 10:05 AM, Nick Thompson  wrote:
> 
> Revolabs FLX2 - http://www.revolabs.com/flx2 
> 
> Revolabs is the OEM for the 8831 conference phone as well.
> 
> 
>> On Apr 9, 2015, at 7:41 AM, Ed Leatherman  wrote:
>> 
>> Hello
>> 
>> I've had a few requests for "Wireless" conference phones lately for some 
>> medium sized rooms. In these cases the executive wants to get any cables etc 
>> off the table, so a wired base station w/ cordless module for the table fits 
>> the bill as much as a WiFi device would. And because "Wireless"!
>> 
>> I'm curious what other people might be using, I see that polycom has the 
>> SoundStation2W that uses a POTS line. I've also had some folks using a Jabra 
>> Speak510 paired with some of the bluetooth capable Cisco Phones, but this 
>> only works well in small rooms.
>> 
>> Thanks!
>> 
>> 
>> 
>> -- 
>> Ed Leatherman
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> 
> -- 
> Ed Leatherman

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[cisco-voip] Cisco Unity Connection 10.5 deleted messages returning...

[Jonathan Charles]

I just migrated a system from CUC 8.6.2 to CUC 10.5 with UM to Exchange
2010.

When two users delete messages, after a few seconds they come back and the
MWI light comes back on.

We have run the EWS Utility to remove the voicemail settings, however, it
has not resolved the issue...

Any ideas?



Jonathan
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Re: [cisco-voip] CER 9.02 IP Subnet ERL and phone using default ERL

[Brian Meade]

Do you see the IP Phone listed in the IP subnet on the CER admin page?

On Tue, Apr 28, 2015 at 1:30 PM, Erick Bergquist  wrote:

> Has anyone seen CER send call using default ERL when it falls within a
> defined IP Subnet ERL?
>
> We have 7925 model phones and most are using the right ERL for the IP
> subnet, but one phone which is in same IP subnet always uses the
> default ERL.
>
> There are no manually configured phones and no traditional ERLs
> configured, only IP sub net ERLs.
>
> Regards,
> Erick
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>
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[cisco-voip] 88XX Actionable Alert and Ignore

[Pawlowski, Adam]

Hi all. Playing around with the 8800 series devices, and I've turned on the 
actionable call alert. It looks pretty nice. There's an "ignore" option that 
appears here, but it doesn't silence the ringer when you press it, it just 
dismisses the toast. If I slap the volume rocker to down it mutes the ringer 
for the call. 

Is this what it is supposed to do? I can't find any description of the "ignore" 
button or what it really should be doing. Since enough features have been 
broken on these and the 7800s I figured I'd ask.

Regards,

Adam P
SUNYAB
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[cisco-voip] CER 9.02 IP Subnet ERL and phone using default ERL

[Erick Bergquist]

Has anyone seen CER send call using default ERL when it falls within a
defined IP Subnet ERL?

We have 7925 model phones and most are using the right ERL for the IP
subnet, but one phone which is in same IP subnet always uses the
default ERL.

There are no manually configured phones and no traditional ERLs
configured, only IP sub net ERLs.

Regards,
Erick
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Re: [cisco-voip] cisco jabber users delays when sending messaging

[Kevin Przybylowski]

It was answered here recently:

On 23-Apr-15 15:39, gen...@ucpenguin.com wrote:

> I haven't personally seen this, but did see this possibly related bug:

>

> https://tools.cisco.com/bugsearch/bug/CSCuu02593

>

> Symptom:

> Slow response 3-5 seconds entering chat

>

> Conditions:

> Jabber 10.x with KB3038314 installed.

>

> Workaround:

> remove KB3038314


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Claiton Campos
Sent: Tuesday, April 28, 2015 1:14 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] cisco jabber users delays when sending messaging

Hello ,

Recently some users have experienced slow during the exchange of messages 
through the jabber . A user sends a message and it takes 2-3 seconds to be sent 
to the other user. Users are using the 10.6.2 version of jabber .

Has anyone had any situation like this ?
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[cisco-voip] cisco jabber users delays when sending messaging

[Claiton Campos]

Hello ,

Recently some users have experienced slow during the exchange of messages
through the jabber . A user sends a message and it takes 2-3 seconds to be
sent to the other user. Users are using the 10.6.2 version of jabber .

Has anyone had any situation like this ?
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Re: [cisco-voip] licensing 8.6 vs 10.5

[Charles Goldsmith]

Yes, separate licensing, but PLM licenses it for us, and to answer Scott's
original question, I don't think there is a server license for it anymore,
same as CUCM and CUC.

UCCX still has it's own license setup, and requires an HA license if you go
down that path.

On Tue, Apr 28, 2015 at 10:39 AM, Heim, Dennis  wrote:

>  CER10 is still outside of CUWL, right?
>
>
>
> *Dennis Heim | Emerging Technology Architect (Collaboration)*
>
> World Wide Technology, Inc. | +1 314-212-1814
>
> [image: twitter] 
>
> [image: chat][image: Phone] <+13142121814>[image: video]
>
> "Innovation happens on project squared" -- http://www.projectsquared.com
>
>
>
> *Click here to join me in my Collaboration Meeting Room
> *
>
>
>
>
>
>
>
> *From:* bmead...@gmail.com [mailto:bmead...@gmail.com] *On Behalf Of *Brian
> Meade
> *Sent:* Tuesday, April 28, 2015 11:38 AM
> *To:* Heim, Dennis
> *Cc:* Scott Voll; cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] licensing 8.6 vs 10.5
>
>
>
> CER10 is managed by PLM10.
>
>
>
> On Tue, Apr 28, 2015 at 10:56 AM, Heim, Dennis 
> wrote:
>
>  I think CER is still on its on island.
>
>
>
> *Dennis Heim | Emerging Technology Architect (Collaboration)*
>
> World Wide Technology, Inc. | +1 314-212-1814
>
> [image: twitter] 
>
> [image: chat][image: Phone] <+13142121814>[image: video]
>
> "Innovation happens on project squared" -- http://www.projectsquared.com
>
>
>
> *Click here to join me in my Collaboration Meeting Room
> *
>
>
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Scott Voll
> *Sent:* Tuesday, April 28, 2015 10:04 AM
> *To:* cisco-voip@puck.nether.net
> *Subject:* [cisco-voip] licensing 8.6 vs 10.5
>
>
>
> we are looking at upgrading soon.  Can someone confirm what I believe:
>
>
>
> in version 10.x our licensing changes to user based.  With this, I can add
> as many nodes as I like? I currently have 2 nodes. if I wanted to add a 3rd
> that would be included?  or adding HA?
>
>
>
> This would be for CM, UC, CER, presence, but not UCCx?  am I correct?
>
>
>
> TIA
>
>
>
> Scott
>
>
>
>
>
>
> ___
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>
>
>
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>
>
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Re: [cisco-voip] licensing 8.6 vs 10.5

[Heim, Dennis]

CER10 is still outside of CUWL, right?

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
[twitter]
[chat][Phone][video]
"Innovation happens on project squared" -- 
http://www.projectsquared.com

Click here to join me in my Collaboration Meeting 
Room



From: bmead...@gmail.com [mailto:bmead...@gmail.com] On Behalf Of Brian Meade
Sent: Tuesday, April 28, 2015 11:38 AM
To: Heim, Dennis
Cc: Scott Voll; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] licensing 8.6 vs 10.5

CER10 is managed by PLM10.

On Tue, Apr 28, 2015 at 10:56 AM, Heim, Dennis 
mailto:dennis.h...@wwt.com>> wrote:
I think CER is still on its on island.

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
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From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net]
 On Behalf Of Scott Voll
Sent: Tuesday, April 28, 2015 10:04 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] licensing 8.6 vs 10.5

we are looking at upgrading soon.  Can someone confirm what I believe:

in version 10.x our licensing changes to user based.  With this, I can add as 
many nodes as I like? I currently have 2 nodes. if I wanted to add a 3rd that 
would be included?  or adding HA?

This would be for CM, UC, CER, presence, but not UCCx?  am I correct?

TIA

Scott



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Re: [cisco-voip] licensing 8.6 vs 10.5

[Brian Meade]

CER10 is managed by PLM10.

On Tue, Apr 28, 2015 at 10:56 AM, Heim, Dennis  wrote:

>  I think CER is still on its on island.
>
>
>
> *Dennis Heim | Emerging Technology Architect (Collaboration)*
>
> World Wide Technology, Inc. | +1 314-212-1814
>
> [image: twitter] 
>
> [image: chat][image: Phone] <+13142121814>[image: video]
>
> "Innovation happens on project squared" -- http://www.projectsquared.com
>
>
>
> *Click here to join me in my Collaboration Meeting Room
> *
>
>
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Scott Voll
> *Sent:* Tuesday, April 28, 2015 10:04 AM
> *To:* cisco-voip@puck.nether.net
> *Subject:* [cisco-voip] licensing 8.6 vs 10.5
>
>
>
> we are looking at upgrading soon.  Can someone confirm what I believe:
>
>
>
> in version 10.x our licensing changes to user based.  With this, I can add
> as many nodes as I like? I currently have 2 nodes. if I wanted to add a 3rd
> that would be included?  or adding HA?
>
>
>
> This would be for CM, UC, CER, presence, but not UCCx?  am I correct?
>
>
>
> TIA
>
>
>
> Scott
>
>
>
>
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Re: [cisco-voip] h323 trunk between cisco and asterisk

[Brian Meade]

Do you have "h323-gateway voip bind srcaddr x.x.x.x" configured on an
interface?

You'll want to run "debug h245 asn1" to see if media negotiations as well.

On Tue, Apr 28, 2015 at 3:55 AM, s m  wrote:

> hello guys,
>
> i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with
> ooh323 module. i configured both side and have successful call from cisco
> to asterisk. but when call comes from asterisk to cisco, my phone rings but
> no audio is heard and call is disconnected after 5 second. i enable "debug
> voice rtp" in cisco and see the source address for receiving rtp packets is
> 0.0.0.0
>
>  Apr 28 07:46:34.765: RTP(50493): ps rx s=0.0.0.0(0),
> d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9
>
> any body knows how should i fix it?
>
> this is my cisco config:
>
> voice service voip
>  allow-connections h323 to sip
>  allow-connections sip to h323
>  allow-connections sip to sip
>  sip
> !
> !
> !
> voice class codec 1
>  codec preference 1 g711ulaw
>  codec preference 2 g711alaw
>  codec preference 3 g729r8
> !
> dial-peer voice 1 voip
>  destination-pattern 2.+
>  voice-class codec 1
>  session protocol sipv2
>  session target ipv4:192.168.0.240
> !
> dial-peer voice 2 voip
>  destination-pattern 1.+
>  voice-class codec 1
>  session target ipv4:192.168.0.71:1720
>
> any comments or hints are really appreciated.
> SAM
>
>
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Re: [cisco-voip] 3rd party conference phones

[Ed Leatherman]

Circling around on this thread,

So we're trying out revolabs FLX2.

They are pretty easy to use and have some nice features. The unit reminds
you to put everything back on the charging station after a call is over,
which is an interesting touch.

Main concern for me right now is that you cannot invoke an ad hoc
conference that uses a cm controlled conference bridge (that I can tell)
from the unit. The conference function can merely join the two line
appearances that the unit has.

We're still going to try it out in one of our rooms where the executive
absolutely had to have "wireless" but I don't see widely deploying it
elsewhere; people use the ad hoc bridges an aweful lot.

Ed



On Thu, Apr 9, 2015 at 10:05 AM, Nick Thompson  wrote:

>
>  Revolabs FLX2 - http://www.revolabs.com/flx2
>
>  Revolabs is the OEM for the 8831 conference phone as well.
>
>
>  On Apr 9, 2015, at 7:41 AM, Ed Leatherman  wrote:
>
>  Hello
>
>  I've had a few requests for "Wireless" conference phones lately for some
> medium sized rooms. In these cases the executive wants to get any cables
> etc off the table, so a wired base station w/ cordless module for the table
> fits the bill as much as a WiFi device would. And because "Wireless"!
>
>  I'm curious what other people might be using, I see that polycom has the
> SoundStation2W that uses a POTS line. I've also had some folks using a
> Jabra Speak510 paired with some of the bluetooth capable Cisco Phones, but
> this only works well in small rooms.
>
>  Thanks!
>
>
>
>  --
> Ed Leatherman
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>
>


-- 
Ed Leatherman
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[cisco-voip] UCCX Script Visualization

[Tanner Ezell]

As some of you may have seen in a previous thread, I brought up the topic
of automatic UCCX script visualizations - that is, generating Visio-like
documentation from AEF files directly.

The responses I've been getting have been great and I want to open it up to
the general forum to get input from the community. I'm asking for a short
survey to be filled out and if possible, roughly 20 minutes of your time
for a quick chat to discuss the solution in greater detail.

I really appreciate all the responses from the community!

Link to the survey is here: http://bit.ly/1Jv5ZoP


Regards,
Tanner Ezell
CTI Logic
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Re: [cisco-voip] licensing 8.6 vs 10.5

[Heim, Dennis]

I think CER is still on its on island.

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
[twitter]
[chat][Phone][video]
"Innovation happens on project squared" -- 
http://www.projectsquared.com

Click here to join me in my Collaboration Meeting 
Room



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Scott 
Voll
Sent: Tuesday, April 28, 2015 10:04 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] licensing 8.6 vs 10.5

we are looking at upgrading soon.  Can someone confirm what I believe:

in version 10.x our licensing changes to user based.  With this, I can add as 
many nodes as I like? I currently have 2 nodes. if I wanted to add a 3rd that 
would be included?  or adding HA?

This would be for CM, UC, CER, presence, but not UCCx?  am I correct?

TIA

Scott


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Re: [cisco-voip] licensing 8.6 vs 10.5

[Haas, Neal]

So start off like this:

1.  Get your Team (Or TAC) to see what you have now in Licenses! Have them 
document it! Because if you upgrade then ask for the new licenses, you will get 
screwed! Sarcasm follows: We lost 600 licenses……poof! Oh did you have more; 
hmmm we don’t think so….. Oh can’t prove it for being a loyal paying customer 
for 16 years. Um we don’t believe you….

2.  Upgrade to 9.1(1). (CUCM, Unity, I think that you will need to do UCCX 
also to version 9.X)

a.  If you do UCCX version 9.x Java goes backwards for the desktop install. 
You cannot select multiple close codes. So be aware.

3.  Disaster recovery to VM 9.1(1)

4.  Upgrade VM to 10.6.(1)? (You do not need to do UCCX, you will need to 
do a COP file for you phones the cop file for 79XX has a transfer bug that just 
hangs up the phone.)

5.  CUCM will become your license manager in 10.X

Yes you can have up to 8 Nodes I believe. Just to let you know that each sub 
can handle 10,000 users as long as you start out with correct VM template. We 
have 4 nodes, at 4 different locations.

You can install HA for Unity, That’s not an extra Cost. HA gave us a smaller 
outage window when we upgraded.

New version of CER sucks! But we have to use it, all of a sudden it will just 
stop (8.6.X was solid) the newer version just stops working……. So watch them.

Presence changes in the 10.x Call Manager does a lot more of this application. 
(I don’t do anything with presence I just know from what I have been told :>)

UCCX you don’t need to upgrade, but in the newer version you get newer and 
better things right From what I have heard 10.x of UCCX works with IE 11. 
You can go all to finesse, this is our next upgrade.

I think I got everything. We just did this 3 months ago.

Neal Haas

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Scott 
Voll
Sent: Tuesday, April 28, 2015 7:04 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] licensing 8.6 vs 10.5

we are looking at upgrading soon.  Can someone confirm what I believe:

in version 10.x our licensing changes to user based.  With this, I can add as 
many nodes as I like? I currently have 2 nodes. if I wanted to add a 3rd that 
would be included?  or adding HA?

This would be for CM, UC, CER, presence, but not UCCx?  am I correct?

TIA

Scott


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[cisco-voip] licensing 8.6 vs 10.5

[Scott Voll]

we are looking at upgrading soon.  Can someone confirm what I believe:

in version 10.x our licensing changes to user based.  With this, I can add
as many nodes as I like? I currently have 2 nodes. if I wanted to add a 3rd
that would be included?  or adding HA?

This would be for CM, UC, CER, presence, but not UCCx?  am I correct?

TIA

Scott
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[cisco-voip] h323 trunk between cisco and asterisk

[s m]

hello guys,

i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with
ooh323 module. i configured both side and have successful call from cisco
to asterisk. but when call comes from asterisk to cisco, my phone rings but
no audio is heard and call is disconnected after 5 second. i enable "debug
voice rtp" in cisco and see the source address for receiving rtp packets is
0.0.0.0

 Apr 28 07:46:34.765: RTP(50493): ps rx s=0.0.0.0(0),
d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9

any body knows how should i fix it?

this is my cisco config:

voice service voip
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
!
!
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
!
dial-peer voice 1 voip
 destination-pattern 2.+
 voice-class codec 1
 session protocol sipv2
 session target ipv4:192.168.0.240
!
dial-peer voice 2 voip
 destination-pattern 1.+
 voice-class codec 1
 session target ipv4:192.168.0.71:1720

any comments or hints are really appreciated.
SAM
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