Re: [cisco-voip] 8861 Default Password issue

[Brian Meade]

Yea, I'd just set it to something and reset the phone.  I'm not sure what
the default is.

On Thu, May 4, 2017 at 6:33 PM, Erick Bergquist  wrote:

> The local phone unlock password is blank.
>
> It stops at 4 characters when "Cisco" is tried and we tried 1234 and
> it stopped taking characters after 3 were entered.
>
> We factory defaulted it twice using proper procedure (saw correct
> light sequence) and same boat.
>
> Attached is a picture.
>
>
>
> On Fri, Apr 28, 2017 at 11:10 AM, Brian Meade  wrote:
> > Seeing this issue too.  It looks like you can set it at "Local Phone
> Unlock
> > Password" on the common phone profile but no idea what the default is and
> > not seeing anything online.
> >
> > On Thu, Apr 27, 2017 at 9:39 PM, Erick Bergquist 
> wrote:
> >>
> >> CUCM.  Other phones work fine on the same switch port and cable, so
> >> that has been checked. Normal switchport config.
> >>
> >> Unable to check settings or network info on phone itself as it is
> >> prompting for this password even after factory reset.
> >>
> >>
> >> On Thu, Apr 27, 2017 at 4:34 PM, Anthony Holloway
> >>  wrote:
> >> > Kind of funny that I have to ask this on Cisco-VoIP now but are you
> >> > registering to CUCME, BE4K, Spark or CUCM? If CUCM, via MRA?
> >> >
> >> > Also, I've deployed many 8800 and have never seen a password issue
> like
> >> > this. Albeit, most of my deployments were with CUCM.
> >> > On Thu, Apr 27, 2017 at 5:08 PM Erick Bergquist 
> >> > wrote:
> >> >>
> >> >> Havng issue getting new 8861 registered.
> >> >>
> >> >> On the phone it prompts for admin password, but stops at the 4th
> >> >> character.
> >> >> Default password is Cisco but we can only enter 4 characters. Doesn't
> >> >> take anything past 4.
> >> >>
> >> >> Anyone seen this before?
> >> >>
> >> >> Have done factory reset to with same.
> >> >>
> >> >> Erick
> >> >> ___
> >> >> cisco-voip mailing list
> >> >> cisco-voip@puck.nether.net
> >> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >> ___
> >> cisco-voip mailing list
> >> cisco-voip@puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
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Re: [cisco-voip] 8861 Default Password issue

[Erick Bergquist]

The local phone unlock password is blank.

It stops at 4 characters when "Cisco" is tried and we tried 1234 and
it stopped taking characters after 3 were entered.

We factory defaulted it twice using proper procedure (saw correct
light sequence) and same boat.

Attached is a picture.



On Fri, Apr 28, 2017 at 11:10 AM, Brian Meade  wrote:
> Seeing this issue too.  It looks like you can set it at "Local Phone Unlock
> Password" on the common phone profile but no idea what the default is and
> not seeing anything online.
>
> On Thu, Apr 27, 2017 at 9:39 PM, Erick Bergquist  wrote:
>>
>> CUCM.  Other phones work fine on the same switch port and cable, so
>> that has been checked. Normal switchport config.
>>
>> Unable to check settings or network info on phone itself as it is
>> prompting for this password even after factory reset.
>>
>>
>> On Thu, Apr 27, 2017 at 4:34 PM, Anthony Holloway
>>  wrote:
>> > Kind of funny that I have to ask this on Cisco-VoIP now but are you
>> > registering to CUCME, BE4K, Spark or CUCM? If CUCM, via MRA?
>> >
>> > Also, I've deployed many 8800 and have never seen a password issue like
>> > this. Albeit, most of my deployments were with CUCM.
>> > On Thu, Apr 27, 2017 at 5:08 PM Erick Bergquist 
>> > wrote:
>> >>
>> >> Havng issue getting new 8861 registered.
>> >>
>> >> On the phone it prompts for admin password, but stops at the 4th
>> >> character.
>> >> Default password is Cisco but we can only enter 4 characters. Doesn't
>> >> take anything past 4.
>> >>
>> >> Anyone seen this before?
>> >>
>> >> Have done factory reset to with same.
>> >>
>> >> Erick
>> >> ___
>> >> cisco-voip mailing list
>> >> cisco-voip@puck.nether.net
>> >> https://puck.nether.net/mailman/listinfo/cisco-voip
>> ___
>> cisco-voip mailing list
>> cisco-voip@puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Re: [cisco-voip] using two 3945s for SRST during upgrades - how to create voip dial-peer statements without causing a loop?

[Lelio Fulgenzi]

Interesting. Will look at that first. I have a simple set of CORs that give me 
what I need now. I'll have to see how I can modify them.

>From what I recall it was a strict "intersection" type relationship for things 
>to work. 

Brought back memories of algebra.

---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519-824-4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1

-Original Message-
From: NateCCIE [mailto:natec...@gmail.com] 
Sent: Thursday, May 04, 2017 3:46 PM
To: Lelio Fulgenzi; 'voyp list, cisco-voip'
Subject: RE: [cisco-voip] using two 3945s for SRST during upgrades - how to 
create voip dial-peer statements without causing a loop?

You can use COR to limit the inbound dial-peer on the router from seeing the 
outbound dial-peer that goes to the other SRST box.  Easy peasy.

-Nate

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lelio 
Fulgenzi
Sent: Thursday, May 04, 2017 11:30 AM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: [cisco-voip] using two 3945s for SRST during upgrades - how to create 
voip dial-peer statements without causing a loop?


Hello folks,

Question regarding voip dial-peers. I've had some experience, but my design 
skills are lacking, especially when it comes to something like I'm trying to do.

Basically, I'd like to take advantage of our two 3945 routers and failover as 
many phones as possible.  Problem is, it will be too difficult to fail them 
over in ranges.

Can I create a dial-peer that says "5 Pointer to router A" on router B, and 
"5 Pointer to router B" on router A and not cause any routing loops?

Is there a built in mechanism that prevents this? Is there something I need to 
configure?

If it's too complicated and requires testing and time, I may have to forfeit 
the idea of using two routers and use just one and selectively pick those who 
failover. I mean, we have to do it anyways, since we have more than the two 
routers could handle. And it's a migration model that I might chose to use 
anyways, i.e. keep one router connected to the cluster and one not.

Thoughts?

Lelio

---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS) University of Guelph

519-824-4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building Guelph, Ontario, N1G 2W1


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Re: [cisco-voip] using two 3945s for SRST during upgrades - how to create voip dial-peer statements without causing a loop?

[Ben Amick]

I'm not sure if two SRST boxes would trigger this, but I know when you create a 
loop which doesn't resolve on both ends (context: a DP to your SIP trunk that 
matches all, followed by a DP to your CM that matches all) it'll loop back and 
forth until a timeout period, after which is goes busy.

I am curious though, why do you need SRST for the upgrades? Do you not have a 
publisher and subscriber that you could fail between?

Ben Amick
Telecom Analyst

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Lelio 
Fulgenzi
Sent: Thursday, May 04, 2017 1:30 PM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net) 

Subject: [cisco-voip] using two 3945s for SRST during upgrades - how to create 
voip dial-peer statements without causing a loop?


Hello folks,

Question regarding voip dial-peers. I've had some experience, but my design 
skills are lacking, especially when it comes to something like I'm trying to do.

Basically, I'd like to take advantage of our two 3945 routers and failover as 
many phones as possible.  Problem is, it will be too difficult to fail them 
over in ranges.

Can I create a dial-peer that says "5 Pointer to router A" on router B, and 
"5 Pointer to router B" on router A and not cause any routing loops?

Is there a built in mechanism that prevents this? Is there something I need to 
configure?

If it's too complicated and requires testing and time, I may have to forfeit 
the idea of using two routers and use just one and selectively pick those who 
failover. I mean, we have to do it anyways, since we have more than the two 
routers could handle. And it's a migration model that I might chose to use 
anyways, i.e. keep one router connected to the cluster and one not.

Thoughts?

Lelio

---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519-824-4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1



Confidentiality Note: This message is intended for use only by the individual 
or entity to which it is addressed and may contain information that is 
privileged, confidential, and exempt from disclosure under applicable law. If 
the reader of this message is not the intended recipient or the employee or 
agent responsible for delivering the message to the intended recipient, you are 
hereby notified that any dissemination, distribution or copying of this 
communication is strictly prohibited. If you have received this communication 
in error, please contact the sender immediately and destroy the material in its 
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Re: [cisco-voip] using two 3945s for SRST during upgrades - how to create voip dial-peer statements without causing a loop?

[NateCCIE]

You can use COR to limit the inbound dial-peer on the router from seeing the
outbound dial-peer that goes to the other SRST box.  Easy peasy.

-Nate

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
Lelio Fulgenzi
Sent: Thursday, May 04, 2017 11:30 AM
To: voyp list, cisco-voip (cisco-voip@puck.nether.net)

Subject: [cisco-voip] using two 3945s for SRST during upgrades - how to
create voip dial-peer statements without causing a loop?


Hello folks,

Question regarding voip dial-peers. I've had some experience, but my design
skills are lacking, especially when it comes to something like I'm trying to
do.

Basically, I'd like to take advantage of our two 3945 routers and failover
as many phones as possible.  Problem is, it will be too difficult to fail
them over in ranges.

Can I create a dial-peer that says "5 Pointer to router A" on router B,
and "5 Pointer to router B" on router A and not cause any routing loops?

Is there a built in mechanism that prevents this? Is there something I need
to configure?

If it's too complicated and requires testing and time, I may have to forfeit
the idea of using two routers and use just one and selectively pick those
who failover. I mean, we have to do it anyways, since we have more than the
two routers could handle. And it's a migration model that I might chose to
use anyways, i.e. keep one router connected to the cluster and one not.

Thoughts?

Lelio

---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519-824-4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1


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[cisco-voip] using two 3945s for SRST during upgrades - how to create voip dial-peer statements without causing a loop?

[Lelio Fulgenzi]

Hello folks,

Question regarding voip dial-peers. I've had some experience, but my design 
skills are lacking, especially when it comes to something like I'm trying to do.

Basically, I'd like to take advantage of our two 3945 routers and failover as 
many phones as possible.  Problem is, it will be too difficult to fail them 
over in ranges.

Can I create a dial-peer that says "5 Pointer to router A" on router B, and 
"5 Pointer to router B" on router A and not cause any routing loops?

Is there a built in mechanism that prevents this? Is there something I need to 
configure?

If it's too complicated and requires testing and time, I may have to forfeit 
the idea of using two routers and use just one and selectively pick those who 
failover. I mean, we have to do it anyways, since we have more than the two 
routers could handle. And it's a migration model that I might chose to use 
anyways, i.e. keep one router connected to the cluster and one not.

Thoughts?

Lelio

---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519-824-4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1

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[cisco-voip] Virtual Telepresence Server droping conference voice calls

[Nilson Costa]

Hello All,

We are using Virtual Telepresence Server to make either conference and
video conference calls. We are facing problems when we have a external call
comming from PSTN and when we try to joinn this call to a conference this
call drops.

We have the following scenário

IOS voice Gateway IOS 12.24T
CUCM 11.0
vTS 4.2(4.23)
Conductor XC4.0

Our PSTN connection is a R2 digital and so:

PSTN ---R2D---> Gateway SIP> CUCM SIP/TLS>
Conductor---> vTS

On the telepresence serve I can see the following logs

529 2017/05/04 12:24:15.734 APP  Info  conference
"0010360100040x6c5eab929a5f3aa2" created
  530 2017/05/04 12:24:16.161 SIP  Info
 Incoming call from 138.132.130.200:58111
  531 2017/05/04 12:24:16.162 APP  Info  call
59: new incoming SIP call from "8633@138.132.129.147"
  532 2017/05/04 12:24:16.852 APP  Info  call
59: "Phone Teste" now joined conference "0010360100040x6c5eab929a5f3aa2"
  533 2017/05/04 12:24:16.948 SIP  Info
 Incoming call from 138.132.130.200:58111
  534 2017/05/04 12:24:16.949 APP  Info  call
60: new incoming SIP call from "0985590026@138.132.129.147"
  535 2017/05/04 12:24:17.104 APP  Info  call
60: "0985590026" now joined conference "0010360100040x6c5eab929a5f3aa2"
  536 2017/05/04 12:24:18.054 SIP  Info
 Incoming call from 138.132.130.200:58111
  537 2017/05/04 12:24:18.055 APP  Info  call
61: new incoming SIP call from "8691@138.132.129.147"
  538 2017/05/04 12:24:18.236 APP  Info  call
61: "Nilson Costa" now joined conference "0010360100040x6c5eab929a5f3aa2"
  539 2017/05/04 12:24:20.096 SIP  Error call
60: Closing call due to failed reINVITE
  540 2017/05/04 12:24:20.096 CMGR Info  call
60: disconnecting, "0985590026@138.132.129.147" - capset error
  541 2017/05/04 12:24:20.096 APP  Info  call
60: tearing down call from "0985590026" - destroy at far end request;
capabilityNegotiationError
  542 2017/05/04 12:24:20.420 CMGR Info  call
59: disconnecting, "8633@138.132.129.147"
  543 2017/05/04 12:24:20.420 APP  Info  call
59: tearing down call from "Phone Teste" - destroy at far end request;
remoteTeardown
  544 2017/05/04 12:24:21.545 CMGR Info  call
61: disconnecting, "8691@138.132.129.147"
  545 2017/05/04 12:24:21.545 APP  Info  call
61: tearing down call from "Nilson Costa" - destroy at far end request;
remoteTeardown
  546 2017/05/04 12:24:22.531 APP  Info
 conference "0010360100040x6c5eab929a5f3aa2":  deleted via API (no
participants)


On conductor I see

2017-05-04T11:38:59-03:00 conferencefactory.controller: Level="INFO"
Event="A conference has been deleted."
Conference_name="0030360100110x6c5eab929a5f3aa3"
Conference_unique_identifier="c7936d82e0d556f4250f361f1c0b3965c69da4f1"
Conference_bridge_address="138.132.130.195"
Conference_bridge_conference_name="0030360100110x6c5eab929a5f3aa3"
Conference_bridge_UUID="488623ee-4cc8-4aa7-a3dc-11853bf8c1f1"
UTCTime="2017-05-04 14:38:59,84"
2017-05-04T11:38:58-03:00 conferencefactory.controller: Level="INFO"
Event="A conference has been marked for deletion."
Conference_name="0030360100110x6c5eab929a5f3aa3"
Conference_unique_identifier="c7936d82e0d556f4250f361f1c0b3965c69da4f1"
Conference_bridge_address="138.132.130.195"
Conference_bridge_conference_name="0030360100110x6c5eab929a5f3aa3"
Conference_bridge_UUID="488623ee-4cc8-4aa7-a3dc-11853bf8c1f1"
UTCTime="2017-05-04 14:38:58,845"
2017-05-04T11:38:58-03:00 conferencefactory.switchboard: Level="INFO"
Event="A management request has been received."
Command="conference.destroy" Conference_name="003036010011"
Requester_(VCS/Unified_CM/client)_address="138.132.129.146"
UTCTime="2017-05-04 14:38:58,823"
2017-05-04T11:38:54-03:00 conferencefactory.controller: Level="INFO"
Event="A request to join a conference was successfully processed."
Conference_alias_description="LC_SPO_Ad-HOC"
Conference_alias_name="LC_SPO_Ad-HOC" Incoming_alias_match="(.*)@.*"
Conference_name_rule="\1" Conference_alias_UUID="adhoc_auto_gen"
Conference_name="0030360100110x6c5eab929a5f3aa3"
Conference_template_name="CUCM Ad-Hoc SP Meeting"
Conference_template_UUID="b78f285a-a1a4-4189-ab5d-26a59a9ee111"
Destination-alias="003036010011@138.132.130.199"
Conference_unique_identifier="c7936d82e0d556f4250f361f1c0b3965c69da4f1"
Pre-configured_endpoint="False" Conference_bridge_address="138.132.130.195"
Conference_bridge_conference_name="0030360100110x6c5eab929a5f3aa3"
Conference_bridge_UUID="488623ee-4cc8-4aa7-a3dc-11853bf8c1f1"
Participant_role="participant" Participant_type="Incoming ad-hoc"

Re: [cisco-voip] no ringback tone for incoming calls

[Brian Meade]

Is this for initial ringback or ringback while a call is being transferred
after selecting an option?  CUCM uses annunciators for playing ringback on
a transfer scenario so you might want to check the gateway MRGL has access
to annunciator resources.

On Wed, May 3, 2017 at 10:00 PM, naresh rathore  wrote:

> hi,
>
>
>
> I am facing the issue in which when i call internal trigger number (CUCM
> ---> UCCX), i can hear the ringback tone. but when i call from mobile to
> the voicegateway (mobile ---ISDN-> voice gateway --H323
> --> CUCM) i cant hear ringback tone when i select the option in
> IVR.
>
>
> progress indicator setup 3 is already configured. see the following
> dialpeer on the voice gateway
>
>
> rap-adl-rc01#sh run | sec dial-peer
> dial-peer voice 2001 voip
>  description Publisher
>  translation-profile incoming Outside-Lines-Inbound
>  preference 2
>  destination-pattern 888..
>  progress_ind setup enable 3
>  session target ipv4:
>  incoming called-number .
>  voice-class codec 10
>  voice-class h323 10
>  dtmf-relay cisco-rtp h245-signal h245-alphanumeric
>  no vad
> dial-peer voice 2002 voip
>  description Subscriber
>  translation-profile outgoing Outside-Lines-Outbound
>  preference 1
>  destination-pattern 888..
>  progress_ind setup enable 3
>  session target ipv4:
>  incoming called-number .
>  voice-class codec 10
>  voice-class h323 10
>  dtmf-relay cisco-rtp h245-signal h245-alphanumeric
>
>
> Regards
>
>
> Naresh Rathore
>
>
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Re: [cisco-voip] CUCM 11.5 license question

[Matthew Loraditch]

Yes this is as Ben stated, contact GLO and tell them you are missing your CUWL 
messaging licenses. They should be able to quickly fix it.

Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
Network Engineer
Direct Voice: 443.541.1518

Facebook | 
Twitter | 
LinkedIn 
| G+

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ahmed 
Abd EL-Rahman
Sent: Thursday, May 4, 2017 7:45 AM
To: Ben Amick ; James Buchanan 
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM 11.5 license question

Hi Gents,

Here is how my PLM looks like:

[cid:image001.png@01D2C4AA.E7C8D9F0]

Despite of having CUWL Standard 7 units available I still see -3 shortage in 
the Basic messaging ??





Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer

From: Ben Amick [mailto:bam...@humanarc.com]
Sent: Wednesday, May 3, 2017 4:21 PM
To: James Buchanan 
>; Ahmed Abd 
EL-Rahman >
Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] CUCM 11.5 license question

I find less than the license usage tab, checking the chart tab is more useful 
when trying to figure out oversubscription.

Also, I’ve run into an issue in the past of something really stupid – while 
CUWLs are a single purchase, it is possible to only install half of the 
license, and they’re allocated separately. See below for what should show in 
your ELM. When I came into ownership of this domain, I actually had 75 of my 
CUWLs that were installed as only the CUWL Standard line, and were missing the 
messaging portion.

[cid:image002.png@01D2C4AA.E7C8D9F0]



Ben Amick
Telecom Analyst

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of James 
Buchanan
Sent: Wednesday, May 03, 2017 6:57 AM
To: Ahmed Abd EL-Rahman 
>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM 11.5 license question

Can you send us a screenshot showing the licenses from License Usage in ELM?

On Wed, May 3, 2017 at 11:42 AM, Ahmed Abd EL-Rahman 
> wrote:
Hi Gents,

I have extra available units in CUWL standard license, and despite of that once 
I add any VM to a user the PLM give me a shortage in the basic messaging 
license, as per the documents CUWL standard license includes VM feature so why 
VM is not consuming from the CUWL standard license and showed separately as 
basic messaging with shortage counts??

Any clues?







Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer


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agent responsible for delivering the message to the intended recipient, you are 
hereby notified that any dissemination, distribution or copying of this 
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Re: [cisco-voip] CUCM 11.5 license question

[Ahmed Abd EL-Rahman]

Hi Gents,

Here is how my PLM looks like:

[cid:image002.png@01D2C4E5.06A1FD40]

Despite of having CUWL Standard 7 units available I still see -3 shortage in 
the Basic messaging ??





Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer

From: Ben Amick [mailto:bam...@humanarc.com]
Sent: Wednesday, May 3, 2017 4:21 PM
To: James Buchanan ; Ahmed Abd EL-Rahman 

Cc: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] CUCM 11.5 license question

I find less than the license usage tab, checking the chart tab is more useful 
when trying to figure out oversubscription.

Also, I’ve run into an issue in the past of something really stupid – while 
CUWLs are a single purchase, it is possible to only install half of the 
license, and they’re allocated separately. See below for what should show in 
your ELM. When I came into ownership of this domain, I actually had 75 of my 
CUWLs that were installed as only the CUWL Standard line, and were missing the 
messaging portion.

[cid:image003.png@01D2C4E5.06A1FD40]



Ben Amick
Telecom Analyst

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of James 
Buchanan
Sent: Wednesday, May 03, 2017 6:57 AM
To: Ahmed Abd EL-Rahman 
>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CUCM 11.5 license question

Can you send us a screenshot showing the licenses from License Usage in ELM?

On Wed, May 3, 2017 at 11:42 AM, Ahmed Abd EL-Rahman 
> wrote:
Hi Gents,

I have extra available units in CUWL standard license, and despite of that once 
I add any VM to a user the PLM give me a shortage in the basic messaging 
license, as per the documents CUWL standard license includes VM feature so why 
VM is not consuming from the CUWL standard license and showed separately as 
basic messaging with shortage counts??

Any clues?







Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer


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Re: [cisco-voip] can make conference call when i use IP communicator (same settings as 7942), but the conference call is not working when i use 7942 phone

[James Buchanan]

I had this very issue about 6 or 7 years ago, and had to upgrade firmware.
The bug was triggered by having G722 disabled on the phone, but having the
phone's built-in-bridge enabled for recording.

On Thu, May 4, 2017 at 9:19 AM, naresh rathore  wrote:

> it started working for one spoke site but other spoke sites are still not
> working.
>
>
>
> will check tomorrow.
>
>
>
> --
> *From:* cisco-voip  on behalf of
> naresh rathore 
> *Sent:* Thursday, May 4, 2017 6:10 AM
> *To:* Ryan Huff
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] can make conference call when i use IP
> communicator (same settings as 7942), but the conference call is not
> working when i use 7942 phone
>
>
> Upgraded the phone firmware and it started working
>
>
> --
> *From:* cisco-voip  on behalf of
> naresh rathore 
> *Sent:* Thursday, May 4, 2017 5:51 AM
> *To:* Ryan Huff
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] can make conference call when i use IP
> communicator (same settings as 7942), but the conference call is not
> working when i use 7942 phone
>
>
> both IP communicator and 7942 access MRGL via same device pool. Also, it
> remote site and using the CUBE at HQ site.  both IP Communicator and 7942
> phone use G729 to make external calls and use CUBE MRGL having hardware
> Conference as first preference and Hardware Transcoding  as second
> preference.
>
>
>
> remote site: region is LID
>
> HQ: region is HQ
>
> its set to use 8Kbps
>
>
> Device Pool on Hardware Conf use HQ Region
>
>
> Device pool on Hardware Transcoder use Moh region (G711 with every other
> region)
>
>
>
> The thing is I used the Same device pool on IP Communicator but conference
> call on IP communicator is working fine. may be i am connected to HQ via
> vpn and the CUBE is also in HQ. but device pool is same as remote /spoke
> site.
>
>
>
> Regards
>
>
> Naresh Rathore
>
>
> --
> *From:* Ryan Huff 
> *Sent:* Thursday, May 4, 2017 5:14 AM
> *To:* naresh rathore
> *Cc:* cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] can make conference call when i use IP
> communicator (same settings as 7942), but the conference call is not
> working when i use 7942 phone
>
> Couple of questions ...
>
> Do both devices have access to the same MRGL/MRG in CUCM?
>
> Does the 7942's audio codec region match the region of where the
> disconnected party comes from (ex. PSTN gateway) or does it need transcoded?
>
> -Ryan
>
>
> On May 3, 2017, at 8:04 PM, naresh rathore  wrote:
>
> hi,
>
>
>
> I am facing this  issue in which, when i use the 7942 phone, the
> conference call fails. it disconnect one of the party and keep it as point
> to point call. but when i use same settings on ip communicator (connected
> via vpn), the conference call works. 7942 firmware version
> SCCP42.9-3-1SR3-1S
>
>
>
> CUCM Version: 9.1.2.11900-12
>
>
> Regards
>
>
>
> Naresh Rathore
>
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Re: [cisco-voip] can make conference call when i use IP communicator (same settings as 7942), but the conference call is not working when i use 7942 phone

[naresh rathore]

it started working for one spoke site but other spoke sites are still not 
working.



will check tomorrow.



From: cisco-voip  on behalf of naresh 
rathore 
Sent: Thursday, May 4, 2017 6:10 AM
To: Ryan Huff
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] can make conference call when i use IP communicator 
(same settings as 7942), but the conference call is not working when i use 7942 
phone


Upgraded the phone firmware and it started working



From: cisco-voip  on behalf of naresh 
rathore 
Sent: Thursday, May 4, 2017 5:51 AM
To: Ryan Huff
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] can make conference call when i use IP communicator 
(same settings as 7942), but the conference call is not working when i use 7942 
phone


both IP communicator and 7942 access MRGL via same device pool. Also, it remote 
site and using the CUBE at HQ site.  both IP Communicator and 7942 phone use 
G729 to make external calls and use CUBE MRGL having hardware Conference as 
first preference and Hardware Transcoding  as second preference.



remote site: region is LID

HQ: region is HQ

its set to use 8Kbps


Device Pool on Hardware Conf use HQ Region


Device pool on Hardware Transcoder use Moh region (G711 with every other region)



The thing is I used the Same device pool on IP Communicator but conference call 
on IP communicator is working fine. may be i am connected to HQ via vpn and the 
CUBE is also in HQ. but device pool is same as remote /spoke site.



Regards


Naresh Rathore



From: Ryan Huff 
Sent: Thursday, May 4, 2017 5:14 AM
To: naresh rathore
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] can make conference call when i use IP communicator 
(same settings as 7942), but the conference call is not working when i use 7942 
phone

Couple of questions ...

Do both devices have access to the same MRGL/MRG in CUCM?

Does the 7942's audio codec region match the region of where the disconnected 
party comes from (ex. PSTN gateway) or does it need transcoded?

-Ryan


On May 3, 2017, at 8:04 PM, naresh rathore 
> wrote:


hi,



I am facing this  issue in which, when i use the 7942 phone, the conference 
call fails. it disconnect one of the party and keep it as point to point call. 
but when i use same settings on ip communicator (connected via vpn), the 
conference call works. 7942 firmware version SCCP42.9-3-1SR3-1S



CUCM Version: 9.1.2.11900-12


Regards



Naresh Rathore

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