CallManager 4;X : PBX Interoperability Tenovis (Bosh) with E1 QSIG

Hi,
Is it possible to connect a Tenovis (Bosh)PBX via a QSIG PRI or IP H323 to a 2800 series router.An experience about that?
Thanks a lot
Cedric

It looks as though the Tenovis I55 supports both QSIG and H.323; this would make it fairly straight forward to connect to CCM, we have not performed any formal testing on it.
In terms of IOS, I would recommend that you use a mainline IOS such as 12.2(10b) or greater. Usually the higher IOS's are more stable.

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The interface is E1 for PBXs, with QSIG ?
I want to make less 10 simultaneous calls. What WIC, VIC or VWIC modules and codecs should be used with router 2801 / 2811
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IP telephony help

Hello,
I currently work on a feasibility study for the deployment of the IP telephony.
We want to distinguish between two specific cases:
1- IP telephony between site, therefore by using the Data line to convey the voice of a site with another, knowing that we use PABX Alcatel 4400 and 3640 routers.
2- IP telephony on the local area networks for a hundred station while keeping the traditional telephone system, therefore the hybrid model.
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Hi Resfat,
Here is some information to help get you started;
Q. What Q.SIG interoperability testing will be performed?
A. Interoperability testing to Avaya Definity, Nortel Meridian, Alcatel 4400, Siemens HiPath (Hicom), Ericsson MD110, IPC IPMX trading turret key system, BT Syntegra trading turret system will be performed.
From this doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a00801f8e18.shtml
Cisco 3640 - PBX Interoperability: Alcatel 4400 Release 3.2 PBX with Cisco CallManager
http://www.cisco.com/application/pdf/en/us/guest/products/ps278/c1237/ccmigration_09186a00801ba721.pdf
Cisco 3600 Series Gateway-PBX Interoperability: Alcatel 4400 with BRI Q.931 Signaling
http://www.cisco.com/application/pdf/en/us/guest/products/ps278/c1237/ccmigration_09186a00800e762d.pdf
Hope this helps!
Rob
Please remember to rate helpful posts........

NEC PBX Interaction with Cisco Call Manager

Hi,
We have a new building that we want to connect to our network and thinking of Callmanager to be installed and linking it to our NEC PBX.
Need some pointers on how am I going to go about doing that. Documents and examples will be really appreciated

Hi!
Here are some documents that might help:
*** NEC 2400 ICS PBX with CallManager using 3640-T1 MGCP Gateway ***
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guide09186a00808b6024.html
*** Cisco Unified CallManager Release 5.0-PBX Interoperability: NEC IPX2400 R15 to a Cisco CMM Using T1-PRI QSIG with MGCP ***
www.cisco.com/en/US/docs/voice_ip_comm/cucme/pbx/interop/notes/522832nt.pdf
The second link includes info on SW and HW requirements as well as features supported.
*** Cisco Unified CallManager Release 5.0-PBX Interoperability ***
www.cisco.com/en/US/docs/voice_ip_comm/cucme/pbx/interop/notes/248705nn.pdf
HTH, if it does please rate!
Tere :)

PBX interoperatibility with router

Dear ALl
I have a confusion regarding the PBX integrations with Router.
Now if i am integrating my 2801 router with any PBX trunk ports (by trunk ports i mean the ports used to connect with PSTN with normal RJ-11 connector) , do i really have to worry about the compatibility between router and PBX.
The reason is that if the PBX can terminate the lines from the PSTN (RJ-11) than it should work with my router too .It is with e1 when we should worry about compatibility .Am i correct.
So what ports do i need on the router side?FXS.i thnok so as when we connect pstn with PBX than PSTN provides the tone ,so now my router should provide the tone to PBX trunks.
My scenario is that eight lines from PSTN are terminated in the norstar PBX and as i am proposing the IPT system the lines will now be terminated into the router and than the router should forward the call to the PBX if it is for any analog phone .
So my real question is do i really need to worry about the PBX interoperability when i am dealing with analog trunks.
Backbone

Hi,
the short answer is if even FXO is barely acceptable to connect to the PSTN (you should really get at least ISDN BRI), FXS/FXO connnection to the PBX will be certainly too limiting. Many problems of double dialing, potential stuck ports, no passing of information, etc. etc. The very miniumum is E&M to the PBX but real good would be again, ISDN BRI or PRI.
If the PBX is missing the necessary cards and is not worth to buy them, the you should at some point look at replacing the PBX completely.
The answer to real question is yes, you should really worry about analog trunks if you want the integration be of a professional level.
Hope this helps, please rate post if it does!

Lync intergration with third party PBX

I want to implement the scenario below, I dont have clear picture whether it is techinicall possible in the state of the art solution or not.
First I have one third party PBX, which is interconnected with Lync server. I configure the binding relationship between one IP Phone registerred in the third party PBX and one Lync clinet under Lync server. Eg, number 1000 is registered in third party PBX
and number 2000 is registered in Lync server. When someboday call 1000, both 1000 and 2000 can ring simutanesouly since this can be configured on the third party PBX, similary whenever 2000 is called, both 1000 and 2000 can also ring at the same time since
this cabapility is provided also by Lync server.
But I want to implement that after the call is initiated from Lync client after some inquery into the contact list and a specific contact number is selected, the IP Phone binded to this Lync number also will ring and act as the caller to call the other
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So, when you set up RCC, Lync sends the command to the IP PBX, the IP Phone knows which number to call because it understands that language from Lync Typically the PBX supplier has this feature that is known to work with Lync. If the PBX
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http://www.estos.com/products/drivers-middleware/callcontrolgateway400.html
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Please help with a configuration I'm trying to get with sp3102

hello everyone,
I need all calls coming from PSTN to ring to XXX in my PBX.
I need all calls coming from PBX to ring in the phone connected to the adapter.
and I need a prefix to dialout from the PSTN line.
to achive this I did the following:
I register with line 1 with an account on my PBX.
I register with the PSTN line on another account on my PBX (555), and put the PSTN line to ring throw line 1,
on the user one I put "Cfwd Sel1 Caller" : 555 Cfwd Sel1 Dest: XXX so calls that comes from 555 (pstn line) will ring to my PBX and not the phone.
in the dial plane I put (xx.|<#,xx.< :@gw0>) so # will get the PSTN line.
but # doesn't works, I'm not sure why but I think its has something to do with PSTN line ring throw line 1.
Can some one help me configure this device so I can achive all 3 tasks in the best way?
Thank you,
Rami.

See these threads and pages about searchqu/searchnu:
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*http://www.ilivid.com/faq.htm#8
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Questions/conditions/needs related to the PBX

Dear Cisco,
I act on behalf of my boss who is planning a larger IT investment, but needs some background information.
So far, we used an open-source free PBX, but after we've recognised its deficiencies, we decided to buy a commercial product.
To tell the truth, we hesitate between some products, and we need some confirmations related to the concerned PBXs. One of them is Cisco.
So our questions/conditions/needs are as follows:
The PBX needs to be an IP PBX.
Basic PBX features that we need: conferencing, video calling, voicemail, call queuing, IVR.
The type of all the desktop phone we use is Grandstream GXP 2000. The PBX should be compatible with this device.
We have a special corporate VoIP SIP phone ("softphone") developed by Ozeki's third party SDK. I am confident that we will continue to use this SDK, so the PBX should be compatible with this software.
And the last one: we would like to use Android smartphones as the part of our telephone system, so the PBX should support mobile extensions.
Please indicate if there is a solution that fulfills the above requirements. If so, please suggest me the proper product/service here or send me an e-mail to [email protected]
I wanted to post this message here intentionally, because any further personal comments/experiences coming from the community would be appreciated.
Best regards,
Donna E. Hill
[email protected]

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Calling ID to PSTN with CME vs Ericsson MD110 and E1

Hi all,
We are having problems with calling party identification in the following scenario: some IP phones managed by a 2800 CME router. This is connected to an Ericsson MD110 PBX through a E1 with QSIG signalling. There is absolutely no problem when identifying both called and calling parties in both directions ingoing and outgoing from the router to the PBX private extensions. It also works fine when calling from the PSTN to direct public numbers redirected from the PBX to the IP phones extensions. However, when calling from these particular extensions, we want the calling party to identify with its particular public number, but instead it shows at the PSTN end with the main number of the ISDN jump group.
Anyone has experienced the same?
Thanks and regards,
Jose Soriano

Hi,
I'll try to be more clear. As I said, we have a CCME connected to an Ericsson MD100 via an E1 Qsig link. Then the MD110 goes to the PSTN also with an E1. The customer has several ISDN public numbers associated with that E1, one of them is the main ISDN group number, with which the originator of all the outgoig calls from the private numbers to the PSTN identify, except for some private extensions associated with one of the other direct public numbers: these can be reached directly through these numbers and also are identified with those numbers when calling to the PSTN.
Let's say we have 30 public numbers, from 555100 to 555129, beig 555100 the main number. Ext 101 is associated to public number 555101, and 102 with 555102. In the situation with no CCME, the call flow is as follows.
Private Ext. 101 (dials a public number) --> PBX EMD110 --> PSTN (displays 555101 as the caller ID)
PSTN dials 555101 --> PBX EMD100 --> Private Ext. 101
Now, let's include the CCME system. Let's say Ext. 102 is an IP Phone extension correctly routed between CCME and MD110. Also, ext. 102 should be identified to the PSTN as 555102.
Case A)
IP Phone 102 (dials a public number) --> CCME --> PBX EMD110 --> PSTN (displays 555100, and NOT 555102 as the caller ID)
Case B)
PSTN dials 555102 --> PBX EMD100 --> CCME --> IP Phone 102
The PRIVATE flow between CCME and MD110 is fine:
IP Phone 102 dials 101 --> CCME --> PBX EMD110 --> Private Ext 101 displays 102 as the CLID
Private Ext 101 dials 102--> PBX EMD100 --> CCME --> IP Phone 102, with 101 as the CLID
So, the problem is in what I called CASE A), calls do not idetify with the especific public number, but with the main group number when they are routed through the PSTN.
Regards,
Jose Soriano

FAX through PBX connected to Router

I do implementation in which PBXs are connected to Cisco routers using E&M ports at several sites. A FAX machine is connected to an extension on the PBX needs to communicate with another FAX machine connected to the PBX at another site. The settings of the E&M ports are here:
voice-port 0/0/0
trunk-group e&m
type 5
dial-type pulse
timing wait-wink 800
description E&M-PBX
A voip dial-peers are configured for the destination router.
dial-peer voice 95 voip
destination-pattern 95..
session target ipv4:192.168.125.2
I tried some chnages but it did not work. I will try to configure the codec to G711 and no VAD and no EC on all ports and test it. But it will cost bandwidth.
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Thank you Paolo
I configured fax protocol pass-through under the service level and it worked. I wait the feedback from the customer if any problems arise.
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How to change backgournd image with 7941 phones

Hello,
I followed the steps indicated to change the background image on my 7941 phones, but the only image I see when I try to change it on the phone, is the default one.
I dont know where to look for answers.
I join my list.xml file and the two background image, just in case.
Thank you

Hi Gabriel,
The 320x212x12 is the wrong file for 7941G phones it should be 320x196x4. Have a look;
Configuring a Background Image
To create custom background images for the Cisco Unified IP Phone, follow these steps:
Procedure
Step 1 Create two PNG files for each image (a full size version and a thumbnail version). Ensure the PNG files comply with the format guidelines that are listed in the "PNG File Requirements for Custom Background Images" section.
Step 2 Place the new PNG files that you created in the following folder on the TFTP server for each Cisco Unified CallManager in the cluster:
C:\Program Files\Cisco\TFTPPath\Desktops\320x196x4
Note Cisco recommends that you also store backup copies of custom image files in another location. You can use these backup copies if the customized files are overwritten when you upgrade Cisco Unified CallManager.
Step 3 Use a text editor to edit the List.xml file. See the "List.xml File Format Requirements" section for the location of this file, formatting requirements, and a sample file.
Step 4 Save your modifications and close the List.xml file.
Note When you upgrade Cisco Unified CallManager, a default List.xml file will replace your customized List.xml file. After your customize the List.xml file, make a copy of the file and store it in another location. After upgrading Cisco Unified CallManager, replace the default List.xml file with your stored copy.
Step 5 To cache the new List.xml file, stop and start the TFTP service by using Cisco Unified CallManager Serviceability or disable and re-enable the Enable Caching of Constant and Bin Files at Startup TFTP service parameter (located in the Advanced Service Parameters).
From this doc for the 7941G/7961G;
http://www.cisco.com/en/US/products/hw/phones/ps379/products_administration_guide_chapter09186a0080724691.html#wp1030692
Hope this helps!
Rob

Should I Upgrade to VoIP, or use PBX

We now use an Alcatel PBX, 15 years old, with Lucent 8410d digital phones. We have 200 users at one site, and 100 users at three remote sites, with data T-1s connecting them. Is VoIP absolutely the best way to go, or should I look at mid-range PBX solutions. If VoIP, is the Cisco brand the worth the extra money? Thanks. jc

Do you like Coke or Pepsi? Are you a Republician or a Democrat? Do you watch the Office or Heroes? The point is that there is alot of personal opinion in what you are asking. And a look of non-quangtifiables.
First choice, to VoIP or not to VoIP. WIth a 15-year only PBX, you are probably going to need a change anyway. The only serious options are a new IP PBX or an upgrade to the Alcatel that will enable IP. The IP-enabled option may end up being cheaper, because it allows you to keep existing handsets. It also reduces the amount of change that your users will face. However, it does tie you down to a single vendor and probably a single topology. And maintenance expenses will probably be higher.
A new IP PBX would probably be most people's choice here, but hey, it's a Cisco IPT forum. (If you go to a Ears, Nose, Throat doctor, he will probably tell you that you have a problem in your ears, your nose, or your throat.) This will probably included a larger initial outlay of capital and services, but in general, maintance will probably be cheaper. Now, if you go with an IP PBX, which do you go with? Obviously Cisco is the popular choice around here. It is expensive, but it also works. Lots of support is available and lots of trained and certified professionals can be found to install and support it. Asterix is the other major option, but this is the opposite of Cisco in a lot of ways. Cheap, difficult to find support, difficult to find people to install. However, when instaleld correctly, it works as well as Cisco. Some would say even better, as Asterix has a broader feature set. But typically the support issue is enough to drive most businesses away.
Aside from Cisco and Asterix, the only other IP PBX worth mentioning is Avaya's IP Office, but that is really more of just an IP enabled PBX/Key system. so probably not a canidate unless you are already using Avaya.

Inter cluster trunk/ ip to ip communication with remote office without ccm

dear sir,
We've both ipipgateway configured to communicate with each other. Attachment pngipipgw1 is our local voice router with destination pattern 3XXXX that need to link with other country Japan with destination pattern 7xxxx . All other country locations is ok except the japan which are using non cisco callmanager to link to our local site with cisco call manager 4.1(3) with destination pattern 3XXXX.
Q1.) Based on the ipipgateway configurations of both parties, please advise any finding/thoughts?
Q2.)At our local site, We have nm-hdv with 2 pvdm simm that configured as dspfarm transcoder. Is it enough to have only 2 pvdm simm out of 5 in the nm-hdv? If we add in new pvdm, do we need any configuration at the voice gateway or we just plug in the hardware?
please advise. thank you very much.
-regards-

Hello chialun
Regarding
Q1. Could you explain a little more about what is the problem you have with the calls between those two sites? Please share which number on one side is calling which other number on the other side and what happens with the call.
Q2. It depends on how are you using your Xcoder resources. If some of your conferences or calls are being terminated because you are running out of resources, it will be advisable to add some more pvdm2 to your router. Also after adding the hardware, you would like to change the configuration to increas the maximum number of transcoding sessions.
Regards
Xabier

CUCM Group Configuration- question about phones failing to secondary callmanager

I wanted to get clarification about how exactly phones will register to the secondary or tertiary servers if the link to the primary fails.
I have a 3 server CUCM cluster. 1 subscriber that some phones register to is separated from the other 2.
When the link to that sub goes down, the phone should then register to the secondary server of its Cisco Unified CallManager Group configuration, correct?
If this is true, then my next question is how long does it take to then failover to the secondary server?
SRST is not involved in this, it's just the CUCM Group Config. Any help clarifying how phones behave in this situation?

I found my own answer in the Cisco Press Troubleshooting Cisco IP Telephony book, page 155:
"The TFTP configuration file received by the IP phone has a list of up to three CallManagers
for registration purposes. The order of CallManagers indicates their priority. The first is the
highest-priority CallManager for the IP phone. It is the first place the IP phone attempts to
register. The highest-priority CallManager is also called the primary CallManager.
When the IP phone registers with its primary CallManager, it also establishes a standby
TCP connection to the next-highest–priority available CallManager, sometimes called the
standby CallManager, secondary CallManager, or backup CallManager.
The IP phone knows that there is an alternative CallManager for quick failover if it loses connectivity
with its primary CallManager.
At any given point in time, the CallManager to which the phone is registered is its active
connection. The CallManager node that has a standby TCP connection for the IP phone is
the standby connection.
Failover occurs under two conditions:
• If the TCP connection between the IP phone and the primary CallManager goes down.
Incidentally, stopping the CallManager service on a server causes all the TCP connections
to be closed and all the IP phones on that server to register to their standby.
• If CallManager has not responded to three consecutive KeepAlive messages sent from
the IP phone.
Any number of network-related issues could cause the TCP connection to go down. When
the IP phone registers with its standby CallManager, it registers with an alarm indicating
why it failed over.
Under the second condition, CallManager fails to respond to three consecutive KeepAlives.
The IP phone sends a KeepAlive message every 30 seconds by default. CallManager should
answer each KeepAlive the IP phone sends with an acknowledgment message. If
CallManager fails to respond to three consecutive KeepAlive messages, the IP phone marks
the connection as “bad.” The IP phone does not tear down the TCP connection, but it does
not attempt to re-register with the “bad” CallManager either. It continues sending
KeepAlive messages to the “bad” CallManager until CallManager tears down the TCP
connection. This delay gives CallManager time to respond if it recovers quickly. After 10
minutes, the IP phone removes the “bad” tag and again tries to establish communication
with CallManager using KeepAlive messages in the process just described. During this
time, the IP phone attempts to establish a connection with its secondary CallManager (or
its tertiary if the secondary is not available) and registers if possible.
If a call is in progress when an IP phone detects the loss of a TCP connection, the IP phone
does not fail over until the call is finished. If CallManager fails to respond to three consecutive
KeepAlive messages while an IP phone has an active call, the IP phone again waits
until the call is finished before registering to its standby CallManager."
Hope that helps someone else in the future.

CallManager 4;X : PBX Interoperability Tenovis (Bosh) with E1 QSIG

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