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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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Cisco Unity Express - Version 8.6.7
Cisco Unified Communications Manager Express - 15.6(3)  / CME 11.5

Internal extension to extension calls will transfer to voicemail fine.

Calls from outside that go to voicemail get the message - "There is no mailbox associated with this extension, wait while I transfer your call."  Then it goes to a busy signal.

If the extension is forwarded to another extension internally, calls from both inside and outside will transfer to the other extension.

If the extension is forwarded to an outside number (we dial 9 to get out and enter it into the number to forward to) calls from outside will be transferred to the outside number.  Calls from inside get a busy signal.

Any help would be much appreciated.
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sing SIP RTP CoS mark 5
    -- Executing [09599259123@numberplan2:1] Dial("SIP/220-00000015", "DAHDI/G1/9599259123,60") in new stack
[Jun 13 17:19:19] WARNING[3910]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [09599259123@numberplan2:2] Hangup("SIP/220-00000015", "") in new stack
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I need to create a procedure in my PBX (running Elastix) that hangup some calls depending on the caller ID and Dial a specific phone number to all other numbers.

At this moment I use Goto to send a call to a queue, but I want to use a direct Dial or a MiscDestination.

This is my current code: exten => 4821,n(message),Goto(ext-queues,5555,1)

So instead the "ext-queues,5555,1" I would like to direct dial a phone for example 8775555555

My dial plan requires me to dial 877 before the phone number.

For example in Misc Destinations, I can put 8775555555 and If I use it I will be calling phone 5555555

Thanks
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Hi, really struggling with dialplans for Snom 300 IP phone at the moment and would appreciate some help.

I need to set a Snom 300 to only allow outbound calls which begin with a "7" but then to drop the lead digit...  Sounds weird I know, but it's the only way I can think of to restrict outbound calling to the speed-dial list which the users will not be able to view, but able to call.

So the plan is that 01234 567890 (for example) is in the speed dial list as 701234 567890, for the phone to then recognise this as an allowed number but drop the lead 7 so that it is able to be dialled.

I can't change anything in the PBX as we are using a hosted solution - already spoken to them and they can't / won't help, saying it needs to be done at handset level.

I hope that makes sense and that someone can help me out! Thanks in advance. :-)
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I have a issue with Dialing out and some inbound.. I have a As5400XM with CT3 card on my TDM side all my b channels are up so are my d channels. My main issue is my dial peer setup when i dial out it will go over 2 dial peers ar the same time and i get no audio. My question is im have 28 T1s and i want them all to do inbound and outbound dialing what is a sample config for some like this. I know a dial peer out going needs a pots dial peer and a voip dial peer just a little confused
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Does anyone know if you can dump the call history log of a sip account to a text file?
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Hi,
I need to log voice calls and in particular call-forward calls made by user, by writing in an external syslog, date, time, internal-number, external-number and user originating that.
Which way could I implement this ?

max
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Hi All,
I have been at this all day to no avail.
I am using Yealink IP Phones. The customer now wants to run his laptops with the phones. So the PC's run through the phones.
The phones use their own gateway on port 1 and the PC's use their own on port 24.
In addition to VID 1 created VID 20 for the Data on all ports and Voice on VID 50 Voice as per this example I found.
Phones and PC's are on all the ports except 1 and 24.
AlI really want to do is give priority to the IP Phones.

[url="http://www.dlink.com/uk/en/support/faq/switches/layer-2-gigabit/dgs-series/es_dgs-1210_como-configurar-voice-vlan"]

The phones don't work and neither do the PC's when activated.
I have also setup the phones WAN port with VID 50 and the PC port with VID20.

Any help is welcome
I have not tried tagging P1 and P24 on all 3 the VLANS.  

Thanks
Ken
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We use twilio and sonic firewall in the company for our voip phone system.
And the issue is packet is drop more frequency and do not know what is the issue.

Most of the drops came from inbound calls.

Twilio has no issue.
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I have an issue that I need resolved.  We are running Cisco Unity v10.5 and have a problem when people call an extension and it goes unanswered.  If the caller hits "0" it automatically transfers the call to our District Office admin.  I found a setting under the User Templates that had this set up to transfer to the District Office admin and set it to Ignore Key under Caller Input Keys but the problem still persists.  Is there another area that I need to be looking into?
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hi all, my company is going to switch to voice over ip phone system within 1 week. before it is switched, do you know there is any stress testing that we can test our network ensure we can handle all incoming phone calls and etc.?

We have around 50 to 70 voice over ip phones will be used.

Thanks
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We are still using Avaya IP Office 403 & 406 control units. Wondering if there is a version of Phone Manager & Softconsole that will work on Windows 10 and with the 403 & 406 control units?
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Hi, I'm testing out Office 365 Skype for Business and PSTN calling with Polycom VVX 601 Phones.

I'm familiar with many Hosted PBX Providers such as RingCentral, 8x8, Vonage, etc. where you can just page and intercom to extensions and groups without much configuration on the Hosted PBX Side.

With Office 365 Skype for Business this seems like quite a hurdle. Per this link:

http://community.polycom.com/t5/VoIP/FAQ-How-can-I-use-PTT-Push-To-Talk-Paging-Page/td-p/49057

I have to manually go into each phone and enable/disable various channels so that anyone set to receive on a channel can get a page?

Is this the only way to do this?

How about Intercom, how's that setup?
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Hi, I have an old door opener system in my building, which lets me speaks when someone rings my department number, and open de door. I'm trying to find a way to control all this with asterisk, perhaps combined with arduino. Not sure if there is such a thing like a door opener with sip conexion, but I'm willing to try to do it myself (with someone else help, obviously :).

My door opener system is like http://www.netyer.com/t3_conexion_basica_con_fuente_generica_4_bornes.html

What I want to accomplish is: When someone is at the door, press my department number and rings in my departmen. Then, I should dial a sip extension, be able to talk, and open de door dialing a code.

Any idea?
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I have run debug ccsip messages in the past to see call setup etc. But on this new router at a new place - if I use that debug and run show log I am only seeing messages like this:

Apr 17 2017 18:25:59.916 UTC: %CALL_CONTROL-6-CALL_LOOP: The incoming call has a global identifier already present in the list of currently handled calls. It is being refused.

So actually - I guess I have two questions. One is why are the messages not showing up as expected in "show log"?
And second - any thought on how to find out what the error message I *am* seeing is about?
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We have old Nortel Meridian PBX and are moving to a Cisco Call Manager environment.  I would like to have a T1 tie line between out PBX and MGCP gateway.  I am able to configure a T1 PRI and connect to the PBX.  I can dial new extension on a PBX phone and my VoIP phone rings but now sound once I pickup.  I am also not able to ring the PBX extension from the VoIP phone.
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Hi,

I am in the process of integrating Cisco CM 6.1 and Avaya SM 7.x.

Has anyone tested this integration? What works , what doesn't?

There is no test bed and integration needs to be done on live devices. any pointers will be appreciated.

thanks,
Rasheed
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I'm wondering if anyone might be able to shed some light on how IP addresses are handled..I have a hacker who has compromised my system and I got some info on him (IP addresses and MAC-probably both spoofed)but I'm filing a motion to get the records subpoenaed but I'm noticing m that some are very similar (all same residential hostname) but a little different.
example: 123.456.78.900 then another 123.456.79.009
what is the reasoning behind this?
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Mohon bantuan nya
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I have a CUCM, Cisco Unity Connection and a gateway 2911. I configured Autoattendant my extension is 5600. The message is functioning since pstn but when i dial the digits don't work. I don't know if I have problems with the gateway or the CUC

Can you help me, please.
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Hi

I have TDA200  I have made no restriction for Landline & Mobile and restrictions for International calls

I want to know...for dialing an international call it should be in such a way that they should be able to dial it with Code only. I want

to know the steps how to do it
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Dear experts,

I have the following issue with Avaya PBX. We have boss (ie. ext 100) who has assistant (ie. ext 200).

The setup we want is that all external calls to ext 100 will be redirected to ext 200 + all internal calls in case of 100 is busy or is not answering will go to 200.

If on 100 I configure coverage path with the conditions as above, it works as expected, however the incoming calls ring on both stations.
Is it possible to configure it somehow, that the calls will be redirected to 200, without being displayed on 100? Call forward is not a solution, as we need to keep internal incoming calls on 100.

Thanks!
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Hi Guys
Just a quick one here: We recently migrated to Call Manager Express and they it is configured now, we are receiving too many complaints from users...The problem is within the keypad...All users need to login in order to make phone calls, and they find it hard (me too) to work with an alpha-numeric keypad...Previously, they only needed to press the keys to input the numbers, but now since it only gives us the Alpha-numeric inputs, it's painful to pursue them...For instance, my username is 00973540, i have to input it the old way we use to text with NOKIA phones to reach the number (have to type 4 times the same key)

We contacted cisco and no solution has been given to us so far
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Hi,

I would like to set up a hunt group that can make calls to multiple external cell numbers (dial 1 first and if no answer dial second)
is this possible to achieve using a hunt group or something with Single Number Reach Feature?
I am running Cisco Unified Communications 520,
Cisco Unified Communications Manager Express version 8.6

Thanks,
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I asked a similar question earlier but realized I need to be more specific about the licensing relative to hardware binding. Currently I have the Publisher as a VM on a UCS C-Series UCSC-C240-M3S. Because the new data center is 12 hours and a state away, it would be ideal to be able
to migrate the Publisher VM from its current site to the UCSC-C240-M3S in the new site. Then renumber. The question is: would everything play ok licensing-wise moving the publisher VM from one ESX/C240 to another?

I got the info on the address change earlier.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/install/10_0_1/ipchange/CUCM_BK_C3782AAB_00_change-ipaddress-hostname-100/CUCM_BK_C3782AAB_00_change-ipaddress-hostname-100_chapter_011.html
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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.