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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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.
0
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!
0
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
0
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,
0
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
0
I am having problems getting calls across CUBE. AT&T calls are arriving to CUBE but CUBE is not sending the calls to CUCM. The same from CUCM getting to CUBE but CUBE not sending to AT&T.



Ive been trying different configurations with the dial peers with no success.



CUCM IP is 172.16.200.3

CUBE IP is 172.16.200.1

ATT IP is 172.31.255.254



My current run conf is:


Building configuration...


Current configuration : 6066 bytes
!
! Last configuration change at 22:59:34 Caracas Thu Feb 16 2017 by admin
!
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no platform punt-keepalive disable-kernel-core
!
hostname localhost
!
boot-start-marker
boot-end-marker
!
!
vrf definition Mgmt-intf
!
address-family ipv4
exit-address-family
!
address-family ipv6
exit-address-family
!
enable secret 5 $1$r9QL$i80izReXxRd1i375cw0lY.
enable password password
!
aaa new-model
!
!
aaa authentication login default local
aaa authorization exec default local
aaa authorization network default local
!
!
!
!
!
!
aaa session-id common
clock timezone Caracas -4 0
!
!
!
!
!
!
!
!
!
!
!


ip name-server 8.8.8.8 4.2.2.2

ip domain name localhost.local
!
!
!
!
!
!
!
!
!
!
subscriber templating
multilink bundle-name authenticated
!
!
!
!
!
!
crypto pki trustpoint TP-self-signed-937083921
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-937083921…
0
Hello All,

In our small office we have a single POTS line from BrightHouse and we are moving to FlowRoute with SIP (or at least I want to) so I have purchased SIP services from FlowRoute and 3 DID's even though I only really need a single line to ring into the phone system. When someone calls now over the single POTs line the AA answers and it routes to the internal team extensions. I setup the new SIP trunk within CCA with the help of FlowRoute to get their settings right. I can call out but actually only to a single line (meaning I can't simultaneously dial outbound from more than one phone) but when I call in I get a busy signal. Something tells me it's because I haven't told the UC that when calls come in on one of our DID's that it needs to go to AA but I appear to be way over my head. Worst off Cisco stopped support for this unit 2 weeks ago :(

Goal is to have a single number dial into our UC and the AA answer as it does now over pots and to have more than one outbound call at the same time.
0
Hi Guys,

I am facing some issue after changing some changes in the CISCO CME, now When we transfer the incoming calls to other extension, its automatically drops the call when the calling person speak.
Note: If the calling person do not speak, the calls remain same. he can also talk smoothly with the first receiver.

Any help will be appreciable.

Good Day,
0
We are just activating mobility and would like to import all if the extensions on the system into contacts..
Any way to to do this through real time connection that will stay updated as extensions change?
0
hi all, please i just installed this panasonic TDA 100 ip intercom telephone system. everything is working fine but i dont know how to programm the call assignment feature, unlike the  1232 where you can do the programming with the operator console. please is there a way to programm this(TDA100) intercom with operator console.
thank you all
0
I have Physical server running with Vicidial & vtiger.
Vicidial working fine & vtiger only install over that. I am unable to config with my vicidial

vicidial version : 2.14
Asterisk : 1.8
vtiger : 6.5.0

Please help me out

Thanks in Advance
0
I'm troubleshooting a strange issue with a Shoretel phone system.  I'm new to looking at Shoretel, so some/any assistance is much appreciated.  HQ is running a shoretel system that shows a Softswitch and SG-220T1A.  The Branch shows a gateway of SG-90.  The two sites are connected via a site-to-site VPN tunnel.  Cisco ASA at HQ, sonicwall at the Branch.  The tunnel has been tested and is working fine.  From HQ I can ping the branch SG-90 at 192.168.1.11, and I can also ping all of the phones registered in their office.  From the branch side, I can ping the HQ softswitch at 192.168.11.10 and the SG-220T1A at 192.168.11.11.  I can also ping all of the phones from the branch to HQ.  

The Branch can call HQ with no problems.  Call sets up and both sides can hear each other.  When HQ tries to call the branch it immediately goes to voicemail.  Being new to this system I'm not sure where to start looking.  In the maintenance section under connectivity the connection shows as a green "C" for connected.

In the windows event viewer logs I do see Event Id 233: TMS has disconnected from switch "Branch" (192.168.1.11). This may be as a result of a network outage, administrative action, or unexpected switch behavior.

I'm watching the traffic across the tunnel and I'm not seeing any packet loss or dropped connections.
0
Hello Dear

Our “Voice Auto Attend” is not working when anybody calls from outside.

I have deleted all the “Visual Voice” from all our AVAYA IP phones (1608L and 1616L).

I have also rebooted, but still it is not working.

We are using “AVAYA IP Office R8.1 Manager”.

Kindly assist me urgently because our whole incoming phone is not working.

Thanks & regards
0
My user ID has just about every permission imaginable including Super User. Yet when I'm trying to
access AuditLog View in RTMT I get "Not an Authorized user: (my user name)." Does anyone know the
specific role or group which governs this? Does it take time for the permission to go into effect? I
did try logging out and back into RTMT but still no joy.
0
In CUCM 10.5 System/Security/Certificate - there are 100 or so certificates. I want to know if any of those certificates are in danger of expiring any time soon. How can this be determined?
0
The client is using Lync in conjunction with Cisco UCM and complains about occasional voice quality issues.
Normally with Cisco phones and CUCM you let the phones use the voice VLAN and tag the traffic that comes
onto the voice VLAN. But with Lync - that traffic is just traveling with all the other data traffic. What's the best
way to tag that traffic as real time EF as near the source as possible?
0

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.