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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When you use Microsoft PSTN calling - can you use your old Cisco 7945 phones provided you load them with the SIP firmware instead of the SCCP?
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the system was setup with an auto attendant and 3 call groups for voice mail and after the auto attendant the caller would choose 1 , 2 or 3  and that would send you to the proper mail box and then you would hear that message. i can not figure out how to rerecord the auto attendant message or change the message for the mail boxes that auto attendant send the caller to. i can do intercom 500 from every extension and that takes me to voice mail management, but I can not figure out how to manage another mail box other than the extension I am on or how to change the recording on auto attendant. I can access both systems with my laptop.
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Hi,

We currently have a Cisco CCM system which runs Vlan 10 as the Voice lan and Vlan 2 as the Data, the hosted platform we are going to can have vlan which is set via portal etc, it gets a normal IP and then boots into the correct lan ..... phone will connect ok but will no allow data pass through .... the Cisco phones are working fine but not sure why the hosted wont work ...... any ideas ?

Thanks
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Got a ip pbx and i want to send the voice mail via e-mail in the office we got a Miicrosoft 2011sbs standar with exchange 2010
altho i have create the account voicemail@xxxxx.org and configure the ip pbx the pbx is not able to send the email with the voicemail. i have testet the email created and work.  The pbx is nec sl1100
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Hey all, i have an issue that external calls has noise, delayed audio from the external side. Internal calls are fine.
We recently changed from the Telstra supplied modem to a Draytek Modem. All ports have been opened up the same as they were on the Telstra one, all lines and SIP registered straight away, however i have not been able to resolve the noise.
As Draytek have alot more advanced firewall settings, QOS, I'm not sure what feature / settings i need to change to test.
We are running freepbx 2.11.

Thanks
Matt
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Hello Experts,

I am in need of options to connect an analog phone to an old, Nortel PBX that is in another building with no physical, wired connection between the two buildings.  There IS network connectivity between the two buildings, and I've been told by our phone support \ vendor that there are devices available that would connect to the analog port on the PBX on one side, and to the phone itself on the other side, and then both devices would connect to the LAN to provide connectivity between the PBX and the phone.  Our vendor, however, has very little info on these.  I am writing to see if anyone here has any familiarity with this concept, and if so, can provide any recommendations or guidance on makes \ models they have used successfully in the past?

Thanks in advance for any help that can be offered,

Russ
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We have ISDN PRI line 30 channel terminated into Digium TE133 card. Sometimes when we make outbound call on any mobile number we got error "All circuits are busy now". Once we redial the same number the call connects.

 How to capture  debugs  ?
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For some project, we are exploring Twilio product "Programmable voice" . The pricing for per minute calling on normal voip is 1.5 cents whereas for SIP it is 0.4 Cents.

Thats makes me wonder how SIP differs from normal voip? Can I practically implement SIP in a small office of 4/5 people ? Can some expert exaplain me in simple English (Without using technical terms).

Twilio pricing I referred is here https://www.twilio.com/voice/pricing
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I have an old CallManager (4.3). it works great and no one wants to upgrade it. I have several small offices and individuals working from home offices and in order to have working phones in their locations I have to do site-site VPN's to each location.
Is there way to create some port forwarding and avoid VPN? Which ports? Any downsides?
The firewall is Cisco ASA5510 and they have Cisco 7941 and 7970 phones if that matters.
Thanks!
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i have Asterisk server running on two servers as Active/Active with 1000 Clients.Now we have configured the Lync Server i need step by step Guidelines for moving the clients from Asterisk to the Microsoft Lync with same extensions and dial plan.
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I am in need to check the best solution for IP Telephony to work with Microsoft Lync Solution.any idea with logical comments will be really appreciated

already running ASterisk but want to replace it.
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Hi Experts,
I installed CME (ISR 4321 IOS version 15.5), with 40 Phones 7821, 1 IP Phone 8841, another 8851, and 2 IP Conference 8831 using SIP Protocol
all of these SIP Phones work fine, but the call transfer doesn't work, I don't know if some config missing, you find below the sh run,
thank you for your help.
CME-EHM#sh run
Building configuration...

Current configuration : 10874 bytes
!
! No configuration change since last restart
!
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no platform punt-keepalive disable-kernel-core
!
hostname CME-EHM
!
boot-start-marker
boot-end-marker
!
vrf definition Mgmt-intf
 !
 address-family ipv4
 exit-address-family
 !
 address-family ipv6
 exit-address-family
!
card type e1 0 1
!
no aaa new-model
!
subscriber templating
multilink bundle-name authenticated
!
isdn switch-type primary-net5
!
crypto pki trustpoint TP-self-signed-246793832
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-246793832
 revocation-check none
 rsakeypair TP-self-signed-246793832
!
crypto pki certificate chain TP-self-signed-246793832
 certificate self-signed 01
  ////////////////////////****////
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
  call start slow
 sip
  bind …
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In RTMT you can view "Gateway Activity" and choose say MGCP PRI. But the thing that's weird is that it shows PRI channels per CUCM server. Given the description I would think that it would make out how many active calls are happening at the router/gateway not the UCM. Is there a way to get this broken out to you see activity per actual gateway?
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If you are using MS Lync to make a phone call to the PSTN and there are multiple SIP trunks to the PSTN (actually these are trunks to Cisco CUCM and then on to PSTN from there) - how does Lync decide which SIP Trunk to use?
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This is a general question to show my ignorant side.  We have construction going on at my business and they need to widen the driveway.  There are wires that need to be re-located for this construction project and the general contractor notes that there is 1 T1 lines and several fiber lines.  The fiber lines, I know, are run by the local cable company to provide us with digital phones and broadband internet.  I am guessing that the T1 is for the PRI for my ShoreTel phone system.  Is it possible to service my phone system some other way via those fiber lines or is the T1 the only way that I will be able to function?
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Hi,

Running CUCM version 9.1.1 and I'm seeing a lot of reverse lookups, they are failing because my AD server is not setup to accept those but what I wonder is it normal to see so many? what causes the CUCM to execute these queries? I can see like 2 million request in the last 8 hours. You can see attached a few examples.

Thanks,
CUCM-queries.jpg
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We recently moved our CUCM 10.5 publisher to another data center. Call have been mostly good.
But we ran into a period where callers were getting this recording
"Call not allowed due to restrictions on your account". Can the Cisco
Unified Communications Manager 10.5 possibly be responsible for
that recording? Or would that indicate a problem at the provider?
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Hello Everyone,

My name is George and I was hoping to get some assistance in configuring a Cisco 7914 expansion module.  I have been thrown into the fire with this Cisco phone system.  I have very experience with the CallManager system but I am making some headway.  The biggest problem is that my company is running very outdated versions of the software.  We are running CallManager Administration 4.1 (0.11).  I know, I know...it's no way to run a business but that decision is above my pay grade!  I am told to make it work, as I am sure many of you can understand and appreciate!

So here is what happened.  We had to do a complete network reboot and I was configuring a new phone when they took everything down.  When the network was brought back up, it took a few minutes and all but one phone came back up.  Of course, the one that did not come back up was the receptionist's.  After some troubleshooting I found that somehow, her phone was no longer showing in CallManager.  I went through the process of adding the phone again and defining it and configured.  Her phone works fine now.  The problem is with the 7914 expansion module.

The module is showing all red lights on the buttons and the display is blank.  I went to Add/Update speed dials, entered all the name and extension information and clicked on Update and Close.  I thought this was all I needed to do but I was wrong.  In the Configure Speed Dial Settings for SEPXXXXXXXXX window, it says 'Speed Dial Settings not …
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VoIP ISP
Why do some people recommend buying business VoIP from an ISP? What are the benefits to my company? What are the costs?
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I need a Google phone number that will ring on my cell number, for my business cards.

That gives me some flexibility if that number gets spammed, I guess.

Do I need Google Voice?

Could you provide me a link to get that number reserved? And was is Google Voice?

Thanks
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How can you monitor DSP usage on a Cisco ISR used for MTP, transcoding, conferencing? Is there an OID which would let  us graph this?

mtp01-sc#sho sccp conn
sess_id    conn_id      stype mode     codec   sport rport ripaddr conn_id_tx

50630987   251788306    mtp   sendrecv g711u   24918 24576 172.28.72.145                                                                
50630987   251788304    mtp   sendrecv g711u   31384 18902 172.28.16.33                                                                  
50631252   50636167     mtp   sendrecv g711u   19210 17128 172.17.254.1                                                                  
50631252   50636166     mtp   sendrecv g711u   18332 50004 172.16.24.142    
Total number of active session(s) 11, and connection(s) 22
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I am using a grandstream IP PBX UCM6202 and GoIP8 as a VOIP Gateway.
when i dial a number in this pattern +960 7788340 from PBX the call goes out to GoIP gsm gateway but from the GoIP outgoing call has a some unknown character due to space in front of digit 7. for this reason call is not dialed. if there is no space then there is no issue with the call going out. i am using an iphone numbers stored in contacts by defaults has some spaces. is there a way to remove the space when dialled out from GoIP gateway
below is a link to GoIP
http://www.dbltek.com/8-Channels-GSM-Gateway-pd6563436.html
UCM6202.png
Goip1.png
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While getting ready to move a CUCM cluster I was reminded the route lists associate with a particular CM Group and register to a member of that group. But the question: Why is that necessary?
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In Cisco UCM 10, how can I get a listing of all members of a specific device pool?
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I have several DT700 to put on my SV8100 switch. I setup all the settings correctly on the phone and can even use the web programming to get to it. What section do I go to for setting it up on the PBX with an extension and such? On the phone's display it shows Full Port.
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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.