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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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Just want to get folks opinions.  Anyone using them?  Any feedback?
Trying to implement sparkboards in every new office and eliminate things like conference phones, polycoms, and all that legacy stuff.

Thanks.
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Nothing ever in the clear!
LVL 1
Nothing ever in the clear!

This technical paper will help you implement VMware’s VM encryption as well as implement Veeam encryption which together will achieve the nothing ever in the clear goal. If a bad guy steals VMs, backups or traffic they get nothing.

Hello, I work with asterisk servers and mainly soft SIP clients. Customers would like to have the possibility to hang up an incoming call. That's the first step...no big deal. But what if the call is a "call group" call i.e. a call to multiple devices at the same time and you want the command from the client not only to hang up his device while the others keep ringing, but to hang up all legs of the call - caller and all callees. Is that possible?
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There is a fellow it guy that keeps forwarding his line to my extension when he is "busy"coding or something else, he always forgets to disable it and I've had enough of it so I want to disable his ability to forward calls, or disable my number from accepting forwards from him only if possible or altogether from anyone, I don't want do disable the function globally just in his ext number forwarding or mine receiving from him/any. ( I have console admin access)
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Hi,

 I am considering "converting cable operator provided phone service to VoIP phone service".
 Can you recommend a vendor and explain why you like them?

 I am aware that there are multiple players - RingCentral, Vonage, 8x8 ... etc.

Thank you for your input in advance.
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I don't much about the Avaya IP-Office 500.  I've attached some pics that may help.  The first pic is the blue cable that connects to the phone but is no longer providing service.  
During our rack equipment move, we may have re plugged in one cable on the IP Office in the wrong port.  I'm not sure if that matters as I'm not proficient in PBX.  I had one of the staff members mark cables based on what phone goes out if we unplug it.  I'm not sure if that's done yet.  I left for another meeting.  
Anyway, are the patch cables placement very fickle for the IP-Office?  We may have accidentally plugged in one cable to the wrong port when moving things and that caused the phone to no longer light up.   Sorry, I'm walking blind here.  I would greatly appreciate some feedback.
blue_cable_from_face_plate_to_phone.jpg
a_and_b_drops.jpg
IP-Office_Ports.jpg
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For the Current PBX system, the Telco delivered an Adtran, off the Adtran there is an amphenol cable that is punched down to a 66 block.  There is also a CAT5 off the adtran that is also punched down to a 66 block.  

I am installing a Cisco VoIP solution,  but I am not sure how to connect my Cisco voice gateway router with a PRI/T1 card to the Adtran.  I'm not sure if the telco will have to hand the circuit off to me another way, or if I can just connect to the adtran using the ethernet port.  I'm having a lot of trouble getting info from telco.    Could anyone lend me some insight on this and their experiences?   I do have some VOIP experience, but no experience with traditional pbx, 66 blocks, etc.  Thanks.
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we create a sip trunk, cisco phones can call to avaya, when a try to call a cisco show "INCOMPATIBLE" im not a expert on avaya, ideas  ?

log avaya -> cisco

274138126mS PRN: ++++ END OF TCP MONITOR CLIENT DUMP ++++
 274160750mS Sip: SIP Line (17): License, Valid 1, Available 15, Consumed 0
 274160750mS Sip: SIP Line (17): sip_trunk_config_items 0002c10c, voip.flags 00040949
 274160750mS Sip: SIPDialog f172d9f4 created, dialogs 1
 274160756mS Sip: 0a3c1e8c0003346c 17.210028.0 33506 SIPTrunk Endpoint(f172f28c) received CMSetup
 274160757mS Sip: 0a3c1e8c0003346c 17.210028.0 33506 SIPTrunk Endpoint(f172d9f4) SetLocalRTPAddress to 10.60.30.140:46754
 274160759mS SIP Call Tx: 17
                    INVITE sip:50528337@10.120.200.20 SIP/2.0
                    Via: SIP/2.0/TCP 10.60.30.140:5060;rport;branch=z9hG4bKfdfadfc4705f6d30aeaf1ab49c1310a3
                    From: "Karina Bolado" <sip:SIPDefault@10.120.200.20>;tag=b25a03e400a8ebb0
                    To: <sip:50528337@10.120.200.20>
                    Call-ID: 554091e18ddc18667746c51e49f9a926
                    CSeq: 740975927 INVITE
                    Contact: "Karina Bolado" <sip:SIPDefault@10.60.30.140:5060;transport=tcp>
                    Max-Forwards: 70
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
                    Supported: timer,100rel
                    P-Early-Media: supported
                    Min-SE: 90
                    Session-Expires: …
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Hi

I've been wrapping my head around this for quite some time - still accomplishing nothing but headache. Here's the setup

Lync 2013 mediation server
Lync PSTN Trunk points to Avaya Aura RED
Avaya Aura RED routes PSTN traffic to Avaya phone central

We have landline numbers in Lync, and cell phone number with a PSTN provider

Lync <-> avaya RED gateway <-> Avaya phone system

Also - we've setup so that calls to cell phone in PSTN, will also simultaneous call Lync number.

so;
1. user call my cell phone (tel: 99 99 99) from his IP-phone/cell phone
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I answer the call with either cell phone or lync - the ringing will stop on the other device

so far so good.

But if this process is repeated, but this time the call is initiated from a Lync client at the internal office

1. user call my cell phone (tel: 99 99 99) from enterprise voice enabled lync client
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I …
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Hello,
Need a question answered being asked of me and our Cisco CUCM 11.1 phone system. I need to know if Cisco CUCM 11.1 will integrate with NetSuite CRM via TAPI and/or CTI. If so, any third party apps needed or can it be done within CUCM?
Thanks in advance!
Steve
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Hi All

I'm in the process of planning out implementing site codes dial codes within my company's CUCM environment. We currently use 4 digit dialing across all sites, but this is no longer scalable with the amount of expansion we're experiencing.

I've set this up in a lab environment, and it works as expected. However, I'm stumbling with the modifying the Calling Party ID when dialing between sites. So when someone in Site A dials an extension in Site B, the Site A caller ID is prepended to include the Site A dial code.

I'm think I have to create multiple translation patterns, intersite partitions, and CSS's to do this; but this also seems messy, so I'm wondering if there's a better way to accomplish this?
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Better audio for more successful meetings
LVL 2
Better audio for more successful meetings

Challenge: S&ME was tired of poor audio quality of Skype for Business calls in mid-sized meeting and training rooms. They were looking for a reliable and cost efficient solution to replace the existing conferencing system.

I want to change time of stages of phones that ring on incoming calls.
Documentation on Incoming call system options. The main two I'm primarily interested in there behavior are 22-01-04 and 22-01-09.

04 - Normal DIL Incoming Call No Answer Time

09 - DID to Trunk to Trunk No Answer Time
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Hello Everyone

I'm currently in the process of migrating our current PBX system away from asterisk to Freeswitch. I am using FusionPBX on Debian 8. I am using the freeswitch webapi to originate calls. I am at the stage where when I execute the command, it rings the call centre agents phone and the customer automatically without the agent manually dialling the number. I would like the ability to manually specify a caller id number for the outbound leg of the call. At the moment it is not sending any caller ID. I can manually specify a caller ID number in the extensions page, and it works statically, however we have a need for the caller ID to be dynamic.

http://X.X.X.X:8080/webapi/originate?{click_to_call=true,origination_caller_id_name='Click to Call',origination_caller_id_number=1000,instant_ringback=true,ringback=\'%(400,200,400,450);%(400,2200,400,450)\',presence_id=630@X.X.X.X,call_direction=outbound,sip_auto_answer=true,domain_uuid=52b92yy9-7fb7-52ae-9e9e0595058bcdaa,domain_name=X.X.X.X}user/630@X.X.X.X &transfer('SOME EXTERNAL NUMBER XML X.X.X.X')

What do i need to add to this web address to get it to send a custom caller ID number to the customer outbound?

Many thanks in advance.
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We are on Avaya IP Office 8.x (?) ....   yes we are planning on an upgrade to 10.? later this summer.  Everything had been fine until about 3 weeks ago when all of a sudden our voicemails stopped going to email.  We are not having any luck with our vendor figuring out the issue.  Have stopped and restarted voicemail services.  I don't understand a lot about Avaya IP Office as I was more familiar with Cisco Call Manager.    We have many other services that send emails through our SMTP server so I don't think that is the issue.  Can anyone give me some insight or point me in the right direction?

Thanks
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Hi

We have a 3CX voip server that is hosted in the cloud and we have a T49G phone we need to configure to work with the 3CX server.

Now, 3CX server doesn't officially support T49G phone but they say it should work as a normal SIP phone without any provisioning.

I have tried simple SIP config by putting in the extension number and its password but the SIP registration keeps failing.

The packet capture on the 3CX server is showing 407 proxy authentication error.

Can someone help me configure Yealink T49G phone on a 3CX Voip server?

Happy to provide packet capture or any other logs you may need.
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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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When ever i use skype for basic chat the output on the screen shows first in non-italic characters and then repeats itself in italic characters.  Is there any way to shut this off and have it all appear as just one non-italic output?
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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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Free Tool: Site Down Detector
LVL 10
Free Tool: Site Down Detector

Helpful to verify reports of your own downtime, or to double check a downed website you are trying to access.

One of a set of tools we are providing to everyone as a way of saying thank you for being a part of the community.

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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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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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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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.