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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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Hi Team,

We have CUCM deployed in our that is integrated with UCCE setup.
We accidentally disassociated all the devices that is defined in the Application user group that maps the devices in CUCM with created agent ID's on UCCE.
We currently associated the devices by manually checking all the available devices in the "Find' field under User-->Application user which is tedious.
We already have a backup scheduler that backs up the entire CUCM configuration that points to a TFTP server.
Kindly confirm the fastest method to restore the specific user configuration from the Application user configuration on available backup.
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Hi,

I`m using avaya ip office 500 v2 and using office manager 10.1

i copied 2 users and gave them their own extensions 3905 and 3906 and their passwords, (its on a vpn) on the other side when i login on the phones using the extension or password it says "extension in use) this extension was never created why is it giving me this error and how to resolve this?

Thanks
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Dear Team, we are planning to deploy the IP PBX (Asterisk) on an Ubuntu server. As my understanding, we can utilize the SIP trunk configurations and lease an VoIP service from ISP, then we do not need to purchase any Voice Hardware card for server, just need a dual Network interface card, am I right?

For example this diagram, the phone calls routes from IP PBX to ISP voice service via an MPLS connection

Internet ----- router ---------- switch ----------- IP PBX (Asterisk) ----------------mpls connection ---------------- ISP for voice service -------> go out for phone Calls

But can we do like this?

go out for phone calls <--------------------- Internet ----- router ---------- switch ----------- IP PBX (Asterisk)

Can the Internet line serve both voice and data? Do we need to inform the ISP for that change?

Many thanks for your help!
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Avaya IP406 PBX third caller goes to fast busy
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snmp not working on shoretel 100da -t1 switch.
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Hello, I have a NEC model ip4ww-24txh-b-tel phone system in our office. We had a new line installed on friday by verizon, and I think something was reset on our phone.
When calls are made to our main line, it doesn't light up but if I pick up the phone the call is there. It can make outgoing calls, but incoming don't ring or light up.
Can anyone help me with this?

Thanks so much.
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IP OFFICE SYS 1 and IP Office SYS 2 configured for Resilience with SCN route on the WAN port connection of IP500 V2 units at Release 9.0We need to add SIP Trunks to take the place of the PRI Trunks. There would be 46 SIP Trunk Voice channels on each IP Office System to replace the PRI Circuits. That is we are adding SIP Trunks for other than SCN Use. These will be
SIP Trunks to replace the PRI circuits and used for connection to the outside world for calls into and out of the IP Office Systems. Does anyone know of a reason that this configuration will not work. Are there any known problems using the WAN port for both SCN Channels and ALSO SIP Voice channels to the outside world?
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All of my Office 365 users have a Office 365 Business Premium license.   We have Conference numbers and Conference IDs assigned by our Audio conferencing service. I see where it takes about using 3rd Party and Import/Export, though I don't actually see where you can enter in the 3rd Party Provider and Import the Numbers for the users.

If I select one of the existing Providers they all want a 6 Digit Conference ID, ours are all 4 Digit.

I would like to be able to use our existing Audio Conferencing system with Skype For Business.

Thanks,
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Please how can i know the verification PIN or extension pin for phone. I am using TDA200 and the phone KX-T7665.

The phone was blocked using *771 and now to open using the code  *770 require the extension pin. How can i know it?
Please help me if possible.
I have access to the PBX system TDA200 through the console
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I have a trouble with Dolby Voice from BT , the below error appears and couldn't  do the meeting :
can't conntect to meeting there are issues with the network connection to avoid missing your meeting, please join by phone BT Meetme Dolby
Dolbyproblem.PNG
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Hi

Currently we having issue to make outbound and inbound call.

We suspect firewall is blocking, how to resolve the issue in Cisco ip phones.
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We have Avaya system installed in our office but the remote phone never worked properly.
they keep on rebooting.
Any expert here who has knowledge of Avaya remote phone setup ?
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Hey Experts, we have a Digium Switchvox VoIP Server. This past weekend our local power company had to upgrade our facilities power. We gracefully shut down everything Friday night, power was restored yesterday afternoon. This morning we have half of our phones not working as they cannot get an IP now. Our LAN and VoIP LAN are attached to our SonicWALL NSA2600, we have 3 Cisco SG500-28-p Stacked switches. What we have found so far is that any phone connected to the Master switch will not get an IP for the phone. Each desk has 1 Ethernet drop, that goes into the phone and the workstation plugs into the phone. The workstations all work fine to phones that don't work. We have rebooted the switches for good measure and nothing changes. Hoping someone can help shed some light on what the problem is.

Here is how the config on the sonicwall looks for the interfaces
Interfaces on SonicWALL NSA2600
Here is the Stack.
SG500-28P Stack
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I'm looking for a guide or handbook on how to program phone buttons on the Avaya IP Office 500 v2.

I'm looking specifically for being able to program buttons to change an incoming call route (for out of office purposes) to change from one hunt group to another.

What I would also be interested in doing would be to have the ability to set the in service/out of service destinations of a specific hunt group and put it in to that mode, which could be an alternative to the incoming call route option above if that is not an option.

Thanks
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We have a user who is moving to another city and we want to set up his Mitel phone so it is like he is still in the office using his same extension as before. I believe we have to setup a vpn so we can do without using teleworker feature on the phone but I am unsure. Looking for advice on how to go about setting up this user to have his phone work remotely. The only VPN we currently use is Cisco AnyConnect but just want to know what steps it would take to accomplish. Thanks in advance for the help.
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I have a Shortetel system with Build 19.42.6503.0 on Server
Shoretel 14.2 same build on the workstations There’s no switches at the offsite facilities, it’s all done through VPN The SG50s are in the same building. Every once in a while my SG50'sgo off line for no reason and have to be rebooted. Is there something that I can do to fix this?
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We have been working for months on correcting issues with our Vonage VOIP implementation. We have replaced switches, created vlans, etc...I am now viewing packet captures using Comm View. Phones are working most of the time but randomly throughout the day, calls will never reach phones...do 1/2 rings and then to voicemail, dropped calls...Using comm view i noticed that ongoing errors since we began capturing ....SIP 401 Unauthorized ...over 1000 since starting capture approx 2 hours ago. Any ideas????
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I am looking for MITEL pbx call accounting software for a time. Through my search i found this website www.expert-coding.com which they have CMar4Pabx call accounting software. Does any one recommend this to me and how good is it ? Your help is greatly appreciated.
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We have an old Asterisk (v.2.x) phone server in our office.  I'm new to the system and need to change an extension number from a rapid busy signal to a working extension.  Also, we have several extension that simple hang-up when dialed (no tones of any sort).  How do we edit those extensions?

I'm new to Linux, but I've figured out how to browse directories and edit conf files.
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We are in the need of a new phone system and there seems to be a mix of vendors pushing a hosted solution.  Has anyone upgraded in the past couple years to a hosted PBX solution and want to share the experience?  Of course the vendors not offering a hosted solution say to stay away from them they are not reliable.  I understand a lot of the bad from hosted is likely a so so Internet connection.  We have 100x100 fiber so I don't think that would pose an issue.  The big downfall I see is you pay for the hosted forever where an appliance based system, you buy it once and usually good for 10+ years.
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Hello There,

I have a a2billing installaed on centos , to witch I want to send the number base64 encoded, and to transalte it from a2billing,

I can do this from asterisk to asterisk , from sipphone to asterisk , but not from sipphone to a2billing,

I tried modifing the a2billing extension :

[a2billing]
exten => _123.,1,SET(EXTEN=${BASE64_DECODE(${EXTEN))})
exten => _123.,2,AGI(a2billing.php,1)
exten => _123.,n,Hangup()

But i dosent work , any idea would be appreciated
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Hi
we have configured h323 Gatway.

when we call landline number 3175 IP Phone to 61871111 and 6187100 are working fine. but other all landline number eg 61871324 and 62300192 etc unable to connect getting busy tone.
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Hi
we have configured h323 Gatway.

when we call landline number 3175 IP Phone to 61871111 and 6187100 are working fine. but other all landline number eg 61871324 and 62300192 etc unable to connect getting busy tone.
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I have a 14 office MPLS link that was working with a Mitel phone system.   We had an emergency need to move the offices so I came up with a solution to use an Cisco ASA to create an IPSec VPN tunnel from the new location to the firewall.

Currently, I only get one-way audio when making calls.  
Additionally, my Mitel Controller is showing a SIP LINK Failure alarm.
I got the tunnel up and running and everything can be pinged.

I am so confused.

Maybe this is a problem with my Access-Control List?  

Here is the top part of my Cisco Config:

ASA Version 8.4(2)8
!
hostname dartmouth-asa
domain-name test.com
enable password OzKLyBY8hcexbQv8 encrypted
passwd 2KFQnbNIdI.2KYOU encrypted
names
!
interface Ethernet0/0
 description INTERNET/OUTSIDE
 switchport access vlan 2
!
interface Ethernet0/1
 description VOICE
!
interface Ethernet0/2
 description DATA
 switchport access vlan 3
!
interface Ethernet0/3
 description DATA
 switchport access vlan 3
!
interface Ethernet0/4
!
interface Ethernet0/5
!
interface Ethernet0/6
!
interface Ethernet0/7
!
interface Vlan1
 nameif voice
 security-level 100
 ip address 192.168.171.1 255.255.255.0
!
interface Vlan2
 nameif outside
 security-level 0
 ip address dhcp setroute
!
interface Vlan3
 nameif data
 security-level 100
 ip address 192.168.172.1 255.255.255.0
!
boot system disk0:/asa842-8-k8.bin
ftp mode passive
dns server-group DefaultDNS
 domain-name test.com
same-security-traffic…
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I have configured asterisk by ./configure.
It has been completed successfully. But "./configure –with-crypto –with-ssl –with-srtp=/usr/local/lib" command not working.

please have a look.

[root@localhost asterisk-13.6.0]# make menuselect.makeopts
make: `menuselect.makeopts' is up to date.

[root@localhost asterisk-13.6.0]# menuselect/menuselect enable format_mp3 enable res_config_mysql enable app_mysql enable app_saycountpl enable cdr_mysql enable EXTRA-SOUNDS-EN-GSM
**************************************************
*** Install ncurses to use the menu interface! ***
**************************************************

but i have seen ncurses has been installed.
[root@localhost asterisk-13.6.0]# rpm -qa | grep ncurses
ncurses-libs-5.9-14.20130511.el7_4.x86_64
ncurses-5.9-14.20130511.el7_4.x86_64
ncurses-devel-5.9-14.20130511.el7_4.x86_64
ncurses-base-5.9-14.20130511.el7_4.noarch


how can be solve. please help.
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.