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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I call an insurance company and there is no "press 1 for english" today but there was yesterday.
And there is a long hold which wasnt there yesterday.

Todays phone call (I tried on 2 phones) goes straight to hold music.

If I dont press 1 for english, how does company know I want to speak in english?
Are buttons set up by company just for "future reasons" and the same operator always picks up?
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VIDEO: THE CONCERTO CLOUD FOR HEALTHCARE

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Hello Experts - Our company switched to the iPhone 6s and we've run into a problem.  The phone does not appear to be correctly passing number presses when dialing in to our phone system and attempting to use the auto attendant directory.  Instead, it will constantly say the name was not located.  It works fine when not using Apple Carplay from the same location.  Any ideas on what I can do to fix this?
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Hello Experts,
We are using Vicidial version 2.6-381a with Asterisk 1.4.44-VICI, I have strange issue one of my user extension 508 voice mail going into ext 801 voicemail box when using DID but when dial extension 508 it goes into right mailbox of 508, i have checked the user phone settings, DID settings to make sure DID rings and send voicemail to right extension from vicidial gui, here is Dial plan for ext 508 which is setup properly can someone please help me figure this out. Thanks

exten => 3508,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 3508,n,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/44048020)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/85100075)
exten => 3508,n,Wait,2
exten => 3508,n,Dial(SIP/508,15,Ttr)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/85100031)
exten => 3508,n,Dial(SIP/508&SIP/801&SIP/505&SIP/507&SIP/509,15,Ttr)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/spool/asterisk/voicemail/default/508/unavail)
exten => 3508,n,VoiceMail(508@default)
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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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I am attempting to troubleshoot our VOIP calls with packet captures and wireshark. Unfortunately, when I choose Telephony->VOIP CALLS, there is nothing displayed. I definitely testing captures during 2 test phones calls and insured protocols are enabled but still nothing. Any help would be greatly appreciated!
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Hello,

We have a branch office using Cisco 8851 phones running the latest up-to-date firmware. When we lose the link to our head office where the Call Managers (v10.5.1) are the phones go into SRST and work. I can see 58 licenses being used on the Voice  Gateway (Cisco 2911).

When the link returns the phones show that they are re-registed to the Call Manager, however, when I look at the phones on the Call Manager they are not registered. When I look at show call-manager-fallback on the router I still see 58 SRST licenses being used.

We have to walk round and reboot all the phones.

The VPN tunnels are connected via Fortinet 300D's and the policies allow ALL services. From what I have read the ACK's to the keepalives may not be getting through

Any help would be appreciated.

Thanks,

Glenn
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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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Granular recovery for Microsoft Exchange
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Granular recovery for Microsoft Exchange

With Veeam Explorer for Microsoft Exchange you can choose the Exchange Servers and restore points you’re interested in, and Veeam Explorer will present the contents of those mailbox stores for browsing, searching and exporting.

Of can it work by attaching directly to CUCM as well? I think it's only with Expressway but way to check. Thank you.
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Have a client with a Toshiba CIX 100 PBX.  For the most part, have been able to fix some old setup issues with the system.  However. faced with an issue that I can't seem to figure out.  I have experience with VOIP systems and translating back to a digital PBX is causing me some frustration.

When an outside call comes in, all phones ring.  With a VOIP system, I would be looking at a hunt group (or at least the ones I work on).  I need to be able to limit which phones ring on an incoming call. So questions are:

1) By default, does the CIX automatically ring all phones on an incoming call?
2) How do I assign a single phone to an outside line?
3) How do I assign a hunt group to an outside line?
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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.
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Hello all,

I need to configure the Sonus gateway to route inbound calls to Toll Free number (main number) based on their area code to their particular response group on my Skype for Business server.

For instance, I have a response group for Chicago with DID 312XXXXXX and I have a main toll free number in which case all the calls come to this particular DID (800XXXXXXX) so I need to route all incoming calls from Chicago to the Chicago response group and the same thing for every other state.

I made the area codes already so all calls coming from Chicago state is directly forwarded to the Chicago response groups but the only thing I couldn't do is route all inbound calls coming to the Toll Free number to the response groups.

I would appreciate any advise on this.
ThanksAreaCode.jpg
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I’m trying to find out if a Cisco PBX can do a global caller id delete. If at any phone on the system, if you delete the caller id it removes it from all phones.
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Hi

I am tasked with setting up a lone staff alert button on a 5320 IP phone that will ring or alert all other phones (12) on the system, is this possible on the mitel 3300 system?
Each phone would hopefully have a key to press that will alert all other phones that the user needs help,

Thank you in advance of any help.
0
How to Use the Help Bell
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How to Use the Help Bell

Need to boost the visibility of your question for solutions? Use the Experts Exchange Help Bell to confirm priority levels and contact subject-matter experts for question attention.  Check out this how-to article for more information.

I'm trying to add a single queue 'csqFabFursEscal' in the working UCCX script (FabFurs_Working.pdf).  I've made changes in an attempt to add the queue in the broken script (FabFurs_broke.pdf) and it doesn't work.  Everytime I load the broken script into my test application I get a system message saying their are problems and to call back.  

This isn't a complex script and I know this is an easy solution for someone better at scripting.  I would really appreciate a UCCX scripting guru take a look.
FabFurs_broke.pdf
FabFurs_working.pdf
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On cisco IP phones (model number 7911) it stores some useful information about placed/received calls in the directories application- is this data stored locally on some storage within the phone, or would this be stored in a central database in a managed voip environment, if so being a cisco device can you elaborate where that information may be stored.
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Hello - im looking for hold music that is free to use. Anyone have a link to download one?
0
Hello,

We are using FreePBX, and have Cisco 525G2 phones, and we added a SPA500s sidecar, how do i configure extensions on it?
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Hi,

I have a question regarding ShoreTel Director setting.
We have about 100 DID numbers and we use "Hunt Groups", "Routes Points", and other settings.

Recently, we receive a number of fax calls on our company main phone number.
For now, we check the history of calls and check on the internet for the company's phone number.
Then, we call the company to inform that it is not the fax number, but it is our company main phone number.

Is there any settings on ShoreTel Director to transfer the fax call to the correct fax number?

I appreciate if you can tell me step by step process for this setting.


Regards,
Aki
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I have one user that is not able to search the GAL for contacts.

He is searching within the “My Contacts” tab but his searches show no results.

He is using SFB on Windows 10. The rest of our users have no issues doing the same.

What am I missing?
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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.