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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.

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?
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?
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://
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)
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!
Of can it work by attaching directly to CUCM as well? I think it's only with Expressway but way to check. Thank you.
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.
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.
Hello - im looking for hold music that is free to use. Anyone have a link to download one?

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

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.

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?
We have a new issue on our phone system.  If we receive a call and try to transfer it to another internal number, but that person cannot take the call...if we go back to the original caller and try to transfer the call a 2nd time, transfer is not available.  This all of a sudden started happening.  We are on CM version  Thank you.
What is the process for adding a ten digit (area code + seven digit phone number) phone number to an Office 365 Skype for Business (Lync) account?
We supply a software application that is run on Remote Desktop servers (each client has their own Virtual Server).  One of the areas I am currently researching is to provide the ability for users to click on a "Call" button in our application, and dial the number (whilst connected to the RDP).  I have found a couple of solutions that will allow me to press the "Call" button, which will then dial the recipient's number (along with the user making the call), which in theory, should be ok - but I am yet to test it and am interested in all options.

For me, the perfect solution would be for users to be able to plug in a headset on their local laptop, run the application on the Remote Desktop and call - without impacting server performance in any way.  Or, if they have a phone system already in place - somehow dial through that.

Any pointers, products, suggestions - would be massively appreciated.
Hello We are using Vicidial version 2.6-381a with Asterisk 1.4.44-VICI, I am trying to set up my dialplan entry so we can call out from Canada to Irland and Trinidad, our current Dialplan allow us to call only witin north america but not outside,
Here I have the country and area codes that need to be called out from Canada
Irland 353-287-xxx-xxxx
Trinidad 868-788-xxxx

Here I have my Dailplan that i have made up to call out but no luck can someone please guide me correct dialplan please

exten => _9353NXXXXXXXXX,1,AGI(agi://
exten => _91868NXXXXXX,1,AGI(agi://
exten => _9353NXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _91868NXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _9353NXXXXXXXXX,3,Hangup
exten => _91868NXXXXXX,3,Hangup

Here is the call_log but no luck (last 4 digits of phone number being modified to xxxx)

[Nov 1 13:48:07] == Using SIP RTP CoS mark 5
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:1] AGI("SIP/298-00017de5", "agi-NVA_recording.agi,BOTH------Y---Y---Y") in new stack
[Nov 1 13:48:07] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Nov 1 13:48:07] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20171101134807_298_1868788xxxx)
[Nov 1 13:48:07] -- <SIP/298-00017de5>AGI Script agi-NVA_recording.agi completed, returning 0
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:2]
Do the Comcast Panasonic KX-TPA65 VOIP phones also have a network jack for plugging in a PC (in addition to plugging the phone into the network)?

We have a Polycom SoundPoint IP 350 and a 450 at a remote office that connect to a FreePBX 2.11/Asterisk 11.7 box in our main office. The two sites talk using a site-to-site VPN via our FortiGate 30E firewalls. When the two remote phones register across the VPN, everything works great and there are no problems. However, if the internet goes out at the remote site, which it does often, the remote phones will never try re-registering again. Even restarting the phones by power cycling them does not let them re-register again. The two things I've found that works is to either upgrade OR downgrade the firmware by 1 version, or to restart the firewall at the remote office.

I've updated the firmware on both Polycom phones to the latest versions, applied the latest firmware to both firewalls (we have other remote sites that do not have this issue) and made sure that SIP ALG and VOIP application control are disabled on both firewalls. I can consistently reproduce the issue by unplugging the modem (not the firewall) at the remote site and then plugging it back in. When internet comes back up, the phones will not be registered and will never try registering again until the firewall is restarted or firmware version is changed. I've also played around with registration expiration and timeout settings on the phones, but this doesn't seem to work, either.

I'm thinking this may be a FortiGate firewall issue, but it's strange that my other 3 remote sites (using the same …
We have a Cisco UC520 that has been in use for many years with no configuration changes apart from user / extension amendments.  We have 8 lines on an ISDN-30 connection with two incoming numbers: one for voice calls, the other for fax.

Currently the fax calls are working fine, and we can make outgoing calls.  However, incoming calls on the main number connect but are immediately dropped.  We have confirmed with the telco that there is no fault, and also rebooted the UC520 and the PoE switch that connects the phones.

How can we troubleshoot this problem?  Thanks in advance.
I'm running IP office manager v8.1
for some reason the call forwarding has stopped working.

I have the system to forward calls at night by manually hitting the "Night forward" button, which is ext 298 and it is forwarded to an outside number.

when i hit forward and hit that number, the calls don't get forwarded and they're not ringing at office either. help?
I've been struggling with VOIP call quality. One thing I notice is where calls to PSTN or Conference are made from work on the west coast or from home on the west coast - the traffic takes a 100ms trip to the east coast to get to the Skype voice control and RTP gateways. Is there any way this can be altered so you can use gateways on the west coast to reduce hops and delay to the Skype voice gateways?
I have a pair of C-level users who both experienced a problem at one of my sites. I'm trying to determine if it's a GoToMeeting problem (not my problem) or a phone problem (definitely my problem). You guys will give me a quick answer, I just know it. :-)

When the users join the audio portion of a GoToMeeting event from their Cisco desk phones (on my CUCM-powered phone system), right after the point where they enter their PIN number for the meeting, they are supposed to press the pound key "#" to join the audio part of the meeting. Every time each of them presses the "#" key on their desk phones, either the meeting or the phone hangs up the call. When they join using their cell phones it works fine.

This just started happening and no changes have recently been made to the phones or the CUCM settings. I'm told that it happened once before but was not reported, and that other occasion occurred several months ago but then the conditions returned to normal operation and they didn't see this again until yesterday.

Can I get some opinions on which party is responsible?

Thanks experts!
Hi All-

Having a weird issue on my Allworx 48x- When incoming calls come from customers and they dial  an ext. say 226 it goes to 222. There is no forwarding set at all on 226 and this happens randomly. Any idea what might be going on? I have replaced both phones, rebooted the allworx server, but issue still remains. It started recently after i setup 222 for a new employee which i selected from the available list of extensions.
got a missed calls from last two months, This evening I got a ring from +381 628302791 at 4:08.Whoever calling rings 1-2 times and disconnects immediately.
searched on google Which country code +381, It's from Serbia I shocked why I got calls from this area.
I would like to know, is this going to be a problem? Like hacking etc??
We have KX-NT700 model, and was working fine with our Panasonic PABX, until we check it today.

The dial tone is missing when we press speaker phone, hence not able to call.

The basic settings are intact as i compared with Phone Manual, such as :

We are trying it through the telephone extension line (Ext: 222) , while this line is being used with KX T7665 without any issue.

For your knowledge, this phone had an extension of 300 while it was working fine before.

Please suggest if there is any configuration issue in the phone or this is a mechanical fault in it ?

Thanks in advance.
I have a ring group containing 3 extensions.
I'm trying to achieve a situation that all of the extensions in this ring group will ring, unless one of them is occupied (in this case, the call will be forwarded to the next destination).

After reading all of the ring strategies available, I couldn't find one that could do it.

Anyone can help ?

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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IP Telephony