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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We have a FreePBX 13.0.196.2 system running Asterisk 13.19.1 and we're using Asternic Pro 2.2.6 for call reporting at a call center. The problem we've had since the system went in is that all of our users are in a queue. If a call comes in, all the phones in the queue ring. Person 1 answers the phone and after talking for 30 seconds or so, transfers (blind or attended makes no difference) to person 2. Person 2 talked for, say, 50 minutes. As soon as the caller or person 2 hangs up, all time associated with the call is attributed to person 1 and person 2 takes no time for the call in the reports. We've tried both blind transfer on the phone as well as just pressing ##extension during the call with the same results. Is there anything I can do to get call reporting to work properly within Asternic so that call times with transfers are accurate?

Thank you
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I'm trying to add a VM to a user in Exchange and I keep on getting this error.  We use 3CX as our VoIP system, in a VM, and Exchange is where the VM is stored.

Starting yesterday, when I try to add a VM to a user, I keep on getting this error. NO changes to my 3cx server or Exchange, so this is odd.

I checked my event logs app/system, and didn't find anything that would point to an issue.  
Also did a wireshark capture for 10 seconds on my exchange during the time I received the error, and still, I didn't see any issues in the wireshark that could be a problem.  
I looked on my 3cx system, in the logs, and nothing shows up there as an issue.

I don't think this would have anything to do with it, but my manager disabled TLS 1.0 on our exchange server to make it more secure, so I doubt that would have anything to do with it?

Not sure what else to do, any thoughts?
exchange VM
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I’m looking for a simple answer to a simple question that it seems our Mitel vendor is having some difficulties answering: are Mitel MiVoice 250s formerly (Mitel 5000) networkable at layer 3 over a private VPN to other Mitel platforms, namely Mitel MiContact servers/Mitel virtual MCD, and Mitel 3300

Our environment consists of the following.
 
Data center:
·         Mitel border gateway
·         Mitel MiContact server
·         Mitel virtual MCD

Corp HQ:
·         Mitel 3300

35 branch locations:
·         MiVoice 250
·         No Mitel border gateways at the branches
 
 
We presently send SIP traffic directly to the 35 MiVoice 250s “over the top” to our broadband circuits which provide excellent call quality.  During primary broadband failure, we’d like to have calls failover to our 4G/LTE cellular service, which we tested, and work almost as well (in terms of call quality) and are not limited in number of calls.  The only way we can have automatic failure to our 4G/LTE backup is through the purchase of an Mitel border gateway (x 35 locations) and additional SIP licenses for the backup cellular route.
 
We’d like to send SIP calls for all 35 branch locations to our data center Mitel border gateway, and have the SIP calls distributed to each of the 35 locations over our private VPN.  For each of the 35 locations, all data traversing the private VPN fails over automatically between the broadband circuit and the 4G/LTE backup cellular route.  Routing our …
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Does anyone know how to unlock a Shortel desktop phones (model 210 for example) to work with Ring Central?
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Hello,

I am wondering if anyone knows what cordless units are compatible with the NEC SV8100 phone system.

I need to replace a couple of cordless handsets but cannot find any, even used. There are plenty of corded DTL-12D-1 BK TEL DT300 units available but I require cordless. Ebay has some base sets but no accompanying hand unit. I'd like to get a couple of more years out of the system which is still working fine.

Thanks!
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Dear Experts, we got this requirement for PABX configuration in a hotel/resort:

ext.png
A master extension will be represented for each villa, then several sub-extensions will represented for blocks in villa. Any incoming/outgoing call will be directed via master extension. Sub-extensions will be used to calling inside a villa (between blocks) only, when calling outside, master extension shall be used.

For example the screenshot is showing Room 202 and its 6 blocks. It will have 1 master ext (8202) and 6 sub-extensions for 6 blocks.

So can NEC and Alcaltel meet this requirement? And what are the things we need to consider? Many thanks!
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I am looking for a way to check the trunk and the DID charged  for the 3cx which is currently setup and working properly.
How can check these information?
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We are moving to Office365 and are looking to slowly move to O365 phone system as well. But we cannot do all of this at once. Does anyone have any experiance with this? and if so, how do the on prem PBX phones work with the cloud phones? Ex, if I wish to call an extension on the PBX system or if I wish to transfer a call to an extension on the PBX system....can anyone advise?
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We are looking at implementing a Shoretel telephone system at one of our remote offices. Today, it only has an internet connection.

We know we need to install a PRI for voice services, but we are being told we need a DVS server for voicemail, administration and call pilot (whether it is on-site or remote). Couple options are being presented to us:

Option 1: Get a windows server and install Shoretel (on site), to be DVS, with a network connection (dedicated for this, separate from the internet mentioned earlier) back to our Head Quarters server (at our datacenter); site to site VPN
Option 2: Same site to site VPN, but DVS in cloud at our datacenter (so it ends up being remote); if site to site VPN goes down, so does the connection to DVS

Question: for a site with no server, what is the best solution (if we are trying to avoid installing a physical server). Is there a Shoretel appliance that can be installed onsite, to incorporate the DVS features? We will still get the site to site VPN, for connection back to Head Quarter server. Ideally, if the StoSVPN goes down, we don't want there to be impact to voicemail and such.

Equipment planning to use:
1x PRI circuit
1x site to site VPN
1x Shoretel T1k
2x Shoregear 50V

User count is up to 10 people (small site)

Recommendations?
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Sometimes SIP channels get "stuck" in Asterisk and cannot be hung up without  restarting Asterisk. FreePBX shows the channels in the "Active Channels"  list under the Asterisk Info "Channels" menu but the "Active Channel" count at the bottom of the list is less than the number of channels in the list.  When trying to hangup one of the "stuck" channels using the channel request hangup <channel> command from FreePBX Asterisk CLI we get a 'channel not found' error. Asterisk version 13.9.1, FreePBX 12.0.76.6
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Running into a problem with CUCM when doing an export of Phones with Specific Details.  I have several phones that have 15 lines (18 lines in the button template total) on them but when exporting, the 2nd-15th lines don't show up until Line Number 30 or later.

What would cause CUCM v11.5 to export the phones lines like that?
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I wish to setup On-Premise IP-PBX Business on my local server using proxmox vm. While being able to sync with an on premise Crm like bitrix24

I was thinking of getting switchvox. Kindly advise.
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Hello,

I have Cisco Call Manager 11x with Unity 10x

Once in awhile I see an issue where I dial a users extension, the phone rings, they don't pick up, and instead of kicking me to their Unity Greeting to leave a voicemail it kicks me to "Your call cannot be completed as dialed. Please consult your directory and call again or ask your operator for help."

I wait 1 minute, dial, it works. it's like call manager doesn't forward it to unity.

Is there a limit on how many connections Unity can accept from Call Manager simultaneously?
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We recently started noticing that a certain floor of a building on campus can only call 1 direction.  When either side tries to call the one side in the bottom of our building cannot hear anything that the other side is saying. This only happens to the 3905's on the first floor of a specific building. If you use a higher model phone on that floor it doesnt happen and if you bring a 3905 up a floor or anywhere else and it functions perfectly.  Strangely, this isn't consistent as some 3905 phones on that floor can call some phones in other areas of the building and there are no problems at all.  I am stumped.  Any suggestions would be appreciated!

Phone system is Cisco
Switches are Cisco 2960S-48fps-L models with IOS 15
Call Manager and Unity versions are 11.5.1
Phone affected on the first floor of the one building are Cisco 3905's

If any log files are needed I can provide those.
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Need a LAN line for my home, but don't want to pay much.  I don't have internet or Cable TV at my home.  I also don't want a cell phone either, just need an actual phone at home for the lowest price possible.
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Please recommend an on-site call recording solution for a UK based call centre. We are using Avaya IP Office 11 phone system (IPO control unit 500 v2).

It needs to be sold and properly supported in the UK (even post Brexit).

We are having major issues with our current solution and are seriously considering a migration to a new product. I'd prefer any recommendations based on honest, hand-on experience rather than a sales pitch.
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We have a voice gateway at a remote location currently directing outbound calls over a pair of POTS lines. The location is very small but even so E911 compliance requires each of the 4 handsets to identify their unique location for caller ID.
I am looking to redirect outbound calls from this location back over the ELAN and out through our Call Manager/SIP connection as we do with other locations. All phones/locations/SRST etc have already been set up in CM, so I suspect this is just a matter of directing the calls.


Can someone give a brief/broad pointer of where to look (in the VG or CM)? At this point if I have the basic starting point I may be able to compare to other locations to figure out, nothing is jumping out at me presently and I'm unsure which settings influence or are irrelevant.

Thanks in advance!
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Hi All, I am attempting to configure an AAPT IP based trunk in 3CX via a dedicated TPG SIP service, and am struggling to get it working.

What I really am after is examples of working AAPT trunk configurations that I can compare my set up to.

If anyone out there could provide some examples of correct trunk configuration, I would be extremely grateful.

Wireshark shows OPTIONS messages successfully hitting the phone system from TPG SIP server, and 200 OKAY messages being sent to TPG SIP server.

EDIT:

 - 3CX on premise
 - Wireshark shows OPTIONS being received and 200 OKAY being sent to/from TPG SIP IP
 - Dual WAN connection with dedicated SIP service on WAN2
 - NAT on WAN2 to local IP of 3CX
 - Static routes are configured to route required SIP traffic in/out via either WAN1 or WAN2 depending on what port is required

Kind regards,

Nick
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Hello,

I have a problem with an asterisk server:

I have a SIP trunk from Vodafone. When I call from another provider ( lets say Orange ) redirecting from Softphone to another extension works. When I call from the same provider ( Vodafone - my sip trunks use vodafone ) and try to redirect from the softphone to another extension, the call is intrerupting.


This is extension.conf related to extension number used for redirecting: (I masked the real number)

exten => _yyy268,1,Set(CALLFILENAME=${CALLERID(num)}_${EXTEN}-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)})
exten => _yyy268,n,MixMonitor(/var/inregistrari/in/${CALLFILENAME}.wav,b)
;exten => _yyy268,n,Goto(ivr-liber,9,1)
exten => _yyy268,n,GotoIfTime(18:00-23:59|mon-fri|*|*?time,5692,1)
exten => _yyy268,n,GotoIfTime(00:00-08:00|mon-fri|*|*?time,5692,1)
exten => _yyy268,n,Dial(SIP/268,20,tk)
exten => _yyy268,n,Dial(SIP/241,60,tk)
exten => _yyy268,n,Congestion()
exten => _yyy268,n,Hangup()

THis is sip.conf related to extension used for redirecting:

[268](sets)
dtmfmode=rfc2833
t38pt_udptl=yes
qualify=yes
notifyringing=yes
call-limit=2
callerid=268
nat=no
mailbox=268@default
secret=sssssssss
canreinvite=no
callgroup=1
pickupgroup=1


[241](sets)
dtmfmode=rfc2833
t38pt_udptl=yes
qualify=yes
notifyringing=yes
call-limit=2
callerid=241
nat=no
mailbox=241@default
secret=rrrrrrrrrrrr
canreinvite=no
callgroup=13
pickupgroup=13



THis is sip.cinf related to the trunk used:

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I'm wanting to setup QoS for Skype for Business Online. When I do a packet capture I see that the real time
ports that come into play are the UDP 50,000 - 59,999. The article below calls the 50,000 - 59,999 as optional.
Is there any way through group policy to tell skype to use the UDP 3478, 3479, 3480, 3481 only or at least
to prefer it? Marking all TCP/UDP 50,000-59,999 for EF classification seems pretty broad.


https://techcommunity.microsoft.com/t5/Skype-for-Business-Blog/Simplified-port-requirements-for-Skype-for-Business-Online/ba-p/77094
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Please tell me I'm wrong:  When using S4B to call a business that has a telephone auto-attendant, our S4B dialpad works just fine.  However, if I and an employee call a business together in a S4B call, the dialpad buttons do not work.  MS tends to suggest that this is a known bug.  We're about to agree and leave it at that... and leave S4B.

But really?  What an obvious thing to need to do.  We REALLY need to do this to train our employees on calling clients, etc.

MS seems to be moving from S4B to Teams.  Teams seems to be entirely geared toward pre-scheduled meetings where all attendees have agreed to join.  This is not our need AT ALL.  We need on-the-fly ability to add a voice call to an existing voice call AND be able to punch a dialpad for auto-attendants.

Therefore, we're looking for economical (5 users or less) solutions for our VOIP needs.  MS Office 365 E3 (which we'll keep) runs us $20/month, but in addition, for Skype PTSN dialing we also need $12/month/user for Domestic Calling Plan and $8/month/user for Phone System.  Therefore, our phone system needs are $20/month/user.

Can someone recommend some economical VOIP solutions? Thank you, yes we've looked - but that process is EXTREMELY unproductive (i.e. false/misleading claims on websites, feature listings are incomplete, etc., etc.)

Thank you!
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Im wanting to host simple PABX for multiple custmers and not sure which vendor to go for
we will host about 40 PABX's each with around 5 phones attached
requirements
support failover we will have instance in 2 separate DataCentres's for redundany/failover
needs to have single SIP trunk to host for all the calling voice channels
need to be low cost around 1$-2$ per extension and scalable as well
meed to be simple to provision and reliable
will be using yealink phones

any recommendation would be great
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One of the Shoretel Server Services showing RED. ShorewareCDRMigration-UPG
Does anyone knows about this service? It seems that server working fine. Thank you!
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I have some new VoIP phones and for some reason they will not configure on my clients network, when i took them home they work perfectly. I tried Wiresharking on a hub to capture the traffic, however i am at a loss as to what it means of what is causing the issue. The DNS is our Win2012R2 server and this then forwards on to the public Google servers.
wireshark-capture.png
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We have a shoretel system that is not under warranty support anymore that we have been asked to do some hopefully minimal support on as part of our day-to-day IT support for a client.

the client has 4 different locations with shoretel IP420/IP480 phones.
I am trying to move a phone from one site to another.....and I can not get the phone to show up in the list of available phones at the new site.
the phone has a static IP on the proper LAN - no VLAN.....i went over the phone menu settings with a working phone to make sure all the settings are correct on the phone.
the phone tries to download a configuration from the server - but says it failed downloading

what am I missing?  maybe licensing?
are there log files on the server side I can look at to see what failed?

as always - any help appreciated!
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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.