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 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 was trying to bring up a SIP over IPSec peering on a CIsco ISR4451-X. This is to complement and already existing
SIP router (CUBE I guess they call it). Anyhow I had a call not connecting and I ran debug ccsip call. One weird thing
I noticed is that the Source IP Media address 136.10.23.97 address on the external Gig 0/1 interface of the router.
The loopback is 136.10.23.98 and that's what I'm expecting to see for Source IP Address (Media). What determines
what interface/address gets used for Source IP Address (Media)? Thank you.

001812: Dec 17 2018 23:26:18.025 PST: //747/E74BC54D82EC/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 136.10.23.97        
Source IP Port    (Media): 17876
Destn  IP Address (Media): 216.25.35.21
Destn  IP Port    (Media): 55414
Orig Destn IP Address:Port (Media): [ - ]:0
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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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Hi Experts,

I am able to access the call manager in our organization, I have a phone device and I can see it under Device --> Phone but I want to know how an anolog phone with DID phone number  will connect to call manager using internal extension usually using the last 4 digits as internal ext,

If the product Type Tye says : Analog Phone , does that mean it is a analog phone.
0
If I have two SIP routes - model 2951 ISRs CUBE - and you want call manager to
failover if one of them can't complete a call - what is required? We currently have
a SIP trunk to one ISR (and the ISR has a TIP trunk to our call center). For redundancy
we want to add a second ISR/SIP Trunk. But the second should only be used in the
event that the SIP peering on the primary goes down. Advice appreciated.
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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!
0
How can I set my Skype default options to be that anyone can present/share files when Skype is opened? My current set up means that I have a Skype meeting and then when the person wants to share I have to change the settings and then re-commence the meeting? It's set so that only people from my organisation can share...thanks
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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 Mitel IP phone system (MiVoice Business release 8.0 SP3) and use 5320e phones.

Is there a way/process that users can have a call on hold, and then still use the phone to make a page?
0
It was ugly with Skype. Still haven't figured how how to add (or invite) external contacts to chat in Teams
0
Is Cisco UCM  Version 12, can I do the following:

  1. Set a user's voicemail to allow breaking out back to the main menu
  2. Monitor Hunt Group and Call Volumes in Real Time
  3. Monitor Agent Login/Logged Out StatE?
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Dear Experts,

I have two FreePBX virtual machines distributed over two different data centers but both are for the same company, I am planning to use sip trunk on those two VMs and seeking to get them to work in failover/load balance mode.

I know that some sip trunk provider provide the capabilities of failover in case my primary Public IP is not responsive however I would like to extend the failover and load balance to the level of the VM as well.

Have anyone tried to do the load balance/failover method on a VM level between two datacenters ? How would VoIP traffic react in case of primary VM down? how about end points configuration ? Can I direct end points to a single FQDN where both IPs can resolve and if one of the VMs are down still the end point would get register and act like nothing is happening ?

I would appreciate any person's comment that have had an experience with such scenario.

Thank you
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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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we are looking for 800 phone no with api. meaning when someone call 800 phone no. we want api to be called immedately to www.myweb.com/phone no
then redirect to 310.222.2222

do you know which provider can. do it? just like something out of the box without too much coding involved
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Looking for a VoIP phone system.
This client currently has a PBX black box supporting over 50+ phones. They are renting 5 lines from a local teleco.
They are building a very modern 100K square foot facility and will be moving to there in the near future. I started researching a new phone system to replace the current one.
I am not a telephone guy, so I would like to hear your opinions. First of all, I guess VoIP is the way to go, correct? I am kind of an old school and did not trust VoIP at the beginning. But after using Skype calling long distance for years, I find it has better voice quality than the landline most of the time.
I heard of 3CX, a cloud based phone system with no need to run an in-house PBX hardware or software. If the PBX is virtually on the cloud, how can we connect the 5 rented lines from the teleco to the virtual PBX?    
What solution would you recommend based on your real world experience? Reliability, voice quality, easy management.

Thanks!!
1
IP address shortage on Class C network.
The company is in manufacturing business. They have Windows servers, office PCs, production PCs, network switches, internal WiFi, IP phones, machines, etc. They all consume IP addresses. Now they wanna add 40 more production PCs while there are only 20 free IP addresses.
What should be done in order to release more IPs on this network?
One thing we are considering is to create a separate network for all 20 IP phones which are used in the "sub-site". (Please see the attached diagram). We are not good at VLAN, but we can learn. Will VLAN help in this situation?  
Are there any other things we can do?
Thanks!
Jack
Map-IP-Phone.png
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In Skype for Business 2015, are there any APIs that I can use to query if a user has Enterprise Voice enabled?

I need to be able to differentiate users that have calling capabilities vs users that do not. Maybe being able to query if a user has a phone number associated with them would suffice too.
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Hi,

We use Mitel 5212 IP Phones. we are trying to get them to work on a custom VLAN setup on a watchguard m500 firewall. We have created the custom vlan and the ip scope which works fine. I have mimicked the DHCP options from our windows based dhcp server, however this didn't work. On the DHCP windows based DHCP server the options are:

128 Mitel TFTP xxxx.xxxx.xxxx.xxxx
129 Mitel RTC xxxx.xxxx.xxxx.xxxx
130 Mitel IP Phone Identifier MITEL IP PHONE
132 VLAN for Mitel IP Phone 0x3
133 priority for Mitel IP Phone 0x6

On the firewall dhcp scop options 9All custom)
Code       Name                                 Type            Value
128         Mitel TFTP                           IP                  xxxxxx
129         Mitel RTC                            IP                    xxxxxx
130        Mitel IP Phone Identifier  Text              MITEL IP PHONE
132        VLAN for Mitel IP Phone   Hex              3
133        Priority for Mitel IP phone Hex             6

When the phone eventually boots it gets a crazy VLAN id. Any clues as to what I am issing, or a how to guide on getting the IP phones to work?

Cheers,
Paul
0
CUCM 10.5 SIP Trunking

    I have a two site Cisco Call Manager Phone System with one server at each site (FL and California).   I have had SIP trunking up and running in Florida for about a year.   We are in the process of migrating our PRI trunks in California to SIP trunks, but we are unable to complete the RTP (Voice) connections on the calls.   Every time we attempt a call the new sip trunk which is mapped through our CA Firewall, the Call Manager Server at that site advertises the RTP IP for the server in Florida.   Since these are not mapped through the other firewall, the call fails with no audio.   I cant seem to find a way to make the secondary call manager server advertise it's own IP address for the RTP instead of using the IP of the publisher.   The calls originate from the CA server, it is just the RTP that keeps requesting to send to the wrong server.   Any help on  how to force the subscriber to advertise it's own IP or how to change it would be greatly appreciated.   At wits end on this one.
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Dear Experts, I have a question related to telephony service. We are using IP PBX Grandstream UCM6510 with SIP trunking from The Provider.

So as my understanding, for example if our number is +AA 710xxxxx; I create a conference room in UCM6510 at ext 8888; then when customers want to join a conference room with us, they will call to +AA 710xxxxx, press 8888. Am I right? (AA is my country code)

But the Boss now have some customers in USA, UK,... and he wants his customers will call to USA, UK numbers, (for example: +1xxxxxxxxx; +44yyyyyyyy) respectively instead of our number (+AA 710xxxxx)  to join our conference room.

Is this feasible? Can you please suggest the solution? Many thanks!
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We have a Polycom VVX D601 IP phone AND we have a VVX D60 cordless phone that is supposed to register with the base phone.  It's not working.

We recently brought in some new fiber to the dry cleaners and with the fiber, our provider supplied IP phones (Polycom).  The dry cleaners need a cordless phone so the girls can walk around looking for clothes while talking to customers, so, we purchased a VVX D601 base and the VX D60 cordless phone that is supposed to work with the D601.

The cordless base will not pull an IP on the network.  We have plugged in the D60 to the router and it will not DHCP.  If we take the phone to our other office, an office with IP phones and a Xircom PBX, it pulls an IP.  If we take the D60 cordless to other networks that do not have phones, it will not pull an IP.  

Now, you're going to ask why we don't plug it into the switch with our other IP phones and the reason is our provider brought in a Juniper switch for the phones and they assign IPs on a static basis.  If we plug the D60 cordless into the Juniper switch provided by our phone provider, it of course will not pull and IP and our phone provider said they will not turn pan DHCP for us.  

So, my question is, since the base pulls an IP on network 1, which has IP phones and a PBX, but the cordless base will NOT pull an IP on any other network, is that because it's a phone and not a PC?

That's my guess.  The cordless D60 pulls an IP when plugged into a network with an IP phone PBX, but…
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Dear Wizards, is there any tool for simulating Grandstream UCM6510?

just like in Network we have GNS3, EVE-NG, Packet tracer,,...Can you please suggest? Many thanks!
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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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