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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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Hello - im looking for hold music that is free to use. Anyone have a link to download one?
0
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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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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://127.0.0.1:4577/call_log)
exten => _91868NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
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]
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the numbers that need transferring are:

44113246xxxx
44121236xxxx

And they both need transferring to:

0113 337 xxxx

How would I port these #'s in cube router  

example:

voice translation-rule 1
  rule 1 /4175209020/ /1000/

!        
!              
voice translation-profile INCOMING
  translate called 1
!        
!

dial-peer voice 200 voip
  description *** Incoming Dial-Peer ***
  translation-profile incoming INCOMING
  session protocol sipv2
  session target sip-server
  incoming called-number 4175209020
  dtmf-relay rtp-nte
  codec g711ulaw
  no vad





Thank you,
Riz
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I have a Cisco UCM cluster ver 10.5.2. We recently migrated all of our outbound calling to an enterprise SIP trunk from a major telecom.
The issue I have is that my inbound calling is on PRI's from a Different major telecom. 3 PRI's in an NFAS group. The calls only come in as 4 digit DNIS. (Previous to UCM the system was on a Nortel PBX) Inbound to my users works fine. However, when a user attempts to forward their line it will not go through.

Trace from RTMT shows 2 call legs. The first as inbound from external number to DNIS with the appropriate routing. The second, rerouting leg show that the call is being sent to the extension again.

"Start Time","Stop Time","Initial Speaker","From","To","Protocol","Duration","Packets","State","Comments"
"59.901472","64.552939","10.136.XX.XX [UMC SUB]",""Internal User x6900" <sip:6900@10.136.XX.XX","<sip:Forwardednumber@10.220.xx.xx"[internalsignalonSBC],"SIP","00:00:04","7","COMPLETED","INVITE 200"
"59.905243","64.606663","10.220.xx.xx"[ExternalSignalOnSBC],""InternalUser x6900" <sip:6900@10.220.xx.xx"[ExternalSignalOnSBC],"<sip:6900@10.138.0.11"[MajorCarrierSignalInterface],"SIP","00:00:04","10","COMPLETED","INVITE 200 200"


I believe that the issue is in the re-routing calling search space. But I am unable to get the correct syntax. Any help?
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hi

i have microsot access 2016 and skype 7.40, and i use this code to dial with skype:
Dim skpSkype As SKYPE4COMLib.Skype
Dim calCall As SKYPE4COMLib.Call

Set skpSkype = New SKYPE4COMLib.Skype
'Set calCall = skpSkype.PlaceCall("+17181234567")
Set calCall = skpSkype.PlaceCall(Me.Phone_number)

Open in new window


is there some way that i can with code program that after a few seconds it's will dial an extention number for example 1#.

thanks
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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.
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Hello Experts,
I am deploying a FreePBX system with Yealink T29G phones. The phones are provisioning no problem but i can't get the EXP20 to work. I have tried the settings on the T39(there are 6 options) but that doesn't work.

Any help would be greatly appreciated.

Many Thanks,
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The Samsung OfficeServ 7200 is behind our Spectrum EVPL (Ethernet Virtual Private Line). We have one Cisco Firewall at the main site. All of our other locations (except for three of them) are also connected to the Spectrum EVPL so no Firewall are needed at these sites. We are able to communicate with our other Samsung OfficeServ systems (100, 500, 7100, 7200) without any issues. But for the three sites that are not on the Spectrum EVPL network, we are using AT&T and Frontier at these sites and using the Cisco ASA 5505 to establish a Site-to-Site IPsec VPN. So when I try to make a call, the phone will ring but when I pick up we have no audio on either side. But here is the strange part. When I created the Site-to-Site IP Sec VPNs at all three locations, I set it up to where they could communicate not only to the main site but also to the remote sites. When we dial the extensions at the remote site, we have audio both ways. My phone vendor keeps telling him that it is a Firewall issue but if that was true, then why am I am able to call the other phone systems. They are going through the same Site-to-Site IPsec VPN tunnel. I think it has to do with the updated firmware that is installed on the Samsung OfficeServ 7200 at the main site. He updated it to a more recent firmware because we are using the Samsung CMS software for reporting. Please help.
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In MS Skype document "Media Quality and Network Connectivity Performance in Skype for Business Online" they say to that in order to estimate the RTT to Skype network edge you should ping Anycast VIP 13.107.8.2. When I do that it's very low at 30ms or so. But when I make actual calls in Skype for Business Online and sniff the wire - I see that real time communications happens with a distant server in the range of 52.112.0.0/24. This is close to 100ms away. What I'm trying to find out is exactly what is the relationship between this Anycast VIP and the actual server you end up peering with for your Skype calls. Any experts on here familiar enough with networking and Skype able to address this? Thank you.

Media Quality and Network Connectivity Performance in Skype for Business Online
https://support.office.com/en-us/article/Media-Quality-and-Network-Connectivity-Performance-in-Skype-for-Business-Online-5fe3e01b-34cf-44e0-b897-b0b2a83f0917
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Free Tool: SSL Checker
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One of a set of tools we are providing to everyone as a way of saying thank you for being a part of the community.

Have been on a Cisco/Linksys E3000 router for a few years.  My Mitel 5360 IP phone has worked flawlessly connecting to my office the entire time.

Replaced my router with a Netgear Nighthawk X4 R7500V2.  The IP phone gets an IP address (DHCP) but hangs at "Contacting Server."  Cannot connect to my office.

Reconnected the old router and the IP phone works fine so it's not the phone.  

Suspect I have to open up a port on the new router but have no idea where to start.

Any suggestions appreciated.
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For Skype for Business O365 MS offers a Call Quality Dashboard that shows quality trends. e.g. 1000 good calls, 20 unclassified, five poor calls. But I'm not seeing a way to search what were those five poor calls and when did they happen, what was the latency or jitter, etc etc. Am I missing something? How do I drill into call quality with this tool? Or are there other tools that would get this done?
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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!
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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.
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I'm setting up an application that integrates with asterisk. From my webapp, a user registers and connects to the asterisk. I have successfully setup the billing of user A calling user B with the a2billing. Now there is another feature of my app that requires B-Party(the callee) to be billed for receiving calls. I've been searching online for any clue but couldn't find any.
Please can someone with the a2billing knowledge help out with how to get this done?
0
Hi,

We have VOIP service through intermedia, they do not offer call recording. Is there some third party service i can use to record calls?
0
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 ?
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I have a small housing community which has 7 IP cameras which currently feed back to a NVR which records streaming in real time. Currently the Enginus wifi antanas. The management company of the property wants to have the NVR removed from the property and placed at their corp office which is about 20 miles away. I can get a good (fast) interent connection to the property and I am pretty sure I will need to setup a router at the current location to hand out IP addreses.

The question is, how can I get a connection from the housing community property all the way to the management property so that the NVR being placed there, will be able to see and record the IP cameras from the HOA property ?

I hope this makes sense.
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Hello, I work with asterisk servers and mainly soft SIP clients. Customers would like to have the possibility to hang up an incoming call. That's the first step...no big deal. But what if the call is a "call group" call i.e. a call to multiple devices at the same time and you want the command from the client not only to hang up his device while the others keep ringing, but to hang up all legs of the call - caller and all callees. Is that possible?
0
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There is a fellow it guy that keeps forwarding his line to my extension when he is "busy"coding or something else, he always forgets to disable it and I've had enough of it so I want to disable his ability to forward calls, or disable my number from accepting forwards from him only if possible or altogether from anyone, I don't want do disable the function globally just in his ext number forwarding or mine receiving from him/any. ( I have console admin access)
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I don't much about the Avaya IP-Office 500.  I've attached some pics that may help.  The first pic is the blue cable that connects to the phone but is no longer providing service.  
During our rack equipment move, we may have re plugged in one cable on the IP Office in the wrong port.  I'm not sure if that matters as I'm not proficient in PBX.  I had one of the staff members mark cables based on what phone goes out if we unplug it.  I'm not sure if that's done yet.  I left for another meeting.  
Anyway, are the patch cables placement very fickle for the IP-Office?  We may have accidentally plugged in one cable to the wrong port when moving things and that caused the phone to no longer light up.   Sorry, I'm walking blind here.  I would greatly appreciate some feedback.
blue_cable_from_face_plate_to_phone.jpg
a_and_b_drops.jpg
IP-Office_Ports.jpg
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For the Current PBX system, the Telco delivered an Adtran, off the Adtran there is an amphenol cable that is punched down to a 66 block.  There is also a CAT5 off the adtran that is also punched down to a 66 block.  

I am installing a Cisco VoIP solution,  but I am not sure how to connect my Cisco voice gateway router with a PRI/T1 card to the Adtran.  I'm not sure if the telco will have to hand the circuit off to me another way, or if I can just connect to the adtran using the ethernet port.  I'm having a lot of trouble getting info from telco.    Could anyone lend me some insight on this and their experiences?   I do have some VOIP experience, but no experience with traditional pbx, 66 blocks, etc.  Thanks.
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we create a sip trunk, cisco phones can call to avaya, when a try to call a cisco show "INCOMPATIBLE" im not a expert on avaya, ideas  ?

log avaya -> cisco

274138126mS PRN: ++++ END OF TCP MONITOR CLIENT DUMP ++++
 274160750mS Sip: SIP Line (17): License, Valid 1, Available 15, Consumed 0
 274160750mS Sip: SIP Line (17): sip_trunk_config_items 0002c10c, voip.flags 00040949
 274160750mS Sip: SIPDialog f172d9f4 created, dialogs 1
 274160756mS Sip: 0a3c1e8c0003346c 17.210028.0 33506 SIPTrunk Endpoint(f172f28c) received CMSetup
 274160757mS Sip: 0a3c1e8c0003346c 17.210028.0 33506 SIPTrunk Endpoint(f172d9f4) SetLocalRTPAddress to 10.60.30.140:46754
 274160759mS SIP Call Tx: 17
                    INVITE sip:50528337@10.120.200.20 SIP/2.0
                    Via: SIP/2.0/TCP 10.60.30.140:5060;rport;branch=z9hG4bKfdfadfc4705f6d30aeaf1ab49c1310a3
                    From: "Karina Bolado" <sip:SIPDefault@10.120.200.20>;tag=b25a03e400a8ebb0
                    To: <sip:50528337@10.120.200.20>
                    Call-ID: 554091e18ddc18667746c51e49f9a926
                    CSeq: 740975927 INVITE
                    Contact: "Karina Bolado" <sip:SIPDefault@10.60.30.140:5060;transport=tcp>
                    Max-Forwards: 70
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
                    Supported: timer,100rel
                    P-Early-Media: supported
                    Min-SE: 90
                    Session-Expires: …
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Hi

I've been wrapping my head around this for quite some time - still accomplishing nothing but headache. Here's the setup

Lync 2013 mediation server
Lync PSTN Trunk points to Avaya Aura RED
Avaya Aura RED routes PSTN traffic to Avaya phone central

We have landline numbers in Lync, and cell phone number with a PSTN provider

Lync <-> avaya RED gateway <-> Avaya phone system

Also - we've setup so that calls to cell phone in PSTN, will also simultaneous call Lync number.

so;
1. user call my cell phone (tel: 99 99 99) from his IP-phone/cell phone
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I answer the call with either cell phone or lync - the ringing will stop on the other device

so far so good.

But if this process is repeated, but this time the call is initiated from a Lync client at the internal office

1. user call my cell phone (tel: 99 99 99) from enterprise voice enabled lync client
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I …
0
Hi All

I'm in the process of planning out implementing site codes dial codes within my company's CUCM environment. We currently use 4 digit dialing across all sites, but this is no longer scalable with the amount of expansion we're experiencing.

I've set this up in a lab environment, and it works as expected. However, I'm stumbling with the modifying the Calling Party ID when dialing between sites. So when someone in Site A dials an extension in Site B, the Site A caller ID is prepended to include the Site A dial code.

I'm think I have to create multiple translation patterns, intersite partitions, and CSS's to do this; but this also seems messy, so I'm wondering if there's a better way to accomplish this?
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.