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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Hey Experts, we have a Digium Switchvox VoIP Server. This past weekend our local power company had to upgrade our facilities power. We gracefully shut down everything Friday night, power was restored yesterday afternoon. This morning we have half of our phones not working as they cannot get an IP now. Our LAN and VoIP LAN are attached to our SonicWALL NSA2600, we have 3 Cisco SG500-28-p Stacked switches. What we have found so far is that any phone connected to the Master switch will not get an IP for the phone. Each desk has 1 Ethernet drop, that goes into the phone and the workstation plugs into the phone. The workstations all work fine to phones that don't work. We have rebooted the switches for good measure and nothing changes. Hoping someone can help shed some light on what the problem is.

Here is how the config on the sonicwall looks for the interfaces
Interfaces on SonicWALL NSA2600
Here is the Stack.
SG500-28P Stack
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I have a bunch of Polycom VVX400/410's, and more and more are becoming Agent phones.

I'm trying to find out if their is a way to setup a softkey or something that will easily show the user their login and possibly away status. For example I've seen on some manufactures the color or shade of the button will change when logged in. or even when away.

I really don't care what it is just something obvious. The best I've been able to do so far it turn on the screen saver and after a minute they can see their status. That of course hides everything else such as DND status, forwarding status etc.

We're using a cloud based provider but I have some pull so if I can find an actual solution (because they aren't) to doing this I can probably get them to customize some config files to make it work.

Any help is appreciated.

Thanks,

Kenneth Morrissey
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I'm looking for a guide or handbook on how to program phone buttons on the Avaya IP Office 500 v2.

I'm looking specifically for being able to program buttons to change an incoming call route (for out of office purposes) to change from one hunt group to another.

What I would also be interested in doing would be to have the ability to set the in service/out of service destinations of a specific hunt group and put it in to that mode, which could be an alternative to the incoming call route option above if that is not an option.

Thanks
0
We have a user who is moving to another city and we want to set up his Mitel phone so it is like he is still in the office using his same extension as before. I believe we have to setup a vpn so we can do without using teleworker feature on the phone but I am unsure. Looking for advice on how to go about setting up this user to have his phone work remotely. The only VPN we currently use is Cisco AnyConnect but just want to know what steps it would take to accomplish. Thanks in advance for the help.
0
I have a Shortetel system with Build 19.42.6503.0 on Server
Shoretel 14.2 same build on the workstations There’s no switches at the offsite facilities, it’s all done through VPN The SG50s are in the same building. Every once in a while my SG50'sgo off line for no reason and have to be rebooted. Is there something that I can do to fix this?
0
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????
0
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.
0
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.
0
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.
0
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
0
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.
0
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…
0
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.
0
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
0
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?
0
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.
0
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.
0
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 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.
0
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
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
0
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: …
0
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

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.