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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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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Veeam® is happy to provide a FREE NFR server license to certified engineers, trainers, and bloggers.  It allows for the non‑production use of Veeam Agent for Microsoft Windows. This license is valid for five workstations and two servers.

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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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??
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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?
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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?
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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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I will have a chance of interview for the subject job position.

Can you share with me what I should look up and prepare before the interview?
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Just want to get folks opinions.  Anyone using them?  Any feedback?
Trying to implement sparkboards in every new office and eliminate things like conference phones, polycoms, and all that legacy stuff.

Thanks.
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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?
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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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Question -
company's IP phones/voice managed by 3rd party.  No PBX on-site.  
Is SIP Trunking still required for the calls or its just calling over the Internet?
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Hi,

 I am considering "converting cable operator provided phone service to VoIP phone service".
 Can you recommend a vendor and explain why you like them?

 I am aware that there are multiple players - RingCentral, Vonage, 8x8 ... etc.

Thank you for your input in advance.
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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 …
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Hello,
Need a question answered being asked of me and our Cisco CUCM 11.1 phone system. I need to know if Cisco CUCM 11.1 will integrate with NetSuite CRM via TAPI and/or CTI. If so, any third party apps needed or can it be done within CUCM?
Thanks in advance!
Steve
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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?
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I want to change time of stages of phones that ring on incoming calls.
Documentation on Incoming call system options. The main two I'm primarily interested in there behavior are 22-01-04 and 22-01-09.

04 - Normal DIL Incoming Call No Answer Time

09 - DID to Trunk to Trunk No Answer Time
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Hello Everyone

I'm currently in the process of migrating our current PBX system away from asterisk to Freeswitch. I am using FusionPBX on Debian 8. I am using the freeswitch webapi to originate calls. I am at the stage where when I execute the command, it rings the call centre agents phone and the customer automatically without the agent manually dialling the number. I would like the ability to manually specify a caller id number for the outbound leg of the call. At the moment it is not sending any caller ID. I can manually specify a caller ID number in the extensions page, and it works statically, however we have a need for the caller ID to be dynamic.

http://X.X.X.X:8080/webapi/originate?{click_to_call=true,origination_caller_id_name='Click to Call',origination_caller_id_number=1000,instant_ringback=true,ringback=\'%(400,200,400,450);%(400,2200,400,450)\',presence_id=630@X.X.X.X,call_direction=outbound,sip_auto_answer=true,domain_uuid=52b92yy9-7fb7-52ae-9e9e0595058bcdaa,domain_name=X.X.X.X}user/630@X.X.X.X &transfer('SOME EXTERNAL NUMBER XML X.X.X.X')

What do i need to add to this web address to get it to send a custom caller ID number to the customer outbound?

Many thanks in advance.
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I'm running CUCM 9 and Unity connection 9.  All screen's on my Cisco IP phones go dim (black) at 5pm.  I know this has to be a global setting in CUCM, as all phones do this, but I can't figure out where to go to change this.  We have recently extended the hours the office is open, so I need to change this.  Does anyone know where this setting is?  Thanks!
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SIPp is a free SIP traffic generating tool for Linux.
http://sipp.sourceforge.net/

SIPp user manual says you can install SIPp under CYGWIN on windows. However I am not experienced  with compiling applications to run under Linux and need help getting SIPp up and running under CYGWIN on a windows10 machine.

I have successfully installed CGYWIN and included the following packages (all successfully)
gcc-core
gcc-g++
gcc
libncurses
make

After the CGYWIN install, I put C:/cygwin64/bin in the win10 systems’ environment variable PATH – so far all ok and CGYWIN seems to be working fine.

In addition, the SIPp install instructions state:

Warning
SIPp compiles under CYGWIN on Windows, provided that you installed IPv6 extension for CYGWIN (http://win6.jp/Cygwin/), as well as libncurses and (optionally OpenSSL and WinPcap). SCTP is not currently supported.


QUESTION 1 -  Do you know what this is???    IPv6 extension for CYGWIN http://win6.jp/Cygwin/ 
is it a CYGWIN package, and entire install version??
What/how do I need to do to check/install?

QUESTION 2 – Nothing happens when I try to run “autoreconf -ivf” ...but this might have to do with Question 1 not being addressed yet.

 /cygdrive/c/Backup/tools/SIPp/3.3
$ autoreconf -ivf
-bash: autoreconf: command not found


+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
FROM DOC


Installing SIPp
•      On Linux, SIPp is provided in the form of source code. You will need to…
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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.