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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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My Storage connect for recording avaya calls was disconnected i noticed today only... last 2 days recordings are not displayed in ip store console(replay)

my questions are

1.  i noticed that after connecting the storage again it is populating some datas in wav xml formats with todays date...maybe that is past 2 days recordings??

2.immediately i made a test call and checked the console replay but it doesnt shows the recording... is it take time to display the recording immediately after the call???

i came home that my shift was over... need to check tommorw between any avaya techs can answer this??
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Free Tool: IP Lookup
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Free Tool: IP Lookup

Get more info about an IP address or domain name, such as organization, abuse contacts and geolocation.

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.

I have an implementation of 3CX with soft phones . Recently i add also telephone (IP/VOIP) except of soft phone. the IP telephones are working properly but now  I need to export the contacts from the soft phone to IP telephone.

I am looking for a step by step instructions how to do it.
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Hello,

Currently running Cisco UCM v11.5 and recently got in some new phones: 8861 series to be exact.

I need a template for this phone and was wondering if these were downloadable from cisco or what?

Thanks,
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Polycom Soundstation 6000 not saving the Provisioning user information.  It saves the URL, but when you try to save the username and password, it "says" saved, but when you reboot or refresh the page, the credentials revert back tot the default one.  I also tried to enter it through the display (not the web UI) and same results.

I also tried to update the firmware, tried almost all of them and still same results.  Anyone have any other ideas?
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Convert TAPI modem dialer to a VOIP dialer for an Access 2010 application.  

Outbound only.  I am thinking I need to send a "ring" to both the caller and the called party, both phones ring, the caller should answer first.  Caller(s) have desktop 8x8 VOIP phone set(s).   And 8x8 accounts.

Please, what are some reference links for a design solution?  Or app so I do not have to reinvent this.

Kind Regards
Sam
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Please I need help on how to create custom trunk/outbound route on Asterisk to A2Billing without GUI. I'm using asterisk 11.25.0 WITHOUT the GUI(only CLI) and A2Billing version 2.2.0

Normally if I'm using freepbx, it's very simple and I would have just created the custom trunk via the GUI with this custom dial string:
Local/$OUTNUM$@a2billing/n

Open in new window

, but how do I do that without a GUI?
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A customer has recently had new phones installed that connect to a cloud based PBX.  They have a Draytek 2860 router.

They have sent me a document with the required entries in yellow (attached).

I presume they are asking that the firewall be opened up for those specific ports to the hostnames/ip addresses provided.

Can someone explain how I do this on th Draytek router
IP-Phones-system----FIREWALL-SETTIN.docx
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Cisco Call Manager Express with Cisco 7970 IP Phones.   All phones are displaying the message "3:Forwarded to 046" in the lower left hand corner of the display.  This happened all of a sudden.  This does not match the display message configured in the CME.  I have tried changing the message and a hard reset of both the phones, CME and IADD.Cisco-7970.JPG
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Our chairman calls his assistant and dictates emails over the phone for his newly hired assistant to type and send out. Sometimes she has difficulties to keep up with him and we really need a device, program or something for her to record his conversation and listen to again ?
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Hi Team,

We have CUCM deployed in our that is integrated with UCCE setup.
We accidentally disassociated all the devices that is defined in the Application user group that maps the devices in CUCM with created agent ID's on UCCE.
We currently associated the devices by manually checking all the available devices in the "Find' field under User-->Application user which is tedious.
We already have a backup scheduler that backs up the entire CUCM configuration that points to a TFTP server.
Kindly confirm the fastest method to restore the specific user configuration from the Application user configuration on available backup.
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Hi,

I`m using avaya ip office 500 v2 and using office manager 10.1

i copied 2 users and gave them their own extensions 3905 and 3906 and their passwords, (its on a vpn) on the other side when i login on the phones using the extension or password it says "extension in use) this extension was never created why is it giving me this error and how to resolve this?

Thanks
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Dear Team, we are planning to deploy the IP PBX (Asterisk) on an Ubuntu server. As my understanding, we can utilize the SIP trunk configurations and lease an VoIP service from ISP, then we do not need to purchase any Voice Hardware card for server, just need a dual Network interface card, am I right?

For example this diagram, the phone calls routes from IP PBX to ISP voice service via an MPLS connection

Internet ----- router ---------- switch ----------- IP PBX (Asterisk) ----------------mpls connection ---------------- ISP for voice service -------> go out for phone Calls

But can we do like this?

go out for phone calls <--------------------- Internet ----- router ---------- switch ----------- IP PBX (Asterisk)

Can the Internet line serve both voice and data? Do we need to inform the ISP for that change?

Many thanks for your help!
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Avaya IP406 PBX third caller goes to fast busy
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Hello -

I'm looking for a solution for a portable conference phone. Currently I'm using a Polycom Soundstation that works with an analog line. The quality of sounds is obviously is not great so I was wondering if there is a more modern solution that could help with our needs. It would be for a room with about 20 people around a long table so I need to make sure there is decent range for them to be heard over the phone. I was looking into something like a Jabra Speak 710 UC Wireless Bluetooth Speaker device that would work with a cell phone but I'm worried about quality of sound over a cell phone connection pushed to a BT device.

Any help would be appreciated.
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Hi guys,
Just got a general question. When I walk into a environment, how do I find the phones are whether digital or analog ? Any way to find by looking at into phones ?
If phones has Ethernet cable , they are ip Phones. Like wise how do I differentiate between analog/ digital ?
What is sip trunk ?
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snmp not working on shoretel 100da -t1 switch.
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Hello, I have a NEC model ip4ww-24txh-b-tel phone system in our office. We had a new line installed on friday by verizon, and I think something was reset on our phone.
When calls are made to our main line, it doesn't light up but if I pick up the phone the call is there. It can make outgoing calls, but incoming don't ring or light up.
Can anyone help me with this?

Thanks so much.
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IP OFFICE SYS 1 and IP Office SYS 2 configured for Resilience with SCN route on the WAN port connection of IP500 V2 units at Release 9.0We need to add SIP Trunks to take the place of the PRI Trunks. There would be 46 SIP Trunk Voice channels on each IP Office System to replace the PRI Circuits. That is we are adding SIP Trunks for other than SCN Use. These will be
SIP Trunks to replace the PRI circuits and used for connection to the outside world for calls into and out of the IP Office Systems. Does anyone know of a reason that this configuration will not work. Are there any known problems using the WAN port for both SCN Channels and ALSO SIP Voice channels to the outside world?
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Cisco call manager soft key templates do not appear to be applying to phones. I have created a new soft key template, have set up a test phone with the template selected both at Device and Device Profile levels and have reset/restarted the test phone. The result is no change to the soft key display on the test phone.

Also..... I have one (long standing) phone that has a different soft key button layout than others, but the soft key template config in both Device and Device Profiles is identical to all.

I have a feeling that there is something key I am missing, and perhaps overriding the soft key template config.

Can anyone point me in the right direction? I have phones to deploy that require a custom soft key config so I'd like to get this sorted out.
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All of my Office 365 users have a Office 365 Business Premium license.   We have Conference numbers and Conference IDs assigned by our Audio conferencing service. I see where it takes about using 3rd Party and Import/Export, though I don't actually see where you can enter in the 3rd Party Provider and Import the Numbers for the users.

If I select one of the existing Providers they all want a 6 Digit Conference ID, ours are all 4 Digit.

I would like to be able to use our existing Audio Conferencing system with Skype For Business.

Thanks,
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Please how can i know the verification PIN or extension pin for phone. I am using TDA200 and the phone KX-T7665.

The phone was blocked using *771 and now to open using the code  *770 require the extension pin. How can i know it?
Please help me if possible.
I have access to the PBX system TDA200 through the console
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I have a trouble with Dolby Voice from BT , the below error appears and couldn't  do the meeting :
can't conntect to meeting there are issues with the network connection to avoid missing your meeting, please join by phone BT Meetme Dolby
Dolbyproblem.PNG
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Hi

Currently we having issue to make outbound and inbound call.

We suspect firewall is blocking, how to resolve the issue in Cisco ip phones.
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We have Avaya system installed in our office but the remote phone never worked properly.
they keep on rebooting.
Any expert here who has knowledge of Avaya remote phone setup ?
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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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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.