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,

We are currently having issues syncing the time/date and setting up the SFB login's on our Polycom trio 8800. Can anyone advise on how to resolve the issues.

Thanks.
0
good morning, i face a big problem with configuration (IP telephone Cisco 7962g) from tow days ago i think my problem in my file .cnf.xml after i register it i can't change phone name and when i change  it  became not register and give me log message can't update local please help me
0
Hello All,

I have integrated Kamailio 4.4 with asterisk 13 LTS and I think its been properly integrated. It also shows me the registered users but when i call from 101 to 102 it gives me the below error

[May  7 12:43:14] NOTICE[19838][C-00000000]: chan_sip.c:25545 handle_request_invite: Call from '101' (192.168.56.103:5060) to extension '102' rejected because extension not found in context 'public'.

I have followed the below for installation and configuration.

http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

The user database is fetching from remote host in which kamailio has been installed. Users are showing in asterisk node as well

asterisk*CLI> sip show users
Username                   Secret           Accountcode      Def.Context      ACL  Forcerport
101                                                          public           No   No
102                                                          public           No   No

So how can i debug this or is there any clue that what might be wrong. Please find below  the extension.conf details as well.

exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,101,Voicemail(${EXTEN},b)
exten => _1XX,102,Hangup

Thanks and looking forward for some clues from this community

Regards,
Atif Ramzan
0
Cisco IP Phone 7962G SCCP to SIP Problem, asking for XMLDefault.cnf.xml and xmlDefault.CNF.XML and SEP<mac_addr>.cnf.xml and CTL<mac-address>.tlv
0
Cisco IP Phone 7962G SCCP to SIP Problem, asking for XMLDefault.cnf.xml and xmlDefault.CNF.XML and SEP<mac_addr>.cnf.xml and CTL<mac-address>.tlv
0
how to configure polycom soundstation ip 7000 in Avaya IP office 500? I have purchased avaya 3rd party licence for Polycom and please share to me steps of configuration
0
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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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?
0
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
0
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?
0
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 ?
0
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.
0
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
0
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!
0
Avaya IP406 PBX third caller goes to fast busy
0
snmp not working on shoretel 100da -t1 switch.
0
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.
0
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?
0
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,
0
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
0
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.
0
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 ?
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.