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,
As I am new to Cisco UC, I have the following design question which I would like to ask on what is required for it to work. There is an existing Cisco Call manager in HQ at Country A.

a. There is a requirement to setup a branch office in country B.
B. There is a secure vpn connection from the branch office in country B to country A
C. When call are initiated from the ip phones in country B locally, it should be called locally and not via country A call manager to avoid overseas call charges
D. When call are initiated from the ip phones in Country B for country A; it should be via country A local call tolls.
E. When call are between the ip phones, it will go via the secure vpn connection.

As I am new to Cisco UC, I would like to ask, is CME required for the router at the branch office or UC license would be sufficient? May I also ask on how this is done/design?

Any suggestions is greatly appreciated
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We recently installed a Polycom  RealPresence Trio 8800 in one of our conference rooms.  We are currently on Office 365 with Skype For Business.  We created a generic conference room 'user account' that we have logged into the device.  The generic account has an Office 365 E3 license assigned.  However, when using Skype, we cannot access the global directory to invite other users.  Even if I log in as myself I only get my Skype Contacts but cannot get out to the Global directory.  What are we doing wrong?
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Hello,

Are cloud phones better than an on premise solution?
We currently have a Siemens 4K phone system used with a PRI.
We have about 230 voice over IP phones and about 15 digital line phones.
Most of the ip phones have a basic setup with add on module to configure additional lines.

The 15 digital lines are for our sales department, the have a ACD routing setup(similar to a hunt group). Each sales person has their prime line and 2 secondary lines. We have it configured so that that each sales person is able to answered each others lines. Each sales person has about 50 lines configured on two phones at their desk that they can answer when any line rings. We it setup this way because our president wants sales to try and answer all calls and not have them go to voicemail.

We recently started to look at cloud phones and providers like ring central, 101voice and others. Our main concern is QOS and if our setup in sales will be supported.

We are not sure if cloud phones will be more reliable and if we will have the same quality of service?

Thank You


Let me know if you need any clarification.
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I've tried every reverse phone number. Lookup that's supposedly free and I get directed to a pay site were can I look up a phone number for free..??? Please help
0
VOIP Assuming that one T1 line can take 42 calls  and per call is 38k.
I have a site with 62 users using 2 T1 lines, how do you calculate if the lines are enough for the site?
0
I have a Panasonic tda50 system and tva50 voicemail.

1. If I’m on a call on line 1 and a second call comes in on line 2, the stations alerts me of the incoming call but doesn’t show the caller id, I constantly have to look on the display of a phone that’s not in use.

2. If the auto attended answers a call, the call does not get logged in my call log button for the icd group - how do I fix.

3. When I press hold the call is laced on hold but then I hear a dial tone - I have to either hang up or shut the speakers phone off - why doesn’t it just put the call on hold and go on hook?
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We are just starting to use Skype For Business for Internal IM.   We have setup Skype For Business to not allow External Contacts to be added. We didn't want Employees to start chatting with Friends and Family while at work.

We are looking to get rid off WebEx for Presentations and wanted to use Skype For Business, though it looks like we can't Invite Customers to the Meetings since we blocked External Access.

Is there a Way to Block External Users for IM, though Allow Inviting External Customers for Presentations?

Thanks!
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Look for outsource call center that they will have to use my five9 subscription to do in and out bounds calls.
https://www.five9.com/

Do u know any places that can provide those services?
1
Is there a Microsoft tool for testing Skype for Business video performance?  There exists for audio NetworkAssessmentTool.ext which allows you to make hundreds of simulated skype audio calls and gain insight into network preparedness for audio. Is there any similar tool that looks at Skype for Business online video?
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I have a project that requires 1000 lines to make outgoing calls.

Inphonex can provide the voips line but I don't know what kind or how many pbx (elastix)  I need or where to get them

Any ideas?

Thx
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Shoretel and SIP Trunk Requirement.  Hi I have shoretel system and Windstream provide PRI connection and want to change over to SIP trunk. What is the requirement for this? Some said that  need different device in order to install SIP trunk.  Thank you in advance!
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I am running FreePBX v13 and need to hangup all current calls from a set of extensions at a set time every day.  
Does anyone have a script or method that could perform this type of function??
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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
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SIP Provider Review, I have quote from Access Point Inc, I've never hear from the company before and bit skeptical where all their servers are. I don't to want SIP server siting on West end of country when our building is at East End.  Is anyone who experience with them? any feed back? or SIP provider who does good what they do?
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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
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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
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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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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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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.