IP Telephony

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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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In Cisco UCM 10, how can I get a listing of all members of a specific device pool?
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I have several DT700 to put on my SV8100 switch. I setup all the settings correctly on the phone and can even use the web programming to get to it. What section do I go to for setting it up on the PBX with an extension and such? On the phone's display it shows Full Port.
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Does anyone know why the switch slows down so much that the VOIP starts to have Jitter and after a reboot it works fine for a week again. The switch runs the PC's through the Phones. There are no VLAN's programmed at the moment. The switch is stock standard. Are there any settings I can change?
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Cisco Unity Express - Version 8.6.7
Cisco Unified Communications Manager Express - 15.6(3)  / CME 11.5

Internal extension to extension calls will transfer to voicemail fine.

Calls from outside that go to voicemail get the message - "There is no mailbox associated with this extension, wait while I transfer your call."  Then it goes to a busy signal.

If the extension is forwarded to another extension internally, calls from both inside and outside will transfer to the other extension.

If the extension is forwarded to an outside number (we dial 9 to get out and enter it into the number to forward to) calls from outside will be transferred to the outside number.  Calls from inside get a busy signal.

Any help would be much appreciated.
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sing SIP RTP CoS mark 5
    -- Executing [09599259123@numberplan2:1] Dial("SIP/220-00000015", "DAHDI/G1/9599259123,60") in new stack
[Jun 13 17:19:19] WARNING[3910]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [09599259123@numberplan2:2] Hangup("SIP/220-00000015", "") in new stack
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I need for the replacement to look just like these?  Notice that they do not have the Plantronics branding on them.plantronics
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I need to create a procedure in my PBX (running Elastix) that hangup some calls depending on the caller ID and Dial a specific phone number to all other numbers.

At this moment I use Goto to send a call to a queue, but I want to use a direct Dial or a MiscDestination.

This is my current code: exten => 4821,n(message),Goto(ext-queues,5555,1)

So instead the "ext-queues,5555,1" I would like to direct dial a phone for example 8775555555

My dial plan requires me to dial 877 before the phone number.

For example in Misc Destinations, I can put 8775555555 and If I use it I will be calling phone 5555555

Thanks
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I have CUCM 10.5 and we use 9+1+(areacode) + number to dial out.  We just introduced Jabber and would like to leverage the dialing from Outlook as well as urls.  

The issue is that it is only the (AreaCode) + Number and it needs the 9+1 in the prefix.
I found something about using \+.! and the PreDot 900  https://supportforums.cisco.com/discussion/11950616/jabber-click-dial-outlook-prefix-9.

I am not familiar with Translation Patterns.  Can someone explain this a bit more.
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I have the above phone trying to VPN with a Dell SonicWall TZ400. When I put in the VPN information, listed below, the phone fails and gives me error codes that Phase 2 no response. I will list the three error codes I also see, if anyone can point me in the right direction.

SonicWALL

SonicWall VPN Settings:

Policy Type: Tunnel Interface
Authentication Method: IKE using Preshared Secret

IPsec Primary Gateway Name or Address: 0.0.0.0

IKE Authentication:

Local IKE ID: Domain Name
Peer IKE ID: Domain Name

IKE (Phase 1) Proposal:

Exchange: Aggressive Mod
DH Group: 2
Encryption: 3DES
Authentication: SHA1
Life Time: 28800

IPsec (Phase 2) Proposal:

Protocol: ESp
Encryption: 3DES
Authentication: SHA1
Enable Perfect Forward Secrecy: Checked
DH Group: 2
Life time: 28800

In advanced tab, the only thing checked is Keep Alive.

PHONE

Server: 50.XX.XX.209
IKE ID: VPNPhone
PSK: *****
IKE Parameters: DH2-3DES-SHA1
IPSEC Parameters: DH2-3DES-SHA1
VPN Start Mode: Boot

Password Type: N/A
Encapsulation: RFC
IKE Parameters: DH2-3DES-SHA1
IPSEC Parameters: DH2-3DES-SHA1

Copy TOS: No
File Srvr: Blank
QTest: Disable
Connectivity Check: Never

Errors

1/3
IKE Phase1 received notify
Error Code: 3997698:18
Module: NOTIFY:305

2/3
IKE Phase2 no response
Error code: 397700:0
Module: IKMPD:353

3/3
IKE Phase2 no response
Error code: 3997700:0
Module: IKECFG:1184
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Hi, really struggling with dialplans for Snom 300 IP phone at the moment and would appreciate some help.

I need to set a Snom 300 to only allow outbound calls which begin with a "7" but then to drop the lead digit...  Sounds weird I know, but it's the only way I can think of to restrict outbound calling to the speed-dial list which the users will not be able to view, but able to call.

So the plan is that 01234 567890 (for example) is in the speed dial list as 701234 567890, for the phone to then recognise this as an allowed number but drop the lead 7 so that it is able to be dialled.

I can't change anything in the PBX as we are using a hosted solution - already spoken to them and they can't / won't help, saying it needs to be done at handset level.

I hope that makes sense and that someone can help me out! Thanks in advance. :-)
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Hello,

We have an On-prem shoretel system configured running the director version 18.xx.  We also have three shoretel switches and use both softphones and deskphones.

Shoretel Switches:
SG-T1k  
SG-220T1A
SG-90

Softphone:
Ran through Communicator on PC's

HardPhones:
Shoretel IP230

Edge Firewall:
Fortigate 100D

T1 Provider:
Level3

New WAN provider:
CenturyLink Fiber

# of Users:
20-30

Currently, our phones use a dedicated T1 connection through level3.  This T1 line connects directly to the SG-T1K.  Due to increasingly high costs, we are considering getting rid of the dedicated T1 line with level3 and routing our shoretel phones through our Primary WAN(century link fiber).  The fiber is connected to our Fortigate 100D edge router.  I have worked with shoretel for several years but I have not had to made a change like this before.  

My question is, can we accomplish this with our existing equipment(phones, switches, etc)?  If we can, how do i implement the changes.  If we can not, what needs to change or be upgraded to facilitate this change?

Thank you everyone in advance for any help that you may be able to offer.
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I have a issue with Dialing out and some inbound.. I have a As5400XM with CT3 card on my TDM side all my b channels are up so are my d channels. My main issue is my dial peer setup when i dial out it will go over 2 dial peers ar the same time and i get no audio. My question is im have 28 T1s and i want them all to do inbound and outbound dialing what is a sample config for some like this. I know a dial peer out going needs a pots dial peer and a voip dial peer just a little confused
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Does anyone know if you can dump the call history log of a sip account to a text file?
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Hi,
I need to log voice calls and in particular call-forward calls made by user, by writing in an external syslog, date, time, internal-number, external-number and user originating that.
Which way could I implement this ?

max
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Hi All,
I have been at this all day to no avail.
I am using Yealink IP Phones. The customer now wants to run his laptops with the phones. So the PC's run through the phones.
The phones use their own gateway on port 1 and the PC's use their own on port 24.
In addition to VID 1 created VID 20 for the Data on all ports and Voice on VID 50 Voice as per this example I found.
Phones and PC's are on all the ports except 1 and 24.
AlI really want to do is give priority to the IP Phones.

[url="http://www.dlink.com/uk/en/support/faq/switches/layer-2-gigabit/dgs-series/es_dgs-1210_como-configurar-voice-vlan"]

The phones don't work and neither do the PC's when activated.
I have also setup the phones WAN port with VID 50 and the PC port with VID20.

Any help is welcome
I have not tried tagging P1 and P24 on all 3 the VLANS.  

Thanks
Ken
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We use twilio and sonic firewall in the company for our voip phone system.
And the issue is packet is drop more frequency and do not know what is the issue.

Most of the drops came from inbound calls.

Twilio has no issue.
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I have an issue that I need resolved.  We are running Cisco Unity v10.5 and have a problem when people call an extension and it goes unanswered.  If the caller hits "0" it automatically transfers the call to our District Office admin.  I found a setting under the User Templates that had this set up to transfer to the District Office admin and set it to Ignore Key under Caller Input Keys but the problem still persists.  Is there another area that I need to be looking into?
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Hi,

Following is the scnerio

Voice VLAN 1 ----->  Building one
Voice VLAN 2 -----> Building two

Both building have separate Avaya telephone systems.

Now there is a third building and in that we need to connect our Avaya IP phones either to building one avaya system or building two avaya system.

In order to achieve this in the third building we need to manually assign the access ports to either voice vlan 1 or voice vlan 2 depending on which building avaya system we need to connect to.

Is there a way i can assign both voice vlans to a single port in the third building and then simply connect the phone on any port i want?
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hi all, my company is going to switch to voice over ip phone system within 1 week. before it is switched, do you know there is any stress testing that we can test our network ensure we can handle all incoming phone calls and etc.?

We have around 50 to 70 voice over ip phones will be used.

Thanks
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We have CallManager (CUCM) 6.1, we want to forward ext. 111 from M-F 4:30 pm to 1:00 am to 555-555-5555.
Also forward ext .111 from M-F 1:00am to 7:45am to (777)-777-7777 and (666) 666-6666.
Can someone help us through it STEP BY STEP.
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I have a 5 user license of UC Desktop suite. How do I release licenses that appear to be hung from previous reboots. I've tried rebooting the phone system and the UC Desktop PC that runs the shared services.

Currently I can only use three before I get the out of license error.

SV8100
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We are still using Avaya IP Office 403 & 406 control units. Wondering if there is a version of Phone Manager & Softconsole that will work on Windows 10 and with the 403 & 406 control units?
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Sip phones do not register to CME when connection to UCM is lost. can any one assist me.

my configs below
cts logging verbose
!
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
!
voice service voip
 no ip address trusted authenticate
 allow-connections h323 to h323
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
  no h225 timeout keepalive
  call preserve
!
voice class codec 1
 codec preference 1 g729r8
 codec preference 2 g711ulaw
!
voice class h323 1
  h225 timeout setup 3
  call preserve
!
voice class custom-cptone exit
 dualtone conference
  cadence 400
!
voice class custom-cptone join
 dualtone conference
  cadence 200
!
!
!
!
voice translation-rule 1
 rule 1 /^.*/ /81\0/
!
!
voice translation-profile 81code
 translate called 1
!
!
!
license udi pid CISCO2911/K9 sn FGL204110KH
hw-module pvdm 0/0
!
!
!
!
redundancy
!
no cdp run
!
ip tcp synwait-time 10
ip tftp source-interface GigabitEthernet0/0
ip ssh time-out 60
ip ssh authentication-retries 2
ip ssh version 1
!
!
!
!
interface Loopback0
 ip address 172.25.12.5 255.255.255.252
 no ip redirects
 no ip unreachables
 no ip proxy-arp
!
interface Embedded-Service-Engine0/0
 no ip address
 shutdown
!
interface GigabitEthernet0/0
 description Suam-suame Link to Radio
 no ip address
 no ip redirects
 no ip unreachables
 no ip proxy-arp
 ip flow ingress
 ip virtual-reassembly in
 duplex auto
 speed auto
 no …
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Hi, I'm testing out Office 365 Skype for Business and PSTN calling with Polycom VVX 601 Phones.

I'm familiar with many Hosted PBX Providers such as RingCentral, 8x8, Vonage, etc. where you can just page and intercom to extensions and groups without much configuration on the Hosted PBX Side.

With Office 365 Skype for Business this seems like quite a hurdle. Per this link:

http://community.polycom.com/t5/VoIP/FAQ-How-can-I-use-PTT-Push-To-Talk-Paging-Page/td-p/49057

I have to manually go into each phone and enable/disable various channels so that anyone set to receive on a channel can get a page?

Is this the only way to do this?

How about Intercom, how's that setup?
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Hi, I have an old door opener system in my building, which lets me speaks when someone rings my department number, and open de door. I'm trying to find a way to control all this with asterisk, perhaps combined with arduino. Not sure if there is such a thing like a door opener with sip conexion, but I'm willing to try to do it myself (with someone else help, obviously :).

My door opener system is like http://www.netyer.com/t3_conexion_basica_con_fuente_generica_4_bornes.html

What I want to accomplish is: When someone is at the door, press my department number and rings in my departmen. Then, I should dial a sip extension, be able to talk, and open de door dialing a code.

Any idea?
0

IP Telephony

6K

Solutions

9

Articles & Videos

5K

Contributors

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.