Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

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Hi Experts,

What I'm trying to do are the following:

Available lines will be shown on the phone as unlit Line Keys.

When I start a call by using one of the following methods:
• Pick up the Handset, I want the first available line to be used, or
• Press an available Line Key then Speakerphone will be used by default, or
• Press the Speakerphone Button (the first available Line will be used).

My current Voip Provider have set me up with 1 phone number that can make 2 simultaneous calls. I would like the phone to indicate to the users what line/channel is currently in use. We currently have four phones used in the office and I was able to configure the BLF. I would like to know how to configure the actual sip line and not  the busy entension feature. I hope I'm making sense.

I don't really know where to even start. Do I need to do something to FreePBX server to allow the phones to know how many sip lines are available or is it just a phone setting I need to configure?

My current network environment:

FreePBX Server Spec:
1. FreePBX v 2.11
2. Asterisk 11.8
3. Debian Wheezy

Phone:
1. Type: Snom320
2. Firmware: 8.7

Please note that making and receiving calls are working correctly.


Please help point me in the right direction.

Many Thanks
Ricky
0
Dear Experts,

I want to use Cisco IP phones in my office, how I can implement and what stuff I need?
0
Hello. I have a Cisco UC540 and there are a couple things happening.

A clean factory config, with a 7945G and side car, two Cisco 9951.

All three phones are registered and can dial each other.

1) The 9951's cannot pull additional wallpapers from the server but the 7945 can.
2) The 9951's do not show up as options to add to the Page group, only the 7945.
3) The Two 9951 are SIP phones. The 7945 is SCCP.

I am looking for any help here. Been quite frustrating.

I would like to have the 9951 access the corporate logo from the UC540 like the 7945 does, and I would like to be able to add the 9951 to the page group (we are using extension 101 as a building page).
0
I have a 3CX softphone app running on my Android phone.  It registers and I am able to receive calls.  However, I am *NOT* able to make calls.  NO Intercom/Internal calls.  No external calls Local or LD.  When I attempt to dial, it just gives me the message "call failed, forbidden".

Thanks!
0
I have a 3CX softphone app running on my Windows 7 workstation.  When I launch the presence information window, I can see my default window & my Queue calls window in a tab view.  But the "Active calls" window that *was* at the top of the screen is missing.  It is not tabbed not floating, etc.  How do I get that window back?

Thanks!
0
I have an issue with my Avaya VoIP phones. I use voice and data vlan on all of my switches. However, when I checked my VoIP phones, it has an IP address from the data side. I talked to the vendor who setup the VoIP system, they say that the IP Office has setup for voice DHCP and something is not configured correctly with my data Microsoft DHCP server. I would expect a bit more info from a vendor who specializes in VoIP. But anyway, my questions are:
- What is the initial process when the Avaya phone boot up?
- Do I need to configure anything special on my data DHCP server to support the phones at the initial bootup?

Thank you in advance.
0
Hi EEs,

I have installed Cisco UC 560 with some phone model no CP-6941,

I simply can not see the missed calls on this...

I can see dialed & received calls but can not see Missed calls! when i go to settings button ---> Call history ---> All Calls .....Here I can not see all calls except Missed calls,

Can any expert share there experience or advice on this...

What all I have to do in UC or phone to see the missed calls on the phone itself,

Many thanks in advance...
0
windows7
skype view user profile
which can be changed by
me, the user, neither, both

1,2,31 2 3
0
Our phones are in a 10.x.x.x range, our data (computers) is in a 192.168.x.x. range.

Would this config allow any typical computer that needs vlan 5 to get through on ports 1-22?

My idea here is to have shortel phones which have computers plug into them go on ports 1-12, while computers only can plug into 1-22.

please ignore vlan1 and vlan10 for now, thanks.



Running configuration:

; J9773A Configuration Editor; Created on release #YA.15.12.0007
; Ver #04:01.ff.37.27:ea
hostname "2530-24G-PoEP-Rad"
snmp-server community "public" unrestricted
vlan 1
   name "mgt"
   no untagged 1-22
   untagged 23-28
   no ip address
   exit
vlan 5
   name "data"
   untagged 1-22
   tagged 23-24
   ip address 192.168.2.12 255.255.252.0
   exit
vlan 10
   name "public"
   tagged 23-24
   no ip address
   exit
vlan 150
   name "voice"
   tagged 1-12
   no ip address
   exit
primary-vlan 5
spanning-tree
0
I want to use Switchvox on-premise PBX
with a cheap 10+ line SIP / VOIP provider.

Who do you know that offers SIP lines under $20/mth/line ?
  ** http://www.voip.com/sip-trunking/sip-trunking-plans.aspx
  ** https://sipstation.schmoozecom.com
  ** etc ?
0
We have a Cisco Catalyst switch with the attached config.  We have two VLANs 2 (Voice), 3 (Data).  The phone system is on FE 2/2 which is handing out DHCP for VLAN 2. Is there any reason with this config that if we plug a phone in to almost any other switch port that it would not get DHCP from the phone system on FE 2/2?  Does the Voice command only work for Cisco Phones?  We are trying to use Mitel phones on this setup.  

Current configuration : 9450 bytes
!
! Last configuration change at 13:54:40 EST Mon Oct 7 2013
! NVRAM config last updated at 13:56:53 EST Mon Oct 7 2013
!
version 12.2
no service pad
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
service compress-config
!
hostname XXXX
!
boot-start-marker
boot system flash bootflash:cat4500-entservices-mz.122-54.SG.bin
boot-end-marker
!
enable password !!!!!!!
!
!
!
no aaa new-model
clock timezone EST -4
ip subnet-zero
no ip domain-lookup
ip name-server 208.67.222.222
ip vrf mgmtVrf
!
!
!
vtp domain Agency
vtp mode transparent
!
!
network-policy profile 1
 voice vlan 2
 voice-signaling vlan 2
!
power redundancy-mode redundant
!
!
!
spanning-tree mode pvst
spanning-tree extend system-id
!
vlan internal allocation policy ascending
!
vlan 2
 name Voice
!
vlan 3
 name Network
lldp run
!
!
!
interface FastEthernet1
 ip vrf forwarding mgmtVrf
 no ip address
 speed auto
 duplex auto
!
interface GigabitEthernet1/1
!
0
I was trying to do a swap of extensions - 2 users were changing their locations.  Logged in as admin2, used the swap wizard and now 1 extension took the other's extension number.  but the other phone / port doesn't work.  Plug the phone in and it says initialize (so it's live / getting power) abd the LCD is blank.  I tried other phones - the problem is with the port.  if you call that extension, you get a busy signal.

any idea how to troubleshoot this?
0
Our Cisco ATA186 is connecting to our network on our data VLAN, and not on our voice VLAN. what this means is that we cannot place calls using the ATA, since it gets no dialtone. How do we force it to connect to our UC560 via the voice VLAN? It used to work perfectly, then it got unplugged and is not working anymore.
Thanks!
0
I captured the traffic for an Avaya VoIP phone with Wireshark. the user phone (10.10.10.100) is calling another user on the remote site (10.10.30.100). I am not sure I understand the flow of UDP. I don't see the conversation between 10.10.10.100 and 10.10.20.100. I only see the conversation between 10.10.10.100 and 10.10.20.1 (the IP Office). 10.10.10.100 and 10.10.20.1 are on the same location.
0
We have set up an office with temporary numbers off of an other building since we won't have the permanent numbers for another couple of weeks.  we set up the temp numbers on dn 3 but the forward all button doesn't work.  message is unable to create a call, the maximum number of calls for this line has been reached.   The dn for line 3 is the primary extension and I've played with increasing the number of simultaneous calls on dn 3 but still receive the same error message.

any ideas or thoughts?  thanks so much
0
Hi

When I call to some mobile number it will display four digit number on receiver phone.
I want to display my number on receiver phone.
Is this possible . Please help .
Also is this possible to send and receive sms in elastix.
0
My boss is considering offering VOIP as a hosted solution to his clients. Just curious to see what would be needed to even begin considering something like this.

All I can think of is that we would need to have space in a Datacenter for sure.

any other ideas?

what kind of software is needed to actually provide users with this option or anything else that you guys can think of.

thanks
0
Hello,
How can I get the dhcp server for the data vlan disabled on the uc560, while allowing it to continue doing dhcp on the voice vlans? Thanks,
Jeff
0
Hi Experts,

I'm currently following Asterisk The Definitive Guide for my setup but can't seem to get a test call to *43.

This is what I have done so far:

1. On a Debian wheezy machine I entered:
    apt-get install asterisk

2. Moved sip.conf and extensions.conf to backup/

3. Edit sip.conf as follows:

[general]
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no

[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw
allow=alaw


[1001](office-phone)
secret=1234


[1002](office-phone)
secret=1234

Open in new window


4. Skip the bit about iax.conf

5. Edit entensions.conf as follows:

[LocalSets]
exten => 1001,1,Dial(SIP/1001)
exten => 1002,1,Dial(SIP/1002)

Open in new window


6. Start asterisk with:

asterisk -c

Open in new window


7. Setup Blink softphone with the following:

Display name: 1001
Sip address: 1001@192.168.0.222 (have setup static ip for this asterisk box)
Password: 1234

8. I am getting the following messages from CLI:

NOTICE[2930]: chan_sip:24850 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1001

Open in new window


9 After performing a fail test call to *43 I get the following from CLI:
WARNING[2930]: chan_sip.c:9187 process_sdp: We are requesting SRTP, but they responded without it!

Open in new window


Please help point me in the right direction.

Many Thanks
Ricky
0
I have a fresh, clean install of Skype 6.15.60.330 installed on my MacBook Pro running Mac OS X 10.9.2. It consistently unexpectedly quits immediately upon application launch. I have completely removed/uninstalled and reinstalled Skype several times. The Skype folder in Application Support has been deleted, as well as the Skype preferences. This seems to have no effect. I have run out of ideas. Typically, uninstalling software completely then reinstalling it from scratch resolves most issues. I am running the latest version of Mac OS X and the latest version of Skype.

I have attached the crash report / error log for your technical review.

Please let me know your thoughts ASAP. Thanks in advance for your timely response.
Skype-Crash-Log-2014-4-1.txt
0
I am using FreePBX 2.11 on Asterisk 11.7 with a 4-port analog card and DAHDI. We would like to be able to allow users to dial *72 and *73 to forward/unforward calls at the provider level instead of through the phone system, and only on line 3.

For example, someone picks up the phone and dials *72. This opens up an outside line to line 3 and dials *72 outbound through line 3. They get Comcast's call forwarding prompts and when finished they hang up. To remove the call forwarding on line 3, they dial *73 which dials out through the third line and uses Comcasts system to remove the call forwarding.

How can I do this?  Thanks
0
I'm testing moving our Voip traffic onto its own voice vlan. We have Univerge 8100 with about 25 DT700's. Each phone has an Eithernet adapter so we just have 1 cable for each phone and pc.

We have a PoE Adtran Netvanta 1500 switch. I have everything working, switch is routing between vlans, trunks are working, pinging between both vlans working, and calls our going out of the gateway and back in.

The issue is the calls coming in and going out connect but I don't get voice. Everything local is working fine. So I figure it's probably something to do with a firewall rule or something. I have a static route on the sonicwall back to the switch vlan gateway for the traffic coming back in the network. We have a Sonicwall NSA.

Note: We didn't have any issues before when everything was on the same vlan.
0
Hi Experts,

I have successfully installed FreePBX on Debian by following this tutorial here http://wiki.freepbx.org/display/HTGS/Installing+FreePBX+on+Debian+Wheezy but having great difficulty in configuring the server and know it's due to my lack of knowledge in the basics of telephony networking. I have a copy of Asterisk The Definitive Guide  and have read the first  5 chapters but the information seems to be over my head. Can you please help provide some guidance on some basic resources I should look at before diving back into the Asterisk guide?

My aim at this time is to just perform a simple test and understand the basics of how telephony networking works.

Many thanks,
Ricky
0
We have implemented an on-premises version of Cisco Jabber which we use for presence and chat etc.
In general it works fine, but for several users they are unable to connect to certain individuals, who are indicated as being offline. Others are able to see presence and connect to the same user. Also if the 'bad' user initiates a  chat their presence is indicated and they can be added as a contact.

As this is isolated to certain individuals connecting with certain others it really looks like some (but not all) are detecting an incorrect/invalid user account.

This does not appear to be a server/AD issue as different people have different results. It is almost like the client is detecting a contact on the individual PC and trying to use the first it finds (although we are pointing to our user OU in AD).

Has anyone witnessed this issue and know of a fix?
0
Need to find out when a workgroup was created in director.

Possible?

Thanks
0

Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).