Voice Over IP

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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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I have a bit of a challenge with a site-to-site VOIP situation.
We are using a Avaya system - PBX hosted on-site - in a 5 location business. They use a SIP trunk provider outbound and that's not a problem. But they also use the system as a sort of "intercom" to communicate between the sites. To make it work, we have setup VPN over the public internet in a "star" pattern, with one of the sites acting as the "hub" - the others as the "spokes". Traffic flows between the sites through the hub, or from the sites TO the hub, depending on who is being called.

Call quality is the problem. Choppy, dropouts or bad voice quality happen but NOT consistently. Just occasionally enough to be a pain. The business uses the "intercom" feature quite frequently, it's becoming a problem.

We use SonicWALL TZ300's at the spokes and a TZ400 at the "hub". We have QoS and Bandwidth management configured and that has helped. We have spoken to Sonic About it and they have put their 2 cents in.

Any suggestions would be appreciated.
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Is it possible to have a call attendant on Google voice , just even greeting or music?
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We have an old Castelle Faxpress 5000 box which currently supports about 40 incoming fax numbers and forwards them to Exchange server. I looked into porting all numbers to a cloud service, but this would cost in the range of $12 per user per month. I was wondering if there are any fax server products out there like Faxpress which would be less expensive.
The Faxpress is out of support and the company no longer exists.
We have Exchange 2013, about 40 fax numbers, onsite AVAYA VOIP phone system.
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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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when a sip call is up,  and the call is terminated by 1 party but the B party does not hang up.  how long will it be before he gets the fast busy tone..
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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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Hello,

we have two sites with skype for business deployed.

between the sites we have 2 subnets enabled via a tunnel.

we've discovered that although calls outbound via either site work calls between the sites don't connect.

looking at snooper logs we see that this makes sense by design as the primary site has a 'voip' subnet for the cx600 phones. when a call is made outwards between the sites and the user attempts to answer the call on the cx600 phones the call would fail.

when a call is made by either site outwards to the pstn calls flow as expected.

setting up the sites in the sfb admin portal i've setup the subnets which each site has.


based on my understanding media bypass should only work when the bypass site ID is the same. if it is not it should establish the call via the mediation gateway.

am i correct in this assumption?

i know i can add the non routable voip phone subnet to the tunnel however i'd rather not.
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i'm setting up COS class-of-service from the attached Shoretel document reference page 22-23 on the EX4600 and im getting the following errors.  I tried applied scheduler-map on these interfaces.
set class-of-service interfaces xe-0/0/0 scheduler-map ethernet-cos-map
set class-of-service interfaces ae0 unit 0 classifiers dscp ezqos-dscp-classifier

interfaces {
    xe-0/0/0 {
        ##
        ## Warning: statement ignored: unsupported platform (ex4600-40f)
        ##
        scheduler-map ethernet-cos-map;
        unit 0 {
            classifiers {
                dscp ezqos-dscp-classifier;
            }
        }
    }
    xe-1/0/0 {
        ##
        ## Warning: statement ignored: unsupported platform (ex4600-40f)
        ##
        scheduler-map ethernet-cos-map;
        unit 0 {
            classifiers {
                dscp ezqos-dscp-classifier;
            }
        }
    }
    ae0 {
        ##
        ## Warning: statement ignored: unsupported platform (ex4600-40f)
        ##
        scheduler-map ethernet-cos-map;
        unit 0 {
            classifiers {
                dscp ezqos-dscp-classifier;
            }
        }
    }
    ae1 {
        ##
        ## Warning: statement ignored: unsupported platform (ex4600-40f)
        ##
        scheduler-map ethernet-cos-map;
        unit 0 {
            classifiers {
ShoreTel-Premises-Services-Data-Netw.pdf
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Dear All,

A friend gave me a cisco cp-8851 ip phone configured in a call manager.I want to upgrade sip firmware .I tried to download cisco sip file but from tftp nothing happens.
Can anyone help me to upload the sip firmware and what files i want???


Best Regards.
0
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Can anyone help me with CUCM 11.5 iso ? I don't have service contract with Cisco so are can't download it.
0
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 are going through a brand new install of Cisco VOIP with a BE6000 system.  One of the features we need is the Single Inbox Reach function where a voicemail is left, and an impersonation account in AD/Exchange allows the voicemail to be transferred to the users email as an attached .wav file.  

From our Cisco Unity server the traffic is leaving the box on port 443 (verified via Wireshark).  But the traffic never appears to hit anything.  We don't get rejections or failure notices.  The Unity server just tries to resend when there is no response from any device.

I have taken a laptop from our VOIP network and talked directly through telnet to our Exchange server.  So I know the path is correct.
Windows Firewall is turned off on Exchange, and I know port 443 works because of all our OWA client access.  

We are at a loss here.  Any thoughts on something I might be overlooking.   Cisco TAC tested and said the Unity box is transmitting as expected on port 443.   Microsoft tech support verified our Impersonation roles are correct with EWS Editor.  

We have a Cisco ASA Firewall in the mix but when trying to filter and view traffic from our Unity server we see nothing, as expected since the switch that the  Unity server is connected to has a direct uplink to our switch with our Exchange Server on it.

This is a pretty vanilla installation for a small company.

Thanks
Matthew
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I am hoping to set up call forwarding on our Samsung officeserv

I have looked at all the call forwarding options, 60? but they do not work.  

I note that this method will not override any pre-set forward settings nor will it work for groups.

Does anyone have any idea on how to make sure these things are not happening so call forwarding works?

officeserv device manager v4.93

cheers
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I have twilio phone no. And just want to use at home using my home network is it possible?
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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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Networking from the perspective of new phone system and QoS – not my area of expertise.   Yet, I have a customer that needs help and part of my job is to ensure that these changes do not impact production.  Which – has already happened.

Prior to my involvement, a 3rd party vendor made a change that caused a Spanning Tree event.  The vendor is Cisco certified.  Familiar with the “phone system”.

The AVPN is a managed network, the phone system is “managed”, and the customer dropped dollars for the third-party to make the necessary changes to the LAN equipment as well.  All Cisco.  Mostly 3750’s.

The network prior was not managed, earlier generation MPLS.  Upgraded to AVPN.  Some circuits higher CIR rates (my recommendations).  

The “old phone system” – working fine.  New phone system…. Static.  Calls dropped.  Voice mail complaints.  (short list).

There is more but I don’t want to reveal too much in the way of who or where.  This is not the first outage caused.  The managed circuit has FIVE 9’s and was down for almost 4 minutes last week – one outage.  No ticket generated.  No alert generated.  Customer of third party had to report it.  (Isn't 5 9's total of 5 minutes and 39 seconds for entire year)?  

This will get fixed.  

In building A, there are 10 floors.  Which they should already know.

One recommendation is to enable QoS on VOIP on the “trunk lines”.  Okay…… But why now? Why was this not done before when LAN changes were made.

I do not “yet” …
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So my setup is as follows
Sonicwall tz215 at each location, they are setup in a 3 way site to site vpn with no filtering in between them.

Location A - Has Voice server
Location B
Location C

I have set up a phone at home to VPN into Location A to which they can establish internal calling to any device at location A however they cannot contact anything at Location B or Location C.

I think something goes through because if I am at home with the VPN settings in the phone for Location A and I bring up cameras I can see the phones screens backlight come up at Location B and C but it will not actually ring or show a call is coming in.


If anyone has an idea of where I should look next it would be greatly appreciated. I'm losing hair.
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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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My apology as to lack of technical knowledge - I am not a tech person but I am in charge of my company's tech spending and decisions. My company is located in a multi-tenant high rise building,  Approximately 45 hours ago our entire internet connection including VOIP went down.  Our internet connection provider CenturyLink first reported that there was a DDoS issue. Now, it appears that the problem has been traced to the DEMARC.  Specifically, our own IT consultant reports that "the DEMARC was touched by CenturyLink for xxx [other tenant] VLAN and in so doing corrupted [our company's] VLAN."

Our IT consultant suggests that there is nothing we can do to reduce risk of this happening in the future. I do not like the risk that my Internet including VOIP is subject to errors by other tenants or their service providers,

Suggestions appreciated.
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We are looking into Skype for Business as our VOIP system. SB is seems working OK on voice side but we are struggling to find solution for paging system for classroom.  We currently suing voip phone system as paging system and will be out once we remove our pbx system out of building.  Any suggestion? Thanks for all of you in advance.
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I have 33 switches in my environment and have lost connectivity between  9 of them. From the shell inquiry command I have found and error on all failed switches stating STATE_WAIT_REBOOTACK. Has anyone experienced this?
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I have been wanting to deploy a firm logo background for all my 7965  Cisco phones for years, but never had the time. Today I started looking into this and found many how to's.  One of the prerequisite is that the png and list.xml file need to be copied to Desktops/320x212x16 directory.  The issue is that the directory is no directory on my CUCM publisher and I do not know how to create this directory.

Would anyone be able to help me create this directory?
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Voice Over IP

8K

Solutions

10

Articles & Videos

7K

Contributors

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).