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

With this config from POE switch s5500 connected to core S5120 i got the phone working but i could connect or ping the S5500 ip address switch ... What's wrong !!

Is it the write config for a voice vlan setup like this (Core switch + poe) phone-pc connection !!

------------------------------------------------------------

Port connected from S5500  to core  S5120 port 10

interface GigabitEthernet1/0/48
 port link-mode bridge
 port link-type hybrid
 port hybrid vlan 210 tagged
 port hybrid vlan 1 untagged

--------------------------------------------------------------------

Phone port on S5500

interface GigabitEthernet1/0/19
 port link-mode bridge
 port link-type hybrid
 undo port hybrid vlan 1
 port hybrid vlan 210 tagged
 poe enable

---------------------------------------------------

Port on core switch S5120 connected to S5500 on  port 48

#                              
interface GigabitEthernet1/0/10
 port link-type hybrid          
 port hybrid vlan 210 tagged    
 port hybrid vlan 1 untagged    
 undo voice vlan mode auto      
 qos priority 6
0
NEW Veeam Agent for Microsoft Windows
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NEW Veeam Agent for Microsoft Windows

Backup and recover physical and cloud-based servers and workstations, as well as endpoint devices that belong to remote users. Avoid downtime and data loss quickly and easily for Windows-based physical or public cloud-based workloads!

Hi Experts,
I installed CME (ISR 4321 IOS version 15.5), with 40 Phones 7821, 1 IP Phone 8841, another 8851, and 2 IP Conference 8831 using SIP Protocol
all of these SIP Phones work fine, but the call transfer doesn't work, I don't know if some config missing, you find below the sh run,
thank you for your help.
CME-EHM#sh run
Building configuration...

Current configuration : 10874 bytes
!
! No configuration change since last restart
!
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no platform punt-keepalive disable-kernel-core
!
hostname CME-EHM
!
boot-start-marker
boot-end-marker
!
vrf definition Mgmt-intf
 !
 address-family ipv4
 exit-address-family
 !
 address-family ipv6
 exit-address-family
!
card type e1 0 1
!
no aaa new-model
!
subscriber templating
multilink bundle-name authenticated
!
isdn switch-type primary-net5
!
crypto pki trustpoint TP-self-signed-246793832
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-246793832
 revocation-check none
 rsakeypair TP-self-signed-246793832
!
crypto pki certificate chain TP-self-signed-246793832
 certificate self-signed 01
  ////////////////////////****////
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
  call start slow
 sip
  bind …
0
In RTMT you can view "Gateway Activity" and choose say MGCP PRI. But the thing that's weird is that it shows PRI channels per CUCM server. Given the description I would think that it would make out how many active calls are happening at the router/gateway not the UCM. Is there a way to get this broken out to you see activity per actual gateway?
0
If you are using MS Lync to make a phone call to the PSTN and there are multiple SIP trunks to the PSTN (actually these are trunks to Cisco CUCM and then on to PSTN from there) - how does Lync decide which SIP Trunk to use?
0
Hi

I got 2 switch, A5120 and S5500 serie switch from h3c, the a5120 is the lan switch and the s5500 is the poe ..
I would like to know the exact config I should use for both switch, I have already do config but not working .

I join the two config switch, the poe (port 48) is connected to port 10 on the a5120 ..

Thanks for the help
s5120.txt
s5500.txt
0
When a user dials in to a conference, they are prompted to enter the conference ID followed by pound.  When they do they system doesn't recognize they are entering the conference ID and the auto attendant states either that it did not get that or that the conference ID is invalid.
This is not with all phones but it is consistent on the ones that it does not work.
This is a three server frontend pool with a ACME Packet SBC and Mitel switches.
It was working perfectly then suddenly this started happening.
What might be causing this?
0
What are the advantages / disadvantages the Office 365 VOIP solution has over other VOIP solutions for small business?
0
Hello Everyone,

My name is George and I was hoping to get some assistance in configuring a Cisco 7914 expansion module.  I have been thrown into the fire with this Cisco phone system.  I have very experience with the CallManager system but I am making some headway.  The biggest problem is that my company is running very outdated versions of the software.  We are running CallManager Administration 4.1 (0.11).  I know, I know...it's no way to run a business but that decision is above my pay grade!  I am told to make it work, as I am sure many of you can understand and appreciate!

So here is what happened.  We had to do a complete network reboot and I was configuring a new phone when they took everything down.  When the network was brought back up, it took a few minutes and all but one phone came back up.  Of course, the one that did not come back up was the receptionist's.  After some troubleshooting I found that somehow, her phone was no longer showing in CallManager.  I went through the process of adding the phone again and defining it and configured.  Her phone works fine now.  The problem is with the 7914 expansion module.

The module is showing all red lights on the buttons and the display is blank.  I went to Add/Update speed dials, entered all the name and extension information and clicked on Update and Close.  I thought this was all I needed to do but I was wrong.  In the Configure Speed Dial Settings for SEPXXXXXXXXX window, it says 'Speed Dial Settings not …
0
Hi,

Ive been given the following to configure on the above network switch for our SIP/VOIP BT Cloud Voice service, can anyone assist on how I can get this done? I have no CISCO OS knowledge but have gained access into the switches gui so just need to implement the rules below.

The BT Cloud Voice Handsets are already live and on the network and they are have PC's also plugged into the pass through ports in most cases.

qos tcp-port 5060 dscp 011000
qos udp-port 5060 dscp 011000
qos type-of-service diff-services
qos dscp-map 101110 priority 5
qos dscp-map 100010 priority 4

qos device-priority 62.239.32.224/28 priority 5            
qos device-priority 62.239.32.240/28 priority 5              
qos device-priority 147.152.35.104/29 priority 5          
qos device-priority 147.152.35.96/29 priority 5                        
qos device-priority 62.7.201.128/27 priority 5
qos device-priority 62.7.201.160/27 priority 5
qos device-priority 213.120.60.128/25 priority 5

Thanks
SycamoreIT
0
hello expert i have problem with DT700 IP phone please give me a way about how to solve double assignment issue at ip phone
thanks for All
0
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I know that I can make free phone calls from Amazon Alexa Echo to another Elexa Echo, but what I want to research fromn my customers is the ability to uise a voice command to Alexa Echo to call a cell number or call a contact.

I want to research how this could be done.  

I'm thinking integration things like a combination of:
  • VPN's to some service,
  • integration with a Personal Contacts app such as Gmail,
  • Use of IOT
  • ITTT

Anything thoughts and directions to get me started would be very helpful.

Thank you,
Robbie.
0
Hi,

I would like to know how I can send messages using an Amazon Echo to a mobile phone as well as being able to call any phone.

In relation to this:

- The phone being called can be landline or mobile
- It would be a call from an Echo to phone and not Echo to Echo
- The Echo should be able to receive message and calls as well as send
- Would it be free to send messages and make calls in this way
- Also, would anything else be needed to make this work

Thanks,
Robbie

My ref: 1029743
0
I am new to Cisco 7940 phones. Have configured 5 phones which are working well in my office. But two more phones I am trying to configure. When I plug in the phone it shows error "Protocol Application Invalid". I did a factory reset. Still no result. Can anyone give me a solution, please?
0
How can you monitor DSP usage on a Cisco ISR used for MTP, transcoding, conferencing? Is there an OID which would let  us graph this?

mtp01-sc#sho sccp conn
sess_id    conn_id      stype mode     codec   sport rport ripaddr conn_id_tx

50630987   251788306    mtp   sendrecv g711u   24918 24576 172.28.72.145                                                                
50630987   251788304    mtp   sendrecv g711u   31384 18902 172.28.16.33                                                                  
50631252   50636167     mtp   sendrecv g711u   19210 17128 172.17.254.1                                                                  
50631252   50636166     mtp   sendrecv g711u   18332 50004 172.16.24.142    
Total number of active session(s) 11, and connection(s) 22
0
I am using a grandstream IP PBX UCM6202 and GoIP8 as a VOIP Gateway.
when i dial a number in this pattern +960 7788340 from PBX the call goes out to GoIP gsm gateway but from the GoIP outgoing call has a some unknown character due to space in front of digit 7. for this reason call is not dialed. if there is no space then there is no issue with the call going out. i am using an iphone numbers stored in contacts by defaults has some spaces. is there a way to remove the space when dialled out from GoIP gateway
below is a link to GoIP
http://www.dbltek.com/8-Channels-GSM-Gateway-pd6563436.html
UCM6202.png
Goip1.png
0
While getting ready to move a CUCM cluster I was reminded the route lists associate with a particular CM Group and register to a member of that group. But the question: Why is that necessary?
0
In Cisco UCM 10, how can I get a listing of all members of a specific device pool?
0
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.
0
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.
0
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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?
0
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.
0
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..
0
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
0
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
0
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

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).