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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Calling on all Cisco CUBE Experts;
CUBE setup for SIP trunking that that talks to the provider's SBC missing SIP port (5060) in the SIP URI, can anyone shine light on why it is happening? Is there a tweak or hack
someone can suggest ? The IP address is coming fine, BTW.

Thanks;
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Hello,

I've a 100mbps L2VPN link to connect two locations in an urban environment. The link utilization is less than 50% and the VOIP traffic is still dropping.

Both sides are connected to L3 switches with QOS policies.

Is there any tweak to overcome this situation.

Thanks,
Chanaka
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amazon connect - I use it as call center and use soft phone as well. it is fine. But now I want to add voip phone with amazon connect in the office.
Do you know how to set up? or where I can find some helps?
Thanks
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Hi guys,

We implement Skype for Business Cloud Connector on-premise  VMs on Hyper-V. (Building 1)
Also take a SIP Trunk from our ITSP in Building 2. There is some questions as below:
1-      Is it possible to forwarding SIP port from IP static in building 1 to IP static in Building2?
2-      How should I edit configuration.ini file? Specific in voice gateway section… put IP static that related to building 2 or what?
3-      Where should I use my SBC IP & IP address which I take with SIP Trunk modem
PS: Network structure and Configuration file is attached. Please help us in this regards ….
Thank you
Drawing1.jpg
CloudConnector.ini
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Hi,

We have a switch stack of 7 3750 switches. One switch just seemed to stop working, still has power. After restart, using the sh switch command, the switch seems to be stuck at initializing, after restart the of the stack, the switch shows ready. Its a POE switch and plugging a phone directly into the switch, no power. However plugging in a laptop works, data is working just not power no data. I used some basic commands, show config, ver, vlan, int and compared the configs to the other switches and everything looks good.  The switch in question has no error using sh int. Any suggestions greatly appreciated. Below is a output from sh int, for the switch in question,  all ports are shown the same.

FastEthernet5/0/20 is down, line protocol is down (notconnect)
  Hardware is Fast Ethernet, address is 0017.94b5.d016 (bia 0017.94b5.d016)
  MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec,
     reliability 255/255, txload 1/255, rxload 1/255
  Encapsulation ARPA, loopback not set
  Keepalive set (10 sec)
  Auto-duplex, Auto-speed, media type is 10/100BaseTX
  input flow-control is off, output flow-control is unsupported
  ARP type: ARPA, ARP Timeout 04:00:00
  Last input never, output never, output hang never
  Last clearing of "show interface" counters never
  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
  Queueing strategy: fifo
  Output queue: 0/40 (size/max)
  5 minute input rate 0 bits/sec, 0 packets/sec
  5 minute output rate 0 …
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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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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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configuring vlans on HP 5406zl
I'd appreciate some advice on configuring a data and voice vlan on a HP 5406zl, the current config is attached.
currently the whole switch is configured on the default vlan, however I want to add a voice vlan for a up coming voip phone system replacing the old analogue pabx.
the goal is to connect the pc's through the phones, phones on Vlan30 and Data on Vlan1.
I have added the vlan30 , however in need of some advice on the tagging and untagging of ports and the routing to enable the vlans to communicate with each other.
this switch also acts as the core switch and has IP routing enabled, it has 6 poe modules (ports A1- F24)
A1 to F22require both vlans , F23/F24 will be used to connect to switches on another floor and need to pass both vlans through. F17 is the link to the FW
appreciate some guidance on this as HP is not mother tongue, when switching.
current-HP-L3-core.txt
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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
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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
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We use five 9 for our main call center and we want to expand to another state but we just want to hire someone or company that take the inbound call and out bound call using five 9

Do u know any call call can do this?
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If I have a Skype for Business Online account  most folks reach me by a regular 10 digit telephone number.
But should people be able to reach me also at a sip address like sip://joebob@acme.com?
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I'm working at integrating Mitel 6867i phones into our network.  
At first, the provider said that there had to be firewall ports open (and, obviously NOT forwarded to any particular IP).  That is, the ports had to be opened incoming and outgoing.
Well, usually outgoing is open by default.  So that would only be mentioned to CYA.
And, if outgoing is all that's needed then saying incoming is needed seems double CYA.
Right?
My hope is that no firewall ports need be specifically opened at all.  

But, I started on this path by accepting what they specified and decided I would NOT run these things through our main firewall at all.
So, there will be a separate firewall as we have enough public IP addresses to accomodate.
And so, there has been a VLAN set up just to service the VOIP phones and to keep the firewall LANs separated.

I hope this much sounds like a reasonable approach.

If you're familiar with these phones (and I hope you are) then you'll know that the phones will accept a trunk line connection and "pass through" the main LAN / e.g. VLAN1 untagged to the computer on the desk.  (This saves running a new cable to the desk assuming the computer is already cabled).  And, the phone functions are supposed to run off a tagged VLAN / e.g. VLAN100 that's combined on the trunk.

All this seems fine so far.  I'm a little uncomfortable with the phones being so integrally connected inside the network but it seems like accepted practice.

As they say, "the devil is …
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Skype (Windows 10). The Video Button has disappeared from its position on right side top - alongside the telephone icon. Have tried for answers without success. What does this signify?
Has it anything to do with paying money.  It is not a problem with my computer, as I have opened the account on a second computer with same result.
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Dear All,

Can anyone have idea on mattermost webRTC integration. I am getting error while connecting video calls.

Thanks
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We have warnings on some of our Cisco small business network switches; network performance/quality seems to be fine.

[21 Cisco PoE and non-PoE switches SG-200, SG-250HP and SG-500 currently on the network].
VoIP system configured throughout our network that has been up and running for the past 6 months without any problems.
The warning messages are recent, and came up on our SG-500 core switches [Flash memory logs]
Warning is declared as "%CDP-W-VOICE_VLAN_MISMATCH: Voice VLAN mismatch detected on interface X" - where X is the up-link port number connecting to core/edge switch.
Ports affected are all (only) up-links, core to core or core to edge.

*         All switches have the latest firmware installed

*         All configured with the same VLAN IDs (Default VLAN is 1 & Voice VLAN 30, configured trunk ports, separated tagged from untagged)

*         Dynamic voice VLAN disabled on all switches

*         Auto voice VLAN disabled on all switches

*         CDP enabled on all switches

*         Global settings for QoS trust mode is CoS/802.1p
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I need a small dual-WAN router to interface with an ISP for VOIP.
It needs to provide VLAN100 tagged on the LAN side (some older routers like RV042 won't do this).
It needs to have some ports opened - not forwarded, opened. (some newer routers won't do this).
I'd rather have a 4-5 port router as ports are unimportant here.
I'd like to add a management port/VLAN and block internet access from that port's connection.

What might you recommend?
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We are having a POE issue on some ports on a Cisco SG500-52MP.  About 5 users phones are not powering up.  We eliminated a cabling issue and a phone issue by plugging in the phones directly to the Cisco switch, however some ports are not powering up the phones.  For instance.

Joe's phone is not working and was plugged into port 25
Mary's phone is working and is plugged into port 41

If we swap phones, Mary's phone does not power up on port 25 and Joe's phone powers up on port 41.  This tells us something is going on with the switch.  We don't want to power cycle the switch just yet as other workstations are working.  The system has been up and running for about a year and no changes were made.

Is there something that we can look for on the switch itself to see why those 5 particular ports are not powering up the phones?
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I always get confused / use firewall / router rules so infrequently that I don't know  what the right way to set these things up.

Can you help?

I have a VOIP service using a Grandstream HT-502 V1.2A gadget.  The call quality isn't always good.  That device is plugged directly into one of the ports on an actiontec router (left over from when we had verizon fios, but now we have cable internet 15down / 5 up speed.

I have lots of other devices plugged into a gig network switch in the basement that might be using the internet at the same time as the calls?  That gig network switch has 1 cable going over to the actiontec also (so there's only 2 cables on the lan side of the router).

To improve call quality, that's a job for QoS, right?

The attached picture is what I did in the actiontec router.  The grandstream has the ip of 192.168.1.52

But then i thought, should this be on the Ethernet/Coax or  Broadband Connection (Ethernet/Coax) sections? Did I at least get outbound rather than inbound correct?

But that just gets the call out of the house with highest priority.  Once it's on the web, it's fighting with all kinds of data / can't prioritize it, right?

Does the VOIP provider have any bearing on the quality of the calls? Iwe are using VOIPO.com).  is there  a way to substantiate / test where the poor qiuality - dropped fractions of a second in the conversation, etc.  NO stuttering, max headroom type things.  Just a m ssing sound here and there.

And most …
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I need to firmware update a CISCO telephone (SPA 504G) to version 7.6.1
The problem is that the firmware update is needed to established as standalone.
Can i firmware update a CISCO telephone SPA504G standalone?
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related question/answer
https://www.experts-exchange.com/questions/29095322/2-phones-ring.html?anchor=a42537973¬ificationFollowed=206569166#a42538269

Using verizon wireless for residential (not business)
is there a way to have both phones ring at the same time

*71 is one phone and then the other
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related question:
https://www.experts-exchange.com/questions/29095078/forward-phone-numbers-with-iphone.html#a42537163

Since you have Verizon, you can do this:

Calli *72 plus the forwarding phone number, including the area code (e.g., *72-555-555-5555). You will hear a confirmation message.

This works with all phones with Verizon service, not just iPhones.

Dial *73 to stop forwarding. You will get a tone or message.

Can I have both phones ring

I this answer only gives me second phone ringing.
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Need a voice support that is not TOO loud

I need to give a tour to about 10-20 people in a public area, but my voice does not carry very far. And if the wind is up, it's even harder.

But I do not want to use one of those megaphones. They are loud and use up one hand. I would prefer some sort of clip-on microphone, that uses BlueTooth to connect to portable speakers.

Can I WEAR a small speaker? Is that possible?

Or I guess I could pull a small speaker behind me, but that would also need a battery.
 
Suggestions?
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can I receive (not send) sms to skype united states phone number


not skype to skype message

receive actual text message from a cell phone
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I have an implementation of 3CX with soft phones . Recently i add also telephone (IP/VOIP) except of soft phone. the IP telephones are working properly but now  I need to export the contacts from the soft phone to IP telephone.

I am looking for a step by step instructions how to do it.
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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).