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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).

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Hey Experts, we have a Digium Switchvox VoIP Server. This past weekend our local power company had to upgrade our facilities power. We gracefully shut down everything Friday night, power was restored yesterday afternoon. This morning we have half of our phones not working as they cannot get an IP now. Our LAN and VoIP LAN are attached to our SonicWALL NSA2600, we have 3 Cisco SG500-28-p Stacked switches. What we have found so far is that any phone connected to the Master switch will not get an IP for the phone. Each desk has 1 Ethernet drop, that goes into the phone and the workstation plugs into the phone. The workstations all work fine to phones that don't work. We have rebooted the switches for good measure and nothing changes. Hoping someone can help shed some light on what the problem is.

Here is how the config on the sonicwall looks for the interfaces
Interfaces on SonicWALL NSA2600
Here is the Stack.
SG500-28P Stack
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What does it mean to be "Always On"?

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Hi all,

I have requested an additional IP address block from my ISP so that I can assign a public IP directly to my VOIP server. I have received and added a nat statement to my router as follows

ip nat inside source static 10.121.50.1 XXX.XXX.XXX.XXX (being one of the static ip's assigned by our ISP)

I can establish a SIP session with my server from outside however still get no audio either way. I ordered the additional IP so I could NAT everything from the external ip to the server to avoid this exact issue however it hasn't worked. To me it looks like no traffic is going back out the nat statement as the debug always shows 0 packets going out but plenty going in

*Jan 15 15:32:53.900: NAT*: s=183.171.81.177, d=58.XX.XX.X->10.121.50.1 [46336]
*Jan 15 15:32:53.960: NAT*: s=183.171.81.177, d=58.XX.XX.XX->10.121.50.1 [28621]
*Jan 15 15:32:54.208: NAT*: s=10.121.50.1->58.XX.XX.XX, d=183.171.81.177 [0]
*Jan 15 15:32:54.212: NAT*: s=10.121.50.1->58.XX.XX.XX, d=183.171.81.177 [0]

183.171.81.177 is my handphone on 4G  
58.XX.XX.XX public IP
Any help Appreciated
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I currently have 1 PRI configured on my voice gateway router.  We have had a few instances where we have had 20-21 simultaneous calls at a time, and as you know a single PRI only allows for 23 simultaneous calls.   I am looking to get another PRI from the same telco.  How does this work?  I have another T1(PRI) port on my router, which will be used to connect to the 2nd PRI, but how does it work on the Teclo side?  Do they  trunk the two PRI's together, so I can now have 46 simultaneous calls?  We are going to order another block of DID's with this new PRI as well.  So right now there are 39 numbers associated with the first T1, and I'm not sure yet how many we are going to get with the second block.    Does the telco tie these two PRI's together somehow, so both PRI's can share all the numbers?
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OK don't laugh. I have a 9 year old Cisco call manager which has run flawlessly for the last 9 years.

Recently it has developed a problem, since it has been end of life'd by Cisco they will not help me with this issue.

Here is the problem.

When a user goes to listen to their voice mail, everything works properly it will tell them they have "X" amount of messages, To listen to your messages press 1...

Once they press one again it works as normal... Saying a message from.... sent on....

Then right when it would normally play that message,

a message will play that says.

"This message contains no recording."  
Then it will go on with the normal  to save it press 2 to delete it press 3

No matter what option you select the next message played is.

"this system is temporarily  unable to complete your call, call gain later, good bye."

On the previous step If you press 2, to save the message. And go into saved messages it is there.

Since I have 90 mailboxes and get over 200 messages a day this is becoming a huge issue.

I'm hoping someone here may have enough knowledge or could at minimum refer me to someone who can help me band aid this until I can work on a replacement plan...

Thanks

Here are some version screen shots.

Show Hardware
https://www.screencast.com/t/WWcVu2wO 

Show System
https://www.screencast.com/t/rgFoIOWBb7
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Can anyone point me in the right direction on how to configure cell phones to  use with Call Manager 11.5 ?

Can this be configured for use with or without VPN ?

I believe express way needs to be configured ?
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using windows 10 and google voice (pick one browser)

how can I record phone calls
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I recently moved a small Call Center from one location to another. They contacted me late in the game to help them get a new call center setup with very little time. This left me with Comcast as the only ISP to choose from that could install services within their timetable. We decided to go with their new Gigabit package, which includes their new Gb modem. The client also chose to go with Jive for their PBX needs and subsequently ordered 20 Yealink T40P IP phones. Let the issues begin...

Comcast's modem would partially default to factory settings once a week. When I say partial, I mean things such as factory internal Gateway IP and Subnet revert back; passwords are set to default, yet the SSID's are still what we created. Comcast claims that they can ping the modem but when I test connectivity inside the modem to the internet I get 0/4 packets received. This test was run by the modem internal troubleshooting software. The Uptime clock inside the modem says 17,500+ days, which is not possible since we have reset the modem within the last 24 hours and we have only had the service for three weeks. We have factory reset the modem, replaced it with a new one, and we are still having the Yealink phones not obtain an IP address, or they have an IP address but say "No Service", even after they have previously worked. Phones continue to drop off of the network in that manner, and PC's that are using pass-through are losing connectivity as well. No matter how many factory resets on …
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We are still attempting to resolve random VOIP issues ( dropped calls, 1/2 rings then straight to voicemail, etc...) We have implemented a new Meraki MR320 switch and placed all desktops/phones (daisy chained) on this new switch.  We have this Meraki switch uplink to physical port on our firewall. There are no other switches connected to the Meraki. Physical port on firewall that Meraki is connected to is part of a bridge to another port on firewall which contains rest of our network switches. We have 2 vlans, voice and data setup on Meraki. I am seeing the messages on the Meraki log below over and over randomly during the day. Is this an issue? I have done some research on RSTP and port settings for the Meraki but still not sure I have the full grasp and if what I am seeing is actually an issue. FYI, we do not have devices being unplugged when seeing the issues. We have newer cabling to all ports.

STP Log
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QoS and roaming policies, that kind of thing.
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I am looking to install a small low cost WIRELESS VOIP phone system for my church.  Requirements are simple - VOIP phones that will operate via the Church's 802.11 wireless network.  We have 2 incoming analog telephone lines.  Will need 5-10 telephones with voicemail on each.  Would like a desktop style phone in most locations (vs. a small handheld).  Recommendations please.
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We have a user who is moving to another city and we want to set up his Mitel phone so it is like he is still in the office using his same extension as before. I believe we have to setup a vpn so we can do without using teleworker feature on the phone but I am unsure. Looking for advice on how to go about setting up this user to have his phone work remotely. The only VPN we currently use is Cisco AnyConnect but just want to know what steps it would take to accomplish. Thanks in advance for the help.
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I am trying to get Zulu UC Softphone to work with FreePBX  13.0.192.16

I have researched virtually every article I can find related to this issue, with no success.

I have the Zulu UC Softphone working on a client.  But cannot get the URL Popup to work.

I get this error on the client window when a call comes in and not sure if it's related or not.


Connection dropped by remote peer.

Any help would greatly be appreciated.


Patrick
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for a small business which conference system will you recommend?

what are the leading brands out there?
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I am using an SSG-140 Firewall at our rack in our data centre. Behind the firewall are several VMWare host servers which house a bunch of VP Servers, one of which hosts several 3CX VOIP PBXes for our customers.

I'm wanting to setup VPN tunnels for each customer so that we can do away with the SBC that 3CX demands we use if remote.

Can someone tell me how to set up a VPN connection to the firewall for our "test" customer (me!) so I can check it works? I'm using the WEB interface on the Juniper as I'm not too clued up with the CLI at this point on the unit.

The VPN tunnel should be ALWAYS UP, LAN to LAN and STATIC IP to STATIC IP.

Any help gratefully accepted.

Cheers
Chris
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i need to know area code of Uk, because need to call up to London.
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Hello, I am using asterisk 13 and I want all my extensions to ring at a time, where anyone can pick up.How can I do this
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Hi

Running FreePBX 13.0.192.19 and using Grandstream phones (latest firmware).  Intermittently (one or twice a week), our phones lose the ability to make calls, either inbound,outbound, or internal.  While the trunks remain registered, all of the endpoints report as being Unreachable.  this condition remains until I restart the FreePBX server.  When this happens there is no issues with the underlying network connections, and there is no firewall between the IP phones and the FreePBX server.  Would anyone be kind enough to point me in the right direction on this one?

[2017-12-15 12:13:00] VERBOSE[12762] res_pjsip/pjsip_configuration.c: Endpoint 233 is now Unreachable
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We have been working for months on correcting issues with our Vonage VOIP implementation. We have replaced switches, created vlans, etc...I am now viewing packet captures using Comm View. Phones are working most of the time but randomly throughout the day, calls will never reach phones...do 1/2 rings and then to voicemail, dropped calls...Using comm view i noticed that ongoing errors since we began capturing ....SIP 401 Unauthorized ...over 1000 since starting capture approx 2 hours ago. Any ideas????
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need to build network for 43 CCTV, 200 data, 200 IP telephone and 84 WIFI for hotel
kindly provide me the suitable part number for the active component as below
24      access switch 24 x GigE, 4 x 1G SFP POE      
2      core switch main and redundant fiber optic interface      
1      WIFI controller server      
84      access point      
1      Firewall
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Hello,

We have a branch office using Cisco 8851 phones running the latest up-to-date firmware. When we lose the link to our head office where the Call Managers (v10.5.1) are the phones go into SRST and work. I can see 58 licenses being used on the Voice  Gateway (Cisco 2911).

When the link returns the phones show that they are re-registed to the Call Manager, however, when I look at the phones on the Call Manager they are not registered. When I look at show call-manager-fallback on the router I still see 58 SRST licenses being used.

We have to walk round and reboot all the phones.

The VPN tunnels are connected via Fortinet 300D's and the policies allow ALL services. From what I have read the ACK's to the keepalives may not be getting through

Any help would be appreciated.

Thanks,

Glenn
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We have an old Asterisk (v.2.x) phone server in our office.  I'm new to the system and need to change an extension number from a rapid busy signal to a working extension.  Also, we have several extension that simple hang-up when dialed (no tones of any sort).  How do we edit those extensions?

I'm new to Linux, but I've figured out how to browse directories and edit conf files.
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We are in the need of a new phone system and there seems to be a mix of vendors pushing a hosted solution.  Has anyone upgraded in the past couple years to a hosted PBX solution and want to share the experience?  Of course the vendors not offering a hosted solution say to stay away from them they are not reliable.  I understand a lot of the bad from hosted is likely a so so Internet connection.  We have 100x100 fiber so I don't think that would pose an issue.  The big downfall I see is you pay for the hosted forever where an appliance based system, you buy it once and usually good for 10+ years.
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We have configured QOS on all HP Procurve LAN switches as CS3/24 & 46 priority 7 for VLAN 90 (Voice VLAN)

VLAN90 is configured on Cisco 3850 and interface on VLAN90 running to additional WAN connection for VOIP traffic.

How do I enable CS3/24 & 46 QOS on the Cisco core switch to maintain the QOS tags across to the WAN gateway (also configured for CS3/24 & 46)

Thanks.
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Hi, our Google voice forwards to main line, all of a sudden it stopped working.

I went through the settings and couldnt find anything that stands out. Any help here?
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I recently tried to upgrade from TZ215W to TZ600. All functions except one worked after the upgrade. The client has a VoIP phone system and we have 3 routes defined to the phone system router.
Traffic for 172.???.???.170, 172.???.???.171 and 10.???.???.0 is routed to 192.???.???.74. I have compared setting from the old router and new and don’t see a difference. The phone vendor says the routed traffic is being Natted. How do I turn this off on the TZ600? I didn’t have to do anything special on the TZ215.
The goal is to have all phone related traffic go out a different internet connection. see attached for network diag.
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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).