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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I am using Freepbx 13 and want to block outbound calls on 911 no only.
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Hello All,

We had a MPLS connectivity between our HQ and branch,  we have recently migrated to IPSec VPN and i have enabled all ports between the HQ Soretel server and the branch Office. We can do a ping and access the Shoretel Branch swithc but Swith fails to Pair up as its sending RST packets back to HQ server.

Will SHoretel works fine in IPsec ? do i need to do anything to avoid this issue.

Shorele guys says there is a network issue Port 5452 seems Filtered intermittently. I have checked the config with Cisco Support, they verified and confirmed ports open and nothing found in packet capture.

can any one suggest over my case ? below are the Logs

tmsncc log shows hq cannot communicate to this switch

00:00:12.321 ( 5072: 6528) cco_cmd: clnt_call error: status= 3 (RPC_CANTSEND)
00:00:12.321 ( 5072: 6528) ncc_connect_setup: --> -20 (RPC_ERROR)
00:00:12.321 ( 5072: 6528) ncc_connect_to_switch: "10.104.6.7", 5452
00:00:12.327 ( 5072: 6528) 9, (19.47.5900.0) "10.104.6.7", "00-10-49-3D-2D-5B", 25(SG4-30) Flash, "en-US", (1.1.3.27),"3 d + 08:12:45",0
00:00:12.327 ( 5072: 6528) ncc_event_connect (2018/06/11 07:00:12.327, +7)
00:00:22.328 ( 5072: 6528) readtcp wfmo timeout
00:00:22.328 ( 5072: 6528) sw_cmd: clnt_call error: status= 5 (RPC_TIMEDOUT)
00:00:22.328 ( 5072: 6528) nec_event_connect_ex: --> -20 (RPC_ERROR)
00:00:26.314 ( 5072: 6528) ncc_connect_to_switch: "10.104.6.7", 5452

since the ip address changed to a different network, this …
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Hello,

I would like to know if it is possible to effectively use voip with private vlan edges, and how.

I have private vlan edges configured, essentially with the switchport protected, switchport block unicast and switchport block multicast, on all my user's workstation ports on the distribution switches. This is to prevent lateral movement in case of compromise. I would like to configure the ports for VoIP in the usual chained jack-to-phone-to-computer format. These catalyst switches are connected to the core catalyst switch via fiber.

I understand that all traffic on a switchport protected interface will be sent to the uplink and that this includes all voice and data traffic from that particular interface. But, I would prefer not to have to disable protected ports to allow phone to phone voice traffic.

Please help.
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We are installing a number ATA's either Cisco SPA's or Grandstream ATA's at a number of different locations. Each location is on a private network, however we do have static WAN IP's for the Firewalls. Each ATA has ethernet Available, but not Copper 2 pair.  Our goal is to pay a VOIP provider for about 5 VOIP Lines, that would be shared by 25 ATA's. Most of the traffic will be outbound dialing, and we would want the ATA's to rollover to the first available outbound line. We do ideally, want to be able to inbound dial to the ATA's using extensions.

Where would we get started in finding a VOIP provider that can help us with this, or would this be best done with on some sort of On Premise Gateway, which we would host at our NOC?

All comments and suggestions are appreciated.
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The current Network setup is customer site connected one SIP trunk each in US and Europe respectively
over MPLS Network. The customer is asking for cross region resiliency in SIP Trunks, is it possible? I'm not sure
if inter continent trunking will cause any issues? Please provide pros and cons.
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Hi
Were experiencing issues with the Shoretel VoIP breaking up.
This is occurring predominately at one of our remote sites, however the main site is being affect, if slightly less.
Internal calls from the remote site to head office along with external calls are frequently causing problems.

Each site has 100/100mb link,
Shoretel switch at both sites.
Director and E1k and Ingate at main site,

Diffserve 467 enabled on HP POE switches
dedicated vlan for voice in place across the sites
sites connected by site to site VPN

Ideas?

Thanks
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I had this question after viewing Microsoft access database make skype call.

Skype4Com API can only be used with the classic Skype version. The new Microsoft Skype for Desktop (the 8.X version) is not compatible with this API.

 

You should also be aware that several functions included in Skype4Com API will now not work even with the classic Skype. You will not find any new documentation of this API. The last available Help file is from 2011, so it's entirely up to you to check if applications based on Skype4Com API are still working or not.

Is there a solution that will allow 'doze 10 users to make VoIP calls from an access database?  I am developing a portable contact solution and I need it to be portable and work from remote locations.
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Hello,

Are cloud phones better than an on premise solution?
We currently have a Siemens 4K phone system used with a PRI.
We have about 230 voice over IP phones and about 15 digital line phones.
Most of the ip phones have a basic setup with add on module to configure additional lines.

The 15 digital lines are for our sales department, the have a ACD routing setup(similar to a hunt group). Each sales person has their prime line and 2 secondary lines. We have it configured so that that each sales person is able to answered each others lines. Each sales person has about 50 lines configured on two phones at their desk that they can answer when any line rings. We it setup this way because our president wants sales to try and answer all calls and not have them go to voicemail.

We recently started to look at cloud phones and providers like ring central, 101voice and others. Our main concern is QOS and if our setup in sales will be supported.

We are not sure if cloud phones will be more reliable and if we will have the same quality of service?

Thank You


Let me know if you need any clarification.
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NEC SL2100 over an Engenius EnStationAC wireless bridge is having issues with Echo on the VOIP phones.  The phones are PoE powered from a Netgear GS724TP.  All equipment is brand new (Phones and network).  The cabling is not new and has not been certified.  The VOIP phones over the internet and not over the wireless bridge work fine.  Only the 7 VOIP phones on the far end of bridge have echo problem.  It is intermittent, sometimes not at all for several minutes.  All equipment has the latest firmware.  All phones are on G711u codec.  A massive amount of monitoring and testing has been done so far, and their is no latency (1ms or less) or dropped packets (0%) ever.  The phones also can monitor packet loss (0%).  The bridge has RSSI of -34, and for testing purposes was lowered intentionally to -45 and -50.  Bridge is 100% stable with no obstructions between the buildings (Clear line of sight).  I have used multiple 5GHz channels, and their is only one other 5GHz broadcast anywhere near this building (That channel is never close in frequency and has been adjusted for testing).  The bridge is currently in 20MHz (have used 80MHz also).  There is no data devices of any kind on this network (Voice only).  The bridge is NOT running in Green mode.  Engenius has been working with me on this and doesn't have any more ideas to test.  I have tested a phone directly connected by 3' cable to PoE switch, and problem exists.  We are going to test directly in front of the bridge (with PoE …
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VOIP Assuming that one T1 line can take 42 calls  and per call is 38k.
I have a site with 62 users using 2 T1 lines, how do you calculate if the lines are enough for the site?
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Having issues with VoIP, we have Vonage and there's some buzzing going on when people make calls, the current environment we have is:

ISP to >>> ASA 5505 >>>> 3560E to the distribution switch for phones 2960X (the phone type is Yealink)


Any ideas?

Thanks so much for your help.
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I have a Panasonic tda50 system and tva50 voicemail.

1. If I’m on a call on line 1 and a second call comes in on line 2, the stations alerts me of the incoming call but doesn’t show the caller id, I constantly have to look on the display of a phone that’s not in use.

2. If the auto attended answers a call, the call does not get logged in my call log button for the icd group - how do I fix.

3. When I press hold the call is laced on hold but then I hear a dial tone - I have to either hang up or shut the speakers phone off - why doesn’t it just put the call on hold and go on hook?
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We are just starting to use Skype For Business for Internal IM.   We have setup Skype For Business to not allow External Contacts to be added. We didn't want Employees to start chatting with Friends and Family while at work.

We are looking to get rid off WebEx for Presentations and wanted to use Skype For Business, though it looks like we can't Invite Customers to the Meetings since we blocked External Access.

Is there a Way to Block External Users for IM, though Allow Inviting External Customers for Presentations?

Thanks!
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Look for outsource call center that they will have to use my five9 subscription to do in and out bounds calls.
https://www.five9.com/

Do u know any places that can provide those services?
1
Is there a Microsoft tool for testing Skype for Business video performance?  There exists for audio NetworkAssessmentTool.ext which allows you to make hundreds of simulated skype audio calls and gain insight into network preparedness for audio. Is there any similar tool that looks at Skype for Business online video?
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I have a project that requires 1000 lines to make outgoing calls.

Inphonex can provide the voips line but I don't know what kind or how many pbx (elastix)  I need or where to get them

Any ideas?

Thx
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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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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
0
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
0
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?
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).