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 have a remote extension (yealink handset) that is dropping outbound calls after 35 seconds. I have diagnosed through logging that the asterisk server is not receiving the proper ACK, and the reason for that is that the asterisk server is looking for the reply from 192.168.1.4, which is the local subnet of the remote phone. Obviously the asterisk server will never receive a reply from that address. What changes do I need to make so that the asterisk server is looking for replies from the WAN IP of the remote network? NAT is setup properly on both ends. For reference, this is CompletePBX from Xorcom that we're dealing with and a copy of the asterisk log error is below.

[2018-08-16 09:27:08] WARNING[5720] chan_sip.c: Retransmission timeout reached on transmission 405286438@192.168.1.4 for seqno 2 (Critical Response)
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Cloud Class® Course: Microsoft Office 2010
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Cloud Class® Course: Microsoft Office 2010

This course will introduce you to the interfaces and features of Microsoft Office 2010 Word, Excel, PowerPoint, Outlook, and Access. You will learn about the features that are shared between all products in the Office suite, as well as the new features that are product specific.

We are transitioning to Skype for Business from an older product called Spark.  With Spark it is fairly easy for each user to populate their own
groups with contacts that are saved and used to IM thru skype.  How can we populate groups without adding them one user at a time. We have some fairly large
groups
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I am converting to SIP at a company site.  I set up the fortigate 60d identical to another one of our company sites where SIP is already implemented.

The SIP Cloud provider tells me that the firewall is not "passing confirmation packets  the phone system (PBX) sends out when it detects an incoming call",
and this is causing their system to transfer the call to our failover number.

I have opened up all requested ports (TCP/UDP etc), configurd policy,and QOS to high priority- everything they suggested, and as I mentioned earlier, it is configured exactly like our other site's fw.

I suspect it is a configuration with one of their servers.  In any case, Logging for all events is enabled in the firewall policy.  How can I tell if the firewall is blocking/not passing back the confirmation packets they SIp provider mentions?
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We have been using Skype for Business (formerly Lync) for IM and video conferencing but are thinking about using Skype PSTN as a replacement for our current PBX.

We are in the process of organising a trial for some users to pilot Skype for Business as their only phone option.

One of the things we need to do is gather some data once their trial has finished and I am looking for help identifying test criteria for the new system.

If you have any thoughts about the sort of questions we should be asking users to assess the suitability of Skype for Business, then please let me know.

Thoughts so far:-

Equipment
Which microphone device do you normally use for S4B phone calls?
Which amplification device do you normally use for S4B phone calls?
Frequency of use
In a typical day, how likely are you to make a phonecall using S4B
In a typical day, how likely are you to receive a phonecall using S4B
Sound Quality
How does the sound quality compare to landline
How does it compare to mobile

Any questions / comments appreciated.


Jon
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Multiforest LDAP Not able to authenticate with remote trusts (Cisco VOIP).

We had to part ways with a vendor that was managing our VOIP system that services multiple school districts. We use a single domain controler for the voip domain that holds trusts to each member district. This allows self service portal and jabber authentication, as well as synchronization of user accounts. Everything LDAP related suddenly stopped working and we were unable to get the current multiforest instance of LDAP to connect to the school districts local domain controller. We were at risk of the voip system removing over 600 accounts, so as quick fix, we configured the Cisco Call Manager , so it connects to each districts LDAP server, directly. This workaround synchronized all user accounts, but the authentication configuration cannot work with our current configuration. We are now looking at the original configuration, in hopes of restoring authentication.

The previous support company had set up the LDAP Manager Distinguished Name as cn=user, cn= multiforest, cn=local, yet the server is not known by that name or domain and ad explorer errs out when attempting to use multiforest.local as the target. We can browse LDAP of the same server with AD Explorer, by IP and can see the trust domains under "users" Domains are appended with $. See screenshot. In short, I need help identifying the proper LDAP Manager DN and search base, so we can restore authentication via our trusts, rather than …
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how to allow IMO messenger through a firewall. viber & whatsapp can allow through firewall ports opening. but IMO seems to connect using port 443. if i have static IP range or port range then I can allow through firewall. I am using PFsense with squid proxy.
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How can I troubleshoot a Cisco IP Communicator phone stuck at "Configuring IP" during boot up? I've configured the device and added an extension and restarted the tftp server on the subscriber. But registration appears stuck. Where can I go to see logging as to what's failing?
CIPCPIC.png
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I'm trying to get an Allworx VoIP system to send email to a Exchange user's mailbox with an attachment of the recorded message (.wav file). I'm struggling getting it to work. During testing I keep getting...

Event 122016, MSExchangeTransport:
"There is no valid SMTP Transport Layer Security (TLS) certificate for the FQDN of MAIL-SVR."mydomain".local. The existing certificate for that FQDN has expired. The continued use of that FQDN will cause mail flow problems. A new certificate that contains the FQDN of remote.mydomain.com.au should be installed on this server as soon as possible. You can create a new certificate by using the New-ExchangeCertificate task."

In the Allworx device it shows the messages queued up but they never go anywhere. Before I was getting an authentication error, but I disabled all authentication in the Allworx Exchange email connector properties.

I have never worked with this before so I'm trying to figure it out as I go. The Exchange Server 2010 is on premises. So it's all internal.

I got a list of all of the Exchange certificates assigned to SMTP (PS: Get-ExchangeCertificate output in attached file). Question... why are there so many certificates?

One thing that bugs me is why is the event referencing the internal FQDN "MAIL-SVR.'mydomain'.local"? Is it because it's trying to communicate through that FQDN? The Allworx Exchange email connector is set to the external FQDN mail."mydomain".org.

Any help or direction …
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Hi, we have a data network here with Cisco switches that we manage. Now, there is a also a VOIP vendor who his own switches in our network. (He didnt want to use ours and VLAN everything).

Both the networks are on different subnets. Now, we noticed all of a sudden PC's started getting IP's from the phone subnet...and it's wreaking havoc internally. I tried to manually trace all 200+ cables in the office to see if someone plugged a phone device into the data network, but no luck..

How else can i troubleshoot this from say a switch level?
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Hi,

We am looking for a camera to buy for our conference/ meeting room which will be used with Skype for business.

The conference room includes a conference table which seats 4-6 people (2 on right of table and 2 on left) and a 55 inch display mounted on a TV stand beside the wall about 4 foot from the table.

We may also need a microphone on the meeting room table for the Skype calls.

1. What would you recommend as the best camera to buy for this room?

2. Is there any other products you would recommend to use with the camera you recommended?

Thanks,
Robbie
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Cloud Class® Course: SQL Server Core 2016
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Cloud Class® Course: SQL Server Core 2016

This course will introduce you to SQL Server Core 2016, as well as teach you about SSMS, data tools, installation, server configuration, using Management Studio, and writing and executing queries.

Hi,
 
 My customer has been using TextBox (a service of Line1 Communications, Inc. https://www.businesstextbox.com/) to send text messages directly from their computers in the past.
 Since they switched their phone system to VoIP system, TextBox failed to run. The tech support of TextBox software company told me that the VoIP phone carrier is the problem.
 TextBox software utilizes main business phone number to send/receive texts from recipients cell phones.
 Having said that, is there a software out there to replace this TextBox software?
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Hi,
As I am new to Cisco UC, I have the following design question which I would like to ask on what is required for it to work. There is an existing Cisco Call manager in HQ at Country A.

a. There is a requirement to setup a branch office in country B.
B. There is a secure vpn connection from the branch office in country B to country A
C. When call are initiated from the ip phones in country B locally, it should be called locally and not via country A call manager to avoid overseas call charges
D. When call are initiated from the ip phones in Country B for country A; it should be via country A local call tolls.
E. When call are between the ip phones, it will go via the secure vpn connection.

As I am new to Cisco UC, I would like to ask, is CME required for the router at the branch office or UC license would be sufficient? May I also ask on how this is done/design?

Any suggestions is greatly appreciated
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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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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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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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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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Cloud Class® Course: Microsoft Azure 2017
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Cloud Class® Course: Microsoft Azure 2017

Azure has a changed a lot since it was originally introduce by adding new services and features. Do you know everything you need to about Azure? This course will teach you about the Azure App Service, monitoring and application insights, DevOps, and Team Services.

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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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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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).