IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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Hey Experts,

I have a FusionPBX Server Installed and everything is working fine at the moment except for the video calling, I've set the global codec to allow it but still nothing, has anyone been able to get this going?
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I have just started experiencing this strange issue with my shoreTel 12.1 Server.
some of the Users Access Licensing are changing randomly from WorkGroup to personal everyday. there is no log shows what happened or no one physically changed it. it started about a month ago.

any help will be highly appreciated.
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I am currently pulling monthly CDR records out of our Cisco Call Manager. However, to get any information out of those logs, we have an individual that spends hours to get some reports and metrics. Does anyone know of a good tool that we can get for free or maybe purchase, that would take the .txt CDR file and convert it into a report.
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We have a 3rd Party TAPI Server, running the NEC 3rd party tapi driver, on Windows server 2012r2

There is an NEC phone system, and a bespoke CRM software package involved.

Client PC's are registered with the third party tapi server and are running Windows 7 Pro

Client PC's should recieve a client information popup from TAPI in their bespoke crm software package, when they answer a phone call.

This works, but 90% there are very long delays before the popup is received.

Does anybody have an opinion on this please?
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We have this toll free line that is masked into 5 different lines for our call center. The center has 5 agents that use x-lite. We have set the queue (hunting ) for each agent 680* code as directed on d Yealink configuration, however,  the calls don't hunting from desk to desk.
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Hi all

We are using our own FreepBx hosted platform out of our datacenter.
I have been used to AASTRA phones and decided to try a new Polycom VVX 410.

The Web interface does not seem to offer the control Aastra does.

I cannot figure out how to setup SOFTKEYS like Transfer etc for the life of me.
In the web config it indicates you have to enable EFK to do anything with soft keys
I thought maybe this was done via SSH to the config file but I don't believe that is the case either?

Also for my BLFs.    The manual indicates I should have ICONS beside with GREEN RED or YELLOW to indicate status
These do not appear either?

Thanks
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Looking for input with anyone who has successes or failures in providing a hosted Cisco UC solution. What were pain points? What is the high-level diagram/design?

My high-level, current environment is as follows: 50+ sites, mostly smaller sites with less than 50 users however there are a few with 600, 400, 300 and 250, 250, 200 and 200). China, Southeast Asia, Europe, US, Mexico, Canada so far, though growth is always occurring.

About half the sites have MPLS right now, with the smaller offices mostly with a quilted solutions of varied ISP's and Voice services.

Again, I know that there are several details that are required such as mpls bandwidth, however I am looking for an overview and understanding of what your solution is, why you chose it over hosted solutions (cm clusters and srst, etc), and ball park pricing..just gathering some information and want to know more about what others are doing...

Thanks in advance.

B
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I have client who wants to disable call history on specific IP Phone.
I tried the command "exclude call-history" but still shows the call history.

We use call manger express
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I have recently setup a sip trunk with net2phone at home in Australia. Extensions are setup from elastics. For some reason I keep having to dial an international prefix even when i call local numbers in Melbourne. I am in Melbourne. For example if i call a melbourne number i still need to dial 00613 8878xxxx. Any ideas ? Has it got something to do with the fact that net2phone is an international sip provider ?

Thanks
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My company has 17 branches, a HQ, and a data center at a colocation.  Our UCCX and Call Manager are located at our data center.  We have a PRI located at each location.  The main number for each branch is translated to a designated trigger on the UCCX on our voice gateway.  When our branches lose data connectivity back to the Call Manager and UCCX, SRST does not function properly.  I know that the phones go into fallback mode and outbound calls can be made, but incoming calls get a fast busy.  I believe there is a problem with the configurations for SRST on the gateway.

Question: Is there a way for the gateway's voice translation to change to an internal extension located at the branch rather than to the UCCX trigger if data connectivity is lost at the branch office?
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Good evening, how i can block transfer call for outside of company, the allow for inside only. For more information, we have two ISP, where one is a normal ISP and other is for SIP Trunk, now our plan is to block all transfer call for outside of company, and permit to transfer call only for inside (LAN) of company.

In future, we want to create a rule that permits a list of numbers to transfer to outside of company and deny the rest. Please, help me to solve this problem because we are receiving a big count to pay to ISPs and people (our BOSS) doesnt understand what real happend, then use to complain everiday.
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I've installed an HP 2530 48 port switch, and I'm having problems getting Cisco 8851 phones with a key management side car / add on.  The KEM won't power up on the phones.  On the rare occasion, I got one phone to power up the KEM but I haven't been able to duplicate the process.  If I were to restart that phone then I'm not sure if the KEM would start or not start.  I cannot tell if it's a config issue on the switch or something with the phone config.  Unfortunately, I don't have access to the phone config.

With the default switch configuration, the phones startup and indicate that the accessory (the KEM) cannot be started. When I look at the phone log, I find a line item indicating that a request for power is made but it is denied.  This is with the POE+ option TLV enabled.

If I turn off POE+ and just use LLDP-MED, the accessory still doesn't power up but there is no message about the accessory not being started.  When I look at the phone log, I find a line item about a timeout occuring when the request for additional power is made.

I've uploaded the results of "show tech all" for the switch.

I don't manage the phones.  I'm trying to get the HP switch in place to avoid having my client purchase the Cisco switches the phone vendor wants to install.  There's almost a 3x price difference !!!

Thanks for your help.
putty-session.log
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I am replacing my 3845 router with PVDM2-64 with a 4331 with PVDM4-64.

In the 3845 I had configured 8 sessions of transcoding, 4 sessions of conferencing. And for MTP " maximum sessions software 128".

With the new 4331 system with PVDM4-64 - what are the maximum number of transcoding and conferencing sessions I could configure? I am thinking to keep the maximum number of participants to ad-hoc conferences to eight.

Thank you.



!

dspfarm profile 2 transcode  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 maximum sessions 8

 associate application SCCP

!

dspfarm profile 1 conference  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 codec g729r8

 codec g729br8

 maximum sessions 4

 associate application SCCP
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I''ve recently been given the responsibility of managing a FreePBX cluster deployed by an outside vendor.  I've been asked to add some extensions to a "hunt group" so when a particular extension in the IVR is pressed, the call will ring on any open extension in the "hunt group"?  The vendor already created several of these extensions which ring on a "hunt group".

In researching about FreePBX, I thought I was supposed to create a "ring group" but when I click on ring groups in the FreePBX UI, it doesn't look like there are any ring groups configured so I'm not sure how the vendor is currently accomplishing this functionality?  How can I accomplish this functionality?  There are queues configured.  Am I supposed to add the extensions to a queue?  If so, how do I do it?
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Attached is the sh tech from the router, also attached is the sh run and sh policy-map for the interface in question, basically we want to confirm the QoS policies we have currently configured, we had an incident yesterday where the traffic for the MPLS circuit had a brief burst which consumed the full allocated bandwidth which is 100mbps, when this occurred the voice RTP traffic was affected along with the call signaling for the Cisco phones. Just like to have a second pair of eyes for the QoS configuration on this router.
sh_policy-amp.txt
sh_tech.txt
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Hi,
I'm getting the below error message when booting my Cisco Call Manager:
*** An error occured during the file system check.
*** Dropping you to a shell; the system will reboot
*** when you leave the shell.
Give root password for maintenance
(or type Control-D to continue):

This link explain how to fixe it but it doesn't fixe the one I'm facing.

Can someone help me fixe this problem?? Give me the root password as I fail to log in??

Regards.
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Hello,
we have a Cisco Unified CM ver 8.6.2 managing our VoIP calls. As test I was asked to configure one (and only one) DN line to use the code g722. Checking around the CUCM configuration I see that the g722 is indeed enabled (see attachment) but how can I configure only a specific DN to use this code without effecting the rest of the lines in production? And is there any configuration change also for the Cisco IP Phone  to use this codec?

thank you
codec.jpg
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I'm getting ready to deploy Skype for Business and replace our existing Lync 2013 Enterprise systems.  We've had an enterprise deployment with 2 front end servers since we migrated to 2013 from 2010.  We've used DNS load balancing as we don't have any hardware/virtual appliance load balancing systems.  Our plan is to use S4B Standard and just create a second server running as a failover pool in case of an issue.  

We have 280 Lync enabled employees, 150 of them with Enterprise Voice.  The Skype for Business server will be going on a server with 64GB of RAM, dual Intel Xeon E5-2650L v3 processors (total 2 Sockets, 24 cores, 48 logical procs, at up to 2.09GHz).

Will I have any issues moving the CMS back to a Standard Edition server?  Any other issues I should expect?
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I am running CUCM 10.5. I need a program that will allow users to click to call from SharePoint.  We are currently not running any IM&P servers and dont plan on it anytime soon. I looked up Cisco click to call but it seems that is no longer available. Does anyone have any suggestions?
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Using RTMT I know that I can view the number of failed registration attempts. But how can I see the details of what mac addresses are attempting to register but failing for whatever reason? Thank you.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/8_5_1/rtmt/RTMT/rtconfcm.html#wp1296686
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I have the Speedport Entry 2i router and have set up both a port forward and a DMZ zone which points to an ubuntu server that has apache2 and asterisk running. Pings and special software confirms that the ports are open (for testing purposes all ports are open) and I can successfully from any point in the world see my "It works" apache2 page. The issue comes when I try to register sip client onto the Asterisk server. The same client and other client register without any issues from LAN, so should not be a config issue. I initially though that the router blocks something, but tcptrack detects connections when I try to register, so not sure where the issue is.
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So to preface, Myself and my underling are not the guys who set this system up, nor are we by trade telecom guys.

Out pbx's digital ports are full, so we're testing adding voip phones to free up digital ports.  So we had extension 336 using port 01a1005, we changed it to a 4612 which changed the port so S00001.  However checking free ports there are still 0 returned.  What are we doing wrong?
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We have a Cisco UCS560 Phone System that has been setup with a auto attendant to welcome all incoming callers a menu:

Option 1 - Sales and Ordering
Option 2 - All other callers

Today the phone system has decided not to route any calls when the option 2 has been selected by the caller. Need immediate advice. TIA!
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I have a Polycom VX300 that was from vonage and now I have a different provider.

I can't reset the phone I tried everything. is there a way to force a flash to the phone to bypass the lock on the phone?
0
I am getting complaints of poor voice quality from my users at a new site. What's new about the
site is that they are using 7821 Cisco phones with SIP instead of 7945 phones with SCCP. I am
stuck at the outset trying to troubleshoot as the product I use to measure call quality - Solar
Winds VOIP and Network Quality - sees no MOS score information. Could that be a problem
of using SIP instead of SCCP? Or is it because of the model of phone? Other possible issue?

Update: I don't think SIP is the problem for no MOS scores as there's a conference phone
there - an 8831 - running sip8831.10-3-1SR2-2 which does report MOS scores ok.
No-MOS-Example.png
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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.