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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Need 2-line analog phones (not VOIP).  Can't seem to find anything reasonably priced for my client.  Their needs are pretty simple:

1.  Telephones (quantity 4) for two-line analog.
2. Must be able to indicate when a call was missed or message was left, by way of an LED on the phone.
3. Must be able to page the other phones (hands free) by pressing a page button.
4. (optional) Intercom capability.

They currently have 2 of the Cortelco 7-series 2740 phones. These have been replaced by the 2750, and they're over $200 each and take 3 weeks to ship.

Anyone have a better idea for 2-line phones?

Thanks.
Dave
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I have one user that is not able to search the GAL for contacts.

He is searching within the “My Contacts” tab but his searches show no results.

He is using SFB on Windows 10. The rest of our users have no issues doing the same.

What am I missing?
0
We have a new issue on our phone system.  If we receive a call and try to transfer it to another internal number, but that person cannot take the call...if we go back to the original caller and try to transfer the call a 2nd time, transfer is not available.  This all of a sudden started happening.  We are on CM version 11.0.1.22900-14.  Thank you.
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We have an HP 2920-48 port POE switch. We also have 2x HP 2920-48 port NON-POE switches. I was surprised to find today that our new VOIP phones are actually get power when plugged into the NON-POE switch?! Is this possible?? If so, are there possible issues I should consider when configured in this manner. I know we are having issues with the phones losing calls before they can be answered...wondering if related.
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When I connect an Avaya 9600 phone to a H3C 802.1x port, the phone just works, it doesn't need or attempt any sort of authentication. I do not have to have a pc connected in line with the phone.  I like this behavior, but I don't understand it.  I have the voice vlan oui's set statically, so it's operating in secure mode.  Could someone explain why 802.1x on the port bypasses for the phone?
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What is the process for adding a ten digit (area code + seven digit phone number) phone number to an Office 365 Skype for Business (Lync) account?
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Hello all, I am in the process of rolling out a VOIP solution for an office of 15 to 20 lines.  The customer is debating between hosted solution like Zultys Hosted with a 3 year contract for $305 recurring fee and zultys 36G phones for $194 per phone, or go with an appliances  and manage it myself a switchvox E510 for $695. and phones are D60 for $139 each and software registration code for $1000, and extension licence and subscription fee 0f $80 a line per year. I am really leaning towards the appliance but just not sure of the switchvox.  I really need something inexpensive but dependable any suggestions??
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Hello We are using Vicidial version 2.6-381a with Asterisk 1.4.44-VICI, I am trying to set up my dialplan entry so we can call out from Canada to Irland and Trinidad, our current Dialplan allow us to call only witin north america but not outside,
Here I have the country and area codes that need to be called out from Canada
Irland 353-287-xxx-xxxx
Trinidad 868-788-xxxx

Here I have my Dailplan that i have made up to call out but no luck can someone please guide me correct dialplan please

exten => _9353NXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91868NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9353NXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _91868NXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _9353NXXXXXXXXX,3,Hangup
exten => _91868NXXXXXX,3,Hangup


Here is the call_log but no luck (last 4 digits of phone number being modified to xxxx)

[Nov 1 13:48:07] == Using SIP RTP CoS mark 5
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:1] AGI("SIP/298-00017de5", "agi-NVA_recording.agi,BOTH------Y---Y---Y") in new stack
[Nov 1 13:48:07] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Nov 1 13:48:07] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20171101134807_298_1868788xxxx)
[Nov 1 13:48:07] -- <SIP/298-00017de5>AGI Script agi-NVA_recording.agi completed, returning 0
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:2]
0
Do the Comcast Panasonic KX-TPA65 VOIP phones also have a network jack for plugging in a PC (in addition to plugging the phone into the network)?
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Anyone know why my log buffer in my functioning voice gateway 2911 is full of this message and how I can stop it? debug is not turned on....:

SIP: Trying to parse unsupported attribute at media level

I see a bug report at https://quickview.cloudapps.cisco.com/quickview/bug/CSCuv36964 but they have no advice for how to stop it from constantly filling up my log buffer .... does anyone know how to make this irritating message stop? Is there any way to determine exactly what the "unsupported attribute" message might be that triggers this or where it is coming from? If someone has an old test sending out something repeatedly, am I going to have to sniff packets all day to find it?
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What are the best types of 5 port POE switches that can have Comcast Power over Ethernet (POE) phones plugged into them?

An office where I work with have two Comcast POE phones but has only one Ethernet port. We need to connect a switch to it so we can plug in both of the Comcast POE phones as well as two laptops.

What are the best types of switches for this?
1
Hi,

We have a Polycom SoundPoint IP 350 and a 450 at a remote office that connect to a FreePBX 2.11/Asterisk 11.7 box in our main office. The two sites talk using a site-to-site VPN via our FortiGate 30E firewalls. When the two remote phones register across the VPN, everything works great and there are no problems. However, if the internet goes out at the remote site, which it does often, the remote phones will never try re-registering again. Even restarting the phones by power cycling them does not let them re-register again. The two things I've found that works is to either upgrade OR downgrade the firmware by 1 version, or to restart the firewall at the remote office.

I've updated the firmware on both Polycom phones to the latest versions, applied the latest firmware to both firewalls (we have other remote sites that do not have this issue) and made sure that SIP ALG and VOIP application control are disabled on both firewalls. I can consistently reproduce the issue by unplugging the modem (not the firewall) at the remote site and then plugging it back in. When internet comes back up, the phones will not be registered and will never try registering again until the firewall is restarted or firmware version is changed. I've also played around with registration expiration and timeout settings on the phones, but this doesn't seem to work, either.

I'm thinking this may be a FortiGate firewall issue, but it's strange that my other 3 remote sites (using the same …
0
I have a client using ShoreTel for their VoIP services.  They have an ASA 5506 as their firewall and I'd like to set up prioritizing on it for the outgoing VoIP communications.  I'm trying to identify a fairly straightforward way to do this.

They don't support DSCP tagging, so those options are out.

It appears that all communications to and from the phones are with three different subnets at the CO.  My plan is to set up a network object for each of these subnets and then a group object (IPPhones) that contains all three of these new objects.  I'd then set up a Service Policy Rule for traffic on the outside interface between any4 on the inside network and ip service on IPPhones on the outside.  The last is to set a Rule Action to "Enable priority for this flow".

Does this sound like a good approach?  It seems pretty simple and straightforward while being thorough.

My assumption is that there is very little traffic with those three subnets other than the phones.

If I want to prioritize incoming traffic, I'd follow what I did above but swap Source and Destination and apply it to the inside interface.  I'm assuming that this is less critical as we have much faster LAN speeds (1G/s) as compared to the internet download speed (100M/s).  My focus is on outgoing traffic as we have 20M/s capability there.

Thanks in advance of any useful comments.
0
Hi all,

We've got a client with 2 managed Cisco routers (not sure what model at this stage and we don't have control of these) that have a leased line into these with different providers and they have asked us to setup failover for them so we were going to get 2 more Cisco's and put these in HSRP on the LAN side behind the managed Ciscos with one of the line as the main line and the other as the backup/failover line but they have SIP coming into one of the lines so is there a way to set a WAN IP to "float" between the 2 Leased Lines to point the incoming SIP trunk to ? They want to avoid having to call the comms company to get them to change the SIP endpoint IP should one leased line fail.
Hope this makes sense.
Thanks
0
I have a client who recently purchased VoIP service from ShoreTel and also installed Jabra 9450 wireless headsets.  They use ShoreTel Flex software on their computers instead of hardware handsets.

The issue is that they are unable to answer incoming calls with the button on the headset;  they have to click on a button in the software, which is not very convenient.

ShoreTel hasn't been much help on this as they say they don't support this feature.

Considering how basic and useful this feature is, I'd be very surprised if there's no way around it.  I'm hoping that someone on EE is familiar with this and can suggest a workaround.  Using different software (such as ShoreTel Communicator) is certainly possible as well as a different brand or model of headset.

Any useful input would be greatly appreciated!
0
I'm running IP office manager v8.1
for some reason the call forwarding has stopped working.

I have the system to forward calls at night by manually hitting the "Night forward" button, which is ext 298 and it is forwarded to an outside number.

when i hit forward and hit that number, the calls don't get forwarded and they're not ringing at office either. help?
0
I've been struggling with VOIP call quality. One thing I notice is where calls to PSTN or Conference are made from work on the west coast or from home on the west coast - the traffic takes a 100ms trip to the east coast to get to the Skype voice control and RTP gateways. Is there any way this can be altered so you can use gateways on the west coast to reduce hops and delay to the Skype voice gateways?
0
I am curious if there may be a better phone system backup strategy other than what I am currently doing. My current backup involves  rerouting the main number to a series of cell phones that are configured in a chain (no answer/busy forwarding enabled on each)

I have noticed there are some online phone system backup solutions. In the event that our onsite phone system goes offline for whatever reason, I need a solution that is cloud based, and does not require any local hardware other than a computer with internet access. It would need to be able to accept at a minimum two telephone numbers that would be forwarded. Does anyone know of any service that could handle this?

Thanks in advance!
0
Hi All,
 
We currently have Business Voice Edge VIOP from Comcast which is their proprietary voice platform. They have provisioned and require a 50Mb circuit over their fiber backbone to our office to services SLA for their voice platform. Thus far, as far as reliability, I have to say that we have had no real issues with call quality over the past year of usage.
 
Management has decided to move offices earlier than expected, and we overlooked Comcasts terms and conditions regarding portability of service to locations that do not currently have a Comcast fiber  backbone in their building – which the location we are moving to does not have Comcast fiber. They are also not willing to work with hhus to temporarily provision over another circuit. At this point, we have three options – ranked in order of preference, and I wanted to know if anyone has experience and any recommendations to help in making the right decision. Here are the scenarios:
 
Upgrade with Comcast to the new location and wait 6 months for them to build out their own fiber (includes city permits) to the new office.
One of two options in this cast to get service to our new office:
                                                               i.      Implement RingCentral month-to-month as a temporary VOIP platform while we wait. Forwarding temporary numbers to main numbers.
                                                             ii.      Implement a Ethernet Dedicated E-Line (point to point) between our …
0
Hi All,

We currently have Business Voice Edge VIOP from Comcast which is their proprietary voice platform. They have provisioned and require a 50Mb circuit over their fiber backbone to our office to services SLA for their voice platform. Thus far, as far as reliability, I have to say that we have had no real issues with call quality over the past year of usage.

Management has decided to move offices earlier than expected, and we overlooked Comcasts terms and conditions regarding portability of service to locations that do not currently have a Comcast fiber backbone in their building – which the location we are moving to does not have Comcast fiber. They are also not willing to work with us to temporarily provision over another circuit. At this point, we have three options – ranked in order of preference, and I wanted to know if anyone has experience and any recommendations to help in making the right decision. Here are the scenarios:

1.)      Upgrade with Comcast to the new location and wait 6 months for them to build out their own fiber (includes city permits) to the new office.
a.      One of two options in this cast to get service to our new office:
i.      Implement RingCentral month-to-month as a temporary VOIP platform while we wait. Forwarding temporary numbers to main numbers.
ii.      Implement a Ethernet Dedicated E-Line (point to point) between our office via layer 2 routing.
2.)      Terminate with Comcast and incur termination charges – something that we don’t want to do.…
0
We're running into call quality problems and I think part of the problem is that the VOIP real time packets are traversing many hops to the other side of the country before they finally make it into Microsoft/Skype network. My question: Is there any way to choose which gateways to use for peering with Skype for Business 365?
0
We just connected a new Cisco SG500-52MP to an existing Cisco SG500-52MP.  We use both VLAN 1 as the default and VLAN 20 for voice.  Via the web interface, on the new switch, we tagged ports 1-48 for VLAN 20.  We connected the new switch port 48 to the existing switch port 1.

When we connect a VoIP phone on the new switch, the phones do not connect.  If we connect the phone to port 1 on the existing switch the phones work so I am sure we are missing a step on the new switch.  What are we missing?
0
I hear that waiting until the URL for a meeting arrives for first set up WebEx can be time consuming. So, is there a way to do this initialization beforehand?

Thanks.
0
I have a pair of C-level users who both experienced a problem at one of my sites. I'm trying to determine if it's a GoToMeeting problem (not my problem) or a phone problem (definitely my problem). You guys will give me a quick answer, I just know it. :-)

When the users join the audio portion of a GoToMeeting event from their Cisco desk phones (on my CUCM-powered phone system), right after the point where they enter their PIN number for the meeting, they are supposed to press the pound key "#" to join the audio part of the meeting. Every time each of them presses the "#" key on their desk phones, either the meeting or the phone hangs up the call. When they join using their cell phones it works fine.

This just started happening and no changes have recently been made to the phones or the CUCM settings. I'm told that it happened once before but was not reported, and that other occasion occurred several months ago but then the conditions returned to normal operation and they didn't see this again until yesterday.

Can I get some opinions on which party is responsible?

Thanks experts!
0
I am attempting to force all ShoreTel IP Phones onto the voice VLAN.  However despite various attempts the connected devices remain on the default VLAN.  Due to computers connected to the pass-through Ethernet ports on the phones we cannot use the primary VLAN for the port.

I have a phone with a computer in pass-through connected to gi6.  I have been attempting to coax the switch to place the phone on VLAN 10.  Yet I consistently receive the following output:
switchb#sh mac add int ge6
Flags: I - Internal usage VLAN
Aging time is 300 sec

    Vlan          Mac Address         Port       Type
------------ --------------------- ---------- ----------
     1         00:10:49:45:8c:26      gi6      dynamic
     1         08:2e:5f:07:b1:7d      gi6      dynamic

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Running Config:
switchb#sh run
config-file-header
switchb
v1.4.8.6 / R800_NIK_1_4_202_008
CLI v1.0
set system mode switch

file SSD indicator encrypted
@
ssd-control-start
ssd config
ssd file passphrase control unrestricted
no ssd file integrity control
ssd-control-end 
!
vlan database
vlan 2,10,65,200-201
exit
voice vlan id 10
voice vlan state oui-enabled
voice vlan oui-table add 0001e3 Siemens_AG_phone________
voice vlan oui-table add 00036b Cisco_phone_____________
voice vlan oui-table add 00096e Avaya___________________
voice vlan oui-table add 000fe2 H3C_Aolynk______________
voice vlan oui-table add 001049 ShorTel
voice vlan oui-table add 0060b9 Philips_and_NEC_AG_phone
voice vlan oui-table 

Open in new window

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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).