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 attempting to troubleshoot our VOIP calls with packet captures and wireshark. Unfortunately, when I choose Telephony->VOIP CALLS, there is nothing displayed. I definitely testing captures during 2 test phones calls and insured protocols are enabled but still nothing. Any help would be greatly appreciated!
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Of can it work by attaching directly to CUCM as well? I think it's only with Expressway but way to check. Thank you.
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I'm adding VOIP on an existing site that uses either Cisco SRW20xx or SG300-xx switches.  I'd like you to comment on my plan for doing this:

The VOIP will be coming in from the internet on it's own connection / firewall and will be using a separate local area subnet.
It will generally be distributed through all the switches unless there's no phone at all, just computers or network devices.

There is a central LAN switch that feeds into other switches in cascade.  I will refer to this as the TOP switch here.

My plan for the downstream switches is this:
Assign VOIP VLAN 100 to all the switch ports along with the Default VLAN 1.
Trunk all the switch ports.
Tag VOIP VLAN 100.

My plan for the TOP switch is this (there being only Default VLAN 1 and VOIP VLAN 100):
Trunk all the switch ports that feed downstream switches.
Trunk any switch ports that directly feed a VOIP phone.
Leave any other ports on Default VLAN 1 in Access Mode.
Assign VOIP VLAN 100 to a single switch port that goes to the firewall.  
Make this a General Mode port joined to VOIP VLAN 100.
Manually tag this port <<< is that right?
Internet Port Setting / TaggedThe VOIP firewall won't have any VLANs set up, just a generic LAN.

Since I've never done this before, I'm a bit unclear as to whether the VOIP firewall port needs to be tagged or not BUT the port sure needs to be part of the VOIP VLAN 100 ONLY with no interVLAN routing / connection.  I want the traffic on the two VLANs to be completely …
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I have a small office with one VLAN.  There are 15 IP phones and 15 PC's that plug into a Cisco 3560G switch.  The phones and PC's are on the same VLAN.  The phones are provided via a hosted VOIP provider.  Lately we have been experience voice quality issues.  The hosted VOIP provider have asked us to prioritize the voice traffic so it has priority over all other traffic out to the WAN.  My network is very simple and looks as follows:

Cable Modem----------Cisco 2911 Router-----------Cisco 3560G switch (phones and PC's plug into this switch.)

Could someone please send me a sample config on how to accomplish this?  Thank You!
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On cisco IP phones (model number 7911) it stores some useful information about placed/received calls in the directories application- is this data stored locally on some storage within the phone, or would this be stored in a central database in a managed voip environment, if so being a cisco device can you elaborate where that information may be stored.
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I am the Office 365 administrator.

What steps need to be followed to record all Office 365 Skype for Business instant message chat conversations?
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Hello - im looking for hold music that is free to use. Anyone have a link to download one?
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Greetings,

I have a site where I have several Yealink T26P phones connected to multiple Aruba (HP) 2530 POE switches.  There is a stack of 4 switches, with the primary switch connected to a SonicWALL TZ300 router.  All the phones on the primary switch work normally.  The phones do not work on the secondary switches.  I have two VLANS (data and voice).  Basically, I have every port tagged with the voice VLAN and untagged with the data VLAN.  The only difference is that I have two ports on the primary switch that are untagged for the voice VLAN only (voice uplink from SonicWALL and phone controller).  In testing with other phone types, I've been able to plug into the POE port on any switch and have it work.  This is my first interaction with the Yealink phones, and it is remote.  As the only difference between the switches is the uplink port config, I'm guessing that I need to have a port untagged for the voice VLAN only and a different port untagged for the data VLAN only?  This would seem like a very poor design, as it would effectively require the use of up to 4 ports per switch.  So, what am I missing?  My SonicWALL is serving as the DHCP server and seems to be working fine and it actually shows the non-working phones actually get an IP.  They just don't connect.  Move them to the first switch and they immediately connect.

Thank you for any assistance that you can provide.

Jeremy
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Hello,

We are using FreePBX, and have Cisco 525G2 phones, and we added a SPA500s sidecar, how do i configure extensions on it?
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I have multiple Cisco 504g phones, that don't seem to take the attendant console changes. I have set up the first 3 extensions as a direct speed dial to other extensions on the network (the extension is set as disabled), and all 3 speed dials with flash orange, even though they are subscribed and working. I know to add c=g in the serv subscribing section of attendant console, but these phones seem to not take any changes I make. Hopefully somebody has an answer for this, as these are for a high end hotel, and I cannot have these lights constantly flashing.
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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?
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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]
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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 …
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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.
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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
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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!
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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).