Questions tagged [voip]
Voice over Internet Protocol (Voice over IP, VoIP) is one of a family of internet technologies, communication protocols, and transmission technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.
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Anybody familiar with the Shoretel database?
I have a copy of the local db from our old shoretel voip system... which is no longer in service.
I've been tasked with trying to locate information pertaining to calls to and from specific numbers ...
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Atxfer does not transfer
I have two agent A and B.
Agent A has a call online and wants to transfer the call to agent B.
I already have axtfer => *2 configured in the features.conf file and I have alltransfer set to YES on ...
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Is using Outgoing SIP Registration the only way to allow incoming calls if you do not allow port forwarding or any new incoming connections?
If I use a SIP provider that uses SIP REGISTER for connections (instead of using an IP address), the outgoing registration originating from my side opens a connection to the SIP provider, and they use ...
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connecting two SIP phones from different LANs over ZeroTier openwrt
i make site to site conncetion by tp-link n600 openwrt zerotier and no problem , i can reach every thing in two sites and ping ok
but i have in site 1 Issable and in site 2 sip phone
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Grandstream wp820 not registering after router/modem switch
Greetings fellow system engineers!
I'm new to the forum and I do have a rather strange problem. Please be aware that I am encountering the following issue as a hobbyist / power-user. I do have some ...
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Get Asterisk IVR inputs to Linphone user
Let's say someone creates an Interactive Voice Response (IVR) system using Asterisk, that prompts the caller to input his customer number using his dial pad, e.g. 1234.
How does one get this ...
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No RTP engine was found. Do you have one loaded? Asterisk-18.10.1
I have been trying to install Asterisk-18.10.1 version on my ubuntu(20.04.4) running inside VM. I was able to maintain connection from GoTrunk SIP endpoint and Zoiper as softphone. Followed https://...
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How do I upgrade Firmware on Cisco SPA50X VOIP Phone?
I downloaded the latest Firmware for my Cisco SPA504G. However, the usual .exe file used to perform the upgrade was not bundled with new firmware. Nor did the release notes provide any clues about how ...
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Lack of audio for SIP call propagated from IPv6 network when using coTurn server
We are using coTurn server to allow our SIP appliance work reliably between client apps and teleconferences. During our test's we have found that one type of calls is not working at all. But at first ...
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Can a VoIP or an L2 switch prevent a PC from accessing a network printer under 2 VLAN circumstances?
My organization's network topology goes like this:
PC -- VoIP -- L2 switch (no IP address, only performs L2 switching)--Network Printer
Also, the same L2 switch is connected to an L3 switch for access ...
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Polycom VVX500 not picking up provisioning server
The company called us regarding needing assistance with setting up a phone line they haven't used in a while Polycom line was initially using a USB wifi dongle so we requested to connect it directly ...
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How to hangup Freeswitch/Sofia SIP-Calls at a 3CX correctly?
We are running a FreeSwitch-instance at host A, that is placing outbound calls to a 3CX-instance running at host B, using the SOFIA module.
Everything is working fine except for the hangup: The BYE ...
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Yealink phones fail to get IP once assigned voice VLAN via DHCP (Mikrotik)
I am deploying Yealink IP phones (T40G, T23G) on following newtork equipment:
CCR1009-7G-1C-1S+ as a router
CRS328-24P-4S+ as switches
Config of the switch in trouble is here.
I set up DHCP option 132 ...
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VoIP and NAT (and blocked ports)
I'm making a VoIP application and I have trouble to make it working properly. On each side there are SIP clients.
In my office, we use 2 differents boxes to access internet. The first one is like a ...
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tel: URI support in Freeswitch(for outgoing calls)
I have integrated a Freeswitch instance with an external SIP server and calls are working without any issues. But now I need to change the SIP server which only supports tel: URI.
In the case of ...
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Latency on VoIP network
our network consists of Aruba and Brocade switches. We have two models of Yealink phones, t21 and T46. Problem seems to be the t21's 10/100 switch with a PC connected behind it. the phones are on ...
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How to use GnuGk as a H.323 gateway/proxy
Please help me by letting me know how to use GNU gatekeeper, I have installed in on a Ubuntu server but I do not know how to use it.
What I need is, now I have a Cisco CMS and enabled H323 bridge on ...
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Issabel-Asterisk- How to handle multi-digit inputs on IVR
I have an Issable server running and integrated with Cisco Call Manager correctly.
I am using Issabel server as my IVR system to answer calls without any direct extension callings.
My Scenario is as ...
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how to run multiple instances of asterisk on one server
hope everybody is safe during challenging time
i have a task to automate our demo installation process.
Our software is mainly a GUI that operates with asterisk via AGI and stores all data to MySQL / ...
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Large Jitter Spikes on a Small Percentage of VoIP calls
On about 5% of our customer's calls, we see large jitter spikes and high delta #s that have caused an noticeable audible impact on call quality. (Stuttering/Breakups/Robotic Audio). We know this from ...
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How to connect an Alcatel-Lucent IP touch phone over VPN?
We have some workers working at home, and we thought about providing them IP phones (Alcatel-Lucent IP Touch) which they could plug into their computers through their Ethernet port (or an external USB ...
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Asterisk: Using Queue() in the h extension
I'm trying to get users to be able to record a message, hangup and have the call continue and dial Queues and playback the recording.
I've gotten most of the way there, but right now when I call ...
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Freeswitch drops calls after 32 seconds
So I've installed Freeswitch on a raspberry PI 3 and it's dropping calls after 32 seconds. I've googled extensively and this appears to be a common problem but all of the people with the problem had ...
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VoIP phone only working in one direction
Ahoy friends.
Currently i got 2 networks connected via routing.
I got a VoIP device connected to one of the networks, to be routed to the other one.
Everything else is working on this configuration of ...
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For SIP trunks, is it possible that DTMF talk-off could mute a call?
My company sells/supports Allworx SIP PBXs. We have a weird edge case with one of our customers who use PRI for dial-tone. On outbound calls, if a user happens to receive an SMS on their iPhone, and ...
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VoIP over VPN (FritzBox) – can register, can call, but no sound
We have a small company network here (192.168.178.x) and a private network in home office (192.168.2.x). These networks are both running a FritzBox router and are connected via VPN trough FritzBox to ...
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Capture the VOIP calls receive to a sip phone that send Notificantions to an API on a call
This is whats I'm trying to achieve I want a SIP phone or (is there is any other way I'm open too) to take the calls it receives from a VOIP provider take the information related to the call when ...
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VMWare & VLANs - why am I able to access one VM but not another?
I have two virtual machines on a VMWare ESXi server - a Sophos Firewall and a Windows Server. On the LAN side of my network, I have a bunch of devices and a Cisco ATA connected to a real switch. This ...
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Provisioning a Unidata ICW-1000G WiFi phone
Apologies if this is off-topic here (if so, is there a better place?).
I just bought a Unidata ICW-1000G WiFi phone (because I prefer to use just WiFi and not to use DECT). I'm attempting to do auto-...
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Freeswitch/fusion 'ghost' calls bypassing F2B/domain acl
We have been receiving 'ghost' calls from non-existent extensions. I've run into this before on asterisk systems and usually just configured the sip profile to disable guest/anon calling. However, ...
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Why does my Netgear Smart Switch randomly untag VLAN on ports?
I have a NETGEAR 48-Port Gigabit PoE+ Smart Managed Pro Switch with 4 SFP Ports (GS752TPv2) that has been sporadically untagging a VLAN on ports. The VLAN in question is an Auto-VoIP VLAN. The ...
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VLAN not working in MikroTik RouterBoard HAP
I'm currently having some problems configuring the VLAN in our network.
Description: get the VOIP phones in to the VLAN2 with the MikroTik RouterBoard HAP. VPN configuration is on this device for the ...
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Freeswitch/FusionPBX trying to list all inbound routes/destinations and their associated actions
I mainly work with Asterisk/FreePBX but just recently started working with Freeswitch on a large scale so forgive me as I'm only now starting to explore a lot of the interface.
Issue: User wants to ...
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Connecting a Fujitsu SS-170A phone to Asterisk
I have recently acquired a Fujitsu SS-170A VoIP phone in order to connect it to my Asterisk server.
First things first, I am able to perfectly use softphones in the same network I am setting up the ...
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IP phones get DHCP NAK from superscope configured DHCP server
We are experiencing a strange problem with our 2012R2 DHCP server.
Setup looks like this:
Superscope with two scopes:
SCOPE 1 - 192.168.100.0/24 (for PC)
SCOPE 2 - 192.168.101.0/24 (for IP phones)
No ...
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in process_sdp: Ignoring video stream offer because port number is zero (ONE WAY)
I have found dead ends for this problem on different forums and sites, it would be time to solve it for everyone's good.
Given 2 devices:
A cisco 894x IP phone with h264 support
B softphone software ...
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Replacing IP address of connections from AWS towards SIP
We have installed a PBX on AWS and connected it to our on-prem Router via VPN.
My on-prem router is connected to the SIP provider via a physical connection with another on-prem MUX device (device ...
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What are RTP packets with zero timestamp and no payload?
I have an RTP traffics which some of the packets have zero timestamps and no payloads.
Packet is :
10.. .... = Version(2)
..0. .... = Padding(false)
...0 .... = Extension(false)
.... 0000 = ...
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Multiple Asterisk Servers Behind Single NAT
I'm debugging an issue we're having with VoIP SIP registration. After looking at packet dumps, I think the issue results from 2 subnets accessing the same web service and the router not knowing how to ...
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Using Asterisk as a gateway to Provider
I have 2 sip servers on different LANs. Freeswitch and another is Asterisk.
Asterisk sits on a VPN with a provider and who has provided DIDs. All users register on Freeswitch. How can I route calls ...
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Server as phone behind eMTA
I've been tasked to research automation options for telephony modem/gateway device testing with a group of DOCSIS voice engineers. For each new DOCSIS device with an eMTA that we receive we need to ...
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Asterisk JITTERBUFFER function details
Asterisk v15 documentation states that JITTERBUFFER function is used to "add a Jitterbuffer to the Read side of the channel". Take this dialplan example:
[from-pstn]
exten => 1234,1,Set(...
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How to screengrab from Cisco SPA504g phone?
Is it possible to screengrab from a Cisco SPA504g phone? I'm wanting to make user manuals which have the exact thing the users will see.
Would be convenient for remote checking of provisioning too, ...
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How to enable asterisk Call Completion on Busy Subscriber (CCBS)?
I cannot enable Call Completion on Busy Subscriber (CCBS) on asterisk, witch is part of the Call Completion Supplementary Services (CCSS), as is the Call Completion on No Response (CCNR) feature.
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Testing quality of sustained call quality to a Public number
While migrating from an old Avaya to Skype for Business (on-prem), I have been asked a few questions about call duration and call quality during maintenance. Questions stem from unclear answers from ...
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AudioCodes 440HD Boss Admin feature configuration
After seeing a demo of how the AudioCodes handle Boss/Admin Delegation I wanted to try deploying this to some users. After some digging I was found the trick was to "flip the SideCar settings". After ...
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Causes of RTP jitter at the server
Investigating some call quality issues (0.5 – 1 second dead spots in calls) I took a packet capture of a phone call between two extensions on the same PBX. Since I was capturing from the PBX, I was ...
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Slow Down Mod_Flite TTS in Freeswitch
I am working with Freeswitch on a debian server. I have an XML file which Flite reads from when someone calls the phone line.
The problem is that she reads it just a little too fast.
Is there a ...
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FreePBX - How to re-write forwarded "follow-me" caller ID?
Sam calls Julie's direct dial line and because she's not at her desk (and because of a follow me rule) the call is forwarded to her mobile.
FreePBX forwards the call through the correct outbound ...
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Do I need STUN if only one party is behind NAT?
One party is connected through a symmetric NAT, other party can be accessed directly, and there is a STUN server.
The parties are going to communicate via WebRTC. Both are configured to use that STUN ...
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