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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Remote site's VoIP phone connection without VPN

We have a small studio overseas. Staff in the remote office have IP phone. Phone exchange in main office. A VPN tunnel connect two sites for data and voice. Recently, the local government forbid to ...
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45 views

How to config router on server over switch

In a working VoIP setup i want to replace the PC(Isabella) with new virtual machine(Isabella ready) on a new server(connected to new switch) using esxi 7 PC is connected to the internet port of router ...
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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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VoiP SIP trunk to Cisco RV345 and SG200-50 Switch epic Fail

Im having some serious trouble with VoiP phones that are sip trunked over the internet. I.E. we do not have an On Premises PBX. Every thing from the provisioning to the calls is done over the internet ...
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50 views

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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1answer
86 views

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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1answer
498 views

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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30 views

Forwarding Caller ID info to a specific IP address

I'm having an issue where I'm supposed to log all the incoming Caller IDs to a specified IP address in the Elastix environment. Can you suggest anything workable as a solution or to what should I ...
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1answer
246 views

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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83 views

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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673 views

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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1answer
125 views

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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2answers
615 views

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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317 views

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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1answer
119 views

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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784 views

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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757 views

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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806 views

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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153 views

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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1answer
166 views

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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2answers
1k views

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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91 views

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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1answer
181 views

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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100 views

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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203 views

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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1answer
210 views

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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1answer
923 views

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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64 views

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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1answer
67 views

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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1answer
399 views

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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630 views

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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1answer
116 views

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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421 views

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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610 views

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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1answer
1k views

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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1answer
100 views

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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1answer
562 views

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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1answer
280 views

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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1answer
3k views

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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42 views

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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1answer
132 views

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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2k views

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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106 views

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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1answer
781 views

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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926 views

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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1answer
268 views

Asterisk (VOIP) port forwarding on TP-LINK [closed]

I have local asterisk server on UBUNTU 16.04. It's address 192.168.0.101:5060 and I would like to be allowed to talk with other users when I am outside local network. I tried THIS but it doesn't work. ...
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1answer
778 views

Connect one SIP phone to multiple Asterisk Servers

I am working on a project where there are multiple apartment complexes that share the same reception desk, it's just one of the buildings. There are VOIP devices in those apartments that have to be ...
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1answer
677 views

MixMonitor applied to every extention in a context

I'm trying to setup call recording in Asterisk applied to a whole context. extensions.conf looks like this: [office] exten => _068.,1,Dial(SIP/sip_trunk_1/${EXTEN}) exten => _073.,1,Dial(...
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1answer
80 views

Does this follows the SIP standard?

We are currently using a Swyx PBX which connect using SIP to a OneAccess box (which translate ISDN <-> SIP). We have trouble to make it works. Here's the deal: The Box uses the SIP protocol on ...

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