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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43 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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24 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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How to detect noise from RTP and clear the voice

Currently, I am developing a VOIP system using Mumble and SIP. I can send the SIP stream to the mumble server, after converting the RTP stream to PCM and the voice quality is fine in mumble client. ...
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66 views

FreePBX dropping calls from Twilio for unknown reason

I have a single FreePBX system connected to Twilio's Elastic Trunk service. Local server IP is 10.0.3.1. Sometimes, when a call is placed, the call comes in, goes through our system, and routes ...
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23 views

Is it possible to connect hardware VoIP phone to the PBX behind IPSec VPN?

I have a hardware VoIP phone, a PC with VPN Client, a PBX (FreePBX) over that VPN connection. I have a softphone that works pretty well through that IPSec connection. Can I use named PC as some kind ...
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669 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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93 views

Asterisk ARI: Get the application name of a channel

I try to access my installation of asterisk via ARI. When I use curl on the endpoint /ari/channels (GET), a list of all existing channels and some detailed information are being returned. According to ...
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1answer
66 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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29 views

Cisco 7912G ringtone add

I'm curious how can i add ringtones on a Cisco 7912 flashed to SIP. I put to tftp server Ringlist.xml, and ringtones in the / directory. I also set on phone alternate tftp server IP to machine where I ...
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2answers
227 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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107 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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58 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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49 views

Cseq RFC comparison and understanding

I have a below call flow, wherein I have some confusion related to CSEQ..... A --INVITE CSEQ-1-> B A <--180 CSEQ-1 --- B A --PRACK CSEQ-2-> B A <--183 CSEQ-1 --- B A --PRACK CSEQ-3-> B ...
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10 views

Asterisk Integration Issue with CRM

I have configured V-tiger CRM. I want to integrate v-tiger CRM from Vici-dial asterisk but in v-tiger CRM there are not showing phone calls extension in extension store. I have configured the call ...
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231 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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384 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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198 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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20 views

NAT on Windows10 with 2 NICs and OpenVPN

Due to a recent pandemia, we have to send our staff into home office. What I'm trying is to enable some of them to use their PBX system IP phones from home. This is a quite easy task using a VPN-...
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63 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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92 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
462 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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1answer
79 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
111 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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66 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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151 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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122 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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622 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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49 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
52 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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263 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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499 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
83 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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304 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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419 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
577 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
91 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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471 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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230 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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2k 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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115 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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1k 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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82 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
636 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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602 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
210 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
596 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
626 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
71 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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1answer
94 views

Creating artificial CPU / memory load on CUCM server

I'm working on a small monitoring application for organization multicluster CUCM environment. For a demo purpose, I have a small lab, that consists of VMware CUCM installation and a couple of hardware ...

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