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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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Forward a single port to KVM Virtual Machine with Libvirt

I have a KVM that is currently running PBX software, and a OpenVPN server. I want to give a friend an IP Phone. I am connecting IP Phones to the OpenVPN Server so that they can communicate with the ...
chris chris's user avatar
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49 views

UDP connection not recognized as UDP stream

A question from the VoIP sector. Several devices behind NAT wants to establish UDP connections to a SenderIP. The NAT settings for UDP are net.netfilter.nf_conntrack_udp_timeout = 30 net.netfilter....
Ralf Viellieber's user avatar
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43 views

VoIP: No ringback on specific provider combination

please help me out with this one: PROBLEM DESCRIPTION No ringback when calling cellphones using VoWIFI/VoLTE Only happens when calling from a specific VoIP telephony provider to a specific cellphone ...
Giorgos Amoiridis's user avatar
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12 views

Traffic flow between network-interface due to VPC peering

We have a scenario, where we have SBC instance which has three network interface attached to it(1,2,3). SBC instance along with network interface 1(management),2 are deployed in VPC A, where as ...
user391's user avatar
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0 answers
90 views

Issue with Asterisk SIP Trunk: Payload Must Originate from Specific IP in Multi-Cloud Setup

I'm facing an issue with an Asterisk SIP trunk setup involving multiple cloud environments and a Session Border Controller (SBC). Our service provider, Airtel, requires that the SIP payload originates ...
George Muddu's user avatar
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84 views

How to Prevent FreeSWITCH from Adding Reason Header to SIP BYE and CANCEL?

I noticed that freeswitch adds a Reason header automatically to SIP BYE and CANCEL requests. I've successfully used the disable_q850_reason parameter to prevent FreeSWITCH from adding Reason headers ...
Mohamed's user avatar
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0 answers
170 views

ED137 SIP & RTP (VoIP) for Air Traffic Control Use with Cisco CUBE

I am working with an air traffic control client who is using a variant of SIP called ED137. There are 2 main uses, Ground to Ground (basically telephone calls) and Air to Ground to Air (radio traffic ...
user10021657's user avatar
-1 votes
1 answer
138 views

How to recover/reinstall Grandstream GXP phones firmware?

I bought a Grandstream IP phone recently and tried to go through first installation, but suddenly most of SIP settings and also settings of firmware update server disappeared and I even can't login to ...
Phree's user avatar
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-1 votes
1 answer
120 views

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 ...
boog's user avatar
  • 240
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0 answers
258 views

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 ...
JMain's user avatar
  • 115
1 vote
0 answers
117 views

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 ------------------...
Best Way's user avatar
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2 answers
541 views

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 ...
xevoryn's user avatar
0 votes
1 answer
429 views

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://...
xenon's user avatar
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2 answers
2k views

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 ...
F1Linux's user avatar
  • 405
1 vote
0 answers
117 views

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 ...
Paweł Madej's user avatar
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1 answer
81 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 ...
Mike Land's user avatar
1 vote
0 answers
249 views

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 ...
Fran's user avatar
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1 vote
1 answer
360 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 ...
Sebastian Schmitt's user avatar
0 votes
1 answer
3k 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 ...
Eugene's user avatar
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1 answer
779 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 ...
Bastien Matthai's user avatar
1 vote
1 answer
418 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 ...
uts9's user avatar
  • 11
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0 answers
719 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 ...
Dean Prince's user avatar
1 vote
1 answer
251 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 ...
fan boyu's user avatar
1 vote
2 answers
2k 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 ...
Zareh Kasparian's user avatar
-1 votes
1 answer
796 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 / ...
d3m0n's user avatar
  • 83
2 votes
1 answer
298 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 ...
AVoIPm8's user avatar
  • 29
0 votes
1 answer
2k 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 ...
Yanick Rochon's user avatar
0 votes
1 answer
1k 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 ...
Mattisdada's user avatar
2 votes
2 answers
2k 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 ...
MikeKulls's user avatar
  • 509
0 votes
0 answers
366 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 ...
Fabian Druschke's user avatar
0 votes
1 answer
246 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 ...
pooter03's user avatar
  • 436
0 votes
2 answers
2k 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 ...
Kevin Lieser's user avatar
1 vote
1 answer
171 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 ...
Luis Javier Davis's user avatar
0 votes
1 answer
671 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 ...
oneslyfox's user avatar
1 vote
0 answers
162 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-...
James Youngman's user avatar
0 votes
0 answers
350 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, ...
Malaxes's user avatar
  • 71
1 vote
1 answer
511 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 ...
Wilson's user avatar
  • 115
0 votes
1 answer
1k 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 ...
Ashwin's user avatar
  • 1
1 vote
0 answers
104 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 ...
Malaxes's user avatar
  • 71
0 votes
1 answer
119 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 ...
akasaka's user avatar
  • 101
0 votes
1 answer
835 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 ...
Eugene K.'s user avatar
  • 101
1 vote
0 answers
826 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 ...
bowre's user avatar
  • 11
1 vote
1 answer
183 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 ...
vichar's user avatar
  • 11
0 votes
0 answers
773 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 = ...
saeed's user avatar
  • 51
0 votes
1 answer
951 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 ...
sleepisfortheweak's user avatar
1 vote
1 answer
2k 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 ...
Ben Kabale's user avatar
1 vote
1 answer
184 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 ...
solarrdev's user avatar
1 vote
1 answer
965 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(...
Nikša Baldun's user avatar
1 vote
1 answer
380 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, ...
jonathanjo's user avatar
2 votes
1 answer
4k 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. ...
Alisio Meneses's user avatar

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