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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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25 views

QOS Switch or Router

I have IP phones on my network along with computers on the same subnet. Traditionally, I would setup QoS on the router to give priority to the voice traffic over the data traffic. My question is ...
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1answer
38 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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24 views

How to remote control a VOIP conferences - FreeSwitch mod_conference

Is there any standard way to remote control conferences in FreeSwitch (or generally in VOIP related standards)? I don't mean by DTMF because i know this method, i'm rather asking about some protocol ...
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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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1answer
23 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
71 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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73 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
39 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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102 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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1answer
104 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
51 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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22 views

No Audio on VOIP Phones

I have a Grandstream UCM6108 Server on my internal (school) network and 30~ Grandstream Phones. The server and phones are not on the same VLANs but are routed. Over the holidays something somewhere ...
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1answer
59 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
158 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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70 views

IPSec and traffic mirroring

I have a VoIP server on which several SIP clients are connected to through IPSec tunnels. This is kind of an automated attendant so I both have RTP and SIP signalling flows. I need to be able to ...
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1answer
83 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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0answers
445 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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0answers
19 views

Testing quality of sustained call quality to a Public number

While migrating from a 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 the ...
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1answer
52 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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2answers
483 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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0answers
36 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
314 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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1answer
130 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
123 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
283 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
413 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
56 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
60 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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1answer
122 views

Shoretel Connect: Create SIP Trunk Backup for Analog Trunk on SG90

I've worked with some VoIP Products before, mostly Asterisk (base version, build from source) management for a small ISP doing in-network VoIP solutions for customers. My current position hasn't ...
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0answers
111 views

Asterisk -rx running the CLI

I've updated my asterisk to 14.4.1 and now I can't run commands outside of CLI. It used to be asterisk -rx <command> but now it runs the CLI and not the command. I've also tried '-x' alone but ...
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1answer
29 views

Where to ask about SNOM870 administrator access?

I updated one of our SNOM870 VOIP handsets to FW 8.7.5.35 and now cannot access the phone's administrative interface. The web access works and the phone finds the SIP proxy but many of the visible ...
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2answers
3k views

Asterisk redirect calls to external number

We have a phone system created on pure Asterisk (no web GUI) and there are the usual Day/Night mode. In the day mode it calls a ring group. However we have a call centre that we use to pick up our ...
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1answer
210 views

How to upgrade HP 4110 VoIP phone firmware.

Actually, I have found the clue here on after googling, but the problem is I can not download the Microsoft Lync Server 2010 or 2013. http://jackstromberg.com/2013/05/configuring-lync-server-to-push-...
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2answers
2k views

Where to download Microsoft Lync Server 2010 or 2013

It tried to google many time to download Microsoft Lync Server 2010 or 2013 evaluation. but the links I got has changed and can not find it. http://technet.microsoft.com/en-in/evalcenter/hh973393.aspx ...
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1answer
53 views

How does buffering work in a chain of Asterisk servers?

I have two Asterisk servers in a chain like this: SIP client (G.729) <-SIP-> Asterisk 1 (converting G.729 to G.711) <-IAX2-> Asterisk 2 -> terminated to analog line. When no buffering was ...
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1answer
1k views

Mikrotik SIP trunk routing

I have a Mikrotik RouterOS v6.36 with my network as follows (I am a newbie to networking in general and have been able to get stuff working half way. I just don't know what to look for.) Router: ...
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0answers
36 views

How can I tell asterisk to initialise each music on hold application when needed rather than constantly

I'm running a custom MoH application for a large number of users under an asterisk 11 box. The configuration is as follows: [customer-100] mode=custom application=/usr/bin/ruby /path/to/script.rb 100 ...
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1answer
3k views

SIP registration timed out

I am facing trouble in registering asterisk to sip trunk. When i do >sip show registry, it shows SIP request is send but never gets response back. I cannot even ping sip.flowroute.com. My firewall is ...
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4answers
12k views

Register asterisk to sip trunk

I want to register my asterisk server to a SIP trunk. I have added following piece of code in my sip.conf and extensions.conf sip.conf [general] register => myusername:mypassword@sip.flowroute....
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1answer
34 views

“Generico” networking issues: how to investigate

Recently I'm experiencing networking issues in my LAN. I have an Ubuntu server doing PDC, mailserver and Asterisk. There's KVM on it, which runs a Windows VM and pfSense firewall. From time to time ...
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0answers
189 views

Can Asterisk's phoneprov module be used securely?

I'm using Asterisk 13.1.0 as packaged by Ubuntu Server 16.04 to run a pure-VoIP phone system. Asterisk has a module – phoneprov – that allows it to template out configuration files for ...
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0answers
279 views

SonicWall QOS for VOIP

I has a Sonicwall NSA 220, I want to enable QOS on the network. But I want any device to be included so any VOIP call gets priority. I've seen guides on setting up QOS or bandwidth management but for ...
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1answer
273 views

HP 4120 phone and Asterisk

I got a task to set up a cheap PBX for HP 4120 phones. I want to try Asterisk, but I can't find any info on connecting HP 4120 to any SIP PBX. So here's the question: are HP 4120 compatible with ...
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1answer
626 views

ASA QoS for VoIP traffic

I'm looking to implement QoS for VoIP traffic using DSCP tags: EF & AF31. LLQ for VoIP: ciscoasa(config)# class-map cm_voip ciscoasa(config-cmap)# match dscp ef ciscoasa(config-cmap)# match dscp ...
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6answers
2k views

Asterisk ghost calls

Problem We have an Asterisk server hosted externally. On four locations we've gotten ghost calls. These are calls with different numbers like 1000, 9999 or 6060. We don't use these numbers, not even ...
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1answer
1k views

How to reset a Cisco ATA 186 to factory defaults?

I have an old Cisco ATA 186, configured by an extincted Phone Service Company, it is password protected and on TFTP client mode. I tried to reset it with IVR FACTRESET code (322873738#) but it ask ...
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2answers
255 views

Polycom SoundPoint IP 4000 won't power on. How to isolate power supply?

I have a Polycom SoundPoint IP 4000 SIP conference phone that has sat unused in a box for a number of years. Today, I pulled it out of the box and powered it on with the conference power supply only ...
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1answer
323 views

Freeswitch configure caller-ID with mod_lcr

I am working to get LCR (Least Cost Routing) working with Freeswitch. Using mod_lcr in Freeswitch 1.4.9. I have set it up and it is working. My install is actually a FusionPBX instance, but this ...
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1answer
236 views

Least Costing Routing in Freeswitch - Cannot Read Config file

We have a Freeswitch installation as part of a FusionPBX GUI system. FusionPBX version 3.7.1, Freeswitch version 1.4.9. Database used by FusionPBX is Postgres. The install compiles the Least Cost ...
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0answers
57 views

VLAN Routing for VOIP

Confession first - my networking skills are not massive but I'm learning fast! We have three VLANs (for the sake of this question - there are actually more but not relevant to the posting). These are:...