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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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....
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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 ...
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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 ...
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route incoming asterisk sip calls - fake auth rejected
we are trying to handle incoming sip_calls on our asterisk server, but somehow we always end up getting either 403 or 603 (which should be the same?)
what should happen if it works:
user calls ...
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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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High availability Asterisk options?
We currently run FreePBX on a single node. I'd like to have a cluster of asterisk instances for sharing the load, but mainly for failover. I'm curious how other people have solved this problem. ...
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SIP Trunk for VOIP behind a SonicWALL
I appreciate this question is quite "out-there" but - has anyone had any success with VOIP over SIP behind a SonicWALL. We are having an issue with one-way audio inbound.
Essentially, the person ...
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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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How can I call my voicemail in Bluebox Freeswitch?
I'm using Bluebox to manage Freeswitch. How can I call my voicemail? If I dial my extension it takes my to the part where I need to record a voicemail.
If I dial 4000, *89 or 879 its just doesn't ...
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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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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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What is the default username and password for the Cisco SPA-942?
I can't seem to find the default admin username and password for the web administration interface for the Linksys/Cisco SPA-942 VoIP phone. What is it?
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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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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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Cooling requirements for POE switches?
Is there any difference between POE switches and normal switches when budgeting for cooling capacity for a network closet?
Normally, when I'm thinking about cooling; I can look at the power draw of a ...
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Do I need VLAN for VOIP phone if my network switches supports IEEE P802.1p QoS?
I have 30 workstation and 30 IP Phones all under one single network 192.168.9.0. The network switches is unmanaged switch but supports IEEE 802.1p QoS (e.g.: DLink DGS 1008D and DGS 1016D).
I ...
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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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Command line SIP dialer
Is there a simple command line SIP dialer for unix which can connect to SIP server, make a call and play some media file (wav/mp3)? In ideal I would look like this:
sip-dailer +1xxxxxxxxxx /path/to/...
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Asterisk dialplan context and label clarifications
I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the web....
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Extension not found error on calling an extension
I have setup a SIP server and connected two polycom soundstation duo devices to it. They both register line with the SIP server fine but when I try to place a call from one device to the other, ...
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Asterisk Call Center Report - Inbound and Outbound Calls
We currently use Asterisk to run our call center which handles incoming calls (both direct dial to extension and queues) as well as a bunch of outbound calls. Up until now, we've used a custom report ...
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Configuration of a Cisco IP Phone 7916 and 7975g
Hi everyone,
I would appreciate some help configuring my Cisco IP Phone 7975.
In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
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Calling out from Asterisk Server
I have an Asterisk server connected with a Legacy PBX. This setup is connected to another Asterisk Server (In another country) over the Internet with an IAX2 Trunk. Inter Office Extensions are ...
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Asterisk / Elastix Address Book and Call Recording
I am trying to build a small asterisk based PBX using Elastix. It'll be having 4 FXO (2 nos.no ISDN, normal analog POTS, 2 nos. GSM connections using a GSM Terminal) and 4 FXS (2 IP Phones and 2 ...
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FreePBX UK Dial Patterns Route Through Specific Trunk
I've read a ton of links and tutorials and for some reason cannot get my dial patterns to work correctly.
I've got Freepbx installed and working properly - no issue there.
I want to create a dial ...
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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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SIP INVITE packet has WAN address rather than call manager LAN IP
I am using SIP between two subnets (192.168.3.0/24 and 192.168.30.0/24) each connected via VPN.
I have a call server on 192.168.3.100, and two phones 192.168.30.118 (Ext. 3128) and 192.168.30.119 (...
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Asterisk & IVR & provider
I'm trying to setup an IVR or to be more specific an automated attendant using Asterisk. I don't want anything fancy besides the simple automated menu system, and instead of dialing an extension (for ...
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How to have dynamic extensions in Asterisk
I have recently swapped VoIP providers. So far so good, but I have ran into one little snag (and I'm honestly not really sure how to ask / Google this).
With my old provider I could have inside my ...
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Factory Reset Cisco SPA504G without admin password
I've been trying to factory reset some SPA50x phones, specifically the 504G but the old provider has locked out everything.
I need to reset about 50 phones for use with a new service. I've man in the ...
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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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How can I simulate a modem over a VOIP connection?
I have a hardware device I need a server to dial into at regular intervals. The problem is I no longer have a POTS line or a modem in any of development computers, and all my production servers are ...
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Dumping configuration from a Sipura 8800
Is there any way to retrieve the configuration from a Cisco Sipura 8800?
And no, I'm not talking about nonsense like saving the HTML pages. I'm talking about saving the proper .xml configurations so ...
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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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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 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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What are some potential uses of Asterisk (PBX) for a "power user"
So I read a lot of good things about Asterisk. I am not however looking to run a call center or small business setup. I am still interested what potential uses it has for me as a "power user" and ...
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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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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 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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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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