Questions tagged [sip]

Session Initiation Protocol is one of the protocols used for controlling sessions (signaling) in internet telephony.

Filter by
Sorted by
Tagged with
0
votes
0answers
16 views

Can you use a regular SIP line as a trunk

I have an asterisk box that can route calls internally. I am trying to determine if I can connect the asterisk box to standard SIP line like AT&T's Digital Voice. I understand that I would only ...
0
votes
0answers
76 views

How to make asterisk call file as user account or extension

I am trying to figure out why the callerid works when I call with twinkle through asterisk but not when making a call file. The callerid with the call file is always unknown. Twinkle account is setup ...
1
vote
0answers
59 views

Using Zyxel Modem As A Simple Voip ATA Device

I have a Zyxel P-2602HWL modem/router which I want to use as a Voip ATA device only. In other words, I already have a router that I am using and I only want to use the Voip functionality of the Zyxel ...
0
votes
0answers
27 views

Weird network issue

We are facing a technical issue with networking, which I'm not able to comprehend. We are using Linphone Client to connect to x.x.x.125(Freeswitch box) via x.x.x.101 i.e .101 being a SIP proxy ...
0
votes
0answers
18 views

How can I direct a chat from a flask based chat app to a live agent

I have a flask based chat app. I want to add a feature to it so that the chat can be transferred to a live person in the middle of the textual chat (say if the app can't answer a question). The live ...
0
votes
1answer
267 views

How to configure instant messaging in freepbx 14 & Asterisk 13

i have a trouble to configure instant messaging using freepbx 14 and asterisk 13 , i want that two sip clients can send and receive messages using their soft phones on smartphone and desktop , can ...
0
votes
0answers
22 views

SIP trunk configuration on FreePBX

How we can configure SIP trunk on Asterisk and FreePBX to re-route the incoming call from mobile/landline over internet.
0
votes
0answers
33 views

Query Regarding routing a normal phone call through PC

I have an urgent requirement of routing a normal phone call to PC. There are additional requirements such as – Call will come from landline/mobile number be to a fixed number, and this call should be ...
0
votes
0answers
37 views

Is there a way to monitor a GrandStream registration status?

I have a provider locked GrandStream device and I had some issues where the device would unregister itself during the night and I would lose my phone line. I cannot afford to not notice when the VoIP ...
0
votes
1answer
114 views

Asterisk how to call by names instead of numbers

I haven't found anywhere clear sample how can I setup alphanumeric alias for internal numbers. For example I am calling now to 504@ip. How can I make it possible to reach 504 number by john@ip by ...
0
votes
1answer
48 views

How does a router know which device handles calls?

How does a home router know which connected device is meant to handle a VOIP call? Is there a specific broadcast that is sent out? My home setup uses an analog telephone adapter (ATA) box in bridge ...
2
votes
0answers
346 views

PJSIP PJSUA python no sound but call OK on Raspberry

I have an Asterisk server on one Rasp and on another I would like to make a SIP client. To do this, I installed PJSIP, and I found this little code from their tutorial to make a phone call. I ...
0
votes
1answer
132 views

How to block a pcap pumped by tcpreplay from one server by iptables on the other?

I am new to stack exchange. What I am trying is: I am pumping traffic from one server by the following command: tcpreplay -i ens3 ~/daniel/sipdump.pcap from server 1 And I am capturing the traffic in ...
0
votes
1answer
238 views

Asterisk sip.conf in MySQL database

I'm able to include a phone inside a database for realtime usage. So, this code (from /etc/asterisk/sip.conf): [phone] type=friend username=phone secret=12345 host=dynamic disallow=all allow=g729 ...
3
votes
1answer
1k views

How to setup Asterisk NAT on docker for Windows

I need to setup a docker container running an Asterisk PBX. The problem is that I receive the wrong ip in sdp replies. In fact Asterisk is using the container internal IP. The second problem is that ...
1
vote
0answers
130 views

On a PBX (Asterisk) determine which phones are connected

I am using Asterisk as my PBX. On my sip.conf I used to have this option enabled: [general] ... qualify=yes more info about qualify can be found in HERE. Thanks to that option I was able to tell if ...
0
votes
0answers
2k views

How do you fix “No Trust List Installed” error on a Cisco 7942G VoIP Phone?

I have a 7942G. I'm trying to connect it to Asterisk. Looking at Settings -> Status -> Status Messages, I see this: Error Updating Locale SEP############.cnf.xml No Trust List Installed I'm only ...
0
votes
2answers
579 views

Converting a P.O.T.S line to VoIP and simultaneously ringing a extension over VPN

Anyone with knowledge please help. I have analog phone service + internet at SITE 1. I only have internet service at SITE 2. When the phone rings at SITE 1 I would like the PC or an attached phone ...
0
votes
1answer
74 views

Ekiga under Windows: registered but calls fail

I have a VoIP account with sub-accounts. Sub-accounts are assigned 3-digit extensions and I can call between sub-accounts using regular desktop phones connected to my ATA. Now I am trying to set up ...
1
vote
1answer
1k views

twinkle configuration for Sangoma

I'm able to run AsteriskNOW virtualized on my local system. Also, I'm using Anveo on two different SIP hard phones. Anveo works fine, indicating that the network handles SIP fine. Now, how do I ...
0
votes
2answers
547 views

How to setup a voip server connected to a ISP voice line

My home fibre bradband ISP provides me with a Huawei HG8240 optical network terminal. This has one ethernet port and one phone port. The phone port is active, but unused at the moment. The ethernet ...
2
votes
1answer
2k views

Asterisk initiate call through sip notify message

I have a Yealink ip phone on my desk and I want to remote control it so that I can place the calls from my computer instead than by typing the numbers directly on the phone. This link explains how to ...
0
votes
1answer
194 views

How will a call get routed with incorrect LRN in SIP request?

I was reading about frauds in telecom. I read about an LRN fraud in which the fraudster would change the LRN in the SIP invite to a cheaper destination. How will the call get routed with an incorrect ...
1
vote
1answer
52 views

IPSec doesn't hide IAX2 informations

I am trying to setup a secure truncked IAX2 connexion with IPSec between routers: But when I make sniffing attack with wireshark, all call's informations are showen ! IPsec is well configured and ...
0
votes
1answer
564 views

(ringinuse disabled) (Invalid) has taken no calls yet

I had a freepbx server in place. I setup a queue with static agents and ring all strategy. Now, suddenly no body in the queue will ring once the call routed to the queue. I check the queue and found ...
0
votes
1answer
219 views

Using a VoIP router behind a DrayTek 2760 router

I'm struggling with the following problem: I use a DrayTek Vigor 2760 as router 1 to access internet by using a VDSL line. The DrayTek behaves as a Modem and DHCP server with IP 192.168.1.1. My ...
0
votes
1answer
256 views

Keeps detecting SELinux as enabled even though I disabled it (When trying to install FreePBX)

I was following this guide to install FreePBX on CentOS 6.9 (minimaslist), everything worked, until the last step to install FreePBX: ./install -n from /usr/src/freepbx, I keep getting the follwing ...
1
vote
1answer
207 views

Default port for calls with Asterisk

I have AsteriskNOW 11.9.0 with FreePBX 2.11.0.43. I moved Asterisk to a non-standard port number, but now I can call extensions only if I specify the port number explicitly in a callee's number, for ...
1
vote
0answers
161 views

SIP based door entry and phone system

For a residential home, we've allowed for a door entry system, where we'd like to the following: Push button at door entry panel calls all intercom stations (handsets) - audio only From any intercom ...
0
votes
1answer
47 views

jitsi account “name” or “default identiy” sticks and cannot be removed

how do I change or remove where it shows a "name" above offline to the right of the silhouette of the user and below the files tools help menu? I removed all accounts. I'm only using Jitsi for SIP ...
1
vote
0answers
157 views

Testing my GNU Ring installation

I've installed GNU Ring on my system, but, frankly, none of my friends use it (none that I know of anyway), so I don't have anyone to call. I'm planning to invite someone to call me up on Ring, but I ...
1
vote
1answer
586 views

PPPoE connection type with VoIP service

Two VoIP ATA adapters is used for VoIP service, adapters are in different offices, but within same VoIP service networks. The PPPoE connection type is used in ATA for VoIP service: is it possible to ...
1
vote
1answer
2k views

Reroute all internet traffic through a tinc tunnel

I want to route all internet traffic (including udp for sip) through a tinc tunnel (https://www.tinc-vpn.org/). To set up the tinc tunnel is quite straight forward. After starting the tinc daemon on ...
1
vote
0answers
273 views

Unable to accept incoming calls on Polycom VVX 410

I'm using a Cisco DPC3848 cable modem running in bridge mode. I have a microtik CRS125-24G-1S-2HnD router behind it. I can make outbound calls on it no problem, but if I try to make an inbound call ...
1
vote
1answer
890 views

Why SIP-ALG in routers and modems?

We currently are using a NetSapiens based VoIP system and have extreme problems with a lot of consumer grade routers and modems due to SIP-ALG not being disabled or not being able disable. So if SIP-...
1
vote
0answers
940 views

Installing Python module sip 4.19.1 on Windows 7

I've got a problem with installing SIP 4.19.1. What I have done by now: - installed Anaconda (for Python 3.6) with compiled SIP 4.18, - pip install sip "Successfully installed sip-4.19.1" - and ...
0
votes
1answer
781 views

Kill a listening port for a specific IP

I have a SIP server with multiple network interfaces. (virtual) When I execute the command netstat -nlput it shows the all listening IP and ports as below. root@DUO-CALLSERVER004:/etc/init.d# netstat ...
2
votes
1answer
116 views

Setting my ISP's modem to let me call via my landline when I'm not home, using its internal hidden SIP server

My ISP offers to his VDSL users the ability to use any smartphone as a cordless home phone, using an official app. They provide no documentation or source code, but someone managed to de-obfuscate ...
1
vote
1answer
324 views

Set sip/tel URL/LSHandler back to OS X Mac Continuity/Wifi Calling instead of Skype

I recently installed skype for business and it hijacked my Mac's ability to make calls via my iPhone (Continuity) or Wifi Calling. I wanted to know how to set it back, but I figured it out before i ...
0
votes
0answers
256 views

Linphone on Windows Server 2012, with multiple users

Im going to explain everything first. We have Windows server 2012, where our users are connecting to (lets say around 60-70 per server). After Linphohe installation and first start of the application (...
2
votes
1answer
898 views

Linphone SIP INVITE providing wrong IP on multi-NIC system

I'm trying to use Linphone on a Windows 7 machine that is connected to multiple networks (via 2 different NICs). It works fine with a single one connected, but when both are connected I only get one ...
2
votes
1answer
1k views

Incomming local SIP calls. What's the cause?

I have Linphone installed on Linux. There is default identity set to me@192.168.1.100. I keep getting incomming calls from numbers like 100@192.168.1.100 or 7070@192.168.1.100. What is the cause of ...
0
votes
1answer
281 views

Connect Normal calls to Asterisk Server [closed]

I am trying to develop a phone system just like vitelity. I am using asterisk as backend. Main problem is to connect pstn call to my asterisk server. that is when i dial a number from my phone(...
1
vote
0answers
865 views

SIP ALG detector

I am trying to create a detector for SIP ALG. I really just need to know what the BEST method of testing this is so I can write a program to test whether SIP ALG is enabled or disabled. Thanks
0
votes
1answer
1k views

is this failed SIP call due to firewall or NAT?

I'm trying to find out why this call failed, with: <--- SIP read from UDP:63.247.69.226:5060 ---> SIP/2.0 100 Trying and then Everyone is busy/congested at this time (1:0/0/1) full debug ...
0
votes
3answers
299 views

how do I provision a hard phone for getonsip?

In provisioning a Cisco SPA 942 hardphone, branded Linksys, how do I finish the setup for getonsip (or onsip)? SIP Address: foo@getonsip.com Username: foo Domain: getonsip.com SIP Password: ...
1
vote
1answer
309 views

SIP server name Defaults to $VERSION for Cisco SPA 942 hard phone

In reading the fine manual for a Cisco SPA 942 IP phone, branded as Linksys: what is, in particular, meant by: SIP Server Name Server header used in responses to inbound responses. Defaults to $...
0
votes
1answer
418 views

How to make international phone calls from my computer' through my SIP server?

I'm currently working for a scientific company, and I'm the only programmer here, and the company wants me to use one of their servers to setup a SIP server, and somehow be able to make calls to ...
0
votes
1answer
87 views

SIP forwarding via DNS. Possible?

Let's say I have an address sip:john.doe@example.com that I would like to use for Internet calls, but my SIP provider is Callcentric and I need to route such calls to sip:17775551234@callcentric.com. ...
1
vote
1answer
1k views

Which folders are affected by System Integrity Protection?

I have trouble writing files with sudo to /Library under OSX v10.11.3. It seems the folder is affected by SIP. Which folders fall under SIP?