Questions tagged [sip]
Session Initiation Protocol is one of the protocols used for controlling sessions (signaling) in internet telephony.
139
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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 ...
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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 ...
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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 ...
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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 ...
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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
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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 ...
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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.
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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
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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 ...
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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 ...
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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
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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
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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 ...
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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
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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
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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
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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
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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 ...
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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 ...
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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 ...
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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 ...
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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 ...
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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-...
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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
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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
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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 ...
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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
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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(...
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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
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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
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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
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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
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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
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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.
...
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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?
