All Questions
57 questions
0
votes
2
answers
3k
views
Blind Transfer in JSSIP
I have been trying to make a blind transfer on a ongoing call.
Below is the code i have implemented:
transfersession(ext) {
this.rtcSession.refer('sip:' + ext + '@' + serveraddress);
}
Can ...
- 147
0
votes
1
answer
740
views
Javascript code to Register SIP endpoint from browser
I just want to Register a SIP or PJSIP endpoint of asterisk from browser. I know some libraries which can do this for me.
But I want to know the core code in javascript to register a SIP end point.
...
- 2,817
1
vote
1
answer
1k
views
Asterisk: WebRTC no audio
I'm having what is probably a simple configuration issue. Calls between two SIP clients (zoiper) are successful.
When I start a call between a WebRTC client (sipml5) and a SIP client (Zoiper) is the ...
- 11
0
votes
1
answer
1k
views
Error: ast_sockaddr_resolve: getaddrinfo("a783543c-1911-44c4-9ba1-52114bbdccb4.local", "(null)", ...): Name or service not known
We connect JsSip to Astersik and long time all worked perfect.
After than unexpectedly voice dissapear without any reason.
We see in astersik log next message
ast_sockaddr_resolve: getaddrinfo("...
- 157
-1
votes
2
answers
2k
views
How to set STUN servers in JsSIP 3.3.0 [closed]
I'm trying to set up a webapp using JsSIP 3.3.0 connection to a Asterisk server.
I can find some documentation regarding TURN servers in an old version (0.3.0), but apparently this feature was removed ...
- 43
5
votes
2
answers
11k
views
webRTC how to tell if there is audio
I am using WebRTC with Asterisk, and getting an error about 5% of the time where there is no audio due to an error with signaling. The simple fix is if there is no audio coming through, then stop the ...
- 2,576
1
vote
1
answer
954
views
Asterisk MessageSend to multiple devices using PJSIP
I currently have a setup using WebRTC -> Asterisk where I can call and send messages. When I make a call from A -> B all of B's registered devices get called (so if he is logged in several times).
...
- 304
0
votes
1
answer
488
views
No video in WebRTC call on smartphone(Android) via Asterisk
I had built a WebRTC system based on Asterisk and sipml5, and I could make audio calls on my smartphone(Android), but when I enables the video, the caller can get callee's video for about 5sec, and ...
6
votes
3
answers
15k
views
Asterisk gives "Strict RTP learning" message and no audio for Chrome WebRTC but works in Firefox
I've been experimenting with WebRTC with an Asterisk server (v13.18) on the same LAN as my computer. I configured the Asterisk extension 6003 to automatically answer and play a certain notorious ...
- 193k
0
votes
2
answers
825
views
No audio in sipML5 with Firefox 58
With the recent release of Firefox Version 58, I have encountered a no audio issue using sipML5, I suspect it has to do with the change they did where they completely removed mozSrcObejct and they ...
1
vote
3
answers
12k
views
Configuration of Asterisk 13 to support WebSockets/WebRTC
I have a virtual machine with debian 9. I have installed Asterisk 13.14.1 through apt-get and I have configured it to have three users two of which are sip users (Zoiper APP) and the other one webrtc ...
- 1,268
0
votes
1
answer
1k
views
How to make a mobile or web app to call to landline or cell phone number
I am going to make a mobile&web app and backend system to setup a call to landline phone number or cell phone number. Like whatsapp or hangout.
I have investigated the paid services such as ...
0
votes
2
answers
767
views
Issue on SIPML5 plugin integration on AWS with Asterisks server- 13 using WebRTC
I have faced an issue on integrating the demo of SIPML5 plugin on the Asterisks server. The Asterisks server version is "Asterisk 13.14.0". The new version of the asterisks server supports SRTP module....
- 41
1
vote
1
answer
3k
views
sipML5 - Negotiate rtcpMuxPolicy
According to this announcement:
"As of the most recent Chrome Canary build, the default RTCP multiplexing policy is "require", instead of "negotiate". This will affect the next Chrome release, M57."
...
- 1,759
2
votes
1
answer
340
views
Conditionally use STUN server
Let's assume we are using Asterisk with WebRTC enabled. When we are trying to establish a connection between two hosts on different networks we need to use
STUN server to correctly determine their ...
- 8,260
0
votes
2
answers
1k
views
WebRTC to PSTN call established but no audio
Basically i set up an asterisk server, connected to a sip provider to make calls to pstn or mobile networks. I have configured SIP to SIP properly because when i make calls from softphone e.g. Zoiper -...
- 52
3
votes
1
answer
3k
views
Asterisk 13.10 + pjsip + WebRTC - Rx buffer overflow (PJSIP_ERXOVERFLOW)
After testing pjsip for a couple of days I finally understood a bit how it works. I hoped it will help me making WebRTC calls from site.
Situation:
I can call and receive usual calls with Asterisk
...
- 21.1k
0
votes
1
answer
644
views
How to run Web-RTC on Elastix?
I would like to have a softphone on my browser. to do this, I'm following this tutorial:
http://www.neomano.com/2015/12/probando-webrtc-en-elastix-4/
because I need to create a ssl cert, I made it ...
- 17.3k
1
vote
1
answer
4k
views
webRTC Asterisk getting: "Media stream permission denied"
We are facing issue while initiating a call from webRTC using chrome "Version 50.0.2661.102 m (64-bit)" to Zoiper SIP Client.
Both users are now connected with webRTC Asterisk server.
Getting error ...
- 337
0
votes
1
answer
980
views
How to get audio from GSM modem - Not to a speaker but as a RTP stream
I have a GSM modem with a serial port and soldered points for MIC and Speaker. Through the serial port I can send AT signalling commands, send SMS and make/receive calls. I had to solder a speaker and ...
- 9,995
1
vote
1
answer
702
views
Asterisk 13.7.2 pjsip 2.4.5 unexpected BYE with SIP cause 58 while answering
I'm upgrading Asterisk 11 to 13 and testing new configuration with WebRTC enpoints. For some reason PJSIP is dropping call sending BYE to replying endpoint after receiving OK packet from that end. No ...
2
votes
0
answers
662
views
WebRTC vs Asterisk 12 (With secure sockets) : No Audio
I am creating a project based on WebRTC and Asterisk. I'm forced tu use HTTPS, WSS, SRTP & DTLS, because new browser don't support non-secure connections ... etc
Asterisk 12.8.2
SipJS 0.7.3
...
- 97
0
votes
1
answer
241
views
Webrtc and Asterisk security
We have an Asterisk server. And we want to create a mobile app with Jssip lib on Cordova. It uses WebRTC.
We will not give to the clients login data for peers, but we need a dynamical accounting. Like ...
1
vote
1
answer
3k
views
InvalidSessionDescriptionError: Invalid description, no ice-ufrag attribute
I'm trying to get asterisk 11.20.0 running with WebRTC (sip.js 0.72 which I believe is a fork of jssip), but I'm seeing the following (and the called party rings, but when the phone is answered the ...
- 701
1
vote
1
answer
143
views
sipml5 givin ns_error_unexpected in firefox 36.0.4 on two simultaneous incoming calls
I am getting ns_error_unexpected when there are two simultaneous "i_new_call" event occurs.
Scenario 1 : When two intercom devices are pressed simultaneously, i receive two "i_new_call" event, after ...
- 339
3
votes
0
answers
6k
views
No audio /// WebRTC + Asterisk + jsSIP in Local Network
I'd appreciate a lot your help with this issue. I'm running a very basic script of JS with a jsSIP User Agent that uses a local Asterisk server for making voice calls. Everything is on a private ...
- 43
0
votes
1
answer
92
views
webrtc no packets from browser
I'm trying to connect webrtc with my asterisk server. Using Asterisk 12.8 and simpl5. I hear no sound on both sides. Wireshark shows udp packets coming from asterisk but no packets from browser. What ...
- 1,313
4
votes
1
answer
6k
views
Configure Asterisk as SIP outbound proxy (as a SIP server relay)
I just installed an Asterisk and i would like to configure Asterisk as a SIP server relay.
I already have a SIP server but this one doesn't accept directly Web Sockets (wss) connections.
The purpose ...
- 9,784
0
votes
2
answers
313
views
Backend for WoIP on WebRTC, asterisk?
I want add feature on my site: receiving phone calls in browser. What is the best way to realize it (backend)? I was read asterisk is bad choise for WebRTC protocol. Do you get me advice, what me do? ...
- 1,100
0
votes
1
answer
396
views
Parameters list send to Event Listener Function in sipml5
Did any one know the parameter list(other than type and session) send to the event listener function when an event occurs.
- 6,040
0
votes
2
answers
2k
views
Can Early Media/Ringtone be heard during sipml5 call to asterisk?
When i called to my asterisk extension from an extension registered using sipml5. I can hear audio when call is connected. But cannot hear the ringtone/early media.
Can anyone have an idea to hear ...
- 6,040
1
vote
2
answers
7k
views
SIP communication with Web socket (Web RTC)
Sip (session initiation protocol) does not understand websocket so we need sip proxy which is basically a translator between sip and websocket.
i am following this architecture for sip handshaking ...
- 880
0
votes
1
answer
2k
views
Asterisk receiving code 503 with SIPML5 and MYSQL
I did make work a audio call between two browsers using asterisk 11.16 and SipML5, when my users are in sip.conf file.
Than I tried one user in sip.conf and another in MYSQL, and it works! So in ...
- 1,494
0
votes
0
answers
3k
views
WebRTC with Asterisk returns "SRTP unprotect" warning
I have this chat system that's using SIP for voice transmission (no phones, only browser to browser in the same server).
The sound goes just fine in both ends, but asterisk gives a warning twice (one ...
- 1,494
-1
votes
3
answers
7k
views
No audio in WebRTC and Asterisk
I have a strange issue with Asterisk (in this case 13.2 version) and WebRTC.
So, I have latest Asterisk 13.2, latest Crome (with Firefox - same problem) and sip.js (also tried with sipml5) and local ...
- 1,025
3
votes
1
answer
2k
views
SipML5 and Asterisk returning 488 in makeCall
Trying to make a videoaudio call with SipML5 and Asterisk13, one user in Chorme and the other Firefox, but right after "Ringing"(180) the caller receives "Not acceptable here"(488).
Asterisk messages: ...
- 1,494
2
votes
1
answer
1k
views
how can we create connection to Asterisk using SIPml5
I want to implement a WebRTC application to be able to make calls over VoIP. My client is running the SIPml5 and in the server side I have installed and confiured the asterisk.
Asterisk was tested ...
- 33
3
votes
2
answers
2k
views
Asterisk 12.6 // TURN // WebRTC // No audio
The WebRTC setup is working on local network. It has been moved to hosting and it doesn't work anymore. I'm looking into SDP but my knowledge of networks and SIP is not useful enough to perfectly ...
- 1,918
0
votes
2
answers
1k
views
How to generate sip address on browser using asterisk
I need to generate a dynamic sip address on browsers so that my asterisk server can place a call on the same sip address.
This way my web browser will become a sip client which can receive call.
I ...
- 340
2
votes
2
answers
23k
views
Websocket connection fails with asterisk 11
I am trying to configure the websocket to work with asterisk 11. But there is some issue.
The steps I have followed are:
In http.conf enabled the following
enabled=yes
bindaddr=0.0.0.0
bindport=...
- 340
0
votes
1
answer
961
views
Is SIP required for webRTC calling to legacy VTC products
I am working on a webRTC application and would like to be able to support multiple calls and be able to call from the browser to legacy VoIP or Videoconferencing systems as well as browser to browser.
...
- 21
1
vote
1
answer
4k
views
Asterisk Webrtc DTLS-SRTP policy
I work with Asterisk 12 and Webrtc ( is use sip.js) . When Call is answered by Chrome browser (caller is zoiper) , the call imediatelly hangup and shows error
[Aug 4 10:45:16] WARNING[30235][C-...
- 123
3
votes
1
answer
5k
views
Asterisk sip.js remote call
I am working with Asterisk 12 and sip.js . I am trying to call chrome browser from zoiper (android phone )
my pears are
[6004]
context=default
secret=6004
type=friend
host=dynamic
[1060] ; This ...
- 876
0
votes
1
answer
5k
views
no audio with chrome using sip.js based webrtc app and asterisk 11.11.0. Working fine with firefox and Opera
I am working on webrtc using sip.js and asterisk. My webrtc application is working fine with firefox 31 and opera 22.0.1471.70. But when i use my webrtc application with chrome (Version 37.0.2062.58 ...
1
vote
1
answer
5k
views
Call SipJs to Asterisk 12
I trying to call from SIpJs to Asterisk 12. my peer is here
[6002]
type=friend
secret=6002
host=dynamic
context=public
transport=ws
avpf=yes
icesupport=no
encryption = no
and my JsSip code is here
...
- 876
-1
votes
1
answer
1k
views
Asterisk trunk, chrome 36, issue with WebRTC
I'm trying to get Asterisk yesterday's trunk and Chrome 36 via WebRTC. The websocket connection is established and the client registers correctly, but when I try to make a call from the browser, I get ...
- 125
2
votes
2
answers
6k
views
webrtc2sip + sipml5 + asterisk no audio issue
I am having a trouble in having a sipml5 to call other sipml5 via webrtc2sip and asterisk.
I have installed configured asterisk(version 11.10.0) + webrtc2sip(latest) + sipml5(chrome version 30.0.1599....
- 1,058
0
votes
1
answer
6k
views
process_sdp: Can't provide secure audio requested in SDP offer
I cannot make a call from Chrome browser to an Asterisk machine in which WebRTC is configured. What should I do? (see details)
I am getting a warning while calling, i.e WARNING [7087] [C-00000005]: ...
- 595
0
votes
1
answer
512
views
Asterisk HOLD functionality workaround
I'm using asterisk with webrtc in chrome (SIPml5 client) and also using their webrtc2sip gateway.
My problem is that I can't send the hook-flash/flash signal to asterisk for some reason.
Is there a ...
- 721