similar to: How to test Websocket support in SIP in Asterisk trunk?

Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc

OK, I need a bit of help here. I'm configuring a new Asterisk 11 system and I accidentally let my firewall rules drop for a day or so. When I logged in today, I found messages like the ones below on my asterisk console. Obviously somebody was trying to take advantage of my carelessness. So can someone explain what would cause these types of messages to show up on my console? I understand

Hi, While compiling 1.8.8.1, I met the following error: [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

Hi All, I am trying to configure webRTC phone example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support . I have asterisk 11.9.0 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout I copied sample code into web root directory and example loaded successfully and also able to register 2 extensions. I

Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct

2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own server or did you use Live demo (

Hi everyone, I'm currently trying to play a little with WebRTC using sipml5 client and Asterisk trunk (370821) It seems the the WebRTC implementation for Asterisk 11 is already available in the trunk? Am I right? http://lists.digium.com/pipermail/asterisk-dev/2012-July/055940.html I'm having trouble to even register to my Asterisk server using sipml5 client. The only reference to

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

Hi, my log is full of errors from this mobile user: -- Registered SIP '0010106' at 212.93.97.135:7759 [2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804 handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms / 10000ms) [2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("212.93.97.135:7759", "7759", ...): Name

Hello list, before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel. Thank you Regards

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

Hello! We recently upgraded one of our customers from 1.4.44 to 1.8.15-cert1. We have several other customers running both versions. The customer in question does not use us as their provider as they?re located in a different country. When they make outgoing calls, there is a 3 second delay between answering the call and the call being established. When debugging this, I found that Asterisk

I am trying to run the CyberMegaPhone demo to see the WebRTC Video Conference demonstration from AstriDevCon 2017 I have been able to make WebRTC work on this same box with SIPML5 demo but not the CMP2K. When I attempt to access the https://myip:8089/cmp2k I am prompted for the unsecure web. I enable unsecure web. (Using the asterisk local certificate generation from the SIPML5 demo). After

Hello all, mmm a bit embarrassing about not having a clue as to why we're getting this error on make of 1.8.3 [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o

Hello All, I am using Asterisk 12 and sipml5 as front-end and when i call from one to another the call will ring on other end but when i allow the camera access call will terminated automatically. I have attached the logs of Asterisk, if some one will get something useful Please reply on the same. Thanks and Regards, Anant == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5

Funny thing happening with an NT Client: it sees all the files in a public Samba share, but when I make any change (like renaming a file or dropping files into it) the change doesn't show up in the window until I press F5 (Refresh). Is it some config in Samba whichs defaults to "lazy" behavior which I'm failing to see? Thanx 4All, Juan --

Hi all, Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works. My network setup by the way: I am working from a cable modem, I created the test setup at digital ocean. From my laptop I also have a direct VPN connection to the

Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: "chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side