similar to: Asterisk 11.4.0-rc1 refuses to use the TURN server

On 11-08-16 18:03, Matt Fredrickson wrote: > On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote: >> My main reason not to upgrade to Ast 13 is because I'm afraid of losing >> functionality as there are certain functions deprecated/replaced. This can >> also cause headache :-) >> >> I will do so if there is no other option.

Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, ????? ?????? wrote: > > Try delete nat from 770000wrtc settings ice should do the same > > > On Aug 11, 2016 10:00 PM, "Jonas

Hello, I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of 2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org/jira/browse/ASTERISK-21696 Currently, I'm systematically going through each Makefile in every directory in pjproject and changing the paths that exist in the pjproject 2.0 included with Asterisk, so that I can successfully build

I was wondering if anyone knows if Asterisk 12 will be supporting the OPUS codec, which is part of the WebRTC standard as the default codec. Thank you, -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.mortensen at voicecurve.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've configured TURN in rtp.conf in Asterisk 11.5. The credentials are correct because I can get Chrome to get relay candidates and attach them to the SDP, but Asterisk doesn't want to play ball. There's little documentation -- at least from what I can tell -- on getting TURN working in Asterisk, other than the samples. STUN debug is also of no help, and when I tcpdump the Asterisk

Please help. I've been searching for days, trying nearly everything I can find that seems relevant, but I can't get this working. I am able to create users, login to Windows systems joined to the SAMBA domain as those users, but filesystem ACLs on Windows Domain Member Servers do not work which I suspect is due to my IDMAP OU is empty. wbinfo -u returns "Error looking up domain

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-) I will do so if there is no other option. But still, I don't see why Ast 13 would differ so much in this case ? If ICE and NAT is working (not causing problems) why should Ast 13 bring me audio and Ast 12 don't

Trying to use gtalk: -- Executing [andy at ipkall:2] Dial("SIP/ipkall-00000000", "gtalk/andy-gtalk/+1xxxyyyzzzz at voice.google.com") in new stack [Apr 1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP client to talk to, us (partial JID) : andy-gtalk gtalk.conf [general] context=google-in ; Context to dump call into allowguest=yes stunaddr =

Having some trouble with sip behind a nat. So tried: stunaddr = numb.viagenie.ca in sip.conf. Didn't help so tried stun debug: asterisk*CLI> stun set debug on STUN Debugging Enabled STUN Packet, msg Binding Response (0101), length: 36 Found STUN Attribute Mapped Address (0001), length 8 Ignoring STUN attribute Mapped Address (0001), length 8 Found STUN Attribute Changed Address (0005),

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(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

The Asterisk Development Team has announced the release of Asterisk 11.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hi After upgrading from 1.4 to 11.4.0, I am able to receive calls and direct them to extensions via defined trunks. However, when making outgoing calls I receive the following error: -- Executing [000441111111 at default:4] Dial("SIP/fixedline-00000004", "SIP/mydevice/00441111111,60,w") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/mydevice/00441111111

Hi, I was attempting to play with LUA in Asterisk, but couldn't compile 11.4.0 stable with LUA on my test box running CentOS 5.5 64bit. I've download lua-5.1 from lua.org, and have used the following as ./configure: ./configure --prefix=/usr/local/asterisk-11.4.0 --with-lua=/usr/src/asterisk/lua-5.1/src This runs well, and creates this output: http://pastebin.com/iKW707Bt Now, when I

I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get iax to work. I've opened 4569 in the EC2 Security Group. I'm using the zoiper client. Using tcpdump I can see the zoiper packets coming in on 4569, but nothing shows on the asterisk cli. Frame 33: 79 bytes on wire (632 bits), 79 bytes captured (632 bits) on interface 0 0000 12 31 3b 12 40 84 fe ff ff ff

Hi After upgrading from 1.4 to 11.4.0, I *am* able to receive calls and direct them to extensions via defined trunks. However, when making outgoing calls I receive the following error: -- Executing [000441111111 at default:4] Dial("SIP/fixedline-00000004", "SIP/mydevice/00441111111,60,w") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/mydevice/00441111111

n active project, than a dead one. Otherwise who is going to patch vulnerab= ilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What

with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part