similar to: allow third party registration/invitaion

Hi all, I'm running some performance tests over my Asterisk,to simplify the test,I want to configure Asterisk to allow several clients registered with different nubmers but using single authentication account,so that Asterisk would not check the "to" "from" head when challenging registartion & invitation,.for example,I want Asterisk to allow following registration :

Asterisk 1.4.31 IAX2/RSA BUG If "A" tries to register on "B" and the RSA key from "A" does not match the key on "inkeys" on "B", "B" do not send a "REGREJ", instead it sends a "REGAUTH" with a new "CHALLENGE", then "A" send a new "REGREQ" for this "CHALLENGE" with the same wrong

Hi all, I encounter an annoying problem using Asterisk. I 'm using SIP. I try to register an Asterisk as a SIP end user with another Asterisk. If I put both asterisk in the same local network, no problem to do it. The asterisk end user registered perfectly with the other (let's call it the registrar). Authentication is enable and works fine. The problem occurs when I put the registrar

Hello. I'm pleased to announce that the registration for the EuroBSDcon 2012 conference in Warsaw, Poland is now officially open! You can find all information about the conference at its official website: http://2012.eurobsdcon.org/ More frequent updates will be posted to the conference's Facebook page: https://www.facebook.com/pages/EuroBSDcon/171013546286700 and on Twitter:

nat is equal to yes, and server definition in ftp provisioning server is correct. i followed packets between phone and asterisk, it seems for some reason asterisk is not happy about challenge response its getting from polycom. why its not happening in the same LAN, beats me! and also NAT device is cisco SIP aware and works flawlessly with Linksys. the only close definition of issue i have found is

We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to 10.0.0.228. But sometimes another of our servers becomes listed as a SIP agent, even though the server's IP address isn't part of our sip.conf, extensions.conf, nor any other config I know of. For example in the log snippet below, the source server experienced an SDP renegotiation in the middle of a call, and seemingly as

------------------------- Call for Participation ------------------------- -------------------------- IEEE/ACM CCGrid 2014 -------------------------- 14th IEEE/ACM International Symposium on Cluster, Cloud and Grid Computing May 26-29, 2014 Chicago, IL, USA http://datasys.cs.iit.edu/events/CCGrid2014/ Upcoming Important Dates: Early Bird Registration: April 15 Registration: May 5 Room

------------------------- Call for Participation ------------------------- -------------------------- IEEE/ACM CCGrid 2014 -------------------------- 14th IEEE/ACM International Symposium on Cluster, Cloud and Grid Computing May 26-29, 2014 Chicago, IL, USA http://datasys.cs.iit.edu/events/CCGrid2014/ Upcoming Important Dates: Early Bird Registration: April 15 Registration: May 5 Room

As some of you may have noticed, the deadline to register for the 2013 LLVM Developers' Meeting has been extended to October 10th. Please register here: https://www.seeuthere.com/rsvp/invitation/invitation.asp?id=/m1312d71-3ZIHARYFRTNUA I know many are on wait lists for other events, and we hope to open up more space shortly. Thank you for your patience. -Tanya -------------- next part

The final registration deadline for the R User Conference is June 20, 2010, one week away. Later registration will not be possible on site! Conference webpage: http://www.R-project.org/useR-2010 Conference program: http://www.R-project.org/useR-2010/program.html Registration: http://www.R-project.org/useR-2010/registration/registration.html The conference is scheduled for July 21-23, 2010,

Dear R-help members, This message is to remind you that the submission deadline for abstracts for the R User Conference, useR! 2010, is one week away: Submission deadline: Monday, March 1, 2010 The deadline for early registration is also Monday, March 1, 2010. I encourage you all to submit abstracts (to give a presentation, or to present a poster). More information regarding the

Hi All, I want to use Asterisk for VoiceMail for a softswitch. I can dial in to leave voicemail and retrieve. Now there are many SIP Endpoints registered to the Softswitch. The Asterisk is sending a NOTIFY msg to the Softswitch on <ip addr>:0 Somehow Asterisk Looses the port from where the INVITE came in, this NOTIFY msg is not going out of the Asterisk, I cannot see in Ethereal.

Hi All; Asterisk version is: 1.8.5.0 But I see at the consol the following warning and really I did google but did not understand if it is bug or related to settings: [Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468) for sip_reinvite_retry for dialog 3c581fa96f2b-53yysntgjmwb in handle_response_invite But actually, we see some SNOM IP

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register => 092779077:XXXX at 85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret=XXXX fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes

I am in the process of trying to port a Windows library to Linux (maybe also to Mac, but let's not get ahead of ourselves). The existing library is essentially a facade layer to a number of third-party dlls (mostly COM-based, although one is .NET-based) over which I have no control. These third-party dlls have no UI of their own. Their purpose is to allow access to proprietary data formats.

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Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121.

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple

Hi All, I have installed Asterisk Successfully and configure a out bound trunk for another SIP server so that if I'll dial 777123 from an asterisk-registered-phone then it will dial to the phone extension(123)-registered in the third party server. But my problem is that the reverse is not happening, that is I am not able to call from Third party SIP server to Asterisk