similar to: 180 Ringing with SDP

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

Asterisk receives a 180 Ringing with SDP from the called side, then it sends 180 without SDP to the calling side. We would like asterisk to sends to the calling side the same response that was received from the called side. This is Asterisk cert 13.1, is that a new behavior, is there a setting to change this ? ?

How does Asterisk (1.2) handle a 180 WITH SDP? I am seeing different behavior when a call is initiated from an Asterisk server and from an alternate point. With Asterisk, I am hearing ringing and with the other origination point, I am getting a message played on the far-end indicating to wait while the call is connected. I am wondering if the ring in the 1st (Asterisk) scenario is being played

I have a box (Genband) expecting the following: 100 trying 180 ringing with SDP Or 100 trying 183 with SDP And asterisk is sending: 100 trying 180 ringing 183 with SDP Any way to modify asterisk to send what he is expecting? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP

On 14/05/2020 08:10, John Hughes wrote: > > I am having a problem with one of my callers who is using either g729 > or alaw.  I can do alaw but not g729 so asterisk should negotiate alaw > right?  In fact from the sip debug it looks like it does, but then I > get the dreaded "channel.c:5630 set_format: Unable to find a codec > translation path: (g729) -> (alaw)"

Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called

Hello. I use libvirtd on my Gentoo development system to manage both QEMU and LXC. When 1.0.3 came out, I updated to it from 1.0.3-r2, but 1.0.4 failed to start my LXC containers. I did not research the issue at the time, so I revert to 1.0.3-r2. Today I updated to 1.0.5 and my LXC containers still fail to start. I have not changed my domain XML at all. I am looking for suggestions on

Hello list, I have a question here that may be a little bit strange for some of you. I would like to send an INVITE from Asterisk to a given client without any SDP anouncement in it. Indeed, that is pretty useful for Click to call applications for instance, where you have no way to know which codecs are supported by the client you try to reach. Moreover, you let the client decide the

I have been sending OPTIONS requests 1) programatically (my own code), 2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes in sip.conf. The trouble is I do not see anything except an ACK 200 come back from endpoints and it does not contain any SDP/codec info. . My goal is to determine audio and video codec capability in advance of a call INVITE. I notice in both 2 and 3

Asterisk Project Security Advisory - AST-2013-004 Product Asterisk Summary Remote Crash From Late Arriving SIP ACK With SDP Nature of Advisory Remote Crash Susceptibility Remote Unauthenticated Sessions Severity Major

Asterisk Project Security Advisory - AST-2016-008 Product Asterisk Summary Crash on SDP offer or answer from endpoint using Opus Nature of Advisory Remote Crash Susceptibility Remote

Asterisk Project Security Advisory - AST-2018-002 Product Asterisk Summary Crash when given an invalid SDP media format description Nature of Advisory Remote crash Susceptibility Remote

Asterisk Project Security Advisory - AST-2018-003 Product Asterisk Summary Crash with an invalid SDP fmtp attribute Nature of Advisory Remote crash Susceptibility Remote Authenticated Sessions Severity Minor

Asterisk Project Security Advisory - AST-2019-001 Product Asterisk Summary Remote crash vulnerability with SDP protocol violation Nature of Advisory Denial Of Service Susceptibility Remote

Asterisk Project Security Advisory - Product Asterisk Summary Re-invite with T.38 and malformed SDP causes crash. Nature of Advisory Remote Crash Susceptibility Remote Authenticated Sessions Severity Minor

Hello, If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip IP address. At this point RTP packets will be coming to the default asterisk IP address. For the machine with multiple interfaces this could be not the right one (not what we want). Could it be configured (in rtp.conf or in sip.conf per context) ? Thank you. Alex Zarubin --------------