Displaying 20 results from an estimated 70000 matches similar to: "Asterisk removes SDP from 180 with SDP"
2007 Jun 18
1
180 Ringing with SDP
We're dialing a disconnected number via Level 3's vector network, and
are receiving this. The response has SDP in it. Apparently, Level 3 is
playing early media. Asterisk doesn't seem to know what to do with SDP
in a 180 RINGING, and just plays ringing. What am I missing here? How
can Asterisk see there's SDP, early media, in the response and act
accordingly?
SIP/2.0 180
2010 Jun 11
3
no ring back 180 with SDP
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
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2009 Sep 04
1
Send 200 OK with SDP instead of 183 with SDP when ringing starts
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
2014 Sep 08
1
Asterisk removes ice lines in sdp when calling between webrtc clients
Hello,
I have a problem with a call between 2 webrtc clients. Asterisk removes the
ice-related lines from the sdp when it sends the INVITE out, and the called
webrtc client rejects the INVITE due to the missing ice lines. Both webrtc
clients are defined exactly the same way, same values in all fields except
the number of the peer.
There's probably something I've changed that causes this
2010 May 17
0
180 with SDP
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
2005 Sep 26
2
Early Media in 180 Ringing
Hello,
I have a problem with the following: When I dial a PSTN number from a
UAC, the call is made through a SIP Trunk (which has a connection to the
PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but
Asterisk forwards the 100 Ringing WITHOUT SDP:
As you can see below, the SIP message from 10.254.254.1 (the PSTN
Gateway) has SDP, while * (with 192.168.0.173) removes the SDP
2013 Aug 28
0
AST-2013-004: Remote Crash From Late Arriving SIP ACK With SDP
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
2013 Aug 28
0
AST-2013-004: Remote Crash From Late Arriving SIP ACK With SDP
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
2018 Dec 12
3
Outbound call: caller gets no ringback on session progress
Hello!
An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the
callee (-> ISP) sends a
100 Trying
183 Session Progress (*without* SDP)
Asterisk now sends to the extension:
183 Session Progress (*with* SDP)
183 Session Progress (*with* SDP) (really two times)
The callee meanwhile sends
180 Ringing (*without* SDP)
which is
2005 Mar 15
9
Asterisk Newbie
Hello all
I have been learning * from almost 1 month now. It looks really powerfull. I
have some problem trying to find previous post, or solutions to common
problems, advice to newbies etc in this mailing list. There is no a
forum-like tool to search thru the posts by keyworks for example. Please
correct me if I am wrong.
That is why I will post my questions here:
1- Transcoding: is this when
2005 Jun 16
1
How to stop Asterisk from changing the SDP?
I'm trying to set up a direct SIP connection and have Asterisk stay
out of the media stream. When I look at the INVITE messages, I see
that Asterisk is changing the Session Description Protocol in the
INVITE message it receives, and send a INVITE message with a different
SDP to the receiver. This is not what I want. Is there any way to make
Asterisk leave the SDP exactly like it is sent from
2014 Aug 22
0
Asterisk rejects sdp from webrtc client
Hello,
I was testing with sdp and something came up worth asking:
While calling from a webrtc client to another (chrome, sip.js) Asterisk
receives the following sdp and rejects it with 488 Not Acceptable. Why does
this happen, what's wrong with the sdp? The second sdp body below is
accepted instead. Both have rtp profile RTP/SAVPF, difference is that the
second one was produced by rtpengine,
2018 Oct 10
2
How to defer SDP in ACK for unit testing purposes
Hello,
I think I met a case similar to the one solved by [1] . Quoting this case :
* res_pjsip: Handle deferred SDP hold/unhold properly.
Some SIP devices indicate hold/unhold using deferred SDP reinvites. In
other words, they provide no SDP in the reinvite.
A typical transaction that starts hold might look something like this:
* Device sends reinvite with no SDP
* Asterisk
2014 Dec 16
1
Asterisk sends CANCEL to the wrong destination
Hi,
I got a weird behaviour in asterisk (original found in 1.8 but it is
still the same in 11.15.0). I have three phones communicating via
OpenSIPs with asterisk. Phone A dials 100 and asterisk calls
SIP/phone-b. Phone B accepts the call. The User on Phone B places the
call on hold, dials 200 and, while hearing the dial tone of ringing
Phone C, places the handset on hook. Phone B sends a REFER,
2005 Sep 26
1
Early Media in 100 Ringing
Hello,
I have a problem with the following: When I dial a PSTN number from a
UAC, the call is made through a SIP Trunk (which has a connection to the
PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but
Asterisk forwards the 100 Ringing WITHOUT SDP:
As you can see below, the SIP message from 10.254.254.1 (the PSTN
Gateway) has SDP, while * (with 192.168.0.173) removes the SDP
2019 Nov 21
0
AST-2019-008: Re-invite with T.38 and malformed SDP causes crash.
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
2003 Oct 23
0
WAS: Call pickup (*8) on SIP devices. Bug #116
I've attached two SIP debugs in reference to bug #116. They are from
today's CVS build.
1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the
call. After which, SIP(2) rings for about 30 seconds then stops.
2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before
the call is answered.
SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom
2023 Apr 28
1
Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.
Hi List
Asterisk 16.28.0 in use.
PJSIP in use
Two endpoints
Both using IPv6
One Endpoint on UDP, the other via TLS.
Both with:
t38_udptl=yes
;fax_detect=yes
;fax_detect_timeout=30
rtp_ipv6=yes
Both sides are T.38 capable and detect fax tone so no need for fax
detection on asterisk.
Voice calls between the two work fine.
But on a Fax call, I see this situation:
A <=> Asterisk
2015 Feb 23
2
Question about Warning message
Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console:
WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write)
We found that line in function "sip_write" inside "chan_sip.c".
In our previous version (11.2.1) we did not see those messages being printed (same verbosity level). We compared
2006 Mar 30
0
SIP: INFO before answer causes disconnect
Hi. We have an odd problem with incoming SIP calls. I have attached a
SIP debug log, with some asterisk verbosity as well, demonstrating the
problem, below.
Is this a known bug?
Vital stats:
- Asterisk 1.2.3
- Sipura SPA-841, SPA-941 phones
- Fedora core 3
The problem manifests itself with these symptoms:
- an internal SIP extension receives a call from our PRI
- the SIP phone answers the