Displaying 20 results from an estimated 1000 matches similar to: "channel not hungup (zombie?) so call limit not reset to zero"
2007 Aug 15
1
why is nonce="584760da" used in sip packets?
Hi all,
There is a parameter called "nonce" included in every register request that
a UA sends to asterisk. I have read sip debug a lot and only found out that
the "nonce" parameter value which is used in register request was generated
by asterisk server in a previous sip response.
As you can see in the sip debug (labled in red).
<--- Transmitting (NAT) to
2010 Jul 28
2
Nat issue one way audio on IP dial
hi there,
i have posted earlier on the list but got no satisfying answer. the problem
is not big.
I have asterisk server directly connected with internet (79.80.x.x) and
clients are behind router. clients/users are registered with asterisk and
are using sipura and xlite softphone.
Now problem is that when a user calls other by dialing his IP:Port (sip
uri), call is connected fine and he can
2007 Aug 09
1
strange warning
Hi all,
I am using an asterisk as a client to connect to another asterisk server by
registering with the register string. Registration is done without any
hassel, but after sometime my asterisk loses the registration with the
server and the server starts displaying the following msgs repeatedly:
[Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth,
but based on stale nonce
2009 Nov 19
7
AXVoice Server Hacked.. accounts info leaked
AXvoice server hacked. Here are few working accounts
USE XLITE to make calls....
Registrar/Proxy
magnum.axvoice.com:9060
Free Sample account....
username=xMaxwellSmartx
secret=thanksapache
username=woodsy
type=friend
secret=haramikuttasala
username=wumingzi
type=friend
secret=kickyourass
Enjoy!
B.R
BaBa Jigger
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2014 Dec 11
2
PJSIP configuration question
Dan Cropp wrote:
> I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)?
>
> PJSIP is including the Contact for the ACK response to the OK.
> Contact:<sip:1234 at xxx.xxx.xx.xxx:5060>
>
There is no configuration option to configure this behavior. What is the
full SIP signaling?
--
Joshua
2009 Oct 23
3
SIREN14 call setup and record/playback
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk
and I'm trying to get it to accept a SIREN14 call from Polycom's softphone.
Having trouble with SDP negotiation, I want to only allow SIREN14 and
nothing else. I also want to record and playback files, any tips on what
the Record function parameters should be?
In sip.conf I have:
disallow=all
2016 Feb 11
2
inconsistency in treatment of USE.NAMES argument
Changing the vapply() behavior makes sense in principle. I analyzed
the CRAN code base using the R parser and found 143 instances of
calling vapply with USE.NAMES=FALSE. These would need to be inspected
to understand the consequences of the change.
For reference:
/AzureML/R/datasets.R:226
/BBmisc/R/toRangeStr.R:33
/DBI/R/DBDriver.R:205
/Kmisc/R/str_rev.R:37
/Matrix/R/diagMatrix.R:98
2012 Jan 09
1
video mail is not store
Hi,
I am facing an issue while testing the video mail service of Asterisk. I have two different setup on one setup client being used is Mercuro while on the other client is Android based.
On the Mercuro setup video mail is stored and retrieved properly while with Android based setup video?mail is not stored (audio is through).
Both the client?use H.264 codec with following sdp information:
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain
2006 Dec 04
2
Odd queue issue
Hi,
I have 2 systems (A and B). I have an 800 number... when someone
calls the 800 number it goes:
IAX2-->A---IAX---B--->SIP PHONE
However.. if the user calling the 800 number is a SIP user that is
registered to A it goes:
SIP--->A---IAX---B--->SIP PHONE
This is the problem... when a call comes in from the IAX2 800
provider, things work fine... however if a SIP user registered to
2016 Feb 08
2
inconsistency in treatment of USE.NAMES argument
Hi,
Both vapply() and sapply() support the 'USE.NAMES' argument. According
to the man page:
USE.NAMES: logical; if ?TRUE? and if ?X? is character, use ?X? as
?names? for the result unless it had names already.
But if 'X' has names already and 'USE.NAMES' is FALSE, it's not clear
what will happen to the names. Are they going to propagate to the
result
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
ping times are fine as well:
[root at freepbx asterisk]# ping sipgate.de
PING sipgate.de (217.10.79.9) 56(84) bytes of data.
64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57
2014 Jun 18
1
Making sense of SDP for debugging of missing audio in SIP trunk
I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the
same setup, the calling party is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to
2007 Mar 20
1
SIP/Polycom Issue, Asterisk 1.2.16, calls dropped
I've been running the 8/1/2004 Head release up until a little over a
week ago. I was forced to due to a card failure to upgrade to 1.2.16
without any advance preparation or testing (most of my connections
are via satellite to all corners of the globe with high latency).
Up until the upgrade I was running with very few issues. Since the
upgrade I have been experiencing strange issues
2019 May 10
4
Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving side.
This is the case in both directions (so it makes no difference which
peer is calling which peer).
Both webRTC SIP peers have opus and H264 codec in their peer definition :
Video
2006 Nov 09
16
Some performance questions with ZFS/NFS/DNLC at snv_48
Hello.
We''re currently using a Sun Blade1000 (2x750MHz, 1G ram, 2x160MB/s mpt
scsi buses, skge GigE network) as a NFS backend with ZFS for
distribution of free software like Debian (cdimage.debian.org,
ftp.se.debian.org) and have run into some performance issues.
We are running SX snv_48 and have run with a raidz2 with 7x300G for a
while now, just added another 7x300G raidz2 today but
2016 Jan 20
2
Incoming webrtc call succeeds in Firefox but fails in Google Chrome
I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue.
My setup is as follows:
Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659
openssl-1.0.1e-51.el7_2.2.x86_64
[root at elx4 ~]#
2014 Oct 07
1
Grandstream GXP2160 + SRTP
Hello,
I am trying to setup a Grandstream GXP2160 IP-phone with secure calling
(SRTP).
Secure signaling SSIP for registration is working great !
I follow this guide :
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
But when I try to make a call with SRTP, I get stuck. There is an
initial INVITE which is anwered with a 401. There should follow a new
INVITE with a nonce,
2006 Jan 17
2
IAX/SIP and openser problem. IAX bug?
Hello.
I am in a strange situation. I have two asterisk. Asterisk "A" makes a
call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers
it to Openser by SIP. The problem is openser printing this in the screen:
ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> .
ERROR:parse_from_header: bad from header
2015 Feb 24
2
having trouble to register cisco 7975 with pjsip
Oh god it works !
to switch cisco to upd I used config:
<transportLayerProtocol>2</transportLayerProtocol>
with udp it works well, thanks for your help :)
> On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote:
>
> If you use UDP with force_rport=no it'll work.
> If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP