Displaying 20 results from an estimated 1100 matches similar to: "Asterisk as a client: can I get the remote SIP server to ignore rport?"
2006 Mar 30
1
'sip show users' shows NAT RFC3581
Ok, this is highly confusing.
hestia*CLI> sip show users
Username Secret Accountcode Def.Context ACL NAT
2944030 2944030 oneeighty_start No RFC3581
2944035 2944035 oneeighty_start No RFC3581
sip users (type=friend) are in sip.conf. I have nat=no
2007 Apr 25
2
No Audio with SIP to only one provider when switching servers
I have been running Asterisk for years on a machine with a public
IP. Most recently, I have been running 1.2.17, from the day it
came out, with no (noticeable) problems.
Yesterday, I switched over to a new server that is on the same
public subnet, one higher than the original server.
I built 1.2.17 from source on that machine (as I did on the old
server). My firewall on the new machine is
2007 Apr 16
2
sip tcp support
Hi all,
i have asterisk 1.2.17 with sip tcp support and i am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the HG3540.
But when i try to place a call from Asterisk to
HiPath, the call fails with SIP/2.0 603 Declined.
The strange thing is that the first INVITE uses tcp
and the response is a 100 TRYING, the next 7 INVITE
are using udp and the respose
2004 Nov 26
2
Uniden UIP200 -- configured, but not working?
Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP
3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the
minor detail that it doesn't work. It registers fine with Asterisk, but
when I copied my Grandstream's sip.conf info and plugged in the Uniden
stuff, no dice. Any ideas?
Thanks...
-Ken
unidencom.txt:
OverwriteLocalSettings
2006 Mar 29
1
Realtime Users/Peers/Friends - Ick
I've been going in circles for a few weeks now with Realtime SIP.
My extconfig.conf has:
sipusers => mysql,dbname,ast_sip_users
sippeers => mysql,dbname,ast_sip_users
When I do a 'sip show peers' I see all my phones. When I do a 'sip show users' I only see a few of them. I can't work out why this is the case. They are also coming up with NAT as
2009 Jul 14
1
Polycom Spectralink 8002 WiFi Phones
Has anyone played with this phone? i cant seem to get it to work
properly, i manged to get it registered and can make calls from it, but
i havent been able to make it receive calls. Weird thing its that if you
make a call from it and while you are on that call you dial its number
does calls go thru in second line, but as soon as you terminate both
calls it wont recieve any calls again.
Heres
2004 Jul 18
4
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......
2010 Nov 06
2
One way voice with Asterisk
Let me explain:
When I dial into Asterisk ( I have a SIP trunk - which I need to make
sure is not faulty), I only get one-way voice communication.
The calling party, from the SIP trunk hears nothing - the extension
rings on the Asterisk server (you can see it in the CLI and hear it at
the computer), and the softphone rings
However, when you answer the SIP softphone , you can only hear the
2009 Jul 15
4
DEVICE_STATE() and Asterisk 1.6.0.10
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
I must be missing something here but I can't figure out why I can't get
DEVICE_STATE() to give me anything other than "NOT_INUSE".
I have two extensions: 6666 and 6668. I used 6668 to make a call to
yet another phone, so I know that it's busy. I then use 6666 to call
6668 and in the dialplan have a noop to see what
2011 Apr 18
2
Registrations stops after 403 FORBIDDEN
Hello list,
I have in sip.conf :
/maxexpiry=60 ; Maximum allowed time of incoming
registrations
; and subscriptions (seconds)
minexpiry=60 ; Minimum length of
registrations/subscriptions (default 60)
defaultexpiry=120 ; Default length of incoming/outgoing
registration
;-----------------------------------------
2006 Mar 28
2
NATted phones transferring calls - BUG0003710
I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107.
IP addresses have been changed to protect the innocent.
It appears this related to bug 3710. It's unclear from the bug if the problem has been fixed or not. If it hasn't, then this seems pretty serious and would I guess affect any NAT-ted phones ability to transfer calls.
Here's the REFER that the phone
2012 Aug 30
0
Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, 10.7.1-digiumphones Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of Asterisk 1.8.11-cert7, 1.8.15.1,
2006 Apr 12
1
SIP call hangup from asterisk CLI
Hi,
We are using Vicidial and sometime even when agent disconnects, outgoing
call originated by dialer is still active. Since call was initiated by
dialer and then bought into meetme conference of agent and we can't corelate
this call to any agent channel.
When agents are dialing, channels doesn't show calls
vicidial2*CLI> show channels
Channel Location
2005 Jan 05
5
Asterisk with MySQL
You are reading the instructions for the STABLE 1.0 version of asterisk and
are using the CVS version.
Goto the wiki and read the instructions for RealTime.
-Matthew
----- Original Message -----
From: "Muhammad Rizwan Khan" <rizwan@advcomm.net>
To: <Asterisk-Dev@lists.digium.com>
Sent: Wednesday, January 05, 2005 12:42 PM
Subject: [Asterisk-Dev] Asterisk with MySQL
>
2005 Aug 31
0
canreinvite=no being ignored?
Am I reading the data below incorrectly, or does it appear that even
though I have the directive canreinvite=no set for the two asterisk
boxes, they are trying to do a reinvite (which fails) anyway?
Is this expected behaviour in this situation? If so, how can I prevent
this?
---- Lots of output ----
Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A)
has a sip ua (2608)
2012 Sep 20
1
chan_motif, xmpp, jabber, jingle
Hi all,
For one of my inverstigations it looks like i'm back to "square one"
I'm trying to accept an incoming xmpp call and forward it conditionally
to a sip, isdn, or voicemail.
No google is involved as i use a local xmpp server (ejabberd)
I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but
some suggested me to have a look at asterisk11,so i did...
I
2006 Apr 20
0
Re: Asterisk-Users Digest, Vol 21, Issue 113
Hi List!!
Thanks for the colaboration, especially to Richard Cavanna who gave me the
necessary support.
I followed your indications and the comunication was better for the test
users. The warning indication is no jumping anymore and the voice is not
delayed. This is my sip.conf:
[general]
context=default
;allowguest=no
;realm=mydomain.tld
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
2011 Jan 10
0
No subject
do not know why. Anybody has a clue what could be wrong ? Is this a
bug ?
[I rebooted asterisk, and now it works.]
Regards
Axelle.
Logs of failed registration:
> sip show users
Username Secret Accountcode
Def.Context ACL NAT
IMSI208011234567890 sip-local
No RFC3581
IMSI208302141472352
sip-external No
2020 Nov 11
0
[cfe-dev] Running LLVMHello Pass from Clang(-cl)
Björn,
Can you try adding -fno-integrated-cc1 to the command-line, see if you get more infos?
De : cfe-dev <cfe-dev-bounces at lists.llvm.org> De la part de Eric Astor via cfe-dev
Envoyé : November 11, 2020 8:39 AM
À : Gaier, Bjoern <Bjoern.Gaier at horiba.com>
Cc : Clang Dev <cfe-dev at lists.llvm.org>
Objet : Re: [cfe-dev] Running LLVMHello Pass from Clang(-cl)
The
2013 Oct 21
3
Asterisk-12 issue after successful installation
Hi Team,
I have installed asterisk-12 Beta but when I try to asterisk start then get
below issue.
*[root at cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r
asterisk: error while loading shared libraries: libjansson.so.4: cannot
open shared object file: No such file or directory
[root at cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#*
--
Thanks and regards
Virendra Bhati
+91-9718500594