Displaying 20 results from an estimated 5000 matches similar to: "Problems w/Asterisk Realtime + MySQL + SIP"
2010 Mar 26
2
need help on setup rtp directly between 2 sip clients
Hi all
my asterisk server, 2 sip client softphones are the same LAN
asterisk ip address : 192.168.1.5
sip client 1 : 192.168.1.4
sip client 2 : 192.168.1.2
asterisk starts ok with sip
setup the sip.conf
[test]
type=friend
username=test
secret=1000
host=dynamic
context=cucku
directmedia=yes
directrtpsetup=yes
[1000]
type=friend
username=1000
secret=1000
host=dynamic
context=cucku
2016 Feb 15
2
Asterisk 13.6.0/The simplest TCP configuration does not work
Thanks for the mighty quick response, Joshua!
I am using Zoiper on Linux softclient:
REGISTER sip:<ipAddr>;transport=TCP SIP/2.0
Changed the port back to 5060.
On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote:
> Sonny Rajagopalan wrote:
>
> <snip>
>
>
> *CLI> pjsip set logger on
>> PJSIP Logging enabled
>> [Feb 15
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
2009 Nov 09
1
Call declined
Dear all,
I'm in basic setup of my network:
I try to do a call from a softphone to an other one but I got the error 603
Declined.
Below the
sip.conf:
*[gianca]
type=friend
username=gianca
secret=pwd_gianca
host=dynamic
context=tutorial*
*[giusy]
type=friend
username=giusy
secret=pwd_giusy
host=dynamic
context=tutorial*
extension.conf:
*[tutorial]
exten => 1234,1,Dial(SIP,gianca)*
*exten
2012 May 04
1
Broadvoice Got SIP response 503 Service Unavailable
Hi,
I'm running Asterisk 1.8.11.1 @office.
The Broadvoice service work fine with all 1.6 version and early 1.8
behind a NAT but about 2 months ago stop working.
No made changes in the firewall NAT rules. Right now I'm @home via my
Xlite softphone working fine without problems
Any suggestions or thoughts?
Alex Celi
This is the info
central*CLI> sip show peers
Name/username
2011 Jan 27
1
chan_sip bug? (Asterisk 1.4)
Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop
working after the upgrade. Here is the sip debug:
---------------------------------------------------------------------------
<--- SIP read from 208.65.xxx.xxx:5060 --->
INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0
Via: SIP/2.0/UDP
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport
Via:
2015 Jan 08
4
Asterisk 13.1.0/PJSIP peer IP address issue
Thank you for your note, Scott.
I set rewrite_contact=yes for both contacts, and I also had to do
remove_existing=yes because I had to remove the existing contact
information (max_contacts = 1 was preventing new contact information)
using pjsip
qualify demo-alice etc., after which the right IP addresses showed in pjsip
show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is
2009 Mar 11
2
how to configure for incoming message-summary SUBSCRIBE
Hi!
AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE -
but how should I handle the SUBSCRIBE in the context?
thanks
klaus
SUBSCRIBE sip:u+431234567 at foobar.at:5160 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.82:39982;branch=z9hG4bK-d8754z-3116e1207913aa4e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:u+431234567 at 11.111.11.11:39982>
To:
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
Hello,
I'd appreciate your comments on the following problem I'm having, please
forgive me if this is something obvious, I've been scratching my head on
this for a while:
I have Asterisk+Kamailio setup where I'm currently testing inbound calls
from outside. I have both webrtc and sip clients, where webrtc peers are
defined according to sip.js instructions (
2005 Jul 06
2
phone comparison matrix
Hi
Is there a phone comparison matrix I could consult
I have a series of features that I would like to evaluate on the most
common phones on the market
example:
dual-ethernet
POE / direct power / both
number of lines
speed dials programmable buttons
BLF LEDS
Headset plug
conference call built in
hands free operation
display size
codecs
communication protocol (SIP, h.323)
price
availability
2004 Dec 15
1
Help with transferring a second call from a snom 190
Hello List-
I'm having a problem getting snom 190 phones to transfer a call to
another local extension.
Here is the scenario:
A call (call1) comes in from the PSTN to (exten1). (via pri, if that
matters)
Another call (call2) comes in to (exten1).
(call1) is put on hold while (call2) is answered.
(call2) is then transferred to (exten2) using the "Xfer" button on the
snom phone. This
2009 Mar 13
2
No reply to our critical packet
Hi,
I?ve installed Asterisk for use as a SIP server. I can call people, but one
strange thing happens: if I call someone with a SIP account outside my server
(for example, sip:enum-echo-test at sip.nemox.net) everything is fine, if I call
any Asterisk extension it also works, but the call gets disconnected in about
20 seconds. To be exact, audio is turned off but the SIP client still thinks
2020 Mar 23
3
Attempting to get BLF working with linphone
On 23/03/2020 18:51, Joshua C. Colp wrote:
> On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com
> <mailto:john at calva.com>> wrote:
>
>
>
> Why is asterisk giving an error 500? I can find no reason, there
> is nothing in any log.
>
>
> The sequence number is from the past. The first SUBSCRIBE is sequence
> number 22 (check the
2006 Dec 28
1
one way rtp stream (Sent alwax to 127.0.0.1)
Asterisk version 1.2.14
I use snom190 and xliteV3 as sip phones.
asterisk send the rtp stream never to the xlite softphone.
Any hits for me?
*CLI> rtp debug
RTP Debugging Enabled
-- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack
-- Called snom
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 answered
2005 Jun 01
7
SNOM 360 extension lights
I recently got a SNOM 360 and have been trying to get the extension lights to work. I can see the subscriptions with sip show subscriptions but I don't see any notifies when a call is made. I must be missing something because I've tried looking to see if anyone else has had this problem but the only solutions I've seen have been to put hints in and I have those. Any suggestions?
2016 Sep 09
2
Asterisk 13 PJSIP with Snom 710
Hi,
I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is
working fine but i cannot dial out. i don't hear anything on the phone and
asterisk CLI also does not show anything. my config is. please advice.
[2001]
type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
2004 May 30
11
New Firefly version
As Promised, I've released a new version of Firefly (ver 1.8) with IAX &
SIP support back in.
Get it from Virbiage site or here's the direct link
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
If it crashes on startup, export your Firefly tree from the registry
(current user -> software -> firefly), then delete tree from your
registry. If that fixes it, send
2010 Oct 07
2
401 Unauthorized with Snom but not with Zoiper softphone
Hello,
I'm having difficulty with registering a SIP account in a Snom 320
IP-phone. This is what sip debug tells me :
[Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42]
<--- SIP read from UDP:public_ip:58697 --->
REGISTER sip:sip.domain.tld SIP/2.0
Via: SIP/2.0/UDP 192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;rport
From: <sip:test3 at
2005 Apr 15
2
sipXphone
Maybe I just woke up too early today. I have SJPhone and X-Lite working
perfectly but I cannot for the life of me get sipXphone working properly
with Asterisk. Its probably something stupid on my part, but does anyone
have a quick setup sheet for it?
-Kerry
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2006 Mar 09
3
OT: Snom 320, displaying text on the scree n from *
try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of
registrar>"
the trick is to specify the "-O desktop" parameter + the "-H <ip of
registrar>" parameter. Sipsak fakes the host-header of the registrar so that
the Snom thinks it is coming from your Asterisk server, then lets the
message through to the