Displaying 20 results from an estimated 2000 matches similar to: "tcpenable"
2014 Apr 24
1
Realtime integration: Unregistered clients showing as registered?
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In Asterisk cli sip show peers shows nothing but for example realtime load
sipusers name 660 shows the
2020 Sep 21
2
Asterisk Drop call
Hello
I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a
drop in call. It does not have a certain time, it is random. The audio
is flowing normally and the call is dropped.
Has anyone ever experienced this?
My settings changed below:
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = no
tcpbindaddr = 0.0.0.0
transport = udp, ws, wss
srvlookup = yes
directmedia = no
2020 Sep 22
3
Asterisk Drop call
Hello.
Thanks for the reply.
Yes. In the traffic analyzed, the BYE is sent by the originator of the
call, but there is no "human" hangup, but the asterisk one.
BYE is sent, received and confirmed.
I don't know how I could investigate the reason for this BYE.
Em 21/09/2020 17:12, Dovid Bender escreveu:
> Is there anything in the Asterisk logs? Which side sends the BYE? Were
2011 May 04
2
Remove "name" part of SIP From header
Relatively new to Asterisk and SIP and am trying to run a proof of
concept using Asterisk to make an outbound call through an Audiocodes
gateway via SIP using Asterisk version 1.6.1.12. The specific
requirements of the gateway in the configuration I am trying to use
specify that the Name part of the From header be blank with the outbound
number that needs to be dialed in the number field of
2017 Jun 06
5
asterisk server - no sound
hello folks,
this might be a simple question...
I just installed asterisk in a debian server.
All seems to be running fine, but the audio sent by the server.
If I have one of my registered peers call and extension (102) that plays
back audio (extension.conf and sip.conf coffee-pasted below), Asterisk
answers and prints no errors.
Its `sip show channels` prints:
Peer User/ANR Call ID
2010 Apr 23
6
RTP over TCP
Hi List,
i have to put an * between two other SIP gateways and due to some
circumstances, i have to use sip over tcp. With 1.6.2.6 this is working
fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B
(ocs) and that's about it. In the other direction however (ocs -> me ->
deverto4) the call setup is complete but there is no audio.
I can see the audio in the form of
2011 Apr 20
1
[IAX] Everyone is busy/congested at this time (1:0/0/1)
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load iax
accounts.
Settings seems fine because apparently the system works as expected.
Yesterday I tried to add another iax account in the iax.conf directly. And I
have a problem with this new account (named 444).
I can authenticate from 444 to the server, and I can receive calls from
2010 Feb 16
6
Asterisk listens on all NICs
Hello List.
I am puzzled and how asterisk listens to calls or connections from clients. When I do a netstat -nat I don't see asterisk listening on port 5060. Now, I'm testing a server with three network interfaces: two to the internet doing load balancing and the other to our LAN. I would like asterisk to only accept connections coming from our LAN but, can't find where to configure
2019 Feb 26
3
Asterisk 1.8.7.0 connectivity to Avaya SM
Hello all, I hope someone can help me with this old Asterisk version. I have to run this version because of a custom IVR written on it. Porting it would take much too long and we'd have to hire a consultant because of all the hooks it has into Oracle databases and real-time information.
We have a brand-new Avaya phone system in place and we will be cutting over to it in late March 2019.
2017 Dec 14
4
SIP trunks going to the wrong context
Hi all,
I'm trying to resolve a weird issue with SIP routing.
I have a number of SIP trunks, from a selection of providers, all of
which are registered in sip.conf:
[general]
context=default
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp
bindport=15060
srvlookup=yes
allowsubscribe=yes
2019 Feb 27
1
Asterisk 1.8.7.0 connectivity to Avaya SM
Thanks for the reply John.
About 85-90% of what this box has to do is just handle calls, but it also has options to transfer calls to the main phone system, which up to now has been another asterisk box. For example, you can hit 6 to be transferred to the Lost & Found Department.
I do have allowguest set to “yes” already, but of course I also have type=peer and the other stuff for a sip
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 (
2015 Mar 03
6
TLS, SRTP, Asterisk11 and Snom870s
CentOS-6.5 (FreePBX-2.6)
Asterisk-11.14.2 (FreePBX)
snom870-SIP 8.7.3.25.5
I am having a very difficult time attempting to get TLS and SRTP
working with Asterisk and anything else. At the moment I am trying to
get TLS functioning with our Snom870 desk-sets. And I am not having
much luck.
Since this is an extraordinarily (to me) Byzantine environemnt I am
going to ask if any of you have gotten
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 Mar 19
2
register => 2345:password@sip_proxy/1234
sip.conf.sample:
;register => 2345:password at sip_proxy/1234
;
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a
; context.
sip.conf:
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
2011 Mar 16
0
Setting up 1.6.2.9 on Debian 6.0 Squeeze
Hello.
I would need some help trying to setup Asterisk 1.6.2.9-2+squeeze1
on a Debian 6.0 system. I'd like to use the Debian packages, hence
the "strange" version number?
Since I'm new to Asterisk, I'm trying to follow "The Asterisk Book"
at http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.html
and created a VERY basic sip.conf; see
2014 Aug 12
0
Asterisk 11.11 with TCP/TLS SRTP and Grandstream gxp1450 not working
Hey there
i'm trying to get an Asterisk 11.11 with encryption working with my
Grandstream phones. But i stuck.
To avoid NAT problems i'm using IPv6
Just with TCP/TLS it's working fine. Only the SRTP funktion is not working.
Asterisk tells me
WARNING[6938]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fa10800f5a0
(len 681) to [2a02:1205::...]:37635 returned -2: Success
and also
SSL
2014 Aug 13
0
SRTP only from asterisk to extention possible
Hello,
trying to implement srtp with already working tls i somehow stuck with
srtp. If the extension has successfully registered a call from asterisk
to that extension works fine. But the other way round nothing happens.
[Aug 13 14:54:16] WARNING[31053]: chan_sip.c:3906 __sip_xmit: sip_xmit
of 0x7fc8880467e0 (len 609) to 123.456.789:36785 returned -2: Success
[Aug 13 14:54:20] NOTICE[31053]:
2020 Sep 21
0
Asterisk Drop call
Is there anything in the Asterisk logs? Which side sends the BYE? Were you
able to capture the traffic with sngrep/wireshark to see if any side
stopped sending/getting RTP? What did the other side see?
On Mon, Sep 21, 2020 at 3:22 PM Roberto <
roberto.medola at gasparimsantos.com.br> wrote:
> Hello
> I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a
> drop in
2020 Sep 22
0
Asterisk Drop call
Roberto
Check your router if ALG or similar feature is enabled. Disable and test.
Also, on SNGREP check if both parties are getting ACK correctly after RTP
starts.
*--*
*Atenciosamente,*
*Luciano Moreira**(85)99974-2750*
*__Logic Telecom*
*0800-085-7799 | (85)4042-7799 | **(11)4210-7799*
Em ter., 22 de set. de 2020 às 13:35, Roberto <
roberto.medola at gasparimsantos.com.br>