Hi all,
I have an issue with asterisk 13 and pjsip. I guess it is somehow
Firewall related, but I'm unsure.
A registration to Sipgate is established successfully:
==
pjsip_sipgate/sip:sipgate.de:50
Very interesting: I have another provider configured, that was not
reachable as well. I disabled the STUN-server (external STUN server),
and now the second registration works fine, BUT with the same "error"
messages (unreachable etc) as the other provider. But in contrast the
number is always r
All other things aside, this stands out immediately:
RTT: 434.393 msec
That's almost half a second round trip for a packet. I'm amazed
anything works at all. For SIP connections, mine are usually about
26ms max, anything above about 35 is bad. Looks like a serious config
issue.
Try pinging and s
ping times are fine as well:
[root@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.
Hmmm, sorry, I can't think of anything except... why do you need the
STUN server? And are you sure that all the ports in your router
definitely match the ones Asterisk thinks it's using?
Then there is always the SIP-ALG problem with some routers, which some
people have been able to overcome by swi
Thanks Jonathan for your support.
I would like to avoid TLS at the moment (in general I am a fan of
secured communication!) because the other provider is not supporting
TLS. And sipgate is just used for testing.
However I can see the following which is quite interesting:
[2016-10-15 11:20:3
hi,
let me explain in detail, what i have configured and what is happening now:
1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to
device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
- configured NAT, eve
Hi,
I don’t see any SIP ACK’s in your trace.
Is the SIP 200 OK reaching the originating caller, or being blocked on
the way through?
Asterisk will tear down the call after ~30secs of audio playing in both
directions if it doesn't receive the SIP ACK.
Regards,
Ian
On 15/10/2016 12:05, An
ok, solved the firewall issue.
A first test call worked fine. Another one now still gets disconnected
after 32s.
But in FW there are no blocked packets anymore?!
And I don't understand why the registration to the same IP and same Port
is working, but not later transmission of further SIP pack
Have you tried setting keepalive(20 seconds) on your sip.conf and on your
phones ?
From: Andre Gronwald
To: asterisk-users@lists.digium.com
Sent: Saturday, October 15, 2016 9:17 AM
Subject: Re: [asterisk-users] Registered successfully, but after a minute or
so no SIP messages anymor
ok, now it is getting weird...
actually i don't see any firewall issues, but i am not able to get a
call from outside to my sipgate account. in asterisk nothing is visible,
core set verbose is activated.
sngrep (on my asterisk server) shows me indeed the invite from sipgate!?
What I see via sn
I have a host 192.168.1.3 that wants to run SIP on 5068 (long story).
My host is 192.168.10.201.
My host needs to stay on 5060 because of all the other devices I have
connected.
I tried putting port=5068 in my SIP extension definition but that did not
work.
So I thought about using iptables to ac
Le 15/10/2016 à 18:17, Jerry Geis a écrit :
I have a host 192.168.1.3 that wants to run SIP on 5068 (long story).
My host is 192.168.10.201.
My host needs to stay on 5060 because of all the other devices I have
connected.
I tried putting port=5068 in my SIP extension definition but that did
not
You're redirecting tcp, sip defaults to udp.
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ht
Hi. Kinda new to the area and I would like some help please. I have asterisk 11 in my system and I have 10 users and 12 DIDs. One did routed to each user and 2 DIDs for faxing. Everything works fine but I do not have call transfer between extensions and feature access codes. I have read somewhere
On Sat, 15 Oct 2016, tux john wrote:
Hi. Kinda new to the area and I would like some help please. I have
asterisk 11 in my system and I have 10 users and 12 DIDs. One did routed
to each user and 2 DIDs for faxing. Everything works fine but I do not
have call transfer between extensions and fea
How can I lock a device state so it can only publish AVAILABE, BUSY, or
RINGNING? (Eg, if the device is not BUSY or RINGNING, its AVAILABLE)
I have a hint published for a fixed phone and a mobile phone. But if the
mobile phone is out of coverage, off or similar, the queue application will
consi
>
> Your correct. I forgot to mention that the other end IS using tcp.
So I have in my SIP trunk.
transport=tcp
So correct my iptables line was specifying "-p tcp"
I also set tcpenable=yes in sip.conf
Thanks.
Jerry
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I have SIP (asterisk 11.23.0) running on port 5060 just fine. udp.
I have another SIP trunk thats wants to run on port 5068 (long story).
I have enabled tcpenable=yes in sip.conf and defined port=5068 in my trunk
definition. It does not seem that anything is listening on 5068?
How can I run SIP t
Thank you for your help! Centos 7 firewall was enable.
systemctl stop firewalld
issue fixed.
Thanks,
On Thu, Oct 13, 2016 at 3:54 PM, Victor Villarreal
wrote:
> Ok.
>
> Please, note that 192.168.1.37 (I suspect) is the internal LAN address Of
> the Polycom hardphone. If this is true, then y
Thanks for the reply. I do know the security practices and I am using VoIP. The problem is that I do not know how to configure the feature access codes including transfer.
On 15/10/2016, 21:42 Steve Edwards wrote:
On Sat, 15 Oct 2016, tux john wrote:
> H
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