Displaying 20 results from an estimated 2000 matches similar to: "Registered successfully, but after a minute or so no SIP messages anymore"
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
2016 Oct 15
2
Registered successfully, but after a minute or so no SIP messages anymore
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, everything to port 51000-55999 to device
192.168.3.99 (same ports)
other direction is totally open.
I
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
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 packets? that
doesn't sound logical to me. What do you think?
regards,
andre
2009 Apr 26
1
sipgate doesn't work with sipgate anymore
Hi,
have some problem with incoming calls from sipgate. This was working in
1.4 but in 1.6 I get a 401 Unauthorized :-(.
Sipgate has mentioned that I have to change the type to friend, but it
is already friend, so what's wrong?
Kind regards,
Michael
Here is the sip.conf:
[sipgate_out]
type=friend
nat=yes
username=1234567
fromuser=1234567
fromdomain=sipgate.de
secret=secret
host=sipgate.de
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
Hello,
a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider
The user calling me is also using Sipgate and is calling my
2013 Sep 18
2
sipgate outgoing calls
Hello
i am trying to setup sipgate gateway
i can get incoming calls fine, but when i dial in and then try to dial
out i get this in asterisk command line
-- Called 01179248615 at sipgate
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on INVITE to
'"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1'
--
2017 Feb 13
2
First SIP-registering succeeds, second doesn't
Hi all,
I have a strange issue, with a some kind complicate architecture...
A router of our internet provider is in front of another bintec rs353j
router, at which my freepbx installation is located. However, NAT etc.
seems to work fine.
BUT: Something is not working...:
When registering my sip-trunk towards my provider (3 different
providers, all behave comparable), everything works at first.
2004 Jul 30
1
SIP connections do not hang up
Hi everybody,
I have strange problem:
I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone
using Zap Channel) using sipgate to a number in public network.
When I'm hanging up before the other person picked up the phone, the line is
not closed correctly.
The phone keeps on ringing until timeout (of Sipgate I assume) and it even
costs my money, if the other person
2005 Mar 13
2
How can I eveluate trailing numbers in extensions.conf?
Checkout
http://www.voip-info.org/wiki-Asterisk+variables
I believe that should have the answer for you.
furthermore assuming that your number is always going to be 12 digits.
exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number.
Hope this helps.
Umar
On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote:
> Hi,
>
> this
2009 Oct 18
1
Call from 'sip-id' to extension 'sip-id' rejected because extension not found ?
I'm trying to setup sipgate on 1.6.1. Following the instructions on the
site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk,
I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf:
[sipgate]
type=friend
secret= ;;SIP_PASSWORD
insecure=port,invite
defaultuser= ;; SIP-ID
fromuser= ;;SIP-ID
context=sipgate_in
fromdomain=sipgate.com
host=sipgate.com
2009 Nov 22
1
transferring SIP call: no voice
I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk
B. Both are behind NAT, but port forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls from Junction and Teliax.
CLI on A looks right:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
==
2005 May 16
4
Asterisk@home 1.0 + Sipgate UK/SIP Provider
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
"8|." to place all calls with a 8 prefix tot he sipgate account the
softphones dial the number, the Asterisk
2004 Dec 23
2
Incoming calls from Sipgate go through the wrong peer
Hi,
I have a few accounts with sipgate.co.uk to get some different DiD
numbers. However, when an incoming call comes in, it seems to pick the
wrong peer from sip.conf which sends the call into the wrong context and
it fails because there is no extension in that context to match the
register.
Using the config's below, if I dial the DiD on account 2222222, it works
fine - picks peer 2222222
2009 Feb 24
2
Multiple SIPGate accounts.
Hi all,
I have two sipgate accounts (numbers), if I have both accounts register only
one will work for incoming calls (which is all i'm interested in). However
if I disable either account the other account will work perfectly. Am I
missing something obvious?
Thanks in advance,
Ray.
Excerpts from sip.conf -
[general]
8<---- SNIP! ---->8
Register => 1212121:aaaaaaaa at
2005 Feb 08
2
Asterisk and Sipgate problem...
Hello all. I'm having an odd problem getting * and sipgate to work
together. From Sipgate support I have gotten this repsonse to my query:
=====
Your Asterisk is registering incorrectly with our servers. It registers
like this: sip:s@217.XXX.XXX.XXX:5076
The "s" should be your SIP ID. Anything else is rejected. I don't know
where you can find this setting, but from our
2006 Feb 25
2
sipgate.de question
Hi,
Anyone here using sipgate.de ?
It worked for months, but for a couple of days now I'm
unable to register with them.
My account is ok, because I can login to the website.
Asterisk keeps showing me:
Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n)
I looked at the sip debug stuff, and all I
2005 Oct 10
1
Incoming SIP getting in, but not ringing.
Hi all.
Just as a quote note, can I thank everyone on this list. I find my
self finding pretty much every answer I am looking for on here. And a
big thanks to all thoughs helping us out. Mass Respect :)
Ok, I'm using a SIP provider (SipGate UK) to do my international
dialing etc, working great from extension 8 on phones. However some
more friends/contacts have started using SipGate also, and
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
2006 Oct 25
2
Call is not coming through sipgate.co.uk+Asterisk
Hi,
I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx.
I configured my Asterisk server with 0207100xxxx. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I
2007 Sep 04
1
SIPBroker vs SIPgate
All,
I've been experimenting with shortcodes for SIPgate etc. Passing calls
to SIPbroker seems a good way to go, but the message I've had back from
SIPgate is "we don't support SIPBroker"...
So whats the easiest way to support SIP <> SIP network calling?
At the moment, I've setup some local shortcodes (eg dial **777. to goto
sipgate.co.uk) based on what Gradwell