Displaying 20 results from an estimated 300 matches similar to: "SIP connections do not hang up"
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
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
2004 Jun 30
0
asterisk: problems with connecting to a (german) sip provider
hello !
My problem is:
Astriks should create a connection to other members using a german Sip
provider (www.sipgate.de).
there are no problems with connections to:
o Sip- Accounts
o national phone numbers
o mobile phone numbers
but connections to international phone numbers DO NOT WORK (see the attached
protokoll).
The connection to international phone numbers does work when I directly use
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.
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
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
2002 Aug 10
2
ps2pdf "Print command" script trouble
hi
2.2.5, Sol 2.8
I have a perl script that I've written to convert a ps file to pdf file.
I want the user to be able to setup a
postscript printer in WNT/2k and convert ps files to pdf by printing to
a [ps2pdf] printer share.
[ps2pdf]
comment = PS to PDF file in Home Directory
path = /usr/spool/public
guest ok = Yes
print ok = Yes
browseable = Yes
printer = ps2pdf
2004 Oct 04
1
How to see CODEC which is in use?
How can I see which codec is in use during conversation. I can see (for
example) which codecs are negotiated before SIP connection, but I don't
know which is chosen:
12 headers, 12 lines
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Peer audio RTP is at port 217.10.79.30:15666
Found description format GSM
Found description format iLBC
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
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
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:
<Registration/ServerURI..............................> <Auth..........>
<Status.......>
==========================================================================================
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'
--
2006 Jan 11
1
[suse-isdn] Major Problems UTStarcom F1000 registering -- pls help
Hi there,
I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with
my asterisk server. I already changed the name of the user to
"anonymous" since it looks like the phone sends that name. The WiFi
Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200
What is it that I am missing? Any help very much appreciated!!!
The error message I get is:
Jan
2009 Mar 18
3
Manager API Originate CDR Problem, all is NO ANSWER
hi, all
asterisk 1.4.24 , zaptel 1.4.10.1 , E1
Manager API Action :
Action: Originate
Channel: ZAP/G1/8888888
Callerid: 12345678
Context: callout
Exten: s
Priority: 1
extensions.conf
[callout]
exten => s,1,Answer()
exten => s,n,Wait(10)
exten => s,n,Hangup()
when the phone 8888888 pick up , it will come to callout context, after hangup, one cdr generate, but the
2004 Jun 27
5
Optipoint 400 Standard Sip
Hi everybody,
I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk.
It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers.
The Optipoint shows "no Server..." (Registrar?) in Display.
Sip debug shows no unusual (to me) Messages.
Sip show
2007 Mar 07
1
sporadic problem using ldap authentication
Hi,
recently some users started complaining about sporadic authentication
failures. When examining the logs I saw the following:
Mar 7 03:20:24 gollum dovecot: auth(default): file auth-request.c: line
472 (auth_request_lookup_credentials_callback): assertion failed:
(request->state == AUTH_REQUEST_STATE_PASSDB)
Mar 7 03:20:24 gollum dovecot: auth(default): Raw backtrace:
dovecot-auth
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
2005 Aug 06
0
SIP rejecting calls?
Hi,
I have researched more into the problem of my Asterisk set-up not answering
calls.
The following error was shown on the CLI, can anyone explain what the
problem causing Asterisk to not answer the SIP calls be?
Information: I have an Asterisk box on a home LAN, behind a D-Link
router/firewall connected to a cable modem. The 82.x.x.x is the IP for my
cable modem. 192.168.0.101 is my
2010 Feb 18
0
ISDN phone not ringing. ISDN PBX not answering?!
Hi,
I've set up an Asterisk as voip gatway:
VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx.
Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset.
I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode.
The msn is set at the dect phone/base station
2006 Jun 09
0
Why are sip-channels too lagged?
Hello,
I am getting lots of messages as the ones attached below. Is this a
problem anybody can explain. (My internet connection is NOT slow or
instable... thus I don't get it.) Maybe does this result from incorrect
registration?
Cheers,
Arik
----- sip.conf ------
[general]
qualify=no
srvlookup=yes
canreinvite=yes
register => xxxxx:xxxx@sipgate.de/xxxx
[sipgate]
type=friend
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