Displaying 20 results from an estimated 1000 matches similar to: "How to see CODEC which is in use?"
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
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
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
2007 Nov 20
0
sl75 wlan not able of being pickuped?
Hello.
I have a strange problem. Its not possible to pickup a call that was placed
with a Siemens SL75 Wlan. When this phone calls an internal number and i try
to pickup (*8) the call from my phone i get nothing. It seems i have the call
for one second or so but after that the call is being cancelled. No problems
with other phones (polycom, grandstream). Attached the complete sip debug log
2005 May 10
1
SIP transfers failing
Hullo :)
I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for
ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from
sipgate.co.uk to any other extension.
My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind
transfer, simply dial the number you want to transfer to, and press 'FWD'...
This is what
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.
2003 Jul 08
0
SIP Problem (previous post) .. information might be relevant
regarding my previous post about SIP outgoing calls
dropping with an error 481 ..
this is my output from a SIP debug.
the call dropped occurs at the end.
Asterisk is mine, Cisco-SIPGateway is the other end (remote) and not in my
control.
help :) please!!
Dave
Signal=0
Duration=250
(no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
2004 Aug 26
0
Asterisk media problem behind NAT
Hello All,
I have a media problem while using sip communicator
user agent with asterisk behind NAT.I had enabled the
debug mode in asterisk and capture the results.I have
attached the results with this mail.Can any one help
me to fix the problem?
Thanks in advance,
Partha
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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
2004 Apr 24
0
Messengers calls dropped (SIP problem?)
I have asterisk with following users;
a) zaphfc ISDN card with two channels
b) two mediatrix FXS gateways with four channels each
c) 1x CISCO 7905G
d) two notebooks with MS Messenger 4.7
Now, it seems that any combination works correctly in all combinations
except when I call from MS messenger and then call is dropped always in
25th second of the call. Any ideas what I did wrong?
here is my
2007 Jun 25
1
Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer
Hello,
I've been racking my brain over this for much of the day so I thought
the list would probably be more helpful. A few days ago I upgraded
from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working
properly.
However, on the first business day, we realized that when transferring
calls (not using call parking, using the built in transfer buttons on
a Cisco 7960) would not
2004 Apr 02
0
SIP call troubleshooting
Can someone help me what went wrong with this call?
This call was initiated from dev/ttyI0 device on my asterisk server to
mediatrix unit. Mediatrix unit user received the call and call started.
I can hear them OK but they can not hear me correctly (cut-off sound,
noise). Call was finally hunged up.
Can anyone point out if there was something wrong?
-*CLI> sip debug
SIP Debugging Enabled
2008 Jul 19
1
Not a valid SIP contact - Asterisk 1.4.21.1 & Mitel SIP phones
Hi,
I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1
Asterisk server (and a couple of previous 1.4 versions). They're
mostly happy with the combination except for this one issue.
For incoming calls only, either originating from other local SIP
phones or from a PRI, calls won't get bridged (remote party get's
hung up) if the call is answer too quickly on the
2008 Apr 11
0
problems in REFER request to a different machine
Hi everyone,
Sorry if I'm repeating the e-mail, but I'm having problems with the
list.
I'm currently trying to enable call transfer to different domains in
asterisk box (Asterisk 1.2.13 running on Debian etch). I have a
configuration that requires me to transfer call to separate domains
like ext at 10.10.10.10:5050. My calls come from a R2 channels in a
board installed in the machine.
2004 Dec 29
1
Polycomm IP500 dropping incoming calls
Hello everyone.
I can place outgoing calls no problem with my IP500 (using teliax as our
provider). Thing is, when a call comes in, 90% of the time when I pick up
the handset it drops the call immediately. I turned on SIP debug, and have
listed my extension config from sip.conf. Any help is greatly
appreciated.... sooo close.... TIA! -Ron
[3004]
type=friend
username=3004
password=XXX
2004 Sep 23
0
Duplicated INVITE in SIP session?
Hello.
I'm trying to use Asterisk in combination with SER, to make the
routing proccess to my PSTN-Gateways. I made a simple test defining some
extension in my extension.conf, when i made a call my SER (SIP) Server
forward the call to Asterisk, this proccess is ok, but when the call is
answered i see an INVITE going out from Asterisk to my SER Server, this
invite is then passed to my
2018 Aug 27
2
feeling n00b again
Retrying, falling of the list some how :-(
-------- Original Message --------
Subject: feeling n00b again
Date: 2018-08-20 09:51
From: asterisk at a-domani.nl
To: asterisk-users at lists.digium.com
Hi all,
Long time ago, I followed a Asterisk training, and both at work and at
home, was able to deploy Asterisk,
make all sorts of internal call (hard/soft voip-phones,
incoming/outgoing,
2003 Dec 21
1
iconnect / asterisk ? calls hang up
hi
i got iconnect to work, works pretty well now except calls sometimes (more often than not) hang up after a couple of minutes.. heres a bit of the debuging
Record-Route: <sip:61892142222@213.137.73.178:5060;maddr=213.137.73.176>
From: sip:61892142222@natrelay.deltathree.com;tag=3281050172-73809
To: "JUSTIN XLITE" <sip:2001@61.95.68.84>;tag=as09766a78
Call-ID: