Displaying 20 results from an estimated 10000 matches similar to: "strange problem with asterisk in media proxy mode"
2005 Aug 30
0
canreinvite = yes with PAP2
Has anyone made this work? For me everything is fine until I switch
canreinvite form no to yes. What happens is that asterisk hangs on
"attempting native bridge" ... from what I understand "attempting native
bridge" means that the RTP is routed through asterisk (just without any
codec translation) But it shouldn't do that ... right? ... canreinvite is
set to yes ...
2007 Jul 12
0
No subject
help me in another issue related also to registering
asterisk with another softswitch:
A) If nat=yes, then I have to set canreinvite=no to be
able to register, correct?
B) In case of using firefly softphone, how it possible
to set it to have nat=yes (at the firefly it self and
not at the sip user configuration section)? As most of
the sip endpoint give an option to set nat=yes and so
on, how it
2010 Nov 19
0
help with annoying warning message: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm)
Hi all,
i have a little problem to understand this warning message, it's annoying
and it cause a lot of spurious in the log files.
Im working with this scenario:
a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are
always routed to this.
a list of sip UAs that potentially can use any codec apart g729/g723.
I setup the asterisk to do as mediaproxy so directmedia=no and
2015 May 28
0
Peer is UNREACHABLE
I think your phone may be trying to register with the username '1234',
while your sip configuration is expecting 'luca'. Can you try changing
your phone registration credentials to use 'luca'? Can you give us a sip
transcript when you try to place a call from it?
On 15-05-28 05:09 PM, Luca Bertoncello wrote:
> Darryl Moore <darryl at moores.ca> schrieb:
>
2007 Jun 22
1
Does Early Media have to be Ulaw?
I have this in sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
progressinband=yes
[19256002182]
type=friend
username=19256002182
callerid="Test hone 1" <+19256002182>
host=dynamic
canreinvite=no
secret=password
context=test
disallow=all
allow=g729
[level3]
type=peer
host=xxx.yyy.16.99
context=default
2010 Dec 22
0
Asterisk 1.8.1.1 Multiple Parking Lots
Asterisk Version: 1.8.1.1
Problem: Multiple Parking Lots
Issue: Not redirecting to the right parking lot. Always uses the first
parking lot from "parkedcalls show" or "features show"
Asterisk Working Version: 1.6.1
Steps Taken:
In features.conf added:
[parkinglot_test]
context => parkedcalls-test
parkext => 700
parkpos => 701-710
parkingtime => 120
findslot
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
2015 May 29
0
Calling from "extern"
Hi list!
Finally I got my wife's phone working in my Asterisk.
Unfortunately I have some problems, too...
Current situation:
- AsteriskNOW with 4 Accounts (00493511111111, 00493512222222,
00493513333333, 5678). This is "for test" and it will be replaced by "the
real world", when I got my Asterisk to work...
- A second Asterisk (Ubuntu-PBX) on another VM, logging in
2006 Jun 05
0
Multiple SIP Accounts Between Asterisk Boxes (Unreachable)
Name/username Host Dyn Nat ACL Port Status
2011/2011 10.1.1.10 5071 UNREACHABLE
2010/2010 10.1.1.10 5070 UNREACHABLE
2009/2009 10.1.1.10 5069 UNREACHABLE
2008/2008 10.1.1.10 5068 UNREACHABLE
2007/2007
2005 Aug 31
0
canreinvite=no being ignored?
Am I reading the data below incorrectly, or does it appear that even
though I have the directive canreinvite=no set for the two asterisk
boxes, they are trying to do a reinvite (which fails) anyway?
Is this expected behaviour in this situation? If so, how can I prevent
this?
---- Lots of output ----
Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A)
has a sip ua (2608)
2007 May 22
3
Dial out issues.
Dear all.
I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received.
Problem:
Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work
2013 May 21
1
Failed to authenticate device "Ext 110"
I'm having a strange problem recently with a Yealink SIP-T28P phone
connected to Asterisk 11.4.0 via openvpn. It was working fine for months,
and now when I dial anything from the phone, it shows "Forbidden", and the
Asterisk console shows:
[May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189
handle_request_invite: Failed to authenticate device "Ext 110" <
2004 Jan 26
0
canreinvite and codec negotations... and NAT
I've gotten canreinvite=yes to work with a sip device behind NAT!! You
*MUST* port forward the SIPPort to in your gateway router to your phone.
This is a MUST.
Okay, now on to my problem.. I have people who will be using ulaw, and I
have people who will be using g729.. I want to set it up so that canreinivte
will work.. I have a single cisco gateway..
Asterisks isn't handling the
2004 Jan 29
0
canreinvite and codec negotations...
Okay, now on to my problem.. I have people who will be using ulaw, and I
have people who will be using g729.. I want to set it up so that canreinivte
will work.. I have a single cisco gateway..
Asterisks isn't handling the negotation between the 2 devices very well..
For example..
[gateway]
type=friend
host=1.2.3.4
canreinvite=yes
qualify=200
dtmfmode=rfc2833
context=default
disallow=all
2007 Jul 13
1
Media Proxy Mode in Asterik: SIP and
Dear Alex;
Thanks for your kindly reply.
Please explain for me what do u mean exactly in "a la"
in the following sentence u wrote it below?
" in SIP, this can be done via
"re-INVITEs" a la the canreinvite= option for SIP
peers in sip.conf"
Another thing, do u mean that it is easier (better) if
we need H.323 endpoint to talk with SIP endpoint then
we use full
2005 Mar 25
2
MGCP issue
Hello List,
I'm trying to setup MGCP channel with a Centile Media Hub box. My
Centile box has 4 ports and I got no dial tone. Can somebody help with
this isuue?
This is my mgcp.conf and extensions.conf
Thanks
Daniel.
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 192.168.11.20
disallow=all
allow=g729
allow=alaw
allow=ulaw
[192.168.11.200]
context=MGCP
2004 Jul 26
1
Nat...again....
This has probably been answered somewhere, but I'm stumped.
I have two Zap channels (FXS and FXO), both working fine. I
can call from Zap/1 to Zap/2 and reverse.
I've also configured SIP channels, both inside and outside of my
firewall. Inside can call outside, and outside can call inside.
Also, both inside and outside can make and receive calls to/from
Zap/1 & Zap/2.
What
2015 May 28
3
Peer is UNREACHABLE
Darryl Moore <darryl at moores.ca> schrieb:
> Ahh. Seen that before! That suggests to me that you don't have your
> sip.conf records setup right.
>
> What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register => 00493511111111:MYSECRET at pbxluca/00493511111111
register => 00493512222222:MYSECRET at pbxfax/00493512222222
register =>
2011 Jul 23
1
One way calling on asterisk to cisco call manager integration
I'm trying to integrate my Asterisk box with my call manager 8 server. I can call from the call manager to a phone on asterisk, but I can't call from a phone on asterisk to call manager. Any help would be greatly appreciated.
sip.conf
[2000]
type=friend
secret=
dtmfmode=rfc2833
host=dynamic
canreinvite=no
context=myphones
allow=ulaw
nat=yes
[2001]
type=friend
secret=
dtmfmode=rfc2833
2007 Feb 16
0
IAX vs SIP - Getting Asterisk out of the media path
If a call comes into my Asterisk server on a DiD provided by an ITSP and the
dialplan sends that call to another external number throught the same ITSP's
network, I don't want the RTP packets to pass through my server once the
call is bridged.
I have had great success getting this to work using IAX, but I have not been
able to get this to work with SIP. The call is bridged OK (media at