Hi Bhavik,
This is sip.conf
[general]
context=public
allowguest=yes
allowoverlap=no
realm=192.168.1.151
udpbindaddr=0.0.0.0
icesupport=yes
dtmfmode=rfc2833
transport=udp,ws
srvlookup=yes
[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ;
Hi Rafael,
It's nothing to worry about -and- you might not be able to fix it. But
it's nothing to worry about.
--
Asterisk is using OPTIONS like a ping, qualify=yes. Since 403 is a
*valid* SIP reply, the remote SIP service is considered reachable.
My carrier replies with 405 Method Not
Hi Sameer,
I think you should try using public ip rather then local and latest chrome
browser.
I have also tried with same configuration and same OS with same asterisk
version and working fine for me.
On Thu, Jul 3, 2014 at 11:59 AM, Sameer Rathod sam...@hostnsoft.com wrote:
Hi Bhavik,
Hi Guys,
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
I think it is some thing related to strp
Could you please send me your configuration file?
That will be helpful for me.
On Thu, Jul 3, 2014 at 3:25 PM, bhavik patel bhavikpatel14...@gmail.com
wrote:
Hi Sameer,
I think you should try using public ip rather then local and latest chrome
I had also tried with asterisk 11.10.2
no I am getting
== Using SIP RTP CoS mark 5
[Jul 3 15:45:10] WARNING[29686][C-0001]: chan_sip.c:10509 process_sdp:
Rejecting secure audio stream without encryption details: audio 9191
UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
followed this
Please, show your dial plan and name your Asterisk version. You might be call the Dial
application with incomplete arguments.
jg
Hi Guys,
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/
--
Dear friends
After spending few days converting my app to PJSIP, today I had to roll back
the upgrade because in the SDP, the Owner section is wrong, or I
misconfigured something
This is what my client said:
That OK message from 1.1.1.1 can not be parsed by our switch due to
address
Steve Edwards wrote:
So are the quotes now a requirement?
It looks like quotes are only a requirement on strings. In the original
post, the only difference I can see, compared to the snippet of my
dialplan without quoting, is that I'm not using the != comparison.
Whereas it seems to work
On Thu, 3 Jul 2014, Andrew Colin wrote:
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/
1) Please choose a more meaningful subject. Lots of errors can be
considered strange. (Note that actually, this is a
Hello,
I know now after some testing that there is no dynamic call parking.
Also explains why you find no example when searching the internet : no
one has a working example.
I have now the following working case :
features.conf :
[general]
parkeddynamic = yes
[parkinglot_77]
findslot
Sound like chan_sip was not build.
Just a guess: check that openssl-dev is available
---
Dennis Guse
On Thu, Jul 3, 2014 at 12:01 PM, Andrew Colin and...@vsave.co.za wrote:
Hi Guys,
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c:
Hello all,
Im using a GSM gateway device for making outbound calls. GSM device is
connected to one of my SIP peer. Now am facing a lot of voice signal
problems. I checked with my vendor and there is no issues with signal and
device. Any settings in asterisk?
Thanks
Arun
--
This one is not fully related but
with asteerisk 11.9.0 and webrtc sipml5 client
I am getting this on client side
1. Failed to set remote offer sdp: Called with SDP without DTLS
fingerprint. tsk_utils.js?svn=224:128
1. tsk_utils_log_errortsk_utils.js?svn=224:128
2.
Hi,
I am using chrome version 36 and opera
with asterisk 11.9.0 and cent os
I am getting the below error
if i do call on sipml5 from blink
1. Failed to set remote offer sdp: Called with SDP without DTLS
fingerprint. tsk_utils.js?svn=224:128
1.
So SIP/2.0 403 Forbidden is a valid response for qualify purpose
Thanks Brian!!
rv
2014-07-03 5:18 GMT-04:00 Brian LaVallee b.laval...@globaltank.jp:
Hi Rafael,
It's nothing to worry about -and- you might not be able to fix it. But
it's nothing to worry about.
--
Asterisk is using
no need.
mixmonitor has a argument that is a script ran just as the recording is
finished.
we use this to move the file from ramfs to final destination.
you can use it to use sox and convert it...
On 2 July 2014 18:54, Dave Platt dpl...@radagast.org wrote:
Problem with this is client
Can you explain?
Sent from Samsung Mobile
div Original message /divdivFrom: Tiago Geada
tiago.ge...@gmail.com /divdivDate:03/07/2014 9:04 PM (GMT+02:00)
/divdivTo: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com /divdivSubject: Re:
Please don't top-post.
Please trim irrelevant posts.
From: Tiago Geada
mixmonitor has a argument that is a script ran just as the recording is
finished.
we use this to move the file from ramfs to final destination.
you can use it to use sox and convert it...
On Thu, 3 Jul 2014, andrew
On Tue, Jul 1, 2014 at 9:39 PM, binary dreamer.bin...@gmail.com wrote:
i would go for recording into wav.
then at regular intervals eg every night at 01:00 i would start a script to
convert the wav to mp3 and then delete the wav files.
it is really easy.
This method works for us too.
We
converting wav to mp3
[sub-Monitor-Init]
exten = s,1,NoOp(Monitor Init)
exten = s,n,Set(_X-SRC_CHANNEL=${CHANNEL})
exten =
s,n,Set(recMonitorFName=${STRFTIME(${EPOCH},,%Y_%m_%d)}/${STRFTIME(${EPOCH},,%Y_%m_%d_%H_%M_%S)}-${FILTER(0-9a-zA-Z,${ARG1})}-${FILTER(0-9a-zA-Z,${ARG2})})
exten =
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