Hello guys.
Have 2 external numbers that required registration on provider server,
trunk1: 73432260005@80.75.132.66
trunk2: 73432260050@80.75.132.66
Thing is I can’t figure out how to route them to different IVRs
by default Asterisk can’t match endpoint
Request from 'sip:+
Le 02/09/2014 08:47, Nick Awesome a écrit :
Hello guys.
Hi
Have 2 external numbers that required registration on provider server,
trunk1: 734322600*05*@80.75.132.66
trunk2: 734322600*50*@80.75.132.66
Thing is I can’t figure out how to route them to different IVRs
by default Asterisk
Am 02.09.2014 07:09, schrieb Bryant Zimmerman:
Hey All
We have several AGI scripts that access databases. These work well
most of the time.
The issue we are having is that on rare occasion our script must fail
to a backup database server.
When this occurs it may take up to two seconds to do
So there is no way to do that with pjsip?
On 02 Sep 2014, at 11:35, Administrator TOOTAI ad...@tootai.net wrote:
Le 02/09/2014 08:47, Nick Awesome a écrit :
Hello guys.
Hi
Have 2 external numbers that required registration on provider server,
trunk1: 734322600*05*@80.75.132.66
Le 02/09/2014 09:38, Nick Awesome a écrit :
So there is no way to do that with pjsip?
Sorry, I didn't read carefully the subject. I can't answer for pjsip. My
bad :-(
On 02 Sep 2014, at 11:35, Administrator TOOTAI ad...@tootai.net wrote:
Le 02/09/2014 08:47, Nick Awesome a écrit :
Hello,
I have a situation where a call comes in to my Asterisk server B. This
call comes from another Asterisk server A. I want to tell to this server
A why my server B hangs up.
So just before hanging up, I add a custom SIP-header :
exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten =
On 01/09/14 12:05, Marie Fischer wrote:
Well, you made me curious - wrote up a little perl script to do a
filtered report by phone number. It takes 2-3 seconds to get a
response from OSX server (Mavericks). Which sure is shorter then
doing a full sync, but still longish. Would be interesting
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be wrote:
So just before hanging up, I add a custom SIP-header :
exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten = s,n,Hangup()
SIPAddHeader only works for INVITE as far as I know.
Steve--
On 02-09-14 11:34, Steven Howes wrote:
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be wrote:
So just before hanging up, I add a custom SIP-header :
exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten = s,n,Hangup()
SIPAddHeader only works
On 2 Sep 2014, at 10:38, Jonas Kellens jonas.kell...@telenet.be wrote:
Then how can I let another Asterisk server know the custom reason of hangup ?
If it is not possible with custom SIP-header, then how ?
As far as I know that’s going to require a source change. May not be the case
with
On Tuesday 02 Sep 2014, Jonas Kellens wrote:
On 02-09-14 11:34, Steven Howes wrote:
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be wrote:
So just before hanging up, I add a custom SIP-header :
exten = s,n,SIPAddHeader(X-My-Hangup:
Nick Awesome wrote:
Hello guys.
Kia ora,
Have 2 external numbers that required registration on provider server,
trunk1: 734322600*05*@80.75.132.66
trunk2: 734322600*50*@80.75.132.66
Thing is I can’t figure out how to route them to different IVRs
by default Asterisk can’t match endpoint
Thats because I call from one to other
here’s logs where I call from mobile
--- Received SIP request (469 bytes) from UDP:80.75.132.66:5060 ---
ACK sip:s@pbx_ip_address:57408;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
Nick Awesome wrote:
Thats because I call from one to other
Then no, you can only match based on IP address. This also applies to
chan_sip. You have to send both to the same context and then within
there you can differentiate them based on the dialed number.
--
Joshua Colp
Digium, Inc. |
On Tuesday 02 Sep 2014, Nick Awesome wrote:
Hello guys.
Have 2 external numbers that required registration on provider server,
trunk1: 73432260005@80.75.132.66
trunk2: 73432260050@80.75.132.66
Thing is I can’t figure out how to route them to different IVRs
by default Asterisk can’t
Tried doing that, but
first: AGI-exten is ’s’ for some reason.
and second its not practical, for example if 80.75.132.66 wound like to
register on my * server - it will not work because I already using that IP with
different endpoint
well, its critical trouble for me, coming back to chat_sip
Nick Awesome wrote:
Tried doing that, but
first: AGI-exten is ’s’ for some reason. and second its not
practical, for example if 80.75.132.66 wound like to register on my *
server - it will not work because I already using that IP with
different endpoint
well, its critical trouble for me,
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
so now in context dialmap (agi application) AGI-agi_channel is
'SIP/10001-0005’
parsing 10001 and checking db for
Try Hangup(123) where 123 is whatever hangup cause you want to send back to
the caller. The calliing Asterisk server will get the valuse back in
HANGUPCAUSE variable.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
On 02-09-14 14:22, Eric Wieling wrote:
Try Hangup(123) where 123 is whatever hangup cause you want to send
back to the caller. The calliing Asterisk server will get the valuse
back in HANGUPCAUSE variable.
Hello,
I have tried sending Hangup(321) on Asterisk server B to Asterisk A but
321 is not a valid Asterisk hangup cause. Valid hangupcauses are 1-127 (Q.831
cause codes) See
https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent:
Nick Awesome wrote:
register = 73432260005:pass@10001
register = 73432260050:pass@10002
[10001]
type=peer
host=80.75.132.66
context=dialmap
[10002]
type=peer
host=80.75.132.66
context=dialmap
Can you provide a sip debug of calls to both of these? I'm confused how
that... works...
--
On 01-09-14 12:31, Chandran Manikandan wrote:
[snip]
I have installed Freepbx server and tried to configure sip extension.
It's working fine.
A better place for FreePBX related questions and to get help is:
http://community.freepbx.org/
Or hire their professional FreePBX support:
On 02/09/14 09:12, Lukasz Sokol wrote:
On 01/09/14 12:05, Marie Fischer wrote:
Well, you made me curious - wrote up a little perl script to do a
filtered report by phone number. It takes 2-3 seconds to get a
response from OSX server (Mavericks). Which sure is shorter then
doing a full sync,
I use in *pjsip.conf *
[sipgate1]
type=registration
transport=transport-udp
outbound_auth=sipgate1_auth
server_uri=sip:sipgate.de
client_uri=sip:555123...@sipgate.de
contact_user=*sipgatefilter* ; *goto the filter in extensions.conf*
retry_interval=60
forbidden_retry_interval=600
expiration=3600
PS all incoming calls are directed to sipgatefilter in extentions.conf
and then filtered.
You maid have to adjust the -11 in *Goto(49${gotoadr:-11},1) ... just
look at the cli output *NoOp( 49${gotoadr:-11} )
Am 02.09.2014 um 17:04 schrieb Rainer Piper:
I use in *pjsip.conf *
upps and delete the 49 in *Goto(49${gotoadr:-11},1) and *NoOp(
49${gotoadr:-11} )
*just look at the cli output*
Am 02.09.2014 um 17:25 schrieb Rainer Piper:
PS all incoming calls are directed to sipgatefilter in extentions.conf
and then filtered.
You maid have to adjust the -11 in
Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi script
On Sep 2, 2014, at 7:04 PM, Rainer Piper rainer.pi...@soho-piper.de wrote:
I use in pjsip.conf
contact_user can be anything and calling an agi is no problem
Am 02.09.2014 um 19:49 schrieb Nick Awesome:
Okay, contact_user seems like do the job. Thanks
is contact_user can be anything, or it should be same as username ?
I would like to use contact_user for transmitting trunk name into agi
so it seems Asterisk Versions does not support video I guess
On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati khalidtou...@gmail.com
wrote:
Any article that goes through this (seems to be tedious) task to add video
support and patents?
On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
username ?
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
--
_
-- Bandwidth and
contact_user in pjsip.conf has to point to the filter or to an agi in
the extentions.conf
like:
pjsip.conf
contact_user=*blablabla
extensions.conf
**exten = blablabla, 1,NoOp(*** ${PJSIP_HEADER(read,To)} ***
${CALLERID(num)} ***)
*
Am 02.09.2014 um 20:11 schrieb Rainer Piper:
contact_user
On 02-09-14 20:18, Khalid Touati wrote:
so it seems Asterisk Versions does not support video I guess
On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the
Bria app on Android and iPhone. With SELinux and the firewall
temporarily disabled I couldn't get it to work with either
As long as you are NOT transcoding video should work in Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 6:39 PM
To: asterisk-users@lists.digium.com
On 02-09-14 21:15, Eric Wieling wrote:
As long as you are NOT transcoding video should work in Asterisk.
Both apps were configured with identical (codec) settings so I don't see
how it would require transcoding. If you did get it to work I would
appreciate it if you could tell me which
Le 02/09/2014 20:18, Khalid Touati a écrit :
so it seems Asterisk Versions does not support video I guess
Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11 with
GrandStream phones (H263, H263+ and H264). Works perfectly
On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati
On 09/02/2014 03:14 PM, Administrator TOOTAI wrote:
Le 02/09/2014 20:18, Khalid Touati a écrit :
so it seems Asterisk Versions does not support video I guess
Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11
with GrandStream phones (H263, H263+ and H264). Works perfectly
A co-worker was doing video, I dislike video. The phones were Polycom VVX, The
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP
Settings / Video section we have Video: Enabled, H.263 and H.263p are the only
two video codecs enabled.
-Original Message-
From:
You might want to check if videosupport=yes in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Tuesday, September 02, 2014 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial
Don't forget videosupport=yes in sip.conf.
j
On 09/02/2014 03:52 PM, Eric Wieling wrote:
A co-worker was doing video, I dislike video. The phones were Polycom VVX, The
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP
Settings / Video section we have Video: Enabled,
On 02-09-14 22:52, Eric Wieling wrote:
A co-worker was doing video, I dislike video. The phones were Polycom VVX, The
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP
Settings / Video section we have Video: Enabled, H.263 and H.263p are the only
two video codecs
core show codecs does not show VP8 on my Asterisk 11. I don't recall why we
are not using H.264. The novelty wore off long ago and few of our staff use
video calling anymore.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi Patrick,
Thanks for your help. Let me try your advise and come back to you if need
further any assistance.
On Tue, Sep 2, 2014 at 9:41 PM, Patrick Laimbock patr...@laimbock.com
wrote:
On 01-09-14 12:31, Chandran Manikandan wrote:
[snip]
I have installed Freepbx server and tried to
Ok, thanks for an answer. That solution works.
On 02 Sep 2014, at 22:36, Rainer Piper rainer.pi...@soho-piper.de wrote:
contact_user in pjsip.conf has to point to the filter or to an agi in the
extentions.conf
like:
pjsip.conf
contact_user=blablabla
extensions.conf
exten = blablabla,
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