Displaying 20 results from an estimated 38 matches for "usereqphone".
2015 May 31
2
Signaling incoming call
...pbxanika/00493513333333
register => 4444444444:MYVERYSECRET at messagenet/4444444444
[pbxluca]
type=peer
defaultuser=00493511111111
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493511111111
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600
[pbxfax]
type=peer
defaultuser=00493512222222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493512222222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite...
2014 Apr 09
1
PJSIP usereqphone setting in config file
Hi everyone,
I am starting to work with PJSIP on release 12.1.0.rc3.
I used to have Asterisk 1.8 with the regular sip channel. I was using the
usereqphone settings in order to set user=phone on from and to URIs.
Is there a similar config in PJSIP?
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2008 May 01
1
http://www.asteriskdocs.org/html/apas02.html
If one of the authors is listening:
http://www.asteriskdocs.org/html/apas02.html
lists usereqphone 2 times. One of the entries should really
be useragent. And the example for usereqphone is wrong.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://ww...
2015 May 28
3
Peer is UNREACHABLE
...ET at pbxanika/00493513333333
register => 4444444444:MYSECRET at messagenet/4444444444
[pbxluca]
type=peer
defaultuser=00493511111111
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493511111111
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxfax]
type=peer
defaultuser=00493512222222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493512222222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxanik...
2015 May 28
0
Peer is UNREACHABLE
...at messagenet/4444444444
>
> [pbxluca]
> type=peer
> defaultuser=00493511111111
> secret= MYSECRET
> dtmfmode=rfc2833
> host=172.16.34.132
> context=luca_incoming
> outboundproxy=172.16.34.132
> port=5060
> fromuser=00493511111111
> fromdomain=172.16.34.132
> usereqphone=yes
> canreinvite=no
> insecure=invite
>
> [pbxfax]
> type=peer
> defaultuser=00493512222222
> secret= MYSECRET
> dtmfmode=rfc2833
> host=172.16.34.132
> context=fax_incoming
> outboundproxy=172.16.34.132
> port=5060
> fromuser=00493512222222
> fromdomain=1...
2015 May 29
0
Calling from "extern"
...t pbxanika/00493513333333
register => 4444444444:MYVERYSECRET at messagenet/4444444444
[pbxluca]
type=peer
defaultuser=00493511111111
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493511111111
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxfax]
type=peer
defaultuser=00493512222222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493512222222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxanik...
2006 Feb 23
6
username as extension
Is there a way to have extensions automatically created for
registered sip users ?
I did some investigation and found some hope in chan_sip with
relation to the somewhat undocumented usereqphone option but i may be
totally off track.
All i want to be able to do is send a call to number@ip_address where
the number is the username configured on the phone that has
registered with asterisk on ip_address.
From what I understand this should be pretty standard sip
functionality no ?
R...
2015 May 31
6
Signaling incoming call
Hi list!
Finally I got my Asterisk works with my two phones...
It was a problem on my Firewall (for the phone of my wife) and on my Dialplan
(for forwarding calls).
Now all works as expected, at least in the simulation I did with AsteriskNOW.
Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP...
Well, now I have some time to spend with "fooling"...
My phone
2015 May 28
4
Peer is UNREACHABLE
Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb:
> The phone you gave your wife is really old. Are you sure it supports SIP
> OPTIONS? Can you make a call in or out to it? If you can, it is more
> likely that it just doesn't support that and you can't use a qualify
> statement.
No, I'm not sure.
And no, I can't make any call, right now... At least,
2016 Feb 17
2
SIP URI set 'telephone-context='
...have it working with chan_sip.
>
> Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10;user=phone)
> But it doesn't include the user=phone at the end when dialling out.
>
> "To: <sip:+4499999999999;phone-context=+44 at 10.10.10.10>".
>
> even adding
> usereqphone=yes
> to the sip.conf doesn't add the user=phone to the end unless I remove the
> the sip uri stuff out of the dial string.
>
> Ideally I would like it to look like this
> INVITE sip:118099;phone-context=+44 at 10.10.10.10:5060;user=phone
> Or
> INVITE sip: 118099 at 10.10...
2020 Jun 13
0
Voice "broken" during calls
...e providers I used).
The settings for Deutsche Telekom are:
[pbxluca]
type=peer
defaultuser=<mylogin>-0001
secret= <myverysecretpassword>
dtmfmode=rfc2833
host=tel.t-online.de
context=luca_incoming
outboundproxy=tel.t-online.de
port=5060
fromuser=0351xxxxxxx
fromdomain=tel.t-online.de
usereqphone=yes
canreinvite=yes
insecure=port,invite
nat=force_rport,comedia
qualify=yes
qualifyfreq=600
disallow=all
allow=alaw
allow=ulaw
and the settings for MessageNet are:
[messagenet]
type=peer
defaultuser=<mylogin>
secret=<myveryverysecretpassword>
dtmfmode=rfc2833
host=sip.messagenet.it
c...
2007 Oct 26
1
Nortel C15K <-> Asterisk
...asterisk to register with it.
Anyone have any ideas?
thanks!
register => username:passwd at 192.168.1.20
[nortel]
type=friend
fromuser=username
username=username
canreinvite=yes
secret=passwd
host= 192.168.1.20
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=rfc2833
qualify=yes
nat=no
usereqphone=yes
context=from-nortel
2008 Jun 21
0
One VOIP Provider Multiple registrations <to> multiple inbound contexts ?
...assword at sip.exampleprovider.net/100 at context2
;example of me trying to get inbound call to 102 in context: context2
[peer-1]
type=peer
context=context1
secret=testpassword
username=7341112222
fromuser=7341112222
fromdomain=sip.exampleprovider.net
host=sip.exampleprovider.net
;register=yes
usereqphone=yes
insecure=very
nat=yes
canreinvite=yes
;call-limit=5
[peer-2]
type=peer
context=context2
secret=testpassword
username=7341113333
fromuser=7341113333
fromdomain=sip.exampleprovider.net
host=sip.exampleprovider.net
;register=yes
usereqphone=yes
insecure=very
nat=yes
canreinvite=yes
;call-limit=...
2020 Jun 13
4
Voice "broken" during calls
Hi!
I have a Asterisk installation to manage my phones at home (provider is
Deutsche Telekom).
It works, but very often the voice is "broken"...
Yesterday during a call it was very difficult to understand what my
partner sayd...
It can NOT be a problem of other downloads/uploads, since in that moment
there were no ones...
I already had the problem in the past, solved it enabling the
2006 Oct 24
1
Basic Conf
...expirey=8600
useragent=Asterisk_Eut
localnet=192.168.1.1/255.255.255.0
[out_eutelia]
type=peer
context=eutelia
secret=xxxxxx
username=<username>
fromuser=<username>
fromdomain=voip.eutelia.it
host=voip.eutelia.it
nat=yes
dtmfmode=inband
usereqphone=yes
[datile3]
type=friend
host=dynamic
username=datile3
context=eutelia
permit=192.168.1.3
default=192.168.1.3
context=eutelia
Is there anybody around who understand where is the problem?
daniel
2010 May 07
0
Issues with remote call setup
...domain=yes
directmedia=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
;entry for phones
[100]
type=friend
context=phones
host=dynamic
[102]
type=friend
context=phones
host=dynamic
;entry for users
[user1]
type=friend
context=on_this_system
secret=password
regcontext=on_this_system
regexten=100
usereqphone=no
host=dynamic
nat=no
[user3]
type=friend
context=on_that_system
secret=password
regcontext=on_that_system
regexten=102
usereqphone=no
host=dynamic
nat=no
And the following entries have been made in extensions.conf:
[general]
static=no
writeprotect=no
autofallthrough=no
[default]
[phones]
incl...
2005 Oct 16
1
iax invtation problem
...61.187.150
the configuration of my sip file was like this:
sip.conf
[callshop]
type=peer
host=213.61.187.150
username=XXXXX
secret=XXXXX
but when ive added these lines on my sip.conf file:
[callshop]
type=peer
host=213.61.187.150
username=XXXXX
secret=XXXXX
fromuser=XXXX
usereqphone=yes
canreinvite=no
nat=yes
insecure=invite
insecure=port
port=5060
disallow=all
allow=g729
it worked and the invitation problem was solved,but
when i have tried to send calls by iax to the same
voip provider the call failed like this:
dial 0017046872001@calls
-- Executing Dial("OSS/dsp&q...
2016 Feb 16
2
SIP URI set 'telephone-context='
Thanks for the reply Trey, should of said I'm using chan_sip.
Regards
Mick
On 16 Feb 2016 18:03, "Trey Hilyard" <kctrey at gmail.com> wrote:
> Are you using res_pjsip or chan_sip?
>
> For PJSIP, it's as easy as passing the parameters to the Dial. For example:
> Dial(PJSIP/${ARG1}\;phone-context=mydomain.com at pjsippeer,60)
>
> I am pretty sure it was
2020 Jun 22
2
Voice broken during calls (again...)
Would you mind repeating the test with canreinvite=no set for all you
phones and mobile phones?
What is your upload bitrate? Is it guaranteed?
I would try also to test the PMTU:
Try:
ping -M do -s 2000 ${ip address of the sip server}
You should receive icmp asking for lowering the packet size.
The LTE phones could have lower MTU and thus overcome PMTU problem.
Marek
2020-06-22 21:48
2007 Nov 28
1
Asterisk <-> Nortel Phone Switch
...time, but still no
go. Can't do calling in either direction. Anyone have any ideas?
Thanks!
Shawn
[nortel]
host=10.0.0.10
insecure=very
type=peer
qualify=no
canreinvite=no
dtmfmode=rfc2833
fromuser=user
username=user
secret=123
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=rfc2833
usereqphone=yes
context=from-nortel
asterisk*CLI> sip debug ip 10.0.0.10
SIP Debugging Enabled for IP: 10.0.0.10
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
Audio is at 192.168.10.2 port 17492
Adding codec 0x4 (ulaw) to...