search for: usereqphon

...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 canreinvit...

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? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140409/65727670/attachment.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://w...

...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 [pbxani...

...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=...

...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 [pbxani...

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 ?...

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

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,

...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 10.1...

...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...

...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

...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...

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

...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

...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] inc...

...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&...

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

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

...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) t...