On Wednesday 17 Feb 2016, imperium broadcast wrote:> I kinda have it working with chan_sip. > > Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10;user=phone) > But it doesn't include the user=phone at the end when dialling out. > > "To: <sip:+4499999999999;phone-context=+44 at 10.10.10.10>". > > even adding > usereqphone=yes > to the sip.conf doesn't add the user=phone to the end unless I remove the > the sip uri stuff out of the dial string. > > Ideally I would like it to look like this > INVITE sip:118099;phone-context=+44 at 10.10.10.10:5060;user=phone > Or > INVITE sip: 118099 at 10.10.10.10:5060; user=phone; phone-context=+44 > > It doesn't matter which way I do it I can only include one extra parameter > and not the two (user=phone;phone-context) as Asterisk ignores the second > one.That's because in the Asterisk dialplan, a semicolon is used to denote a comment (on account of the comment mark being a valid DTMF digit). So you will have to insert a backslash before the semicolon before user=phone . -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .
Agree. All you have to do is:
Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10\;user=phone)
I am actually surprised that the dialplan reload would work without it...
On Wed, Feb 17, 2016 at 5:51 AM A J Stiles <asterisk_list at
earthshod.co.uk>
wrote:
> On Wednesday 17 Feb 2016, imperium broadcast wrote:
> > I kinda have it working with chan_sip.
> >
> > Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10;user=phone)
> > But it doesn't include the user=phone at the end when dialling
out.
> >
> > "To: <sip:+4499999999999;phone-context=+44 at
10.10.10.10>".
> >
> > even adding
> > usereqphone=yes
> > to the sip.conf doesn't add the user=phone to the end unless I
remove the
> > the sip uri stuff out of the dial string.
> >
> > Ideally I would like it to look like this
> > INVITE sip:118099;phone-context=+44 at 10.10.10.10:5060;user=phone
> > Or
> > INVITE sip: 118099 at 10.10.10.10:5060; user=phone; phone-context=+44
> >
> > It doesn't matter which way I do it I can only include one extra
> parameter
> > and not the two (user=phone;phone-context) as Asterisk ignores the
second
> > one.
>
> That's because in the Asterisk dialplan, a semicolon is used to denote
a
> comment (on account of the comment mark being a valid DTMF digit). So you
> will have to insert a backslash before the semicolon before user=phone .
>
> --
> AJS
>
> Note: Originating address only accepts e-mail from list! If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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imperium broadcast
2016-Feb-17 13:50 UTC
[asterisk-users] SIP URI set 'telephone-context='
I swear I tested it like that and it didn't work. But its working now so thanks guys for your help. On 17 February 2016 at 13:13, Trey Hilyard <kctrey at gmail.com> wrote:> Agree. All you have to do is: > > Dial(SIP/+${EXTEN}\;phone-context=+44 at 10.10.10.10\;user=phone) > > I am actually surprised that the dialplan reload would work without it... > > On Wed, Feb 17, 2016 at 5:51 AM A J Stiles <asterisk_list at earthshod.co.uk> > wrote: > >> On Wednesday 17 Feb 2016, imperium broadcast wrote: >> > I kinda 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 10.10.10.10:5060; user=phone; phone-context=+44 >> > >> > It doesn't matter which way I do it I can only include one extra >> parameter >> > and not the two (user=phone;phone-context) as Asterisk ignores the >> second >> > one. >> >> That's because in the Asterisk dialplan, a semicolon is used to denote a >> comment (on account of the comment mark being a valid DTMF digit). So >> you >> will have to insert a backslash before the semicolon before user=phone . >> >> -- >> AJS >> >> Note: Originating address only accepts e-mail from list! If replying >> off- >> list, change address to asterisk1list at earthshod dot co dot uk . >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/4f10c787/attachment.html>