Hi,
I've run into a minor snag trying to pass on CALLERID presence from one
Asterisk to another via SIP (both running 13.16.0)
I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
its own callerid values and presence. I pass on those calls to PBX_B via
SI, and I'm trying to pass on this CALLERID info to PBX_B as well.
My relevant dialplan snippet on PBX_A is:
exten =>
1,1,Dial(SIP/pbx_b/5555555555,,f(${CALLERID(all)})u(${CALLERID(pres)})))
*the u() value being dynamically taken from the channel itself.
On pbx_b, I have a simply verbose line like this:
exten => 5555555555,1,Verbose(1,Presence information :
${CALLERID(num-pres)} - ${CALLERID(name-pres)} - ${CALLERPRES()})
Here is my experience with this: whenever "prohib_not_screened"
(tested
via a cell phone with hidden caller id info) is sent in the u() value of
the Dial application, pbx_b always gets "allowed_not_screened" as
presence
state. Short version: the callerid presence seems lost on the SIP link.
The callerid info isn't, name and number are fine.
I'm clearly missing something to pass on the callerid presence state via
the SIP link, but I can't figure out what.
Any help or hint would be appreciated.
Michael
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Actually, a correction: the callerid isn't passed on properly either: on
SIP_B I get "Anonymous <anonymous>" instead of "
<514-555-1234>" that my
dial app is sending.
The exact dial command that is used, once variables are evaluated, is
this:
Dial(SIP/pbx3/55555555555,,f(""
<5145551234>)u(prohib_not_screened))"
While the log value found on the other end of the sip link are evaluated,
I get this:
Callerid name: Anonymous callerid number: number: anonymous Presence
information : allowed_not_screened - allowed_not_screened -
allowed_not_screened
Somewhere in this Dial(SIP/) command callerid info is changed. An
asterisk verbose check does not show me anything that would change
callerid info.
Mike
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike
Sent: June 14, 2017 10:18
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] CallerId presence issue
Hi,
I've run into a minor snag trying to pass on CALLERID presence from one
Asterisk to another via SIP (both running 13.16.0)
I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
its own callerid values and presence. I pass on those calls to PBX_B via
SI, and I'm trying to pass on this CALLERID info to PBX_B as well.
My relevant dialplan snippet on PBX_A is:
exten =>
1,1,Dial(SIP/pbx_b/5555555555,,f(${CALLERID(all)})u(${CALLERID(pres)})))
*the u() value being dynamically taken from the channel itself.
On pbx_b, I have a simply verbose line like this:
exten => 5555555555,1,Verbose(1,Presence information :
${CALLERID(num-pres)} - ${CALLERID(name-pres)} - ${CALLERPRES()})
Here is my experience with this: whenever "prohib_not_screened"
(tested
via a cell phone with hidden caller id info) is sent in the u() value of
the Dial application, pbx_b always gets "allowed_not_screened" as
presence
state. Short version: the callerid presence seems lost on the SIP link.
The callerid info isn't, name and number are fine.
I'm clearly missing something to pass on the callerid presence state via
the SIP link, but I can't figure out what.
Any help or hint would be appreciated.
Michael
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On Wed, Jun 14, 2017 at 10:18:19AM -0400, Mike wrote:> I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. > PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has > its own callerid values and presence. I pass on those calls to PBX_B via > SI, and I'm trying to pass on this CALLERID info to PBX_B as well. > > My relevant dialplan snippet on PBX_A is: > exten => > 1,1,Dial(SIP/pbx_b/5555555555,,f(${CALLERID(all)})u(${CALLERID(pres)})))...> I'm clearly missing something to pass on the callerid presence state via > the SIP link, but I can't figure out what.Never heard of this method, are you sure this works for SIP, sound more like for ISDN (look at packet captures). But the/a standardized method is to use the P-Asserted-Identity and Privacy headers (rfc3325). This should work if you set in the peer configs in sip.conf on both sides: sendrpid=pai trustrpid=yes Or you can do header manipulation/getting/setting manualy if desired.
Thank you - At first glance it seems to have done the trick. Mike -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Daniel Tryba Sent: June 14, 2017 10:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CallerId presence issue On Wed, Jun 14, 2017 at 10:18:19AM -0400, Mike wrote:> I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. > PBX_A gets PRI calls on a 4 port Digium card, and each call naturally > has its own callerid values and presence. I pass on those calls to > PBX_B via SI, and I'm trying to pass on this CALLERID info to PBX_B aswell.> > My relevant dialplan snippet on PBX_A is: > exten => > 1,1,Dial(SIP/pbx_b/5555555555,,f(${CALLERID(all)})u(${CALLERID(pres)}) > ))...> I'm clearly missing something to pass on the callerid presence state > via the SIP link, but I can't figure out what.Never heard of this method, are you sure this works for SIP, sound more like for ISDN (look at packet captures). But the/a standardized method is to use the P-Asserted-Identity and Privacy headers (rfc3325). This should work if you set in the peer configs in sip.conf on both sides: sendrpid=pai trustrpid=yes Or you can do header manipulation/getting/setting manualy if desired. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users