I'm a little confused by Freeswitch's support for the paging feature of some phones... The wiki has a page about "Intercom" that suggests that you just prefix the extension with an "8". But the way our dialplan works is that a 3 digit extension number (eg. 302) is picked up and then a tenant-specific bridge command is performed (eg. to ${sofia_contact(0095302)}) so how could that be extended to paging?
OK, so some more research reveals this example: <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^pagegroup$|^7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> Then there's a suggestion of using a loopback: <action application="bridge" data="loopback/3456/YOUR_CONTEXT/XML"/> /<extension name="rtp_multicast_page"> // <condition field="destination_number" expression="^3456$"> // <action application="set" data="absolute_codec_string=PCMU at 20i <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>"/> // <action application="set" data="jitterbuffer_msec=500:1000:20"/> // <action application="answer"/> // <action application="esf_page_group" data="234.5.6.7 23458 30"/> // </condition> /<//extension>/ This is beginning to be in the right direction for what I want. What I'm trying to do here is really aimed at callback. The user hits a web button and then Freeswitch pages his phone and then places an outbound call. "api originate user/0095302 02070601234 XML dp0095" Does the trick, by calling 0095302 and then on answer calls 02070601234. What I'm trying to do is to not require the user to lift the handset and it has been suggested that paging is the way to do it. It's not a very true use of multicast, but might it work?
Alex, What you are looking for is sip_auto_answer ( <http://wiki.freeswitch.org/wiki/Variable_sip_auto_answer> http://wiki.freeswitch.org/wiki/Variable_sip_auto_answer) that tells the phone to auto-answer the call as long as your phone supports it and is configured properly. If the variable doesnt work but your phone supports it, then it is a matter of adding a SIP header the phone understand. The SIP header is added by prepending {sip_h_...} On my Aastra phones, the SIP header is Call-Info so it should look like this: {sip_h_Call-Info=;auto-answer=0} Benoit Raymond AMI Concept Inc. Solutions VoIP Affaire / Création Web Tel: (450) 553-1231 <http://www.amiconcept.com/> http://www.amiconcept.com De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Alex Lake Envoyé : 15 mars 2013 09:31 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Paging OK, so some more research reveals this example: <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^pagegroup$|^7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> Then there's a suggestion of using a loopback: <action application="bridge" data="loopback/3456/YOUR_CONTEXT/XML"/> <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^3456$"> <action application="set" data="absolute_codec_string=PCMU at 20i <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> "/> <action application="set" data="jitterbuffer_msec=500:1000:20"/> <action application="answer"/> <action application="esf_page_group" data="234.5.6.7 23458 30"/> </condition> </extension> This is beginning to be in the right direction for what I want. What I'm trying to do here is really aimed at callback. The user hits a web button and then Freeswitch pages his phone and then places an outbound call. "api originate user/0095302 02070601234 XML dp0095" Does the trick, by calling 0095302 and then on answer calls 02070601234. What I'm trying to do is to not require the user to lift the handset and it has been suggested that paging is the way to do it. It's not a very true use of multicast, but might it work?
Keep in mind one feature of "auto-answer" that can potentially be annoying. At least on our Polycom SoundPoint IP 320 phones, if you are having a conversation and somebody calls your intercom number, then the party you were talking to is unexpectedly placed on hold. If you didn't know about the intercom feature, then you may think you just got disconnected. -- Steve
Brilliant - thanks. I may have to do a little "see also" in the places I was looking in the wiki
Alex,
What you are looking for is sip_auto_answer (http://wiki.freeswitch.org/wiki/Variable_sip_auto_answer) that tells the phone to auto-answer the call as long as your phone supports it and is configured properly.
If the variable doesn't work but your phone supports it, then it is a matter of adding a SIP header the phone understand. The SIP header is added by prepending {sip_h_...}
On my Aastra phones, the SIP header is Call-Info so it should look like this:
{sip_h_Call-Info=;auto-answer=0}
*Benoit Raymond*
AMI Concept Inc.
Solutions VoIP Affaire / Création Web
Tel: (450) 553-1231
http://www.amiconcept.com <http://www.amiconcept.com/>
*De :*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *De la part de* Alex Lake *Envoyé :* 15 mars 2013 09:31 *À :* FreeSWITCH Users Help *Objet :* Re: [Freeswitch-users] Paging
OK, so some more research reveals this example:
<extension name="rtp_multicast_page"> <condition field="destination_number" expression="^pagegroup$|^7243"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension>
Then there's a suggestion of using a loopback:
<action application="bridge" data="loopback/3456/YOUR_CONTEXT/XML"/>
/<extension name="rtp_multicast_page">/ / <condition field="destination_number" expression="^3456$">/ / <action application="set" data="absolute_codec_string=PCMU at 20i <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>"/>/ / <action application="set" data="jitterbuffer_msec=500:1000:20"/>/ / <action application="answer"/>/ / <action application="esf_page_group" data="234.5.6.7 23458 30"/>/ / </condition>/ <//extension>/
This is beginning to be in the right direction for what I want.
What I'm trying to do here is really aimed at callback. The user hits a web button and then Freeswitch pages his phone and then places an outbound call.
"api originate user/0095302 02070601234 XML dp0095"
Does the trick, by calling 0095302 and then on answer calls 02070601234.
What I'm trying to do is to not require the user to lift the handset and it has been suggested that paging is the way to do it. It's not a very true use of multicast, but might it work?
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participants (3)
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Alex Lake
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Benoit Raymond
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Steven Schoch