Searching for Asterisk Tel Uri Support information? Find all needed info by using official links provided below.
https://reviewboard.asterisk.org/r/3250/
This patch is filed on behalf of Geert Van Pamel as filed against Asterisk-12 on ASTERISK-17179. It was cleaned up by me. The patch should allow incoming INVITEs with a tel: uri. An "IMS" server apparently uses it. Geert would appreciate it if this was looked at and checked in, …
http://forums.asterisk.org/viewtopic.php?t=76432
Jul 11, 2015 · Actually I found out that Asterisk 1.6.1.20 is indeed not conform to the RFC 3966 standard. I have written a patch for Asterisk 1.6.1.20 chan-sip.c to …
https://reviewboard.asterisk.org/r/3349/
Still this phone context identifier could be needed for subsequent outgoing calls (return calls, callbacks, etc.). I agree that ${SIPDOMAIN} will remain reserved for SIP invites, and is untouched for TEL URI invites. I perfectly understand that this TEL URI context has nothing to do with dialplan context.
https://stackoverflow.com/questions/812243/problem-using-tel-url-to-initiate-a-call
As @bentford said one might get miscarried because the simulator does show an alert when you try to click on a phone on the contacts app, this is just an alert that gets generated because the app checks whether or not the tel: protocol is supported on the device or not.. Adding to what he writes you might want to also add support to escape any special characters or spaces as in:
https://asteriskfaqs.org/2019/01/31/asterisk-users/tel-uri.html
, Using Asterisk 16.1.1, with PJSIP, Im asked to build a SIP trunk to a system that uses exclusively tel: uri on inbound and outbound calls. I could not find
https://github.com/lancethepants/tomatoware/issues/29
Apr 16, 2016 · @lancethepants Hi Lance,. I'm having the same problem on my asterisk system. Too early for me to want to upgrade the whole system and I don't know if I can get the telco to change 'tel:' to 'sip:'.
https://social.technet.microsoft.com/forums/lync/en-US/8faf43a9-2539-4248-b4e4-32cf09dd466d/how-to-forward-sip-call-from-asterisk-to-a-lync-ivr-or-hunt-group
Mar 20, 2011 · Support TechCenter ... You create an ivr with sip uri [email protected] and tel URI +123456789. In Asterisk you create a lync trunk point to the mediation server. You have a normalization rule defined in Lync that translates 00XXXXXXXXX to +XXXXXXXXX In Asterisk you configure the extension as 00123456789 and let it forward to Lync.
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