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meetecho / janus-gateway

Janus WebRTC Server
https://janus.conf.meetecho.com
GNU General Public License v3.0
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Attended and blind transfers #978

Closed iclems closed 4 years ago

iclems commented 7 years ago

Add support for attended and blind transfers. This is a requirement to build a web softphone.

bhakimi commented 7 years ago

yea would love to have this too!!! additionally if possible have multiple lines and conferencing too =)

iclems commented 7 years ago

I tried to be reasonable in my request :-) I suspect the SIP library used behind the scenes does support this. From what I've read, conferencing / merging would be more complicated if you consider that you already need to instantiate different plugins to make simultaneous calls.

bhakimi commented 7 years ago

what you requested can be done by passing the right sip info from JS =) its a reinvite i believe, attended transfer requires multiple lines from what i understand, and then you bridge them and walk away, but blind transfer should be fairly simple

bizzr3 commented 7 years ago

currently i have a full setup with all features including transfer. 5000 calls per day.

for transfer i used AMI and node js.

bhakimi commented 7 years ago

Would you mind sharing your code ?

On Aug 18, 2017 12:03 AM, "Mouschti Bakhtiaree" notifications@github.com wrote:

currently i have a full setup with all features including transfer. 5000 calls per day.

for transfer i used AMI and node js.

— You are receiving this because you commented. Reply to this email directly, view it on GitHub https://github.com/meetecho/janus-gateway/issues/978#issuecomment-323276666, or mute the thread https://github.com/notifications/unsubscribe-auth/ADQXbTfJDEnPBfd4q2hqQAZvfOMpd-voks5sZTc7gaJpZM4O6fO0 .

bizzr3 commented 7 years ago

@bhakimi sure give me some times, i will find a free time and i will make a repo for that.

bhakimi commented 7 years ago

Thank you very much !!!

On Aug 21, 2017 9:22 PM, "Mouschti Bakhtiaree" notifications@github.com wrote:

@bhakimi https://github.com/bhakimi sure give me some times, i will find a free time and i will make a repo for that.

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iclems commented 7 years ago

Examples as to how to best manage simultaneous calls and call merge would be amazing. I'm working on this separately.

lminiero commented 7 years ago

Add support for attended and blind transfers

You forgot to say "please"... :wink:

iclems commented 7 years ago

Oh pretty please! Sorry for the short issue created on the go. I've been working with Janus for the past week and it's really an impressive project. I'm loving it. Until now, I've mostly been using the HTTP transport combined with the SIP plugin.

lminiero commented 7 years ago

No worries, I was just kidding :slightly_smiling_face: To be fair, I'm not familiar with the feature you guys are asking for. If you can elaborate, that might help. Of course, it the same can be done via some tweaks on the SIP server as @bizzr3 suggested, even better...

andresico commented 6 years ago

In Asterisk you can do attended and blind transfers using its built-in features. So you can do transfers sending DTMF sequences with the SIP plugin.

fiveways commented 6 years ago

does the sipre plugin support attended and blind transfers ?

lminiero commented 6 years ago

No.

Il 01 ott 2017 11:36 PM, "fiveways" notifications@github.com ha scritto:

does the sipre plugin support attended and blind transfers ?

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ibc commented 5 years ago

Hi guys, instead of requesting the SIP leg of Janus to be a full featured SIP endpoint (with blind/attended transfer), just consider it a "dumb SIP trunk". All the cool SIP features should be placed in an intermediary proxy or PBX.

lminiero commented 4 years ago

Feature implemented here: #1815

I hope I'll get more feedback than the (non-existing) one I got for multiple calls. Thanks

iclems commented 4 years ago

Sounds great. Hope I can give it a try soon. Don't have a good setup right now as it's been a while I haven't used the default SIP.

lminiero commented 4 years ago

Any feedback?

jmordica commented 4 years ago

This seems to be working fine for me. Using Kamailio as SBC ---> Asterisk.

lminiero commented 4 years ago

This seems to be working fine for me. Using Kamailio as SBC ---> Asterisk.

Excellent, thanks for confirming!

Just merged in master.