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Attended and blind transfers #978
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yea would love to have this too!!! additionally if possible have multiple lines and conferencing too =) |
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. |
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 |
currently i have a full setup with all features including transfer. for transfer i used AMI and node js. |
Would you mind sharing your code ?
…On Aug 18, 2017 12:03 AM, "Mouschti Bakhtiaree" ***@***.***> wrote:
currently i have a full setup with all features including transfer.
5000 calls per day.
for transfer i used AMI and node js.
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@bhakimi sure give me some times, i will find a free time and i will make a repo for that. |
Thank you very much !!!
…On Aug 21, 2017 9:22 PM, "Mouschti Bakhtiaree" ***@***.***> 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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Examples as to how to best manage simultaneous calls and call merge would be amazing. I'm working on this separately. |
You forgot to say "please"... 😉 |
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. |
No worries, I was just kidding 🙂 |
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. |
does the sipre plugin support attended and blind transfers ? |
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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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. |
Feature implemented here: #1815 I hope I'll get more feedback than the (non-existing) one I got for multiple calls. |
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. |
Any feedback? |
This seems to be working fine for me. Using Kamailio as SBC ---> Asterisk. |
Excellent, thanks for confirming! Just merged in master. |
Add support for attended and blind transfers. This is a requirement to build a web softphone.
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