jami-daemon issueshttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues2023-11-12T14:17:59Zhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/826problem connecting tio SIP account on startup2023-11-12T14:17:59Ztomo90problem connecting tio SIP account on startupI have configured a simple SIP account. The application does not automatically log in to the account when it starts. However, all I need to do to connect is go into the settings and click in the name settings field, for example, and then...I have configured a simple SIP account. The application does not automatically log in to the account when it starts. However, all I need to do to connect is go into the settings and click in the name settings field, for example, and then exit the settings and the application will connect to the account immediately. It doesn't make any sense to me.
Generally speaking, it will connect to the account after editing any unrelated settings.
Furthermore, the application does not seem to save some of the settings i make. After quitting, the switches and settings are at their original values. This is not a problem with, for example, write permissions to the configuration file, because some of the settings made are preserved and some are not when the application is restarted.
You can see for yourself that on the attached video.
![vid](/uploads/e0e18d538065a0cffa977629dbfcf594/vid.mp4)Sébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/790SIP ip 2 ip issues2022-12-08T19:39:33ZSébastien BlinSIP ip 2 ip issues+ One account on 5060, one on 5061 => 5061 receives all calls even if :5060 is specified in the URI
+ ~~text messages doesn't work~~
+ IPv6 addresses not supported
+ Calling back doesn't work+ One account on 5060, one on 5061 => 5061 receives all calls even if :5060 is specified in the URI
+ ~~text messages doesn't work~~
+ IPv6 addresses not supported
+ Calling back doesn't workhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/694ffmpeg: add G711, G729 codecs (SIP)2022-07-06T00:18:18Zovariffmpeg: add G711, G729 codecs (SIP)Please add the following codecs to Jami:
* [G.711a](https://en.wikipedia.org/wiki/G.711)
* [G.711u](https://en.wikipedia.org/wiki/G.711)
* [G.729a](https://en.wikipedia.org/wiki/G.729)
What do you think?
Thank you
> MyNetFone uses th...Please add the following codecs to Jami:
* [G.711a](https://en.wikipedia.org/wiki/G.711)
* [G.711u](https://en.wikipedia.org/wiki/G.711)
* [G.729a](https://en.wikipedia.org/wiki/G.729)
What do you think?
Thank you
> MyNetFone uses the **G.711a** codec, it is the standard codec used in Australia and Europe. The **G.711u** codec is used within the US, and may be present in PBXs or UC platforms from US vendors. The **G.729a** codec is often used for conference calling services.<br>
> https://www.mynetfone.com.au/support/faq/question/477
> The patents for G.711, released in 1972, have expired, so it may be used without the need for a licence
> https://en.wikipedia.org/wiki/G.711#Licensing
> As of January 1, 2017, the patent terms of most licensed patents under the G.729 Consortium have expired, the remaining unexpired patents are usable on a royalty-free basis.<br>
> https://en.wikipedia.org/wiki/G.729.1
https://git.jami.net/savoirfairelinux/ring-daemon/issues/200Backloghttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/109SIP account TLS/SRTP configuration is old fashioned/confusing2022-11-11T16:13:35ZMaxim CournoyerSIP account TLS/SRTP configuration is old fashioned/confusingTested on: Android 20190103 (f-droid)
Recently, voip.ms (finally) added support for calls encryption using TLS for signaling and SRTP for media. I was thrilled to try it with Jami, but couldn't get it to work:
1. Add SIP account basic ...Tested on: Android 20190103 (f-droid)
Recently, voip.ms (finally) added support for calls encryption using TLS for signaling and SRTP for media. I was thrilled to try it with Jami, but couldn't get it to work:
1. Add SIP account basic detail (alias, hostname, username & password), registered OK.
2. Went to Security tab (android client), and enable TLS transport. As voip.ms is using a trusted SSL certificate, I wouldn't expect to have to do anything else, but:
a) the greyed out options below suggest that only the client certificate is going to be verified (I don't care about my cert, but I do want to authenticate the SIP server). So I checked the "Verify Server" box, and unchecked "Verify Client" and "TLS Require Client Certificate".
b) I have no idea why there's a "Server Name" field; this should at least defaults to my SIP hostname, if required?
c) There are other options which are nice for a self signed certs setup I guess, but overly complicated for the more straightforward CA signed use case. Perhaps they could be hidden under an "advanced" section?
d) I put my hostname in Server Name, just in case, and left the other options empty/default.
Expected result: SIP account is re-registered using TLS.
Actual result: TLS seems to fail silently, option is reverted to disabled when visiting the security menu.Sébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/185Call Forward* { on Busy ; No Answer ; Always } for Jami accounts2024-02-09T20:01:33ZovariCall Forward* { on Busy ; No Answer ; Always } for Jami accounts[Improving SIP Support for Ring](https://ring.cx/en/news#improving-sip-support-for-ring) shared news about the `Call Transfer` feature.
Please add the following:
* `Call Forward on Busy`: Forward all calls if you are on the phone to a n...[Improving SIP Support for Ring](https://ring.cx/en/news#improving-sip-support-for-ring) shared news about the `Call Transfer` feature.
Please add the following:
* `Call Forward on Busy`: Forward all calls if you are on the phone to a number of your choice. Standard call rates may apply.
* `Call Forward No Answer`: Forward all unanswered calls to a number of your choice. Standard call rates may apply.
* `Call Forward Always`: Forward all calls to your SIP (VoIP) phone number to another number of your choice. Standard call rates may apply.
![image](/uploads/4df8d0a3947eff09dd5747925ac97034/image.png)
The idea for this feature (which also shows icons in the image above) came from https://www.exetel.com.au/phone/voip-features
[**Difference Between Call Forward and Call Transfer**](https://learningnetwork.cisco.com/thread/85291)
<b><i>Call Forwarding</i></b>
Call forwarding allows you to send all your incoming calls to another landline or cell phone number. Call forwarding overrides the ability to answer the phone forwarded from the original line. This service requires a subscription through your phone service provider and may incur an additional monthly fee.
<b><i>Call Transfer</i></b>
Call transfer allows you to send a call from one phone to another telephone without the need to disconnect the phone call. This feature is usually activated by the push of a button followed by dialing an extension.
Thank you