savoirfairelinux issueshttps://git.jami.net/groups/savoirfairelinux/-/issues2021-11-10T20:20:32Zhttps://git.jami.net/savoirfairelinux/jami-client-gnome/-/issues/1241account settings, connectivity, "Auto registration after expired" is only for...2021-11-10T20:20:32ZSébastien Blinaccount settings, connectivity, "Auto registration after expired" is only for SIPAmin BandaliAmin Bandalihttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1020Allow adding a SIP account without creating a Jami user first2022-10-21T11:49:18Zarkanoid87Allow adding a SIP account without creating a Jami user firstI've found Jami inserted in https://en.wikipedia.org/wiki/List_of_SIP_software#Mobile_clients but on first download I've been quite surprised to have to create jami user, add sip account, delete jami user, to make it workI've found Jami inserted in https://en.wikipedia.org/wiki/List_of_SIP_software#Mobile_clients but on first download I've been quite surprised to have to create jami user, add sip account, delete jami user, to make it workhttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1054Android SIP client not working after update to Taranis2022-11-11T18:35:52ZFietzeAndroid SIP client not working after update to TaranisBug report form
---------------
## Environment
Version: Taranis 20211210-01
Device model: Nexus 5X
Network connection: WiFi FritzBox 7530
Android version: LineageOS 17.1 (Android 10)
Build: F-droid
## Steps to reproduce
1...Bug report form
---------------
## Environment
Version: Taranis 20211210-01
Device model: Nexus 5X
Network connection: WiFi FritzBox 7530
Android version: LineageOS 17.1 (Android 10)
Build: F-droid
## Steps to reproduce
1. tap on app logo to open Jami application
2. choose SIP account
3. tap on any of the existing conversations
4. conversations opens, showing the call history
5. tap on receiver icon
- Actual result: Screen turns dark grey. After a few second it falls back to the call history, showing "Missed outgoing call"
Another few seconds later, screen switches back to the conversations list.
Sometimes an error message appears "Jami keeps stopping"
Quite often, after some seconds Jami just closes down.
- Expected result: Jami should call the requested contact; I should hear a dailing tone.
## Additional information
SIP provider: sip.diamondcard.us
On 14th of November 2021 I had made a successful call of 10 min, 35 secs
I have not deliberately changed any of the settings - not the router nor the phone.
To verify the devices function, I have installed and set up the SIPdroid app. It works like a charm.https://git.jami.net/savoirfairelinux/jami-daemon/-/issues/947Blind transfer doesn't work2024-01-25T19:27:07ZSébastien BlinBlind transfer doesn't work# Scenario
+ Create SIP account
+ Do a blind transfer (transfer to another contact without an active call)
# Expected
The transfer is successful
# Current result
No transfer happen (however attended transfer works fine)# Scenario
+ Create SIP account
+ Do a blind transfer (transfer to another contact without an active call)
# Expected
The transfer is successful
# Current result
No transfer happen (however attended transfer works fine)Sébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/377Bugfix - SIP authentication username option missing in desktop clients2022-07-04T18:39:37ZRobinBugfix - SIP authentication username option missing in desktop clientsI'd like to refer to my request on [Twitter](https://twitter.com/jami_social/status/1217490812847763456)... BTW thanks for the quick reply :thumbsup:
---
I'm unable to set up my SIP account since there is no option for `authentication...I'd like to refer to my request on [Twitter](https://twitter.com/jami_social/status/1217490812847763456)... BTW thanks for the quick reply :thumbsup:
---
I'm unable to set up my SIP account since there is no option for `authentication username`. I have something like this:
```config
username: 1234
registrar: example.net
authentication username: johndoe
password: ***
```
Could you please fix this?Ming Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-project/-/issues/794Bugfix - SIP authentication username option missing in desktop clients2021-04-16T14:17:34ZRobinBugfix - SIP authentication username option missing in desktop clientsI'd like to refer to my request on [Twitter](https://twitter.com/jami_social/status/1217490812847763456)... BTW thanks for the quick reply :thumbsup:
---
I'm unable to set up my SIP account since there is no option for `authentication...I'd like to refer to my request on [Twitter](https://twitter.com/jami_social/status/1217490812847763456)... BTW thanks for the quick reply :thumbsup:
---
I'm unable to set up my SIP account since there is no option for `authentication username`. I have something like this:
```config
username: 1234
registrar: example.net
authentication username: johndoe
password: ***
```
Could you please fix this?Ming Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1289Bundle same contacts with different spelling2021-08-19T19:54:53ZmokkinBundle same contacts with different spellingThe following examples shows three different spellings of the same contact:
![image](/uploads/564ff72a0adad7f06a3055ba20dd53f6/image.png)
1. International format with country code and city code
2. National format with city code
3. Local...The following examples shows three different spellings of the same contact:
![image](/uploads/564ff72a0adad7f06a3055ba20dd53f6/image.png)
1. International format with country code and city code
2. National format with city code
3. Local format without any prefix
All of them are successful for calling, because the pbx/sip knows its region. For a better overview these contacts should be recognized as the same and bundled.https://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 youhttps://git.jami.net/savoirfairelinux/jami-client-ios/-/issues/109Calling a number with spaces in SIP fails2021-02-15T15:23:02ZStéphane GuillouCalling a number with spaces in SIP failsWhen calling a number with an SIP account, if the number includes spaces (which iOS does automatically to increase readability), the call fails.
For example, +32 123 45 67 89 vs +32123456789: the first one would do nothing (while it see...When calling a number with an SIP account, if the number includes spaces (which iOS does automatically to increase readability), the call fails.
For example, +32 123 45 67 89 vs +32123456789: the first one would do nothing (while it seems it is trying to connect, showing the call screen, but no sound), whereas the second one would work.
I am not sure how dependent this is from the SIP service it connects to (I am with DiamondCard.us), but I think Jami should remove the spaces if they exist in the first place anyway, before placing the call?Kateryna KostiukKateryna Kostiukhttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/1073CallOverlay: incorrect button shown sometimes2023-06-13T12:43:10ZSébastien BlinCallOverlay: incorrect button shown sometimes# Scenario
+ Do a call with a SIP account, hangup
+ Do a call with a Jami Account, hangup
+ Do a call with a SIP account
# Expected
+ Pause button, call forward, DTMF panel MUST be shown
# Current result
+ Sometimes pause/forward/dt...# Scenario
+ Do a call with a SIP account, hangup
+ Do a call with a Jami Account, hangup
+ Do a call with a SIP account
# Expected
+ Pause button, call forward, DTMF panel MUST be shown
# Current result
+ Sometimes pause/forward/dtmf are not shown
# Note
CallActionBar.qml is the root cause. This file seems badly designed depending on a lot of reset() instead of just showing properties correctly. I'd recommend to re-do this classAline Gondim SantosAline Gondim Santoshttps://git.jami.net/savoirfairelinux/jami-project/-/issues/744Chat using SIP/XMPP2021-08-19T18:48:42ZAdevurChat using SIP/XMPPHello, I currently have three Jami clients (latest nightly version on Ubuntu 18.04), and they are all connected to the same SIP exchange (specifically: FreePBX 15 based on Asterisk 16). I'm using Jami as a SIP client in order to do video...Hello, I currently have three Jami clients (latest nightly version on Ubuntu 18.04), and they are all connected to the same SIP exchange (specifically: FreePBX 15 based on Asterisk 16). I'm using Jami as a SIP client in order to do video-conference calls between the three clients, and everything works well so far.
However, I noted that the clients are not able to send text messages between them, so my question is: does Jami support chat via SIP and XMPP? If yes, how can I configure it on Jami?
NOTE: I've already enabled chat/XMPP support on FreePBX for all three users (following this [guide](https://wiki.freepbx.org/display/ZU/Enabling+Chat+for+a+User)).
NOTE 2: in case Jami does support XMPP, the problem could be a misconfiguration of FreePBX (in particular, the XMPP domain, that is currently blank). I have little knowledge of XMPP on FreePBX, so maybe some of you know how to configure FreePBX in order to enable chat support in Jami.
Thanks very much.https://git.jami.net/savoirfairelinux/jami-client-gnome/-/issues/964Connection to external SIP provider fails2020-10-02T20:17:21Zkr428Connection to external SIP provider failsHi all, tried using Ring/Jami with our corporate SIP provider. Can't connect, but unfortunately I don't get any error message on what's exactly wrong here. Linphone and some other open-source SIP clients work so it should *generally* wor...Hi all, tried using Ring/Jami with our corporate SIP provider. Can't connect, but unfortunately I don't get any error message on what's exactly wrong here. Linphone and some other open-source SIP clients work so it should *generally* work. Any ideas on how to get more detailed error messages?
Thanks,
KristianMing Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-client-gnome/-/issues/1118copy/paste sip number if already in contacts = crash2020-11-20T00:18:36ZSébastien Blincopy/paste sip number if already in contacts = crashSébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1126[Debian 10 sid] SIP Incall on FritzBox2021-06-08T11:08:26ZPeter Maier[Debian 10 sid] SIP Incall on FritzBoxIncalls internal from FritzBox with IP phone are not working. Outcall on internal IP phones is working.
**Problem**
Add call to conversation with "**611"
"slotCallStateChanged (call: 8738021371634479), from Incoming to Talking"
(jami...Incalls internal from FritzBox with IP phone are not working. Outcall on internal IP phones is working.
**Problem**
Add call to conversation with "**611"
"slotCallStateChanged (call: 8738021371634479), from Incoming to Talking"
(jami-gnome:180595): Gtk-WARNING **: 12:57:54.298: Theme parsing error: <data>:1:79: Not using units is deprecated. Assuming 'px'.
(jami-gnome:180595): Gtk-WARNING **: 12:57:54.381: Calling org.xfce.Session.Manager.Inhibit failed: GDBus.Error:org.freedesktop.DBus.Error.UnknownMethod: No such method “Inhibit”https://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/19DTMF should be sent with keyboard2023-01-11T15:15:49ZSébastien BlinDTMF should be sent with keyboardpressing the keyboard should send DTMF in sip callspressing the keyboard should send DTMF in sip callsSébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-project/-/issues/762[Feature request] Allow multiple concurrent SIP accounts2020-12-17T15:10:36ZJSmith[Feature request] Allow multiple concurrent SIP accountsAt the moment I can add more than 1 SIP account to jami, but only one can be used at the time.
Example if I have more than 1 SIP account, only the "Default" can receive incoming call at the moment (on Windows at least).
The suggestion i...At the moment I can add more than 1 SIP account to jami, but only one can be used at the time.
Example if I have more than 1 SIP account, only the "Default" can receive incoming call at the moment (on Windows at least).
The suggestion is to allow all SIP accounts created in jami can receive incoming call.Ming Rui ZhangMing Rui Zhanghttps://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-product-backlog/-/issues/40ffmpeg: add G711, G729 codecs (SIP)2022-02-03T18:28:56Zovariffmpeg: 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/201ffmpeg: add G711, G729 codecs (SIP)2021-12-29T21:26:12Zovariffmpeg: 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-client-gnome/-/issues/1255[GNOME] Dialpad2023-05-26T13:59:32ZJami Bot[GNOME] DialpadIssue generated from Tuleap's migration script.
**Originally submitted by: Óvári (ovari)**
Please add a Dialpad which has numbers (0,1,2,3,4,5,6,7,8,9) and star/asterisk (\*) and hash (\#).
https://support.skype.com/en/faq/FA22/how...Issue generated from Tuleap's migration script.
**Originally submitted by: Óvári (ovari)**
Please add a Dialpad which has numbers (0,1,2,3,4,5,6,7,8,9) and star/asterisk (\*) and hash (\#).
https://support.skype.com/en/faq/FA22/how-do-i-dial-an-extension-during-a-call
Thank you
ring-gnome 2018-03-23 23:25:11 UTC
Linux Mint 18.3 Cinnamon 64-bitMing Rui ZhangMing Rui Zhang