savoirfairelinux issueshttps://git.jami.net/groups/savoirfairelinux/-/issues2022-10-26T15:13:31Zhttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1111SIP account can't receive incoming calls with TLS activated2022-10-26T15:13:31ZLem MingSIP account can't receive incoming calls with TLS activatedWhen TLS is enabled in the SIP account, no incoming calls are received.
Environment:
- OS: Android 11
- Jami version: Taranis - 20220615-01 (F-Droid)
- Device: Teracube 2e
- SIP provider: voip.ms
Steps to reproduce:
1) Create a SIP ac...When TLS is enabled in the SIP account, no incoming calls are received.
Environment:
- OS: Android 11
- Jami version: Taranis - 20220615-01 (F-Droid)
- Device: Teracube 2e
- SIP provider: voip.ms
Steps to reproduce:
1) Create a SIP account following this [post](https://forum.jami.net/t/how-can-i-create-a-sip-account-in-the-android-version-of-jami/1294)
2) Follow the [Jami wiki on voip.ms](https://wiki.voip.ms/article/Jami) for the basic configuration
3) Activate: Encrypt media streams (SRTP)
4) Activate: Enable SDES as key exchange protocol
5) Activate: Use TLS Transport
6) Permutate between 5061 (voip.ms TLS port) and the default value for whatever port options available
Expected and actual behaviors:
- On Jami, the account is online
- Echo test work (voip.ms: 4443)
- On the portal, it is registered with secure transport
Expected behavior:
- I can receive incoming calls
Actual behavior:
- I can't receive incoming calls
Following steps:
7) Deactivate "Use TLS transport"
Actual behaviors:
- On Jami, the account is online
- Echo test work (voip.ms: 4443)
- On the portal, it is registered but **no with secure transport**
- I **can** receive incoming calls
Following steps:
8) Install Linphone for Android
9) Follow the Linphone for Android tutorial specific for voip.ms [here](https://wiki.voip.ms/article/Linphone_Android)
Expected and actual behaviors:
- On Linphone, the account is online
- Echo test work (voip.ms: 4443)
- On the portal, it is registered with secure transport
- I can receive incoming calls
For the same environment:
- same network
- same voip.ms portal configuration
- same device
it works for Linphone but not for Jami.Sébastien BlinAntoine NoreauSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1021SIP messaging failure causes authetication failure2022-11-11T16:34:31Zarkanoid87SIP messaging failure causes authetication failureI'm aware that this could be a ticket that could really be multiple ones, but I'm not sure what's causing this and it really feels like an uroboro.
I've an asterisk server running pjsip module to provide SIP calls and messages to vpn co...I'm aware that this could be a ticket that could really be multiple ones, but I'm not sure what's causing this and it really feels like an uroboro.
I've an asterisk server running pjsip module to provide SIP calls and messages to vpn connected clients (no direct connections). I'm already successfully using it with other Android clients like linphone and zoiper but I'm finding issues with Jami.
- Jami version: Maloya - 20211004-01 (Play Store)
- Device model: Samsung Note 10+
- Android version: 11
How to reproduce:
1) I setup my sip account in Jami -> authentication successful
2) I call myself or others -> call works
3) I try to send a message to myself or others -> no message is delivered, no sip packages are received on the server, app goes in "connection error" state.
4) I try to call call me again -> 401 not authorized
5) I try to enable and disable sip account -> 401 not authorized
6) Open menu "Account > advanced > Local Interface" and re-select already selected option (tun0 in my case) -> back to step 2
There are alternative steps "6" to rollback to step 2, like enabling and disabling "Published same as local"https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/434Strip whitespace from SIP phone numbers2022-11-11T16:36:06ZJami BotStrip whitespace from SIP phone numbersIssue generated from Tuleap's migration script.
**Originally submitted by: Dennis Schridde (devurandom)**
Currently it is impossible for me to call phone numbers directly from my phone book using Ring for Android, because they usually c...Issue generated from Tuleap's migration script.
**Originally submitted by: Dennis Schridde (devurandom)**
Currently it is impossible for me to call phone numbers directly from my phone book using Ring for Android, because they usually contain whitespace - it appears my SIP provider simply rejects that call. When I enter the phone number manually without whitespace into Ring, I can call them without any problem. This issue also affects text messages, which are impossible to send, because there is no way to enter the phone number manually, once a contact has been called and Ring figured out the corresponding phone book contact.
I am using Ring for Android 20160816 from the Google Play Store.https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1312[SIP-Account] Can't answer a SIP-call2023-07-14T18:17:25ZElys[SIP-Account] Can't answer a SIP-callusing Jami 20230616-01 on Android 13 (Samsung Galaxy A53 5G)
might be related: https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1302
Step to reproduce:
- In the past you've created a SIP-account, used it successfully ...using Jami 20230616-01 on Android 13 (Samsung Galaxy A53 5G)
might be related: https://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1302
Step to reproduce:
- In the past you've created a SIP-account, used it successfully and disabled it for a few weeks (never used again; however other clients were used for the same SIP-account; but the SIP-Account in Jami has been always disabled; so the phone has been probably rebooted many times)
- (Now you disable all other clients for the same SIP-account) Now enable the SIP-account in Jami again
- Make a call (Phone 1 (SIM) -> Phone 2(SIP-Account; using Jami) )
- You'll get a notification "Decline" or "Answer in audio"
- If you tap "Answer in audio" nothing will happen but another noficiation "Answer in audio" will appear
- You can tap like 4 next notification "Answer in audio" but nothing will happen
- After that the call will fail
- You'll get a "missed call" notification
NOTE:
- Enabeling/Disabeling the STUN-option didn't change anything
- Enabeling/Disabeling the SIP-account didn't change anything
- Switching between wifi and mobile networks didn't change anythingPierre NicolasPierre Nicolashttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1302SIP Account: Number shown twice2023-07-13T16:56:12ZElysSIP Account: Number shown twiceusing the newest beta version of Android
Steps to reproduce?:
- Add a SIP-Account
- You'll view your number
- But you get the same account twice? (You can only disable/enable one account; the other will be automatically enabled / disab...using the newest beta version of Android
Steps to reproduce?:
- Add a SIP-Account
- You'll view your number
- But you get the same account twice? (You can only disable/enable one account; the other will be automatically enabled / disabled too)
![SIP-issue](/uploads/bd45559eccb0be3c9c9f4907aa6bef5c/SIP-issue.png)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/1173[SIP] Delete account and then Jami disappears (closes or quits)2023-07-31T12:53:10Zovari[SIP] Delete account and then Jami disappears (closes or quits)1. Have one Jami account and one SIP account
2. Delete the SIP account, then Jami disappears (closes or quits) and the Jami icon in the system tray also disappears
Are you able to reproduce and please fix?
Thank you
Linux Mint 21.1 Ci...1. Have one Jami account and one SIP account
2. Delete the SIP account, then Jami disappears (closes or quits) and the Jami icon in the system tray also disappears
Are you able to reproduce and please fix?
Thank you
Linux Mint 21.1 Cinnamon<br>
Jami Version: 202305271132
P.S. Unable to verify once fixed as no longer have any SIP account. Hopefully the fix will be beneficial to Jami and JAMS users who have SIP accounts.Aline Gondim SantosAline Gondim Santoshttps://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-daemon/-/issues/846SIP call on android: plain RTP session impossible (intentionally unencrypted ...2023-04-14T12:23:34ZTobias HuberSIP call on android: plain RTP session impossible (intentionally unencrypted LAN-PBX)Hi, since Android 13 doesn't offer native SIP support anymore, I thought this great Jami thing could step in...
Unfortunately, it was quiet inconvenient to get at least some idea why this doesn't work with my LAN/VPN PBX
_(Inconvenience...Hi, since Android 13 doesn't offer native SIP support anymore, I thought this great Jami thing could step in...
Unfortunately, it was quiet inconvenient to get at least some idea why this doesn't work with my LAN/VPN PBX
_(Inconvenience: LAN/VPN PBX is asterisk 1.6, which doesn't use identical string for account identification and user authentication; the user corresponding to the password for authentication is different to the name of the SIP account.
It's possible to set Jami up that way, but it's quiet hidden - besides some more not so minor UI nits on android, but that will be a different issue report.)_
I checked that all Security->Security switches are off (SRTP and TLS transport).
1.) Initiating (outgoing) call fails because:
`"Rejecting secure audio stream without encryption details: audio 19686 RTP/SAVP 104 9 2 112 111 110 8 0 101"`
> <--- SIP read from UDP:172.21.97.226:5060 ---> [115/1865]
> INVITE sip:11@pbx.example.net SIP/2.0
> Via: SIP/2.0/UDP 172.21.97.226:5060;rport;branch=z9hG4bKPjaf1c522c-6e92-4539-b899-15c2bad55ad1
> Max-Forwards: 70
> From: <sip:tobimob_line1@pbx.example.net>;tag=b2110c22-6e14-4381-8a99-cc1e40d3872f
> To: <sip:11@pbx.example.net>
> Contact: <sip:tobimob_line1@172.21.97.226:5060>
> Call-ID: 4b970369-1b13-44da-b798-3a74f51d8c39
> CSeq: 629 INVITE
> Subject: Phone call
> Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
> Supported: replaces
> User-Agent: Jami Daemon 13.7.0 (android)
> Authorization: Digest username="tobi.huber", realm="pbx.example.net", nonce="3b54e054", uri="sip:11@pbx.example.net", response="4070f25aadda2536ee24058551dfb64
> 1", algorithm=MD5
> Content-Type: application/sdp
> Content-Length: 1047
>
> v=0
> o=localhost 3890394360 0 IN IP4 172.21.97.226
> s=Call ID 8706183892147327
> c=IN IP4 172.21.97.226
> t=0 0
> a=ice-ufrag:42a09c4e
> a=ice-pwd:04afb39763f13c7f32a67b91
> m=aud--- (15 headers 27 lines) ---
> Sending to 172.21.97.226:5060 (no NAT)
> Using INVITE request as basis request - 4b970369-1b13-44da-b798-3a74f51d8c39
> Found peer 'tobimob_line1' for 'tobimob_line1' from 172.21.97.226:5060
> Found RTP audio format 104
> Found RTP audio format 9
> Found RTP audio format 2
> Found RTP audio format 112
> Found RTP audio format 111
> Found RTP audio format 110
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 101
> Found unknown media description format opus for ID 104
> Found audio description format G722 for ID 9
> Found audio description format G726-32 for ID 2
> Found audio description format speex for ID 112
> Found audio description format speex for ID 111
> Found audio description format speex for ID 110
> Found audio description format PCMA for ID 8
> Found audio description format PCMU for ID 0
> Found audio description format telephone-event for ID 101
[Apr 13 19:05:59] ERROR[101987][C-000005d5]: chan_sip.c:33575 int setup_srtp(struct sip_srtp **): No SRTP module loaded, can't setup SRTP session.
[Apr 13 19:05:59] WARNING[101987][C-000005d5]: chan_sip.c:10417 int process_sdp(struct sip_pvt *, struct sip_request *, int): Rejecting secure audio stream witho
ut encryption details: audio 19686 RTP/SAVP 104 9 2 112 111 110 8 0 101
Why does it send RTP/SAVP although I disabled SRTP?
2.) Incoming call signalling works, but not possible to establish RTP session:
```
"Ignoring audio media offer because port number is zero" and
"Failing due to no acceptable offer found"
```
Ringing:
> <--- SIP read from UDP:172.21.97.226:5060 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 172.21.98.12:5060;received=172.21.98.12;branch=z9hG4bK0d2bf1b6
> Call-ID: 1e4bf56a62264ba6792450e4506bed52@pbx.example.net
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as48fb14ca
> To: <sip:tobimob_line1@172.21.97.226>;tag=6c456d21-e6ae-4c85-bf04-da0aaf56e758
> CSeq: 102 INVITE
> User-Agent: Jami Daemon 13.7.0 (android)
> Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
> Contact: <sip:tobimob_line1@172.21.97.226:5060>
... (snipped)
Accepting call on Jami:
> <--- SIP read from UDP:172.21.97.226:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.21.98.12:5060;received=172.21.98.12;branch=z9hG4bK0d2bf1b6
> Call-ID: 1e4bf56a62264ba6792450e4506bed52@pbx.example.net
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as48fb14ca
> To: <sip:tobimob_line1@172.21.97.226>;tag=6c456d21-e6ae-4c85-bf04-da0aaf56e758
> CSeq: 102 INVITE
> User-Agent: Jami Daemon 13.7.0 (android)
> Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH
> Supported: replaces
> Contact: <sip:tobimob_line1@172.21.97.226:5060>
> Content-Type: application/sdp
> Content-Length: 142
>
> v=0
> o=localhost 3890395686 1 IN IP4 172.21.97.226
> s=Call ID 5890041505306313
> c=IN IP4 172.21.97.226
> t=0 0
> m=audio 0 RTP/AVP 9 8 0 3 101
> <------------->
chan_sip.c:10008 int process_sdp(struct sip_pvt *, struct sip_request *, int): Ignoring audio media offer [5/1992]
port number is zero
chan_sip.c:10438 int process_sdp(struct sip_pvt *, struct sip_request *, int): Failing due to no acceptabl[3/1992]
found
**Here's a working internal call, where sipdroid is the user agent on the same phone:**
> <--- SIP read from UDP:172.21.97.226:40739 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 172.21.98.12:5060;branch=z9hG4bK04f5274c
> To: <sip:tobimob_line1@10.26.229.169:40739;transport=udp>;tag=fb8b4ff3c7a73ee8
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as686012bd
> Call-ID: 3547fb4e68213eb90d8295ae221e36ea@pbx.example.net
> CSeq: 102 INVITE
> Server: Sipdroid/6.3 beta/Pixel 7
> Content-Length: 198
> Content-Type: application/sdp
... (snipped)
Accepting call on SIPdroid:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.21.98.12:5060;branch=z9hG4bK04f5274c
> To: <sip:tobimob_line1@10.26.229.169:40739;transport=udp>;tag=fb8b4ff3c7a73ee8
> From: "T. Huber (22)" <sip:thbuero@pbx.example.net>;tag=as686012bd
> Call-ID: 3547fb4e68213eb90d8295ae221e36ea@pbx.example.net
> CSeq: 102 INVITE
> Contact: <sip:tobimob_line1@10.26.229.169:40739;transport=udp>
> Server: Sipdroid/6.3 beta/Pixel 7
> Content-Length: 198
> Content-Type: application/sdp
>
> v=0
> o=tobimob_line1@pbx.example.net 0 0 IN IP4 10.26.229.169
> s=Session SIP/SDP
> c=IN IP4 10.26.229.169
> t=0 0
> m=audio 21000 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> <------------->
**Significant difference here is "m=audio ...." lines.**
With SIPdroid (the working incomming call), it reads
> m=audio 21000 RTP/AVP 9 101
while with jami (not working)
> m=audio 0 RTP/AVP 9 8 0 3 101
I'd very much appreciate if somebody could take care and bring back originating strenghts of Jami in that it's usable as a working SIP UA - for plain RTP too!
Especially due to dropped native SIP support in recent android versions!
The android settings UI is broken too, like already mentioned, but ther's the workaround to use it in landscape mode, which makes it possible to sroll the account-enabler switch out of overlapping tab selection area... Will tell in a different issue report.
Thanks in advance,
-Tobihttps://git.jami.net/savoirfairelinux/jami-client-android/-/issues/1207SIP settings are partially broken2024-01-30T14:53:27ZPierre NicolasSIP settings are partially broken## Describe your environment
- Device model: Samsung Galaxy Tab S5e SM-T720
- Android version: 11
- What build you are using: your own, commit 5be1de17c790bddfb30dde74c8a1364e213f9847
## Steps to reproduce
- Can you reproduce ...## Describe your environment
- Device model: Samsung Galaxy Tab S5e SM-T720
- Android version: 11
- What build you are using: your own, commit 5be1de17c790bddfb30dde74c8a1364e213f9847
## Steps to reproduce
- Can you reproduce the bug: at will
- Steps:
1. Open a SIP account
2. Go to settings ("Account Settings")
3. Scroll horizontally
- Actual result: Tab headers are unclickable; settings are displayed in front of headers
## Additional information
![sip_settings_broken](/uploads/cfc122de3fa6faa6ee298c7748c21cee/sip_settings_broken.mp4)Pierre NicolasPierre Nicolashttps://git.jami.net/savoirfairelinux/jami-client-ios/-/issues/242SIP: adding a call doesn't add any history2023-04-18T18:17:01ZSébastien BlinSIP: adding a call doesn't add any history# Scenario
+ Create a SIP account (e.g. internal 69X)
+ Call someone (e.g. a phone)
# Expected
+ Conversation should be shown with a "Ongoing call" message# Scenario
+ Create a SIP account (e.g. internal 69X)
+ Call someone (e.g. a phone)
# Expected
+ Conversation should be shown with a "Ongoing call" messageBinal AhiyaBinal Ahiyahttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/809[SIP] unable to call a contact in conversation list2022-08-23T08:11:18Zovari[SIP] unable to call a contact in conversation list## Scenario 1
1. Type a number to call in SIP account<br>
![image](/uploads/25c6f87f5b7b229b0003ac514a5301e8/image.png)
1. Call the number
2. Buttons to call that contact are no longer available<br>
![image](/uploads/e6979f8b74157359665...## Scenario 1
1. Type a number to call in SIP account<br>
![image](/uploads/25c6f87f5b7b229b0003ac514a5301e8/image.png)
1. Call the number
2. Buttons to call that contact are no longer available<br>
![image](/uploads/e6979f8b7415735966539b23f7f23edb/image.png)
## Scenario 2
1. Type a number to call in SIP account
2. Click `Add to conversations` button (located in top right)
3. Call button disappears
Are you able to reproduce and please fix?
Thank you
Linux Mint 21<br>
Jami Version: 202208152318Sébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/786SIP connections are not gracefully terminated2022-11-23T14:59:36ZMaxim CournoyerSIP connections are not gracefully terminatedWhen using a SIP account, I noticed that upon either disabling the account in the settings or quitting the client, the server hasn't received the BYE and assumes the client is still connected (that's at least the case for voip.ms).
linp...When using a SIP account, I noticed that upon either disabling the account in the settings or quitting the client, the server hasn't received the BYE and assumes the client is still connected (that's at least the case for voip.ms).
linphone doesn't exhibit this problem.Antoine NoreauAntoine Noreauhttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/720Support DTMF2023-06-16T18:56:18ZSébastien BlinSupport DTMFTo be definedTo be definedBackloghttps://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-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-product-backlog/-/issues/12Support DTMF2022-02-16T21:37:18ZSébastien BlinSupport DTMFTo be definedTo be definedBackloghttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/662Support DTMF2021-12-29T17:07:16ZSébastien BlinSupport DTMFTo be definedTo be definedBackloghttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1329Settings: move use STUN/Stun address into SIP Account2022-11-03T02:34:32ZSébastien BlinSettings: move use STUN/Stun address into SIP Accountas it's unnecessary for Jami accounts (we have the DHT)as it's unnecessary for Jami accounts (we have the DHT)https://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 work