jami-project issueshttps://git.jami.net/savoirfairelinux/jami-project/-/issues2022-11-03T02:34:32Zhttps://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-project/-/issues/1224re-register to late2023-03-13T13:31:51ZLukas Wallischre-register to lateAs the logs suggests the re-register timer is to long... as far as i know the reregister should come before the old register expires, not 10s after...
```
[1618330072.037|43625|sipaccount.cpp :958 ] Start keep alive timer for accoun...As the logs suggests the re-register timer is to long... as far as i know the reregister should come before the old register expires, not 10s after...
```
[1618330072.037|43625|sipaccount.cpp :958 ] Start keep alive timer for account 7e9411aaf521bc4b
[1618330072.037|43625|sipaccount.cpp :977 ] Registration Expire: 119
[1618330072.037|43625|sipvoiplink.cpp :776 ] Register new keep alive timer 2099486383 with delay 129
```
OS : Ubuntu 20.04
Version: snap ( downloaded today)
`jami -v` gives:
```
Testing for explicit PulseAudio choice...
Testing for ALSA permissions...
...and using ALSA.
Jami Daemon 9.8.0-a6d5ad32d7-dirty, by Savoir-faire Linux 2004-2019
https://jami.net/
[Video support enabled]
[Plugins support enabled]
```
i can only find the settings for the expire-timer, but the not the reregister-timer. am I missing something?
I'm using this client behind a Hardware-PBX, because of the late reregister my calls keep getting abortedSébastien BlinSébastien Blinhttps://git.jami.net/savoirfairelinux/jami-project/-/issues/1215Please detect status 488 from SIP server and offer suggestions to change codec2023-05-29T13:37:07ZreubenfirminPlease detect status 488 from SIP server and offer suggestions to change codecWhen a SIP server returns 488, it can mean a codec incompatibility. Your UI buries that 488 is returned. Please detect the status and pop up the settings asking the user to check the codecs vs what the voip provider allows.When a SIP server returns 488, it can mean a codec incompatibility. Your UI buries that 488 is returned. Please detect the status and pop up the settings asking the user to check the codecs vs what the voip provider allows.https://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-project/-/issues/1057No early media after SIP 1832020-10-02T21:14:46ZPaweł BogusławskiNo early media after SIP 183Hello,
When using
* jami_20200829.1.d8786b1~dfsg1-1_amd64.deb
* jami-daemon_20200829.1.d8786b1~dfsg1-1_amd64.deb
from
https://dl.jami.net/nightly/debian_10/pool/main/j/jami/
in Debian 10, Jami does not play ringback when calling ext...Hello,
When using
* jami_20200829.1.d8786b1~dfsg1-1_amd64.deb
* jami-daemon_20200829.1.d8786b1~dfsg1-1_amd64.deb
from
https://dl.jami.net/nightly/debian_10/pool/main/j/jami/
in Debian 10, Jami does not play ringback when calling external number (which uses SIP 183 message when waiting for call to be picked up and early media/SDP); when call is picked up - voice is played correctly; dring debug log of such call:
```
[1600086622.792|15229|manager.cpp :581 ] ----- Switch current call id to '2159493006188118' -----
[1600086622.792|15230|sipcall.cpp :963 ] [call:2159493006188118] fill SDP with ICE transport 0x5633ef823740
[1600086622.792|15230|sipaccount.cpp :375 ] contact header: "test" <sip:login@192.168.1.100:5060> / "test" <sip:login@sip.mydomain.loc> -> <sip:111222333@sip.mydomain.loc>
[1600086622.792|15230|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 1 (CALLING): cause=0, tsx@0x7fd6dc048e88 status 0 (Default status message)
[1600086622.793|15230|call.cpp :259 ] [call:2159493006188118] state change 0/1, cnx 0/2, code 0
[1600086622.793|15230|call.cpp :286 ] [call:2159493006188118] emit client call state change CONNECTING, code 0
[1600086624.793|15230|call.cpp :112 ] Call 2159493006188118 is still connecting after timeout, sending fallback request
[1600086625.482|15233|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 3 (EARLY): cause=0, tsx@0x7fd70800d278 status 183 (Session Progress)
[1600086625.482|15233|sipvoiplink.cpp :1009 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 media update: status 0
[1600086625.482|15233|sdp.cpp :383 ] Local active SDP Session:
v=0
o=ibnb003 3809075422 0 IN IP4 192.168.1.100
s=Jami Daemon
c=IN IP4 192.168.1.100
t=0 0
a=ice-ufrag:1d26a479
a=ice-pwd:33f8ac4760b4c6501dd61dfa
m=audio 41884 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:41885 IN IP4 192.168.1.100
a=sendrecv
a=candidate:H44e08c2a 1 UDP 2130706431 [...] 42537 typ host
a=candidate:Ha030271 1 UDP 2130706431 192.168.1.100 39418 typ host
a=candidate:Hc0a87b64 1 UDP 2130706175 [...] 39418 typ host
a=candidate:Hc0a87a01 1 UDP 2130705919 [...] 39418 typ host
a=candidate:H44e08c2a 2 UDP 2130706430 [...] 42180 typ host
a=candidate:Ha030271 2 UDP 2130706430 [...] 35281 typ host
a=candidate:Hc0a87b64 2 UDP 2130706174 [...] 35281 typ host
a=candidate:Hc0a87a01 2 UDP 2130705918 [...] 35281 typ host
m=video 0 RTP/AVP 96 97 98 99
[1600086625.482|15233|sdp.cpp :383 ] Remote active SDP Session:
v=0
o=ib 1858813877 1858813877 IN IP4 192.168.100.2
s=IB VoIP Server
c=IN IP4 192.168.100.2
t=0 0
m=audio 23562 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=maxptime:60
a=sendrecv
m=video 0 RTP/AVP 96 97 98 99
[1600086625.482|15233|sipcall.cpp :1230 ] [call:2159493006188118] medias changed
[1600086625.482|15230|sipcall.cpp :1242 ] [call:2159493006188118] no remote ICE for medias
[1600086625.482|15230|sipcall.cpp :1182 ] [call:2159493006188118] stopping all medias
[1600086625.482|15230|sipcall.cpp :1021 ] [call:2159493006188118] startAllMedia()
[1600086625.482|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41884
[1600086625.482|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41885
[1600086625.482|15230|socket_pair.cpp :349 ] SocketPair: local{41884,41885} / 192.168.100.2{23562,23563}
[1600086625.482|15230|audio_input.cpp :53 ] Creating audio input with id: 2159493006188118
[1600086625.483|15230|audio_input.cpp :270 ] Switching audio source to match 'camera://04f2b3ec200901010001'
[1600086625.483|15230|audio_sender.cpp :72 ] audioEncoder_->openOutput rtp://192.168.100.2:23562
[1600086625.483|15230|media_encoder.cpp :268 ] Not using hardware encoding for speex
[1600086625.483|15230|media_encoder.cpp :574 ] [libspeex] Using 4 threads
[1600086625.483|15230|media_encoder.cpp :621 ] [libspeex] Frame size 160
[1600086625.483|15230|sipcall.cpp :1065 ] [call:2159493006188118] [SDP:slot#1] Missing local codec
[1600086625.483|16375|media_decoder.cpp :134 ] Trying to open device dummyFilename with format sdp, pixel format , size 0x0, rate 0.000000
[1600086625.483|16375|media_decoder.cpp :146 ] Using format sdp
[1600086625.503|16374|audio_input.cpp :76 ] Switching audio input to '04f2b3ec200901010001'
[1600086628.306|15233|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 4 (CONNECTING): cause=0, tsx@0x7fd70800d278 status 200 (OK)
[1600086628.306|15233|sipvoiplink.cpp :1009 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 media update: status 0
[1600086628.306|15233|sdp.cpp :383 ] Local active SDP Session:
v=0
o=ibnb003 3809075422 1 IN IP4 192.168.1.100
s=Jami Daemon
c=IN IP4 192.168.1.100
t=0 0
a=ice-ufrag:1d26a479
a=ice-pwd:33f8ac4760b4c6501dd61dfa
m=audio 41884 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:41885 IN IP4 192.168.1.100
a=sendrecv
a=candidate:H44e08c2a 1 UDP 2130706431 [...] 42537 typ host
a=candidate:Ha030271 1 UDP 2130706431 [...] 39418 typ host
a=candidate:Hc0a87b64 1 UDP 2130706175 [...] 39418 typ host
a=candidate:Hc0a87a01 1 UDP 2130705919 [...] 39418 typ host
a=candidate:H44e08c2a 2 UDP 2130706430 [...] 42180 typ host
a=candidate:Ha030271 2 UDP 2130706430 [...] 35281 typ host
a=candidate:Hc0a87b64 2 UDP 2130706174 [...] 35281 typ host
a=candidate:Hc0a87a01 2 UDP 2130705918 [...] 35281 typ host
m=video 0 RTP/AVP 96 97 98 99
[1600086628.306|15233|sdp.cpp :383 ] Remote active SDP Session:
v=0
o=ib 1858813877 1858813877 IN IP4 192.168.100.2
s=IB VoIP Server
c=IN IP4 192.168.100.2
t=0 0
m=audio 23562 RTP/AVP 110 101
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=maxptime:60
a=sendrecv
m=video 0 RTP/AVP 96 97 98 99
[1600086628.306|15233|sipcall.cpp :1230 ] [call:2159493006188118] medias changed
[1600086628.306|15230|sipcall.cpp :1242 ] [call:2159493006188118] no remote ICE for medias
[1600086628.306|15230|sipcall.cpp :1182 ] [call:2159493006188118] stopping all medias
[1600086628.306|15233|sipvoiplink.cpp :861 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 5 (CONFIRMED): cause=0 (TX_MSG)
[1600086628.306|15233|sipcall.cpp :910 ] [call:2159493006188118] onAnswered()
[sdp @ 0x7fd6bc0206c0] Could not find codec parameters for stream 0 (Audio: speex (libspeex), 8000 Hz, mono): unspecified sample format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[1600086628.387|16375|media_decoder.cpp :481 ] Decoding audio using libspeex Speex (libspeex)
[1600086628.387|16375|media_decoder.cpp :491 ] Not using hardware decoding for speex
[1600086628.387|15230|sipcall.cpp :1021 ] [call:2159493006188118] startAllMedia()
[1600086628.387|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41884
[1600086628.387|15230|socket_pair.cpp :172 ] use local address: 0.0.0.0:41885
[1600086628.387|15230|socket_pair.cpp :349 ] SocketPair: local{41884,41885} / 192.168.100.2{23562,23563}
[1600086628.387|15230|audio_input.cpp :270 ] Switching audio source to match 'camera://04f2b3ec200901010001'
[1600086628.387|15230|audio_sender.cpp :72 ] audioEncoder_->openOutput rtp://192.168.100.2:23562
[1600086628.387|15230|media_encoder.cpp :268 ] Not using hardware encoding for speex
[1600086628.387|15230|media_encoder.cpp :574 ] [libspeex] Using 4 threads
[1600086628.387|15230|media_encoder.cpp :621 ] [libspeex] Frame size 160
[1600086628.387|15230|sipcall.cpp :1065 ] [call:2159493006188118] [SDP:slot#1] Missing local codec
[1600086628.387|16376|media_decoder.cpp :134 ] Trying to open device dummyFilename with format sdp, pixel format , size 0x0, rate 0.000000
[1600086628.387|15230|call.cpp :259 ] [call:2159493006188118] state change 1/1, cnx 2/4, code 0
[1600086628.387|15230|call.cpp :286 ] [call:2159493006188118] emit client call state change CURRENT, code 0
[1600086628.387|16376|media_decoder.cpp :146 ] Using format sdp
[1600086628.387|15230|manager.cpp :2007 ] [call:2159493006188118] Peer answered
[1600086628.387|15230|manager.cpp :1631 ] Add audio to call 2159493006188118
[1600086628.387|15230|manager.cpp :1645 ] [call:2159493006188118] Attach audio
[1600086628.387|15230|ringbufferpool.cpp:175 ] Bind call 2159493006188118 to call audiolayer_id
[1600086628.387|15230|ringbufferpool.cpp:156 ] Bind rbuf '2159493006188118' to callid 'audiolayer_id'
[1600086628.387|15230|ringbufferpool.cpp:156 ] Bind rbuf 'audiolayer_id' to callid '2159493006188118'
[1600086628.387|15230|audiolayer.cpp :130 ] Hardware audio format available : {s16, 2 channels, 44100Hz} 0
[1600086628.387|15230|audiostream.cpp :53 ] Playback: trying to create stream with device alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600086628.387|15230|audiostream.cpp :148 ] Stream is creating...
[1600086628.387|15230|audiostream.cpp :53 ] Ringtone: trying to create stream with device (44100Hz, 2 channels)
[1600086628.387|15230|audiostream.cpp :148 ] Stream is creating...
[1600086628.387|15230|audiostream.cpp :53 ] Capture: trying to create stream with device alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600086628.387|15230|audiostream.cpp :148 ] Stream is creating...
[1600086628.403|16374|audio_input.cpp :76 ] Switching audio input to '04f2b3ec200901010001'
[1600086629.465|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600086629.465|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.pci-0000_00_1b.0.analog-stereo
[1600086629.465|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600086629.465|15245|pulselayer.cpp :371 ] All streams ready, starting !
[1600086629.588|16376|media_decoder.cpp :481 ] Decoding audio using libspeex Speex (libspeex)
[1600086629.588|16376|media_decoder.cpp :491 ] Not using hardware decoding for speex
[1600086634.112|15233|sipvoiplink.cpp :853 ] [call:2159493006188118] INVITE@0x7fd6dc010a48 state changed to 6 (DISCONNCTD): cause=200, tsx@0x7fd70800d278 status 200 (OK)
[1600086634.112|15230|manager.cpp :2040 ] [call:2159493006188118] Peer hungup
```
Problem does not occur on same Debian system and SIP account when using Twinkle SIP client.
Problem does not occur in Jami when local VoiP number is called (which uses SIP 180 message when waiting for call, without early media/SDP); dring debug log of such /not answered/ call when ringback was played correctly:
```
[1600085689.195|15229|manager.cpp :581 ] ----- Switch current call id to '5190575654160253' -----
[1600085689.195|15230|sipcall.cpp :963 ] [call:5190575654160253] fill SDP with ICE transport 0x5633ef824e50
[1600085689.195|15230|sdp.cpp :718 ] addIceCandidates failed: cannot access media#1 (may be deactivated)
[1600085689.196|15230|sdp.cpp :718 ] addIceCandidates failed: cannot access media#1 (may be deactivated)
[1600085689.196|15230|sipaccount.cpp :375 ] contact header: "test" <sip:login@129.168.1.100:5060> / "IB" <sip:login@sip.mydomain.loc> -> <sip:login2@sip.mydomain.loc>
[1600085689.196|15230|sipvoiplink.cpp :853 ] [call:5190575654160253] INVITE@0x7fd6dc0460b8 state changed to 1 (CALLING): cause=0, tsx@0x7fd6dc0286b8 status 0 (Default status message)
[1600085689.196|15230|call.cpp :259 ] [call:5190575654160253] state change 0/1, cnx 0/2, code 0
[1600085689.196|15230|call.cpp :286 ] [call:5190575654160253] emit client call state change CONNECTING, code 0
[1600085689.374|15233|sipvoiplink.cpp :853 ] [call:5190575654160253] INVITE@0x7fd6dc0460b8 state changed to 3 (EARLY): cause=0, tsx@0x7fd70800d278 status 180 (Ringing)
[1600085689.374|15233|call.cpp :259 ] [call:5190575654160253] state change 1/1, cnx 2/3, code 0
[1600085689.374|15233|call.cpp :286 ] [call:5190575654160253] emit client call state change RINGING, code 0
[1600085689.374|15230|manager.cpp :2029 ] [call:5190575654160253] Peer ringing
[1600085689.374|15230|audiolayer.cpp :130 ] Hardware audio format available : {s16, 2 channels, 44100Hz} 0
[1600085689.374|15230|audiostream.cpp :53 ] Playback: trying to create stream with device alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600085689.374|15230|audiostream.cpp :148 ] Stream is creating...
[1600085689.374|15230|audiostream.cpp :53 ] Ringtone: trying to create stream with device (44100Hz, 2 channels)
[1600085689.374|15230|audiostream.cpp :148 ] Stream is creating...
[1600085689.374|15230|audiostream.cpp :53 ] Capture: trying to create stream with device alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback (44100Hz, 1 channels)
[1600085689.374|15230|audiostream.cpp :148 ] Stream is creating...
[1600085690.454|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600085690.454|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_output.pci-0000_00_1b.0.analog-stereo
[1600085690.454|15245|audiostream.cpp :156 ] Stream successfully created, connected to alsa_input.usb-GN_Netcom_A_S_Jabra_PRO_930_027608F15805-00.mono-fallback
[1600085690.454|15245|pulselayer.cpp :371 ] All streams ready, starting !
[1600085697.904|15229|manager.cpp :1665 ] [call:5190575654160253] Remove local audio
[1600085697.904|15229|ringbufferpool.cpp:242 ] Unbind call 5190575654160253 from all bound calls
[1600085697.904|15229|sipcall.cpp :368 ] [call:5190575654160253] Terminate SIP session
```
It seems that Jami has problem playing early media from SDP after 183 SIP message (opening sound device problem maybe?) and has not such problem when self playing ringback tone after 180 SIP message.
Sound configuration is pulseaudio, rindtone device = default (uses standard speakers), output and input devices = Jabra PRO 930 Mono and works ok in other functions (i.e. incomming ringing on standard speakers but ringback tones and speech on Jabra).
Regards,
Pawełhttps://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-project/-/issues/847IPv6 for SIP unavailable2022-11-11T16:26:42ZPavel PolyakovIPv6 for SIP unavailableIt looks like Jami doesn't support IPv6, neither does Jami listen on IPv6 when it is started up or does it allow calling an IPv6 host (truncates the address then says Bad URI).
This is pretty sad knowing that SIP will usually perform be...It looks like Jami doesn't support IPv6, neither does Jami listen on IPv6 when it is started up or does it allow calling an IPv6 host (truncates the address then says Bad URI).
This is pretty sad knowing that SIP will usually perform better in IPv6 since hosts can have a public address of their own and establish direct connections without relying on clumsy NAT bypass techniques and port redirections.Sébastien BlinAntoine NoreauSébastien Blinhttps://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/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-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.