savoirfairelinux issueshttps://git.jami.net/groups/savoirfairelinux/-/issues2021-03-26T18:57:25Zhttps://git.jami.net/savoirfairelinux/jami-client-qt/-/issues/300Swarm: Remove SIP contact => crash2021-03-26T18:57:25ZSébastien BlinSwarm: Remove SIP contact => crash```
#0 0x00007fffee236d74 in QVariant::QVariant(QString const&) () at /lib64/libQt5Core.so.5
#1 0x000000000044aee2 in SmartListModel::getConversationItemData(lrc::api::conversation::Info const&, lrc::api::account::Info const&, int) con...```
#0 0x00007fffee236d74 in QVariant::QVariant(QString const&) () at /lib64/libQt5Core.so.5
#1 0x000000000044aee2 in SmartListModel::getConversationItemData(lrc::api::conversation::Info const&, lrc::api::account::Info const&, int) const ()
#2 0x0000000000449db0 in SmartListModel::data(QModelIndex const&, int) const ()
#3 0x00007fffef187acd in QQmlDMAbstractItemModelData::value(int) const () at /lib64/libQt5QmlModels.so.5
#4 0x00007fffef181f54 in QQmlDMCachedModelData::metaCall(QMetaObject::Call, int, void**) () at /lib64/libQt5QmlModels.so.5
#5 0x00007fffee8789a6 in loadProperty(QV4::ExecutionEngine*, QObject*, QQmlPropertyData const&) () at /lib64/libQt5Qml.so.5
#6 0x00007fffee879b17 in QV4::QObjectWrapper::getQmlProperty(QV4::ExecutionEngine*, QQmlContextData*, QObject*, QV4::String*, QV4::QObjectWrapper::RevisionMode, bool*, QQmlPropertyData**) () at /lib64/libQt5Qml.so.5
#7 0x00007fffee85f664 in QV4::QQmlContextWrapper::lookupInParentContextHierarchy(QV4::Lookup*, QV4::ExecutionEngine*, QV4::Value*) () at /lib64/libQt5Qml.so.5
```
# Scenario
+ Sip account add a contact
+ Right click
+ remove contactSwarm-chatAlbert Babí OllerAlbert Babí Ollerhttps://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/567No ringing tone feedback when making a SIP call2023-04-24T15:32:32ZMaxim CournoyerNo ringing tone feedback when making a SIP callBug report form
---------------
## Describe your environment
Please specify the following:
- Ring version: 20190103 (F-Droid)
- Device model: HTC U-Play
- Android version: 6.0
- What build you are using: F-droid
## Steps to r...Bug report form
---------------
## Describe your environment
Please specify the following:
- Ring version: 20190103 (F-Droid)
- Device model: HTC U-Play
- Android version: 6.0
- What build you are using: F-droid
## Steps to reproduce
- Can you reproduce the bug: at will
- Steps:
1. Configure SIP account
1. Place call to a landline phone number
1. Observe the lack of audible feedback while the other side is ringing.
- Actual result:
No audible feedback of ringing callee.
- Expected result:
There should be some ringing sound as a feedback to the caller.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/965Please have way to display caller id2020-06-23T02:04:39ZShem PasambaPlease have way to display caller idWould it be possible to display the caller id of the sip account instead of just the number?Would it be possible to display the caller id of the sip account instead of just the number?Ming Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/179[GNOME] Dialpad2021-03-10T17:54:39ZJami 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 Zhanghttps://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 Zhanghttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/180Missed call after one call2020-12-15T21:14:36ZMuhammad FattahiMissed call after one callUsing freepbx 15(using Asterisk 16)
After one call i must kill process jami then try to call.
Otherwise after one call any calls are missed call and nothing happened.
Other client like linphone work currectly.
OS: Ubuntu 18.04Using freepbx 15(using Asterisk 16)
After one call i must kill process jami then try to call.
Otherwise after one call any calls are missed call and nothing happened.
Other client like linphone work currectly.
OS: Ubuntu 18.04Ming Rui ZhangMing Rui Zhanghttps://git.jami.net/savoirfairelinux/jami-daemon/-/issues/186SIP proxy field is not used correctly2020-12-13T23:57:25ZJami BotSIP proxy field is not used correctlyIssue generated from Tuleap's migration script.
**Originally submitted by: Raman Gupta (rocketraman)**
A detailed description of the bug. Use following fields for description field:
\* Environment:
Ring v1.0.0 on Linux Fedora 25 ...Issue generated from Tuleap's migration script.
**Originally submitted by: Raman Gupta (rocketraman)**
A detailed description of the bug. Use following fields for description field:
\* Environment:
Ring v1.0.0 on Linux Fedora 25
\* Reproduce steps:
Using the hostname and proxy fields to connect to a domain mydomain.com and proxy server proxy.versature.com. The configuration should set up Ring to connect to proxy.versature.com, passing the mydomain.com in the subsequent SIP request(s).
\* Expected result:
I would expect ring to connect successfully using this configuration. As a point of comparison, Zoiper will connect successfully.
\* Actual result:
The actual result is that Ring attempts to make a SIP connection to the hostname, ignoring the proxy field. The debug logs at SIPLOGLEVEL=4 are:
```
[1496261573.159|18869|sipaccount.cpp:698 ] doRegister mydomain.com
[1496261573.159|18869|sipvoiplink.cpp:1220 ] try to resolve 'mydomain.com' (port: 0)
16:12:53.159 resolver.c Transmitting 38 bytes to NS 0 (127.0.0.1:53): DNS SRV query for \_sip.\_udp.mydomain.com: Success
16:12:53.190 \_sip.\_udp.redo DNS SRV resolution failed for \_sip.\_udp.mydomain.com: DNS "Name Error" (PJLIB\_UTIL\_EDNS\_NXDOMAIN)
16:12:53.190 \_sip.\_udp.redo DNS SRV resolution failed for \_sip.\_udp.mydomain.com, trying resolving A/AAAA record for mydomain.com
16:12:53.190 resolver.c Transmitting 28 bytes to NS 0 (127.0.0.1:53): DNS A query for mydomain.com: Success
16:12:53.190 resolver.c Transmitting 28 bytes to NS 0 (127.0.0.1:53): DNS AAAA query for mydomain.com: Success
[1496261573.220|18869|sipaccount.cpp:815 ] Creating transport
16:12:53.220 udp0x18cc530 SIP UDP transport started, published address is 192.168.1.6:5062
[1496261573.220|18869|siptransport.cpp:357 ] Created UDP transport on default : 0.0.0.0:5062
[1496261573.221|18869|siptransport.cpp:82 ] SipTransport@0x190acc0 {tr=0x1919ec8 {rc=2}}
[1496261573.221|18869|sip\_utils.cpp:87 ] Adding route proxy.versature.com
[1496261605.223|18869|sipaccount.cpp:1033 ] SIP registration failed, status=408 (Request Timeout)
[1496261605.223|18869|sipaccount.cpp:2018 ] Scheduling re-registration retry in 53 seconds..
[1496261658.316|18869|sip\_utils.cpp:87 ] Adding route proxy.versature.com
[1496261690.321|18869|sipaccount.cpp:1033 ] SIP registration failed, status=408 (Request Timeout)
[1496261690.321|18869|sipaccount.cpp:2018 ] Scheduling re-registration retry in 302 seconds..
```
IF the system is configured with \_sip.\_udp SRV records on the domain provided in the hostname to point to the proxy, and the proxy field is left blank, then Ring successfully looks up the SRV record and connects to the proxy given there. Explicitly providing the proxy does not work as shown above.
The way I think it should work (and the way I think Zoiper works) is that you have a "domain/hostname" [1] setting and a "proxy" setting. The logic would be:
1) If proxy is set, Ring connects to the proxy and then passes username@domain in the SIP header.
2) If proxy is not set, then Ring looks for the proxy in the DNS SRV records for domain and if it exists, Ring connects to it and then passes username@domain to it.
3) Lastly, if the DNS SRV record does not exist, Ring attempts to connect to the A record of the domain/hostname setting, and again passes username@domain to it.
That way the auth information is completely configurable (username + domain) and the server that handles the request is completely configurable (proxy). The server to physically connect to is configurable either by DNS record on the domain, OR by explicit configuration in the proxy field.
[1] The "domain/hostname" config value would replace the current "hostname" config value.
The codepath to make the SIP connection as described above already exists -- its just that its impossible to configure the UI currently to trigger it. Given an SRV record of \_sip.\_udp.mydomain.com pointing to proxy.versature.com port 5060, Ring makes a connection to proxy.versature.com and passes all information with domain @mydomain.com. Here is a trace https://pastebin.com/raw/5z39MRu8.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.https://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-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-android/-/issues/680[sip] SIP text messages (SIMPLE) are not received/sent.2022-12-02T16:31:03ZMaxim Cournoyer[sip] SIP text messages (SIMPLE) are not received/sent.Tested using the latest F-Droid build (20191208-01):
1. Register a SIP account, e.g. from the voip.ms provider, which implements a SMS <-> SIP SIMPLE gateway.
2. Send a text message using another SIP account, or alternatively if using v...Tested using the latest F-Droid build (20191208-01):
1. Register a SIP account, e.g. from the voip.ms provider, which implements a SMS <-> SIP SIMPLE gateway.
2. Send a text message using another SIP account, or alternatively if using voip.ms with SMS support, a SMS from a mobile phone.
Actual: No SIP message gets received nor sent.
Expected: SIP messages should be received/sent.
FWIW, I believe the protocol used for the text messages over SIP is SIMPLE (see: https://en.wikipedia.org/wiki/SIMPLE_%28instant_messaging_protocol%29).Antoine NoreauSébastien BlinAntoine Noreauhttps://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/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-client-gnome/-/issues/1157Outgoing call sometimes not working2020-04-15T09:46:12ZmokkinOutgoing call sometimes not workingStarting in the morning incoming and outgoing calls are working. After a couple hours working suddenly the ability to place an outgoing call disappears. Incoming is still working.
```
[New Thread 0x7fffc37fe700 (LWP 2540)]
"slotCallStat...Starting in the morning incoming and outgoing calls are working. After a couple hours working suddenly the ability to place an outgoing call disappears. Incoming is still working.
```
[New Thread 0x7fffc37fe700 (LWP 2540)]
"slotCallStateChanged (call: 4894118646942841), from Suchen to Verbindet"
"slotCallStateChanged (call: 4894118646942841), from Verbindet to Abgeschlossen"
"slotCallStateChanged (call: 4894118646942841), from Abgeschlossen to Abgeschlossen"
(jami-gnome:2302): Gtk-CRITICAL **: 16:40:21.706: gtk_application_uninhibit: assertion 'cookie > 0' failed
Renderer "4894118646942841" not found
[Thread 0x7fff5bffd700 (LWP 2507) exited]
[New Thread 0x7fff5bffd700 (LWP 2549)]
[Thread 0x7fffc37fe700 (LWP 2540) exited]
```
On the asterisk/FreePBX system I cannot even see the call attempt in the logfiles.
How can I log jami SIP locally?https://git.jami.net/savoirfairelinux/jami-daemon/-/issues/213No SIP registration2020-11-20T00:22:23Zedonkey2001-jamiNo SIP registrationI'm running Fedora 31 x86_64, and I've installed Jami from the jami.net rpm repo, version 20200414.
I'm trying to use my SIP account with Jami. It is from an ISP also providing VoIP calls.
Since I don't know whether this is specific to ...I'm running Fedora 31 x86_64, and I've installed Jami from the jami.net rpm repo, version 20200414.
I'm trying to use my SIP account with Jami. It is from an ISP also providing VoIP calls.
Since I don't know whether this is specific to my provider, or a more general issue, I'm providing specific details about this setup.
My ISP setup is: UDP transport, no TLS nor SRTP, they provide an username/password and one should connect via their proxy. I'm not sure one can even connect to it from outside their network.
They have two proxies actually (IPv4 only):
srvrm.p.ims.tiscali.net (SRV record)
core1.p.ims.tiscali.net (A record)
213.205.21.8 (actual IP address)
srvmi.p.ims.tiscali.net (SRV record)
core2.p.ims.tiscali.net (A record)
94.32.130.112 (actual IP address)
These are the connection settings I'm using on a Grandstream ATA where I can place and receive calls:
Primary SIP Server: ims.tiscali.net
Outbound Proxy: 213.205.21.8
SIP User ID: $MYPHONENUMBER
Authenticate ID: $MYPHONENUMBER@ims.tiscali.net
Password: $MYPASSWORD
These are the connection settings I'm trying to use with Jami:
Name: $MYPHONENUMBER
SIP Server: I've tried using all of SRV name, A name and IP address.
Password: $MYPASSWORD
Proxy: I've tried using all of SRV name, A name and IP address.
What should happen: I should be able to register with my provider and be able to place and receive calls.
What happens actually: Jami is not sending a single SIP packet anywhere. At most it's performing DNS resolution via SRV or A records, receiving responses with the correct IP address.
I've started Jami with "jami-gnome -d" but I saw no relevant debug info. I'm running wireshark on the box running Jami, and by filtering for "udp.port==5060 || sip" I see no packets.Ming Rui ZhangMing Rui Zhanghttps://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-client-qt/-/issues/234Send a text message, SIP account can't call after this2022-11-01T17:12:46ZMing Rui ZhangSend a text message, SIP account can't call after thisPossible sip failure. For SFL's SIP test account
if you send a text message I think the account can't call after thisPossible sip failure. For SFL's SIP test account
if you send a text message I think the account can't call after thisSébastien BlinAntoine NoreauSébastien Blinhttps://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 Kostiuk