mumsi Registeration to SIP

[satti001]

hello I have installed mumsi with all the dependencies and mumble-server already running working. i have created user in asterisk with name mumsi and password foobar. When i start mumsi its say request time out not regsitering.

I need help to setup and test call from asterisk softphone like zoiper to mumble server user..
Any help would be appreciated.
Thanks in advance

[mrscotty]

Are you running asterisk and mumsi on the same server? You might be having UDP port conflicts.

I'm using mumsi to connect to my SIP account at sipgate.de and it works like a charm. In fact, I'm able to run 5 connections to the same SIP account, which allows for multiple regular telephone callers to dial into the Mumble channel at the same time as if it were a telco bridge.

[satti001]

Thanks for your response I will change mumsi port and will get back there soon .. Many thanks

[satti001]

Hello,
I am still having problem with registration mumsi sip setup. if any one can please write down sample setup to just setup one sip extension to make call to mumble.

Asterisk mumsi and mumble all three are running on same machine. can any one suggest that what exactly i am doing wrong.

Regards

[satti001]

ok finally got it working :)

[satti001]

hello mrscotty

I'm using mumsi to connect to my SIP account at sipgate.de and it works like a charm. In fact, I'm able to run 5 connections to the same SIP account, which allows for multiple regular telephone callers to dial into the Mumble channel at the same time as if it were a telco bridge.

i would love to know your configurations if you want to tell please regarding multiple connections to the same sip.

[mrscotty]

Great to hear you got it working. Were you able to assign alternate SIP ports or did you have to separate asterisk and mumsi on different servers?

To use the multi-session support, take a look at my pull request from May [https://github.com//pull/18]. It adds a couple of conference-call-like features such as playing WAV files and prompting the caller for a PIN via DTMF. An example config file is included.

The idle mumsi users in Mumble sit in an extra "offline" channel (the name is configurable) and when the caller dials in, they are prompted for a PIN and you can have them assigned to a specific channel based on the PIN they enter.

If you use ACLs to manage your user access on your Mumble server, you'll need to generate user certificates, one per mumsi user, and configure them in the mumsi config. Once logged in, just add the mumsi users to the groups in your ACLs as needed. You will not be able to have the callers moved to a specific channel via PIN unless the mumsi user has the corresponding ACL access.

[satti001]

Hi i am stuck at adding multiple mumsi users. How to add multiple asterisk users to access mumble. Currently I have default one mumsi user register to asterisk and mumble and all asterisk users are calling using mumsi user. How to access mumble by all asterisk user identifiers not only single mumsi.

[satti001]

Any instructions regarding adding multiple users in config.ini file in mumsi

[satti001]

@mrscotty I have tried to compile new commit files regarding multi-sessions support but unable to compile it says some functions not declared in mumlib. Where I can find new functions added to mumlib as well to compile and test new features

[mrscotty]

Hi, You need one Mumble user per simultaneous session you want to support. Depending on your Asterisk configuration, you'll either need just a single user that is allowed to have multiple SIP sessions or one Asterisk user per Mumble user. In my configuration, my account at sipgate.de supports multiple connections, so I use this connected to 5 Mumble users. Scott

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On Dec 20, 2017, at 12:53, satti001 ***@***.***> wrote: Hi i am stuck at adding multiple mumsi users. How to add multiple asterisk users to access mumble. Currently I have default one mumsi user register to asterisk and mumble and all asterisk users are calling using mumsi user. How to access mumble by all asterisk user identifiers not only single mumsi. — You are receiving this because you commented. Reply to this email directly, view it on GitHub, or mute the thread.

[satti001]

I want one asterisk user per mumble user but don't know how to map that using mumsi configuration

[mrscotty]

Hi, Ah, sorry. You'll need a couple of additional commits that were submitted last May in my pull request [slomkowski/mumlib#14]. When you build mumlib, be sure you have a recent version of pjproject--the original mumlib code uses a version that's newer than the debian release I did my initial work on. Best regards, Scott

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On Dec 20, 2017, at 12:57, satti001 ***@***.***> wrote: @mrscotty I have tried to compile new commit files regarding multi-sessions support but unable to compile it says some functions not declared in mumlib. Where I can find new functions added to mumlib as well to compile and test new features — You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub, or mute the thread.

[satti001]

Thanks for that will be back after testing but the documentation is really limited regarding configuration can you please tell the mapping users just a sample syntax or may be Readme file

[mrscotty]

Sorry, but after looking at the source, I noticed that configuring multiple SIP users is not supported. I had implemented just the multi-session support for a single SIP user because this is what I needed and it allows me to have just a single dial-in phone number for our telco bridge. You could try running multiple mumsi sessions, but you'll probably have to make sure your SIP UDP ports don't conflict. A better solution is to configure Asterisk to allow multiple sessions for the single user.

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On Dec 20, 2017, at 12:59, satti001 ***@***.***> wrote: I want one asterisk user per mumble user but don't know how to map that using mumsi configuration — You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub, or mute the thread.

[satti001]

Multiple sessions for single user means sip trunking like solution is that you are suggesting. Sorry for lot of questions