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SIP und Multipath

WolfgangS

Hallo zusammen,

SIP, Multipath und ich werden wohl keine Freunde.....

Also.. 

Telefonie funktioniert soweit, bis auf die Kleinigkeit, dass ich mein Gegenüber nach 1 Min. nich mehr höre.

Jetzt habe ich die div. Threads hier durch und wahrscheinlich mehr kaputt gemacht als Probleme gelöst

2 Uplinks 

1x DSL > festes IP Netz für meine Mail und Webserver

1x Glasfaser für Familie , Streamen und co.

Ich häng mal ein paar Bilder rein. Ich glaube ich hab das so verbastelt, dass es ein Wunder ist, dass es überhaupt noch läuft.

Mag mich da mal jemand in die richtige Richtung schieben ?!

Danke !



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  • 0

    Hallo Wolfgang,

    (Sorry, my German-speaking brain isn't creating thoughts at the moment. Frowning2)

    Before commentating on your question, I admire the self-deprecating humor in your post and your use of familiar wording instead of strict Hochdeutsch.  I laughed while reading your creative post.  Have you ever thought of doing stand-up comedy for extra fun?

    In Multipath rules 1 and 3, why use "Any" instead of "Internet" as the destination?

    In your DNATs:

    • What are sip. and tel.?  subnets?
    • Why use the "(Network)" objects in of the "(Address)" objects?
    • You don't need to change "SIP" to "SIP" - see #4 in Rulz.

    MfG - Bob (Bitte auf Deutsch weiterhin.)

     
    Sophos UTM Community Moderator
    Sophos Certified Architect - UTM
    Sophos Certified Engineer - XG
    Gold Solution Partner since 2005
    MediaSoft, Inc. USA
  • 0 in reply to BAlfson

    Hi Bob,

    thanks for the compliment, but just "normal" conversation for me... i mean i think so...

    well, to be honest, i never got the point of "any" and "internet". so i guess that's why i used "any". but after your question i guess the firewall changes routing for any or internet.

    tel.myinternal.domain = 192.168.0.x/24

    sip = sip.plusnet.de

    read that in a thread where a user put in all sip hosts/ip's from the sip provider into a netgroup. so i tought i do the same and see what happens.

    well, nothing happend. was the same as bevor. after 1 min. can't hear anything on the phone.

    So i guess i can dump that.

  • 0 in reply to WolfgangS

    Calls that are usually dropped after one minute are some kind of NAT issue.

    In one of your DNAT screens, can you edit one and screenshot your edit window and paste here?

    OPNSense 64-bit | Intel Xeon 4-core v3 1225 3.20Ghz
    16GB Memory | 500GB SSD HDD | ATT Fiber 1GB
    (Former Sophos UTM Veteran, Former XG Rookie)

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  • 0 in reply to WolfgangS

    Two things you can try here:

    1.  Remove the entry in the 'And the service to:', as it really doesn't  need to be in there.  I've had this create a problem for me a long while back.

    2. If that doesn't work, then uncheck the Automatic firewall rule.

    OPNSense 64-bit | Intel Xeon 4-core v3 1225 3.20Ghz
    16GB Memory | 500GB SSD HDD | ATT Fiber 1GB
    (Former Sophos UTM Veteran, Former XG Rookie)

  • 0 in reply to Amodin

    Well, no success....

    After 1:24 i cant hear anything on the phone

    Did 1 and 2 with checked and unchecked with same result.

  • 0 in reply to WolfgangS

    I wonder if there is something being blocked while you are on the phone.  Can you look in your logs while testing out SIP and see if you get anything being blocked or something not right in the logs?

    OPNSense 64-bit | Intel Xeon 4-core v3 1225 3.20Ghz
    16GB Memory | 500GB SSD HDD | ATT Fiber 1GB
    (Former Sophos UTM Veteran, Former XG Rookie)

  • 0 in reply to Amodin

    Hi @all, 

    sorry for the later answer, but i was down with corona.

    SO, u guys won't belive it, so i make it sort.

    i had a small PI , forgotten in my network which had the same IP as ths SIP device. yeah.... 

    and best part, i did not put that thing into my ip adress list for IPs@home 

    So solved, because of....

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