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   alt.bbs.synchronet      Rob Swindell fetishistic worship forum      26,287 messages   

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   Message 26,000 of 26,287   
   bbsing to Charles Blackburn   
   Re: dialup issues   
   18 Sep 24 11:33:31   
   
   From: user@net396.fidonet.org   
      
   + User FidoNet address: 1:229/317   
   AREA:ALT-BBS-SYNCHRONET   
      
   On 08 Oct 2022, Charles Blackburn said the following...   
      
    CB> From: "Charles Blackburn"    
    CB>   
    CB> Hi ALL   
    CB>   
    CB> im struggling getting some SIP dialups going. sometimes they work, but   
    CB> most often they dont.   
    CB>   
    CB> for example....   
    CB>   
    CB> I have two USR 56k Sportsters on a grandstream SIP thingy,   
    CB>   
    CB> one minute i'll connect fine  and then the next time i get something   
    CB> like the following:   
    CB>   
    CB> CONNECT 24000/V34/NONE   
    CB> Ü]êC                       ìs   
    CB>   
    CB> **** BUNCH OF OTHER GARBAGE TILL I HANG UP **   
    CB>   
    CB>   
    CB> other times it will work and i get to the login prompt and it will just   
    CB> "hang"   
    CB>   
    CB>   
    CB> i've even just tried tying minicom to mini com as far as software goes,   
    CB> and it's the same.   
    CB>   
    CB> i'm sure it has something to do with either the init strings or (most   
    CB> likely) the grandstream.   
    CB>   
    CB>   
    CB> i'm just at a loss. I havent figured out a way to force a specific DTE   
    CB> speed (57k6 or 115k2) and also a way to force it not to do compression   
    CB> or error correction. Ideally i want to be able to have a max connect at   
    CB> 57600 and no v92, v44 et al   
    CB>   
    CB> anyone got any thoughts (other than pay for a bunch of real pots lines)   
    CB> LOL   
    CB>   
    CB> regards   
    CB>   
    CB> Charles Blackburn   
      
   Hi Charles,   
      
   Did you ever get your USR 56k to connect via your SIP setup?   
      
   I'm looking for info how to get my modems to connect togther.   
      
   I have a cisco ata spa122 and I'm using asterisk and freepbx (front end).   
   I have two extensions and they each can call and detect rings but never   
   connect.   
      
   I'm using pjsip (not legacy).   
      
   Someone has to have found the solution to this issue with modems over a VOIP   
   system.   
      
   .. The only place I want data loss is on my credit card!   
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