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Brahman

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Everything posted by Brahman

  1. I just read in your FAQ: If you want to suggest an improvement to Express Talk or find a bug, let us know at our suggestions site. If we implement the suggestion or fix the bug, we will give you a free upgrade. Does this mean I will receive a free upgrade to the Business Edition? Thank you. Kind Regards, Brahman
  2. Solution to "Error call attempt timed-out." problem with static IP address: If you can make outgoing calls with Xlite or any other VOIP software, but not with Express Talk, get smartsniff software and capture the packets when Xlite sets up a call successfully and in comparison Express Talk not working capture an attempt also (Express Talk snif log for comparison only so that you might find it easier to replace your old values in setup with the new ones). Check Xlite sniffer log (where it works) after "Via: SIP/2.0/UDP " you will find the correct port for static port, f.e.: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.143:44168;branch=z9hG4bK-d87543-4b136f65eb19dc01-1--d87543-;rport=44168;received=192.168.0.27 To: "10000"<sip:10000@sipgate.de>;tag=b11cb9bb270104b49a99a995b8c68544.bf95 Correct port is in this example 10.0.0.143:44168 In "Express Talk - Settings - Network" enter 10.0.0.143 in "Public IP Address (if static)" and 44168 in "External SIP Port". Now go to the end of your sniff log and look in the section below for the numbers after ":" and "/" respectively. Do you see the rport :44168 changing at the end into :5060 in the example below? v=0 o=root 14784 14784 IN IP4 217.10.79.30 s=session c=IN IP4 217.10.79.56 t=0 0 m=audio 39274 RTP/AVP 8 0 3 98 101 a=rtpmap:8 PCMA/8000 <<<<<<<<<<<<<<<<HERE a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=nortpproxy:yes BYE sip:XXXXX@192.168.0.27:44168 SIP/2.0 <<<<<<<<<<<<HERE STILL EXTERNAL PORT Record-Route: <sip:217.10.79.9;lr=on> Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK91ac.f8e0ae33.0 Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK1fa11386;rport=5060 <<<<<<<<<HERE From: "10000"<sip:10000@sipgate.de>;tag=as705193e3 To: "Sipgate"<sip:XXXXXX@sipgate.de>;tag=f03fad16 Contact: <sip:10000@217.10.79.30> Call-ID: M2ZjMGIwMzUxYjY0ODc0MjQ4MDdhYTQ1NjBlOWUxNzc. CSeq: 102 BYE User-Agent: sipgate asterisk Max-Forwards: 16 Content-Length: 0 In this example you now would put 5060 in "Express Talk - Settings - Network - Local SIP port to listen on" and for "Local RTP ports to listen on ..." as well as for "External RTP port starting from ..." field you would enter 8000. Now it should work, because you entered the correct port that the OUTSIDE sip network provider sees (here 44168)which will then get translated by your router's NAT to the standard port (often 5060 and 8000 for the rtp port). Sorry if it is a bit confusing, but it will work when you work with static IPs and get the "Error call attempt timed-out." problem. Hope this solves your problem! Kind Regards, Brahman
  3. Super, thank you. Re: USB-friendly: Wonderful, that is definitely the trend. And encrypted SIP passwords for security ... Re: TIMEOUT: Yes, some callers do not pick up within that time frame. 40 seconds would be great. Great software, appreciate it more and more. Regards, Brahman
  4. Addendum: This only happens when I dial via address book entries in the dial number dropdown list (at the bottom is a list of the address book numbers) not from the real address book Greetinmgs, Brahman
  5. Hi, in the log window as well as in the provider call list of my voip supplier I notice that quite often only the first three digits of the numbers in the address book entries are dialed (v2.02). Could you please fix this bug ASAP? Thank you. Really nice software (but please increase ringer call time-out value and make sip passwords encrypted in an ini file so it is USB stick compatible and secure). Ciao, Brahman
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