It would probably help if you used a packet sniffer (eg, ethereal) to look at the traffic, or at least provide the list with a useful clue other then it doesn't work.
------------------------ > same here, when i recive an incoming call from x100p to line 1 on > sipura, i can hear them but people can't hear me im using 1.0.24 on my > firmware > > Miguel > On Sun, 2004-01-25 at 20:54, Chris Higgins wrote: > > Frankie Gravato wrote: > > > > > > > > I've been beating my head for 5 hours to figure out why my asterisk > > > server or sipura isn't passing my voice over to the caller. It seems i > > > can hear the caller but they can't hear me it seems either the > > > asterisk or the sipura isn't passing this information. > > > > > > Here's my setup specs > > > > > > asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - > > > Voicepulse Service and DID's > > > > > > when i get Phone call using the Voicepulse or Pstn the caller can't > > > hear me or barely hear me. The Sipura is running Firmware 1.20 and > > > calls are being passed using Ulaw Codec? Anyone out there in the > > > asterisk community please oh please help me before i do something that > > > my asterisk server won't like. > > > > > > > > > > I just received my Sipura on Friday and have been testing it extensively > > over the weekend. I have noticed an issue similar to what you mention > > above. For the record, the sipura tells me I'm running software version > > 1.0.20. Also, there is NO nat configuration that is causing my problem. > > > > When I receive a call over my X100P and dial my 3 SIP phones (one gs > > budgetone 100, two analong phones through sipura), if I answer the > > analong phone connected to line 1 of the sipura, the caller cannot hear > > anything. I've only noticed this problem in this exact scenario. The > > other situations listed below have no problems whatsoever and audio > > works in both directions: > > > > 1. Call from sipura line 1 to any internal SIP phone. > > 1. Call from any internal SIP phone to sipura line 1. > > 2. Call from sipura line 1 out through X100P. > > 3. Call into my X100P from outside and answer sipura line 2. > > 4. Call into my X100P from outside and answer sipura line 2 and THEN > > transfer to sipura line 1. > > 5. Call into my X100P from outside and answer sipura line 1 (the caller > > cannot hear audio for this leg of the conversation), TRANSFER to any > > other line, and transfer back to sipura line 1. After the second > > transfer, the caller can hear audio from sipura line 1. > > > > I don't know what is special about line 1. I've switched my analog > > phones across the two ports on the sipura to make sure it wasn't one of > > my phones (not that I thought it was anyway). > > > > Frankie, have you tried the same experiment, but pulled your analog > > phone from line 1 and put it in line 2? > > > > Has anyone else seen issues like this with line 1 on a sipura? > > > > Thanks.. > > > > -- Chris > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---------------End of Original Message----------------- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
