I have a similar set up, I don't have a separate sip phone, but I have the same exact problem with the line 1. I don't know if my config files aren't right, but I can't transfer between exts yet, but my issues is with line one on an incoming call from an X100P as well.
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Chris Higgins > Sent: Sunday, January 25, 2004 3:54 PM > To: [EMAIL PROTECTED] > Cc: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No > Sound being passed to caller > > > 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/aster> isk-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
