I added the music on hold feature. I answer on line 1, flash for a sec and come back and transmission both way is fine, just can't answer initially.
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Miguel Cavazos > Sent: Sunday, January 25, 2004 11:55 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No > Sound beingpassed to caller > > > 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/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
