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
