I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality.
I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country, but this made no improvement. We had disallow=all allow=g726 in the [general] section of sip.conf. In the section for one of the phones, I added allow=alaw and then inserted Set(SIP_CODEC=alaw) in the relevant part of extensions.conf. For good measure, I also added NoOp(Codec was ${SIP_CODEC}) in the "h" extension. The messages in the Asterisk CLI appeared to show that the audio codec was correctly being set to "alaw", and on hangup I got "Codec was alaw", but there was no improvement to the sound quality. Is there something I am doing wrong, or do I need to get in touch with our SIP trunk provider? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users