Hi Guys, i have recently connected my (working) asterisk 1.2 server, with two 1.4 asterisk servers (one using SIP the other using IAX), since then (i believe) people starts complaining about a high background noise when using the handset on Polycom phones (but when using the speaker it's fine, and i noticed that my self), my question is, can anybody tell me any step to begin diagnosing the issue, to be honest i don't know from where to begin!! -- Abdullah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100325/23409139/attachment.htm
Hi!> i have recently connected my (working) asterisk 1.2 server, with two > 1.4 asterisk servers (one using SIP the other using IAX), since then > (i believe) people starts complaining about a high background noiseThe best idea is probably to start out by looking at the codecs. If you happen to have either gsm or g726 somewhere along your media path then I would strongly suggest you start to get rid of them so see if that eliminates the issue. g726 is one big mess, and gsm had some compiler issues in the past. The next thing to look at is if the issue only appears when IAX is involved. Maybe looking at the Polycom firmware, and release notes in particular (no clue how detailed those are), would also be a good idea. Philipp
Hi Philip, So i looked at the codecs in the device (polycom) it says only G.711 and ulaw can be used, i made an internal call using two phones that are configured just with sip (so IAX not involved) but the static noise is there, i typed show sip peer <username> and this is the only thing i got: Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none) it should be some commands that can give me a better idea about the codecs, if anyone know them, please help! 2010/3/25 Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de>> Hi! > > > i have recently connected my (working) asterisk 1.2 server, with two > > 1.4 asterisk servers (one using SIP the other using IAX), since then > > (i believe) people starts complaining about a high background noise > > The best idea is probably to start out by looking at the codecs. If you > happen to have either gsm or g726 somewhere along your media path then I > would strongly suggest you start to get rid of them so see if that > eliminates the issue. g726 is one big mess, and gsm had some compiler > issues in the past. > > The next thing to look at is if the issue only appears when IAX is > involved. > > Maybe looking at the Polycom firmware, and release notes in particular > (no clue how detailed those are), would also be a good idea. > > Philipp > > > -- > _____________________________________________________________________ > -- 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 >-- Abdullah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/daade096/attachment.htm
Hi!> it should be some commands that can give me a better idea about the > codecs, if anyone know them, please help!Use "sip show channels" and "iax show channels" and look at the Format column. About the Polycom devices: Others will have to help you there. I have no good guess why you might have the issue only on speakerphone, but not in handset mode. Could it maybe be some kind of electrical grounding issue (instead of something caused by transcoding)? Philipp
:) all users are having the same issue, even those connected to this server from abroad! 2010/3/29 Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de>> > i have the same model polycom phone configured with another server > > (asterisk 1.4), and guess what no noise at all. any guess! > > Replace the handset? > > > -- > _____________________________________________________________________ > -- 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 >-- Abdullah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100329/af17a713/attachment.htm