Scott Scecina
2006-Oct-18 10:27 UTC
[asterisk-users] random one way audio and noise betweenSIP phoneson same LAN
Giorgio, I'll answer in reverse order: I've not had reports of "noise" from my users. However, when I went down to get the s/w version from the phone that seems to be acting up the most, the user reported that earlier they were actually on a call that was ok then spontaneously dropped the audio. Per my instructions (based on another similar report I read on Digium's site), my user hit a digit on the phone which brought back the caller's audio. I've also had them attempt to put the call on hold, and then resume, but that did not bring the audio back. As far as the S/W versions: One of the phones that acts up (and they all should match): Polycom 501 BootRom: 3.1.3.0131 BootBlock: 2.5.0 SIP: 1.6.6.0036 My phone, on which I've never experienced the problem: Polycom 601 BootRom: 3.1.3.0131 BootBlock: 2.6.0 SIP: 1.6.6.0036 - Scott -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Wednesday, October 18, 2006 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN Hi Scott, seems that we have the same problem...I have canreinvite=no and polycom phones. I do not have cisco switch and qualify=yes but I think that is not the problem. I've got 2 questions: 1) my polycom firmware is: sip.ver: 1.6.5.0043 bootrom.ver: 2_6_2 what are yours? 2) have you got one way calls only or noise on sip calls conversations too? TIA Giorgio Incantalupo P.S.: for configuration/monitoring apps I'm still on it...I hope to find useful tools asap. In case, I'll let you know. Scott Scecina wrote:> I'm having the same "random" problem. > > I have "canreinvite=no" on all extensions. I have "qualify => yes" on all > non-NAT extensions. I do have several NAT extensions, but I've not had > reports of problems from those. 95% of my extensions (all polycom 501/601) > are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. > > In all cases, the called party cannot hear the calling party. The calling > party has the "still ringing" icon on their phone, but can hear the called > party talking. I've got call monitoring turned on, and asterisk isrecording> both sides of the conversation. > > The problem occurs on SIP->SIP and Zap->SIP calls. > > I've tried enabling sip debug on a particular extension that seemed to be > experiencing the problem more than others. However the problem did notoccur> when the debugging was on. > > Sip debug generates so much noise I've been hesitant to turn it on > system-wide. Is there a way I can turn on sip debug and have all that > logging go to a specific file (and not in the asterisk console)? > > Also, are there any other configuration/logging tricks I can try? > > Thank you, > > Scott Scecina > >
Scott Scecina
2006-Oct-18 10:59 UTC
[asterisk-users] random one way audio and noise betweenSIP phoneson same LAN
Yes, 10000-20000 are open. This "phenomenon" is random. Most calls work fine most of the time. - Scott -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Derek Whitten Sent: Wednesday, October 18, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN Scott Scecina wrote:> In all cases, the called party cannot hear the calling party.do you have the RTP ports open?
Giorgio Incantalupo
2006-Oct-19 00:29 UTC
[asterisk-users] random one way audio and noise betweenSIP phoneson same LAN
Hi Scott, so it seems that are polycom phones not working well... have you tried with other IP phones or only with polycom? Giorgio Incantalupo Scott Scecina wrote:> Giorgio, > > I'll answer in reverse order: > > I've not had reports of "noise" from my users. However, when I went down to > get the s/w version from the phone that seems to be acting up the most, the > user reported that earlier they were actually on a call that was ok then > spontaneously dropped the audio. Per my instructions (based on another > similar report I read on Digium's site), my user hit a digit on the phone > which brought back the caller's audio. I've also had them attempt to put the > call on hold, and then resume, but that did not bring the audio back. > > As far as the S/W versions: > > One of the phones that acts up (and they all should match): > > Polycom 501 > BootRom: 3.1.3.0131 > BootBlock: 2.5.0 > SIP: 1.6.6.0036 > > My phone, on which I've never experienced the problem: > > Polycom 601 > BootRom: 3.1.3.0131 > BootBlock: 2.6.0 > SIP: 1.6.6.0036 > > - Scott > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Giorgio > Incantalupo > Sent: Wednesday, October 18, 2006 11:12 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] random one way audio and noise betweenSIP > phoneson same LAN > > Hi Scott, > seems that we have the same problem...I have canreinvite=no and polycom > phones. > I do not have cisco switch and qualify=yes but I think that is not the > problem. > > I've got 2 questions: > 1) my polycom firmware is: > sip.ver: 1.6.5.0043 > bootrom.ver: 2_6_2 > > what are yours? > 2) have you got one way calls only or noise on sip calls conversations too? > > TIA > > > Giorgio Incantalupo > > P.S.: for configuration/monitoring apps I'm still on it...I hope to > find useful tools asap. In case, I'll let you know. > > > Scott Scecina wrote: > >> I'm having the same "random" problem. >> >> I have "canreinvite=no" on all extensions. I have "qualify => yes" on all >> non-NAT extensions. I do have several NAT extensions, but I've not had >> reports of problems from those. 95% of my extensions (all polycom 501/601) >> are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. >> >> In all cases, the called party cannot hear the calling party. The calling >> party has the "still ringing" icon on their phone, but can hear the called >> party talking. I've got call monitoring turned on, and asterisk is >> > recording > >> both sides of the conversation. >> >> The problem occurs on SIP->SIP and Zap->SIP calls. >> >> I've tried enabling sip debug on a particular extension that seemed to be >> experiencing the problem more than others. However the problem did not >> > occur > >> when the debugging was on. >> >> Sip debug generates so much noise I've been hesitant to turn it on >> system-wide. Is there a way I can turn on sip debug and have all that >> logging go to a specific file (and not in the asterisk console)? >> >> Also, are there any other configuration/logging tricks I can try? >> >> Thank you, >> >> Scott Scecina >> >> >> > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
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