Re: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water
On Tue, Feb 17, 2004 at 09:24:53AM -0500, Rana Dutt wrote: I was able to solve the audio quality problem by going to www.grandstream.com/BETATEST and downloading the latest beta firmware, version 1.0.4.46. This version has at least one problem I'm aware of : when you dial an external number (via an ISDN PRI), if you hangup the SIP phone the message doesn't reach Asterisk, which keeps on ringing the ISDN leg. -- Nicolas Bougues Axialys Interactive ___ 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
RE: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water
On Tue, 17 Feb 2004, Rana Dutt wrote: I was able to solve the audio quality problem by going to www.grandstream.com/BETATEST and downloading the latest beta firmware, version 1.0.4.46. I wish their Beta Releases actually had a file that showed what changes/fixes/updates have been made to the firmware. It's kind of like the blind leading the blind. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Tuesday, February 17, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water Hi! You need to add this to EACH and EVERY sip user, not just in [general]: disallow=all allow=ulaw allow=alaw See also: http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone Cheers, Philipp [200] type=friend username=200 host=dynamic context=home reinvite=no canreinvite=no [201] type=friend username=201 host=dynamic context=home reinvite=no canreinvite=no I turned on sip debug, and noticed the following in the output: v=0 s=SIP Call c= IN IP4 192.168.2.29 m= audio 5004 RTP/AVP 0 a=rptmap:0 PCMU/8000 a=ptime:20 Found audio format UNKN Found description format PCMU Capabilities: us - 4, them 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined 0 Does anyone know why this could be happening? Thanks, Ron ___ 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/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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
RE: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water
The release before the latest has a list. And the latest release has actual fixes for asterisk with phones unregistering and now supports config by MAC address. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Wednesday, February 18, 2004 7:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water On Tue, 17 Feb 2004, Rana Dutt wrote: I was able to solve the audio quality problem by going to www.grandstream.com/BETATEST and downloading the latest beta firmware, version 1.0.4.46. I wish their Beta Releases actually had a file that showed what changes/fixes/updates have been made to the firmware. It's kind of like the blind leading the blind. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Tuesday, February 17, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water Hi! You need to add this to EACH and EVERY sip user, not just in [general]: disallow=all allow=ulaw allow=alaw See also: http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone Cheers, Philipp [200] type=friend username=200 host=dynamic context=home reinvite=no canreinvite=no [201] type=friend username=201 host=dynamic context=home reinvite=no canreinvite=no I turned on sip debug, and noticed the following in the output: v=0 s=SIP Call c= IN IP4 192.168.2.29 m= audio 5004 RTP/AVP 0 a=rptmap:0 PCMU/8000 a=ptime:20 Found audio format UNKN Found description format PCMU Capabilities: us - 4, them 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined 0 Does anyone know why this could be happening? Thanks, Ron ___ 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/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water
Is this also true for iax.conf ? On Tue, 2004-02-17 at 12:14, Philipp von Klitzing wrote: Hi! You need to add this to EACH and EVERY sip user, not just in [general]: disallow=all allow=ulaw allow=alaw See also: http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone Cheers, Philipp [200] type=friend username=200 host=dynamic context=home reinvite=no canreinvite=no [201] type=friend username=201 host=dynamic context=home reinvite=no canreinvite=no I turned on sip debug, and noticed the following in the output: v=0 s=SIP Call c= IN IP4 192.168.2.29 m= audio 5004 RTP/AVP 0 a=rptmap:0 PCMU/8000 a=ptime:20 Found audio format UNKN Found description format PCMU Capabilities: us - 4, them 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined 0 Does anyone know why this could be happening? Thanks, Ron ___ 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 -- | http://www.opusvl.com | T: 08717 50 40 02 | F: 08717 50 40 03 | E: [EMAIL PROTECTED] ___ 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
RE: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water
I was able to solve the audio quality problem by going to www.grandstream.com/BETATEST and downloading the latest beta firmware, version 1.0.4.46. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Tuesday, February 17, 2004 7:14 AM To: [EMAIL PROTECTED] Subject:Re: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water Hi! You need to add this to EACH and EVERY sip user, not just in [general]: disallow=all allow=ulaw allow=alaw See also: http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone Cheers, Philipp [200] type=friend username=200 host=dynamic context=home reinvite=no canreinvite=no [201] type=friend username=201 host=dynamic context=home reinvite=no canreinvite=no I turned on sip debug, and noticed the following in the output: v=0 s=SIP Call c= IN IP4 192.168.2.29 m= audio 5004 RTP/AVP 0 a=rptmap:0 PCMU/8000 a=ptime:20 Found audio format UNKN Found description format PCMU Capabilities: us - 4, them 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined 0 Does anyone know why this could be happening? Thanks, Ron ___ 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/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speech between Grandstream phones sounds like talking under water
When I make a simple phone call from one Budgetone 101 to another, the speech sounds slurred and slow, sort of like the person is talking under water. Both phones and the Asterisk server are on the same subnet. Both phones are configured to use the PCMU (ulaw) codec as first choice, and the Voice Frames per TX parameter is set to 2. Incidentally, if I directly IP dial from one phone to the other (bypassing Asterisk) the speech sounds excellent. I'm running a CVS build from Feb. 1, 2004, and there is a Digium X100P card with one incoming CO line in my machine. The first part of my sip.conf looks like this: [general] port=5060 binaddr=0.0.0.0 disallow=all allow=ulaw [200] type=friend username=200 host=dynamic context=home reinvite=no canreinvite=no [201] type=friend username=201 host=dynamic context=home reinvite=no canreinvite=no I turned on sip debug, and noticed the following in the output: v=0 s=SIP Call c= IN IP4 192.168.2.29 m= audio 5004 RTP/AVP 0 a=rptmap:0 PCMU/8000 a=ptime:20 Found audio format UNKN Found description format PCMU Capabilities: us - 4, them 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined 0 Does anyone know why this could be happening? Thanks, Ron attachment: winmail.dat
