Re: [Asterisk-Users] Polycom config and DTMF problems
On Wednesday 05 October 2005 08:02, Douglas E. Warner wrote: > I'm going to try to capture the RTP stream and > see if it's being sent inband, So, I've captured the RTP stream and played it back (nothing but payload type 0 packets, btw), and there was no DTMF in there at all - I held down the zero key a lot of the time, and talked into the handset the rest of the time; the time where I was talking could be heard, but there was nothing during the times when I was holding down the zero key. Could this be a firmware bug? Is anyone else having problems w/ Sip 1.5.3? Defective phone? -Doug -- Douglas E. Warner<[EMAIL PROTECTED]> Network Engineer CTI Networks, Inc. http://www.ctinetworks.com+1 717 975 9000 pgpCNSyPgq8KD.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Polycom config and DTMF problems
I have dtmfmode=rfc2833 in my sip.conf for each registration, and this gobbledegook (hey - that passed a spellcheck!) in my ipmid.cfg: tone.dtmf.offTime="50" tone.dtmf.chassis.masking="0" tone.dtmf.stim.pac.offHookOnly="0" tone.dtmf.viaRtp="1" tone.dtmf.rfc2833Control="1" tone.dtmf.rfc2833Payload="101"/> I'll save you the trouble of checking - it's the same as yours ;-) Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Oct 5, 2005, at 5:02 AM, Douglas E. Warner wrote: On Tuesday 04 October 2005 18:04, Anthony Rodgers wrote: > I found the best reference to be the SoundPoint IP / SoundStation IP > Admin Guide - SIP 1.5 from the Polycom web site - > http://www.polycom.com/common/pw_item_show_doc/1,1276,4349,00.pdf. > You're right - that admin guide is much more useful that I had initially thought - thanks! > Not sure about the DTMF issue - I used the config files at > http://www.krisk.org/asterisk/pcom/, if that helps Yeah, I have no idea either. I'm going to try to capture the RTP stream and see if it's being sent inband, but I clearly have my sip.cfg file set to rfc2833: And I've already tried "dtmfmode=inband" in my asterisk sip.conf, so I'm not sure what's going on. -Doug -- Douglas E. Warner <[EMAIL PROTECTED]> Network Engineer CTI Networks, Inc. http://www.ctinetworks.com +1 717 975 9000 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Polycom config and DTMF problems
On Tuesday 04 October 2005 18:04, Anthony Rodgers wrote: > I found the best reference to be the SoundPoint IP / SoundStation IP > Admin Guide - SIP 1.5 from the Polycom web site - > http://www.polycom.com/common/pw_item_show_doc/1,1276,4349,00.pdf. > You're right - that admin guide is much more useful that I had initially thought - thanks! > Not sure about the DTMF issue - I used the config files at > http://www.krisk.org/asterisk/pcom/, if that helps Yeah, I have no idea either. I'm going to try to capture the RTP stream and see if it's being sent inband, but I clearly have my sip.cfg file set to rfc2833: And I've already tried "dtmfmode=inband" in my asterisk sip.conf, so I'm not sure what's going on. -Doug -- Douglas E. Warner<[EMAIL PROTECTED]> Network Engineer CTI Networks, Inc. http://www.ctinetworks.com+1 717 975 9000 pgpSma0yCYIDV.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Polycom config and DTMF problems
Actually, there is a very good guide on voip-info.org at http://www.voip-info.org/tiki-index.php?page=Polycom+Soundpoint+IP+501 --- Anthony Rodgers <[EMAIL PROTECTED]> wrote: > Hi Douglas, > > I found the best reference to be the SoundPoint IP / SoundStation IP > Admin Guide - SIP 1.5 from the Polycom web site - > http://www.polycom.com/common/pw_item_show_doc/1,1276,4349,00.pdf. > > Not sure about the DTMF issue - I used the config files at > http://www.krisk.org/asterisk/pcom/, if that helps > > Regards, > -- > Anthony Rodgers > Business Systems Analyst > District of North Vancouver > Web: http://www.dnv.org > RSS Feed: http://www.dnv.org/rss.asp > > On Oct 4, 2005, at 1:43 PM, Douglas E. Warner wrote: > > > I've just got a batch of 301s and 501s in and am trying to get them to > > work. > > I'd like to manually configure everything via FTP rather than the web > > or > > phone interfaces, but I can't seem to find a good source of > > definitions for > > all the options in the sip.cfg or phoneX.cfg files. Anyone know of > > any? > > > > Also, I'm having quite the problem getting the Polycom SP 501 to send > > *any* > > DTMF. Running tethereal, I'm just seeing G.711 packets; no other RTP > > packets > > being sent (using RFC2833, supposedly). > > > > Relevant info: > > Asterisk 1.2.0 beta1 > > PolyCom SP 501, sip 1.5.3, bootrom 2.6.2 > > > > Let me know what other info is needed to debug this, or any insight > > anyone can > > provide would be great. > > > > -Doug > > > > -- > > Douglas E. Warner<[EMAIL PROTECTED]> Network Engineer > > CTI Networks, Inc. http://www.ctinetworks.com+1 717 975 9000 > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users> ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Polycom config and DTMF problems
Hi Douglas, I found the best reference to be the SoundPoint IP / SoundStation IP Admin Guide - SIP 1.5 from the Polycom web site - http://www.polycom.com/common/pw_item_show_doc/1,1276,4349,00.pdf. Not sure about the DTMF issue - I used the config files at http://www.krisk.org/asterisk/pcom/, if that helps Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Oct 4, 2005, at 1:43 PM, Douglas E. Warner wrote: I've just got a batch of 301s and 501s in and am trying to get them to work. I'd like to manually configure everything via FTP rather than the web or phone interfaces, but I can't seem to find a good source of definitions for all the options in the sip.cfg or phoneX.cfg files. Anyone know of any? Also, I'm having quite the problem getting the Polycom SP 501 to send *any* DTMF. Running tethereal, I'm just seeing G.711 packets; no other RTP packets being sent (using RFC2833, supposedly). Relevant info: Asterisk 1.2.0 beta1 PolyCom SP 501, sip 1.5.3, bootrom 2.6.2 Let me know what other info is needed to debug this, or any insight anyone can provide would be great. -Doug -- Douglas E. Warner <[EMAIL PROTECTED]> Network Engineer CTI Networks, Inc. http://www.ctinetworks.com +1 717 975 9000 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom config and DTMF problems
I've just got a batch of 301s and 501s in and am trying to get them to work. I'd like to manually configure everything via FTP rather than the web or phone interfaces, but I can't seem to find a good source of definitions for all the options in the sip.cfg or phoneX.cfg files. Anyone know of any? Also, I'm having quite the problem getting the Polycom SP 501 to send *any* DTMF. Running tethereal, I'm just seeing G.711 packets; no other RTP packets being sent (using RFC2833, supposedly). Relevant info: Asterisk 1.2.0 beta1 PolyCom SP 501, sip 1.5.3, bootrom 2.6.2 Let me know what other info is needed to debug this, or any insight anyone can provide would be great. -Doug -- Douglas E. Warner<[EMAIL PROTECTED]> Network Engineer CTI Networks, Inc. http://www.ctinetworks.com+1 717 975 9000 pgp0jyh5XuT1a.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
