Im working on getting 1.3.4.. Will post.
On Thu, 02 Dec 2004 13:15:29 -0500, Andrei (MPI) <[EMAIL PROTECTED]> wrote: > I would sniff UDP packets with tcpdump and see what is going on: > separately in 10.24.102 and 10.24.100 segment. > > Do you have any general NAT settings in sip.conf, like externip and > local network etc? How about RTP port range? > > It would be easier to understand your setup if you post your sip.conf, > as well as phone's <MAC>.cfg file (which may be too big for this > newsgroup - email is fine). > > Also, asterisk sip registration and error messages would give us some > more info. > > Andrei > > > > Tim Jackson wrote: > > >I've already added nat=yes. Nothing fancy/special on the routing between > >these. > > > >-Tim > > > >-----Original Message----- > >From: [EMAIL PROTECTED] > >[mailto:[EMAIL PROTECTED] On Behalf Of Andrei > >(MPI) > >Sent: Thursday, December 02, 2004 10:51 AM > >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [Asterisk-Users] Polycom IP500 > > > >Hello Tim, > > > >You are saying that: phone is on "10.24.102.0/24" and Asterisk resides > >on "10.24.100.0/24". Honestly, I see at least one hop forwarding here > >and possible network issues right away. > >At one moment, not in a NAT environment, but having phone IP > >172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes > >in sip.conf in order to get the phone to work. > > > >Sincerely, > >Andrei > > > >Tim Jackson wrote: > > > > > > > >>Theres no NAT going on here. Just 1 router in between, phones reside on > >>10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT > >>problem. Any other ideas? > >> > >>-Tim > >> > >>-----Original Message----- > >>From: [EMAIL PROTECTED] > >>[mailto:[EMAIL PROTECTED] On Behalf Of Andrei > >>(MPI) > >>Sent: Wednesday, December 01, 2004 11:15 PM > >>To: Asterisk Users Mailing List - Non-Commercial Discussion > >>Subject: Re: [Asterisk-Users] Polycom IP500 > >> > >>Tim, > >> > >>You may see description of new 1.3.4 firmware at polycom.com (check - > >>http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.p > >> > >> > >d > > > > > >>f > >>) released in October. > >> > >>Though, it was proven over time that troubles with a SIP phone like > >> > >> > >"not > > > > > >>hearing" one side or the other is NAT related problems. You may want to > >> > >> > > > > > > > >>investigate firewall setup. I am not saying it is not phone related, > >> > >> > >but > > > > > >>the phone would be the last one to blame. > >> > >>Also, may I express my feelfings about Cisco and Polycom - not allowing > >> > >> > > > > > > > >>direct firmware download for their phones - that sucks big time. I will > >> > >> > > > > > > > >>get the firmware this way or the other. They just force me to waste my > >>time again and again contacting their dealers and searching the > >>internet. That should just enrage customers, in my opinion. Are they so > >> > >> > > > > > > > >>big, they do not even care? > >> > >>Sincerely, > >>Andrei > >> > >>Tim Jackson wrote: > >> > >> > >> > >> > >> > >>>Any idea if 1.34 fixes the problems with the phones being up for long > >>>periods of time and weird call problems (I cannot hear remote caller, > >>>but they can hear me) ? > >>> > >>>-Tim > >>> > >>>-----Original Message----- > >>>From: [EMAIL PROTECTED] > >>>[mailto:[EMAIL PROTECTED] On Behalf Of Paul > >>> > >>> > >>> > >>> > >>Hales > >> > >> > >> > >> > >>>Sent: Wednesday, December 01, 2004 6:00 PM > >>>To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > >>>Subject: RE: [Asterisk-Users] Polycom IP500 > >>> > >>> > >>>Any idea if 1.34 makes Daylight Savings work for us people in > >>> > >>> > >>> > >>> > >>Australia? > >> > >> > >> > >> > >>>PaulH > >>> > >>>-----Original Message----- > >>>From: Andrei (MPI) [mailto:[EMAIL PROTECTED] > >>>Sent: Thursday, 2 December 2004 9:30 AM > >>>To: Asterisk Users Mailing List - Non-Commercial Discussion > >>>Subject: Re: [Asterisk-Users] Polycom IP500 > >>> > >>>Hi Chris, > >>> > >>>First of all, you need to configure ftp or tftp and watch syslog > >>>closely - > >>>what the phone is looking for at boot time. You would need to put > >>> > >>> > >>> > >>> > >>config > >> > >> > >> > >> > >>>files into (t)ftp directory, named according to MAC address of you > >>>phone. > >>>XML and Web is really weird - they do not even share same config data. > >>>For > >>>example, I had to change address of SNTP server (clock) and it still > >>> > >>> > >>> > >>> > >>was > >> > >> > >> > >> > >>>showing as 'clock' on Web admin page for the phone. I will email you > >>> > >>> > >>> > >>> > >>the > >> > >> > >> > >> > >>>config files I got from a good fellow from this list not so long > >>> > >>> > >ago... > > > > > >>>and > >>>those config files do really help! > >>> > >>>Also, I suggest that you upgrade your SIP firmware if you have not > >>> > >>> > >done > > > > > >>>it > >>>yet. I got the Polycom 500 with firmware which was very old and > >>>incapable to > >>>work with asterisk. Mine is 1.3.1 now > >>>(http://www.freedomphones.net/polycom/files/). > >>> > >>>If anyone has SIP firmware 1.3.4 - please make it downloadable > >>> > >>> > >>> > >>> > >>somewhere > >> > >> > >> > >> > >>>on > >>>the internet? > >>> > >>>Thank you, > >>>Andrei > >>> > >>>Chris Cherry wrote: > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>>>Hey All, > >>>> > >>>>First Time Writing. > >>>> > >>>>I'm trying to set up my IP500 phones to register SIP with *. I input > >>>>all the (I assume) correct data in to the fields on the Web > >>>> > >>>> > >Interface. > > > > > >>>> > >>>> > >>>> > >>>> > >> > >> > >> > >> > >>>>And I get no notification that the phone is even attempting to > >>>>register, no failed messages etc. I have read that the Web interface > >>>> > >>>> > >>>> > >>>> > >>is > >> > >> > >> > >> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>> > >>> > >>> > >>> > >>>>crap and the XML config files is the way to go. Does anyone have a > >>>>basic config file that doesn't change any defaults? I couldn't seem > >>>> > >>>> > >to > > > > > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>find > >>>one. > >>> > >>> > >>> > >>> > >>> > >>> > >>>>Extra Info: > >>>>Server is 192.168.0.3 > >>>>Phone name/ext I want to be 301 > >>>> > >>>>Thanks, > >>>>Chris Cherry > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>_______________________________________________ > >>>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 > >>>CAUTION: This email message and accompanying data may contain > >>>information > >>>that is confidential. If you are not the intended recipient, you are > >>>notified that any use, dissemination, distribution or copying of this > >>>message or data is prohibited. 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