Tim, I've experienced routing problems before where I could dial a phone on a different subnet and I could hear them, but they couldn't hear me. Is it possible that the phone "learns" the route, then loses it later? I ended up setting the default gateway on all of my phones to a router that knows about all of my subnets, instead of my internet gateway.
Ty -----Original Message----- From: Tim Jackson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 02, 2004 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 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.pd 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. 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