Jason Lixfeld
2006-Sep-12 08:34 UTC
[asterisk-users] Problems getting 7970G upgraded to SIP
I have a 7970G with 5.0.3.0S Skinny (Load File: TERM70.5-0-3-0S) on it and I'd like to get it up to 8.x. - With the SEP<MAC>.cnf.xml in place (which was taken from voip-info (http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP under "This worked for me...")), I get "Load ID Incorrect" on the phone display after it boots. The loadInformation line in the SEP file reads like this: <loadInformation>SIP70.8-0-4SR1S</loadInformation> - If I remove the SEP file, the phone requests XmlDefault.cnf.xml. I create the xml file based on the example from the same link above, the phone grabs the file, but doesn't upgrade. It just sits in a loop of: release IP => renew IP => look for SEP, fail => look for XmlDefault, find and load XmlDefault => release IP... The loadInformation line in the XmlDefault.cnf.xml file reads like this: <loadInformation6 model="IP Phone 7970">SIP70.8-0-4SR1S</ loadInformation6> - Here are TFTP server logs to illustrate that I'm using the correct case'd XmlDefault.cnf.xml file: Sep 10 21:57:55 bubbles tftpd[89195]: jalc7970.sip : read request for SEP00131A4D39F4.cnf.xml: File not found Sep 10 21:57:55 bubbles tftpd[89197]: jalc7970.sip : read request for //XmlDefault.cnf.xml: success - All the files from the .cop are 100% unmodified. I just tar -zxvf cmterm-7970_7971-sip.8-0-4SR1.cop and the files are extracted into the tftpd root directory, which is the same place the SEP and XmlDefault file are located. Anyone have any ideas?
Richard Klingler
2006-Sep-12 12:14 UTC
[asterisk-users] Problems getting 7970G upgraded to SIP
Hi Jason> <loadInformation6 model="IP Phone 7970">SIP70.8-0-4SR1S</loadInformation6>1. Stick with the 8.0.2 SIP image as it works best with asterisk... at least for me (o;> > - Here are TFTP server logs to illustrate that I'm using the correct > case'd XmlDefault.cnf.xml file: > > Sep 10 21:57:55 bubbles tftpd[89195]: jalc7970.sip : read request for > SEP00131A4D39F4.cnf.xml: File not found2. I thought you created your SEPxxxx file? And still it can't be found?> Sep 10 21:57:55 bubbles tftpd[89197]: jalc7970.sip : read request for > //XmlDefault.cnf.xml: success3. Wondering what messages are coming after that...or is it the point where it starts over again?> > - All the files from the .cop are 100% unmodified. I just tar -zxvf > cmterm-7970_7971-sip.8-0-4SR1.cop and the files are extracted into the > tftpd root directory, which is the same place the SEP and XmlDefault > file are located.4. So you have all those: -bash-2.05b$ tar tzvf cmterm-7970_7971-sip.8-0-2SR1.cop 644 Mar 22 23:49 SIP70.8-0-2SR1S.loads 2538161 Mar 22 23:49 apps70.1-1-1-15.sbn 411264 Mar 22 23:49 cnu70.3-1-1-15.sbn 1996 Mar 23 00:06 copstart.sh 2401588 Mar 22 23:49 cvm70sip.8-0-1-18.sbn 483105 Mar 22 23:49 dsp70.1-1-1-15.sbn 465288 Mar 22 23:49 jar70sip.8-0-1-18.sbn 71 Mar 23 00:06 load119.txt 72 Mar 23 00:06 load30006.txt 0 Mar 23 00:06 signed/ 4046848 Mar 23 00:06 signed/cmterm-7970_7971-sip.8-0-2SR1.cop 644 Mar 22 23:49 term70.default.loads 644 Mar 22 23:49 term71.default.loads> > Anyone have any ideas?5. Not yet. But might be you need to go with a firmware in between first before going with 8.0.x. cheers rick
For more info please visit cyber-telecom.net The price per unit is ?199 ex delivery fee. CT-375 VoIP GSM Gateway is the conformity of advanced GSM Interface and VoIP Gateway. The CT-375 combines the both devices; it brings ease of use and simplicity to both knowledge and beginner users. It also allows features like call termination (VoIP to GSM) and origination (GSM to VoIP), and effectively utilize same circuit of both, but without FXO and FXS interface transformation, this in term save in material cost as well as improving the voice quality during communication. Key Facts: ? GSM termination (VoIP -> GSM) ? GSM origination (GSM -> VoIP) ? Dual band: 900/1800MHz or tri band: 900/1800/900MHz Description: 50 sets of LAN -> mobile Routes Setting 50 set of Mobile -> LAN Routes Setting Voice response for setting and status (Dial in from mobile) Series connection to save bill Standard SIP Protocol (RFC2543, RFC3261) Protocol to communicate with other Gateway or PC All function can be set on web SPECIFICATIONS: TCP/IP: IP/TCP/UDP/RTP/RTCP, CMP/ARP/RARP/SNTP, DHCP/DNS Client, IEEE802.1P/Q, ToS/DiffServ, NAT Traversal, STUN, uPnP, IP assignment, Static IP, DHCP, PPPoE Codec: G.711u-Low, G.711a-Low, G.723.1(5.3k), G.723.(6.3k), G.729A, G.729A/B Voice quality: VAD, CNG, AEC, LEC, Packet loss CDMA (SC-375C): 800MHz CDMA, support cdma2000 1xRTT air interface Backward compability from IS2000 to IS95A/B Support Voice for both 13k QCELP and EVRC Support both full rate and 1/8 gating on the reverse link GSM (SC-375): BAND EGSM/900 DCS/1800MHz or 900/1800/1900MHz Speech service with EFR (Enhance Full Rate)/FRFull Rate)/ HR(Half Rate) Codec Sam
Tzafrir Cohen
2006-Sep-12 21:26 UTC
product annocements [was: Re: [asterisk-users] Finally a VoIP GSM Gateway single port..]
On Wed, Sep 13, 2006 at 03:45:57AM +0800, Sam Tam wrote: [snip] Would you have any idea about the content of the message from the subject? Some people believe that annoucements of new products related to Asterisk are on-topic in this list. In such a case they should follow some simple rules in order to be useful to the lists readership: 1. Informative and descriptive subject I would like to find the message later on. It should include the name of the company and the name of the product(s). 2. Useful details first, press release later This is a non-commercial list. Please be informative. Be sort on superlatives. A direct link to a page describing the product (that should remain over time) is also a bonus. The message I was replying to basicaly followed (2) but not (1). -- Tzafrir Cohen sip:tzafrir@local.xorcom.com icq#16849755 iax:tzafrir@local.xorcom.com +972-50-7952406 jabber:tzafrir@jabber.org tzafrir.cohen@xorcom.com http://www.xorcom.com