just saw this: http://michigantelephone.wordpress.com/2011/05/12/if-you-have-cisco-phones-do-not-upgrade-asterisk-to-version-1-8-4/
Now I need to do more testing…
That’s a really old blog post. Really old. Almost two years ago. No one uses that version of asterisk now
I realize that is an old post; I also realize I may have jumped the gun on that one - I am just banging my head against a wall here. I literally did a clean PIAF install and dumped the cnf.xml files in the new tftp directory, changed the IP of the new server to what the old one was, recreated several extensions exactly as they were in TB, and nothing works. If I add in the new TCP phone, that works fine. I just dont get it. I can dial out and have the phone behave that way OK, but dialing the phone itself results in a fast-busy, and there’s a X over the extension on the phone itself (yet I can call out from it).
Really missing something - and yes, nat=never is set in the extension. Getting frustrated now.
I’ve tried new install of FPBX, every different firmware version on the phone, and several versions of the SEP9mac).cnf.xml files, all to no avail. Still getting 401 Unauthorized. Tried it on the same subnet as the server, same results. This can’t be a networking issue. Nat-never and qualify-no are set in each extension. Latest TCP software does NOT work on the 79x1 phones, but 69x1 phones register fine. If this is not a bug in Asterisk, does anyone have an idea here? I don’t want to just be using TB forever just because of this issue, and I have a full set of Cisco phones, that worked on TB, so purchasing new phones is not an option at this point.
I have 296 of them working fine:
296 sip peers [Monitored: 251 online, 41 offline Unmonitored: 3 online, 1 offline]
chagrin-backup*CLI> sip show peer 4478
* Name : 4478
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox : 4478@default
VM Extension : *97
LastMsgsSent : 0/8
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "Care Walter" <4478>
MaxCallBR : 384 kbps
Expire : 1890
Insecure : no
Force rport : No
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : Yes
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 10.5.36.14:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 4478
SIP Options : (none)
Codecs : 0xe (gsm|ulaw|alaw)
Codec Order : (ulaw:20,gsm:20,alaw:20)
Auto-Framing : No
Status : OK (35 ms)
Useragent : Cisco-CP7941G/8.3.0
Reg. Contact : sip:[email protected]:5060;transport=udp
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
chagrin-backup*CLI> core show version
Asterisk 1.8.14.0 built by root @ jenkins3.schmoozecom.net on a i686 running Linux on 2012-07-11 15:36:04 UTC
chagrin-backup*CLI>I believe that Scott, and I believe that it can work, I just am at the point I think where I’ve looked at everything so many times it’s all starting to run together. would it be possible for you to send me a sample SEP[mac].cnf.xml file to compare mine to? I’m assuming I don’t need to do anything special after install other than set the NAT=never and qualify=no in the extension. I have also set the external IP in the SIP Settings module. Phones are running 8.5.3 for the 7965s and 8.4.0 for the 7971s and 7961s. 6941 is on 9.3.1.5.
I do nothing special at all. I know that getting 9.x to work is not for mortal’s. If you try my config you need to run the same version (take a look in my peer info).
Send me a PM with your email and I will send you a copy.
That’s kindof funny because a few years ago when I set up the 79x1s, I had no problem. Now I had no issue setting up the 9.x but I can’t get the 8.x phones working! I think I need a vacation…
Thanks again
I know at one point above you mentioned some phones partially register? We had an issue where someone was using Crossover network cables by accident to connect their Polycom phone. When they removed the crossover network cable, and installed a proper straight through network cables the phones registered properly.
I only read the first couple of replies but I just wanted to let you know a little tidbit that might effect you. I installed all my extensions with Nat=never and nothing would register for some reason and after pounding my head against the wall forever I switched a phone to nat=no an boom! it registered. Might not help in your case but something to think about.
I have been trying for over a week with no success in registering this phone on pbxiaf, i have setup lots of softphones, an android phone. The phone asks for the files from the tftp server I have and processes them but just says phone not registered. I can ping the phone from the ipbx server and from the tftp server. I think it may be the scripts that i have that are wrong. Im on sip version 9.3.3.2 please could anyone help or send me their scripts for this phone.
Mike
I installed linux through virtual box in my pc which is having windows7 os after that i installed tftp in windows7 ,after that tftp is not working it is giving too many retry errors can anybody suggest solution?
Don’t use windoze?
There is are several functional tftp servers available for linux. Most distributions have one set up already and ready to rock.
there is no internet connection coming for ubuntu (but windows have) how can i install tftpd-hpa packages can anybody help?
fix the network then
apt-get install tftpd-hpa
i got the solution i have downloaded package of tftp in windows and transfered the file to ubuntu system and installed there. now tftp is working for me.