When i put PPM file to my http server. Emergency button coming up on my phones but any other services not running like Favorites , Contacts etc. I just want to run Contacts. Any help in here?
If you dont have the correct mods to astrisk then all you will get is the emergency button if I remember the below links are working again
https://www.slockner.com/RDWeb/Pages/chan_sip.c
https://www.slockner.com/RDWeb/Pages/sip.h
I really need your help to work with this issue. I can give you more information from log files or if you want to connect to system i can give access to you . Waiting for your answers 
I am sorry I don’t have a lot of free time in front of a computer to do debugging so you just have to be patient… I am glad its compiling for you now. … I cant seem to see your screen shot. … now that its compiling what does your pmm and your extention config files look like?
Thank you for your answer. I am trying to upload files to another website.
Could you see my files now ?
Its so hard to upload files in here 
Maybe send files to you by email ?
I just filled Contacts file because i just want to use Contacts only. But other files are there in folder but they are empty.
Any update?
Any help would be greatly appreciated.
are you using the emulator or the phone? my phones seem to be getting their contact list for some reason my emulator is failing
Also what firmware version are you running as now I got my 6.5 emulator working but 7.1 emulator does not.
Just found this thread. Am trying to configure a FreePBX Raspberry Pi system to use Avaya 9650 Phones. Have managed to get them to connect and work as single extension phones with limited features. I assume this is SIPPING-19 vs Avaya SIP. I was excited to try this approach but discovered in the very first steps that the Raspberry Pi version of FreePBX does not appear to have a /usr/src/asterisk-*** directory. From skimming through this page I also assume that this directory is pre compiled files which I am suspecting may not exist on the Raspberry Pi version. Can anyone confirm this? If they are there where would I find them? If they are not, is there a way for me to add the features as described? Sorry for the dumb questions, kinda new to this…
Hello Shawn,
I have tried with 7.1 emulator and 7.1.6.0 with sip firmware on physical phone.
Maybe downgrade to 6.5 firmware and give a try ?
UPDATE:
Downgraded phone’s firmware to SIP 6.5.0 but no work 
I think my problem is chan_sip and sip.h file . There is something wrong on my configuration. Because if i use your file , asterisk do not compiling. But if i add your codes to one by one its compiling but gives me error. you can see on pbx_asterisk.png form my files.
Hi,
I am trying in your footsteps but I can’t seem to be able to compile asterisk-13.7.2 with the modified files you posted. Can you tell me which version of asterisk I should be using with these files?
Thanks for the great work. I hope I can my phones up an running soon.
I managed to load SIP firmware on an avaya 9611G finally but now my challenge is now registering on asterisk it is just displaying acquiring services after entering sip server IP transport tcp, port 5060 and entering the extension and password
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In FreePBX, is CHAN_SIP set to 5060 and TCP Enabled under “Asterisk SIP Settings>CHAN SIP Settings”
Hi! Just found this thread and have a few Avaya 96X1 sets I want to use. Like @Shesakillatwo, I do not have the /usr/src/asterisk-*** directory. I’m running FPBX 15 with Asterisk 16.6.2 on SangomaOS, which I believe is a variant of Debian.
Any thoughts?
i think since you dont actually complile the code… and use a pre build os … you dont have the files
Thanks @slockner, and thanks for all of your work on this.
I just compiled Asterisk on CentOS7 and found the necessary files. I’ll dig in and give it a shot!
I do not have an http server - do you think I could use httpd?
Thanks again Shawn!
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Hi shawn, I have installed my freepbx on linux. all is perfectly installed and all my Avaya9608G phones register wonderfully. I’m able to retreive incoming call as well as dialing outgoing calls successfully.
But I have an issue implementing the contactlists for instance. I have setup everything according to your guideance.
in the 46xx file the config is pointing to my freepbx server to make php work. php is working on my freepbx server as well and also the ppm file is loaded in version of ppm.txt and ppm.php in oder the phone can pick it up. Nothing happens. What I’m doing wrong.
Thanks for your support.
Best Regards
Andy.
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Hi Shawn.
I’m trying to compile asterisk 13.7.2 on centos 7 and ubuntu server 16.04 with your mod but the compilation fails when it tries to compile the file chan_sip.c. I have the same error on various files. The error is: error: storage class specified for parameter
Asterisk 13 was current when this thread was started but is now no longer receiving even security fixes. Don’t use it. 13.7.2 is over six years old.
chan_sip is deprecated. Unless you are forced to use (e.g. ITSP who insists on using tel: URIs), don’t use it. It will be removed completely in next year’s release of Asterisk.
Having the error message without the line that generated it is not much use.