sorry been swamped … have you compared your code to theses
Just have a doubt, is there anyone able to make the conference and join works after provision with PPM?
As after provision with PPM, the SIP flow of conference join seem like changes to send a SIP REFER like below.
And Asterisk seem like do a transfer instead of conference join.
When there is no PPM provision, the conference can be host within the phone. (the phone locally mixed)
I also moved back to Asterisk 13.7.2. But it seems that the original code is much different to your chan_sip.c. Thats makes comparsion difficult.
One thing I noticed is this missing code in the description above:
With this I was able to dial out the full number. Before the phone was directly dialing just after the first digit.
Never the less I still get an incoming call, shown with a =, on the second line. It seems that some code/changes is/are still missing.
I tried also the Soft Sip Phone. But I can see the same phenomenon there, as soon the Avaya Backend is enabled in the phone.
May somebody is able to find the root cause somewhere in my SIP log.
It looks like, that the NOTIFY is causing this issue.
I ran the Avaya phone and the software phone in parallel under the same phone number. Then I dropped a call from Softphone.
This NOTIFY message came then for both phones. Both phones then had also the “=” on the display.
This is the only entry with the IP address of the Avaya phone. The softphone has the IP address .20,
.16 is my FreePBX
I start now another try with 14.5.0. I saw that you also use this version.
— (7 headers 0 lines) —
Reliably Transmitting (no NAT) to 192.168.178.19:5060:
NOTIFY sip:email@example.com;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.178.16:5060;branch=z9hG4bK09580239
CSeq: 111 NOTIFY
I am sorry I have been traveling and have not had a chance to dig in to this
There is code that handles the Notify issue. I will try to look at soon can you post you 2 modified files?
update… very messed up site… was finally able to get them…looking them over now.
Here are my files. In the meantime I switched back for some reasons to 13.20.0. The filename isn’t correct. Sorry.
Never the less, the problem with the incoming = call, where I also can see the bridge function/button on the phone, still exists.
The rests looks OK so far:
18.104.22.168 - - [18/Sep/2018:15:20:46 +0000] “GET /96xxupgrade.txt HTTP/1.1” 200 4865 “-” “Mozilla/4.0 (compatible; MSIE 6.0)”
192.168.178.19 - - [18/Sep/2018:15:20:46 +0000] “GET /46xxsettings.txt HTTP/1.1” 200 1332 “-” “Mozilla/4.0 (compatible; MSIE 6.0)”
192.168.178.19 - - [18/Sep/2018:15:20:46 +0000] “GET /Mlf_S96x0_German.xml HTTP/1.1” 304 129 “-” “Mozilla/4.0 (compatible; MSIE 6.0)”
192.168.178.19 - - [18/Sep/2018:15:20:47 +0000] “POST /axis/services/PPM HTTP/1.1” 200 938 “-” “Mozilla/4.0 (compatible; MSIE 6.0)”
192.168.178.19 - - [18/Sep/2018:15:20:50 +0000] “POST /axis/services/PPM HTTP/1.1” 200 1012 “-” “Avaya one-X Deskphone 22.214.171.124 (43258)”
192.168.178.19 - - [18/Sep/2018:15:20:50 +0000] “POST /axis/services/PPM HTTP/1.1” 200 6865 “-” “Avaya one-X Deskphone 126.96.36.199 (43258)”
192.168.178.19 - - [18/Sep/2018:15:20:50 +0000] “POST /axis/services/PPM HTTP/1.1” 200 5498 “-” “Avaya one-X Deskphone 188.8.131.52 (43258)”
192.168.178.19 - - [18/Sep/2018:15:20:50 +0000] “POST /axis/services/PPM HTTP/1.1” 200 12899 “-” “Avaya one-X Deskphone 184.108.40.206 (43258)”
192.168.178.19 - - [18/Sep/2018:15:20:51 +0000] “POST /axis/services/PPM HTTP/1.1” 200 149 “-” “Avaya one-X Deskphone 220.127.116.11 (43258)”
192.168.178.19 - - [18/Sep/2018:15:20:51 +0000] “POST /axis/services/PPM HTTP/1.1” 200 743 “-” “Avaya one-X Deskphone 18.104.22.168 (43258)”
192.168.178.19 - - [18/Sep/2018:15:20:51 +0000] “POST /axis/services/PPM HTTP/1.1” 200 1431 “-” “Avaya one-X Deskphone 22.214.171.124 (43258)”
Could the problem be due to any settings in the PMM? Without activated Avaya backend the SIP calls work without problems.
I have also just seen that my default language with enabled backend has not been selected correctly.
I would be very grateful for your help.
where is your copy of sip.h?
Please wait a moment. I think I have it now.
However, the phone currently shows the settings such as address book from the PPM no longer. I’ll check.
The PPM also writes me a proxy server with the name 03 automatically in the phone. The stored setting is TCP 5060.
Somehow I have weird problems. But the SIP telephony is currently running
It looks like, that the parameter SET SIPDOMAIN 192.168.178.16 turns on the “=” calls on my phones. I tried this with my 9650 and the softphone 126.96.36.199. Both phones showing the same behavior when the SIPDOMAIN is set.
But when I skip this parameter I have no PPM features in the phones. The files are pulled from the server but ignored somehow. With enabled parameter the phonebook is on the phone. In manuall way the behavior is the same. SIP works only with a blank SIPDOMAIN field in the phone for me.
I checked the code in the PPM. Normally I’d expect that the SIPDOMAIN will be set by this script.
Here is my HomeServer.soap from the softphone: https://nopaste.xyz/?528f02d8c975e6cc#8ucxfMa7N21lfxpzxds4qnXsLkMpZ1DrNtlBdiBpW9c=
function SendgetHomeServer https://nopaste.xyz/?ae7c83a4a18ca782#gXsUGMDEAEZBZUfI4lcMsC18LnnJ1CKsufmmjxRQB6U=
Looks strange for me. Without the SIPDOMAIN, the softphone puts all the file in the folder e.g. 303 without the domain. That would may explain the missing PPM settings for me.
Did you have a chance to look at my files? I turn in a circle and unfortunately can not continue.
I would be really grateful for help
i am not sure how your posting those file but its not allowing me to download them… i just keep getting exe files and rar files with exe in them. …
update finaly got the files looking over them…
i dont see anything wrong with you code. … can you enter astrisk and enable sip debuging… ,
sip set debug on
and send me that out put after you attempt a call
durning the sip debugging you should see a lot of my loging code show up like “handle_request_invite Check point 2.1 \n”);
also if you could send me the full sip converstion that would be good.
Here is a log from a call.
Do you have any news/hints on this topic?
I have been away for awhile but I came back to testing this recently.
@comtech I believe the exclamation mark at the top left is directly related to the response the phone gets back from SUBSCRIBE for Event: avaya-cm-feature-status, if it’s anything other than 200 OK with the correct headers, that icon will be there
@Unchained77 I only got that = button when I allowed the phone to SUBSCRIBE to Event: dialog and only in certain situations like trying to receive a transferred call
@d3951768 In my testing, when the phone successfully is SUBSCRIBEd to Event: avaya-cm-feature-status, I can’t accept attended transfers to an Avaya phone and I can’t complete a conference from an Avaya phone. If anyone happens to have SIP traces of what these look like in a proper Avaya environment, I’d be interested to see them.
@slockner I’m curious if you’re able to accept an attended transfer to one of your Avaya phones.