Alcatel Unleashed

Dear all
I have a lot of OXO-OXE systems joined against Asterisk servers with SIP Trunk.
If I use the last release of OXO (7.0), even the last patch, I have a problem: The Ars table apears "active" in the line against Asterisk server, but I can,t call from OXO to Asterisk, the message "unavailable" (Inaccesible in spanish) is seen in the display of phones.
This problem is resolved if I use R5.x or 6.x .
Anyone knows if the R7 in Oxo has changes in SIP protocol? If yes if is posible to know what changes have been implemented may be I can resove it.................... I hope!
Thank you guys!!

Hello. Why don´t you make a report to Alcatel?, if it is working in rel 5 and 6, it is a problem of the rel 7 that must be reported.

As I know, there is no changes on SIP protocol.

Regards.

AFLORES

Dear aflores
I have made one eservice request to Alcatel about it.
The answer is : "In R7 in fact there are some changes in SIP (adaptation to any RFC without more details)".
I know among Alcatel PBX all is right, in the other side (Asterisk for example) is where Alcatel says we have to correct the problem.
Thank you for your answer.

Dear all
Don,t use R7.0, with 7.1 the problem is resolved. Alcatel says "in 7.1 are some new changes in SIP protocol". I don,t know what kind of changes, but the ALCATEL-ASTERISK union works fine again.
Thank you

Hi all,
I get a strange issue with release 7.1 and asterisk. The external call is forworded from OXO to Asterisk for some queue management and then come back to OXO, well in the second stage the voice from oxo to external caller start with a delay of 30sec. In wireshark trace I get the correct rtp flows and segnaling is ok. Have you some ideas? Thanks

Ok, after the upgrade of Asterisk platform to 1.5.0 stable, we solve the problem but... yesterday voip card of oxo crash.... I hope well...