Attended transfer issue on generic SIP phone [Solved]

[javiherbar]

Hello everyone,

I am new to the Alcatel PBX environment and have been working my wat arround some basic scenarios in a testbed environmet I have.

Lately, I am trying to validate if a generic sip phone (Grandstream GXP1628) can interwork ok with the OXE Release 11. Didnt find the phone in the IWR I have, so decided to give it a go.

So far, I have been able to perform all the basic functions of the phone, but now I found an issue that puzzles me, and maybe some of you can give me a hint on what can be happening.

Scenario:
Phone A - GXP1628, Sip Extension 1201
Phone B - External phone, number 2000
Phone C - 8008, Extension 1150

A calls B, once call is established, A attempts to perform atended transfer to C. Call to C returns 488. A attemtps the transfer again, the call between A and C is established, and the transfer can be performed correcly.

So, at first I thougth maybe some codecs issue, or some permission issue, but given that the second attempt is succesful, Im not really sure.

Any ideas? ever happen to anyone?

Attached is the trace of the scenario playing out, please let me know if there is some more information I can provide.

Thanks a lot in advace, have been learning a lot reading the formum, regards to everyone

Transfer_488.rar

[haroun]

192.168.50.10 IS THE PROXY
192.168.10.20 IS duplicated for the GS and IP touch
"Start Time","Stop Time","Initial Speaker","From","To","Protocole","Duration","Paquets","State","Comments"
"16.321579","56.688914","192.168.50.10","<sip:1201@192.168.50.20;user=phone>","<sip:02000@192.168.50.20;user=phone>","SIP","00:00:40","30","COMPLETED","INVITE 407 200 200 200 200"
"26.608672","67.398763","192.168.50.20",""1201" <sip:1201@192.168.50.20;user=phone>","<sip:2000@192.168.50.28;user=phone>","SIP","00:00:40","19","COMPLETED","INVITE 200 200 200 200 200"
"40.863017","40.901529","192.168.50.10","<sip:1201@192.168.50.20;user=phone>","<sip:1150@192.168.50.20;user=phone>","SIP","00:00:00","8","REJECTED","INVITE 407 488"
"49.279214","56.768913","192.168.50.10","<sip:1201@192.168.50.20;user=phone>","<sip:1150@192.168.50.20;user=phone>","SIP","00:00:07","19","COMPLETED","INVITE 407 200 200"

[javiherbar]

Hello haroun, thanks for checking my traces.

Unfortunately, I dont quite understand your answer, see:

192.168.50.10 is the GS phone.
192.168.50.20 is the SIP Gateway, where the phones are registered to, as SIP Extensions.

Again, I am new to the Alcatel world, but in order to register SIP phones, the way to do it isnt creating a SIP Gateway and then authenticating the extentions against it? So why the domains being duplicated would be incorrect?

Thanks

[javiherbar]

If anyone finds this post with a similar problem, in my case the issue was resolved.

Really was not an issue, I was just performing the transfer too fast in the test scenario. If you wait a few seconds (~4s) after the call is established, you can set up transfers or conferences without any issue.

If you compare the same scenario, but performing the transfer from a 8008 for instance, you will find that during the first seconds of the call you actually cannot input any number to perform another call.

I dont know deeply the why´s of this yet, but helped me to solve my issue.

Regards to everybody.

[haroun]

may be a Timer issue ?!