Re: [Freeswitch-users] Multi FS behind same NAT, PRACK goes to wrong port

[x y]

Hy!

You were right about the contact in 183, its port 5060 in there.
I've tried turning of 100rel, it seemed to work with calls, but caused some 
problems with others things, so I would really appreciate if there is another 
option.
Btw, I have mentioned that, that I had gateway problems too. Setting up ext-ip 
as stun.freeswitch.org has seemed to work, but after 5 days, the gateway has 
went down again with the same 503 error. Is there any common in the two issues?

Thx for your advices.

Cheers,
Viktor






U xxx.xxx.xxx.xxx:56965 -> yyy.yyy.yyy.yyy:5060
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0.
Via: SIP/2.0/UDP  xxx .xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" ;tag=7185D258-BB0.
To: .
Date: Thu, 27 Nov 2008 16:28:41 GMT.
Call-ID: [EMAIL PROTECTED]
Supported: timer,100rel.
Min-SE:  1800.
Cisco-Guid: 1234720477-3151434205-2374282417-4237312787.
User: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
NOTIFY, INFO.
CSeq: 101 INVITE.
Max-Forwards: 10.
Remote-Party-ID: 
;party=calling;screen=yes;privacy=full.
Timestamp: 1227803321.
Contact: .
Expires: 180.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 264.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 5202 8450 IN IP4 xxx.xxx.xxx.xxx.
s=SIP Call.
c=IN IP4 xxx.xxx.xxx.xxx.
t=0 0.
m=audio 16732 RTP/AVP 3 8 101.
c=IN IP4 xxx.xxx.xxx.xxx.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.

#
U yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" ;tag=7185D258-BB0.
To: .
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE.
Timestamp: 1227803321 0.000388.
User-Agent: agent
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -> zzz.zzz.zzz.zzz:1352
INVITE sip:[EMAIL PROTECTED]:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F.
Max-Forwards: 8.
From: "" ;tag=D6gypX8vH4raQ.
To: .
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
Contact: .
User-Agent: agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, 
REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, 
include-session-description, presence.winfo, message-summary.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 398.
Remote-Party-ID: "" ;screen=yes;privacy=full.
.
v=0.
o=FreeSWITCH 6476130113585053783 8141266268953030291 IN IP4 yyy.yyy.yyy.yyy.
s=FreeSWITCH.
c=IN IP4 yyy.yyy.yyy.yyy.
t=0 0.
a=sendrecv.
m=audio 17068 RTP/AVP 3 98 8 9 0 18 101 13.
a=rtpmap:3 GSM/8000.
a=rtpmap:98 SPEEX/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U zzz.zzz.zzz.zzz:1352 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "" ;tag=D6gypX8vH4raQ.
To: .
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
User-Agent: agent2
Content-Length: 0.
.

###
U zzz.zzz.zzz.zzz:1352 -> yyy.yyy.yyy.yyy:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "" ;tag=D6gypX8vH4raQ.
To: ;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
Contact: .
RSeq: 2093511444.
User-Agent: agent2
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, 
REFER, UPDATE, REGISTER, INFO, PUBLISH.
Require: 100rel.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, 
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 264.
.
v=0.
o=FreeSWITCH 6247558966294607749 119302723364474833 IN IP4 zzz.zzz.zzz.zzz.
s=FreeSWITCH.
c=IN IP4 zzz.zzz.zzz.zzz.
t=0 0.
m=audio 24756 RTP/AVP 3 101 13.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U yyy.yyy.yyy.yyy:5060 -> zzz.zzz.zzz.zzz:5060
PRACK sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK5DH0ryBH70FSB.
Max-Forwards: 70.
From: "" 

Re: [Freeswitch-users] Multi FS behind same NAT, PRACK goes to wrong port

[x y]

The whole situation:

xxx.xxx.xxx.xxx:56956---INVITE--->yyy.yyy.yyy.yyy:5060
yyy.yyy.yyy.yyy:5060---100-Trying--->xxx.xxx.xxx.xxx:5060
yyy.yyy.yyy.yyy:5060---INVITE--->zzz.zzz.zzz.zzz:1352
zzz.zzz.zzz.zzz:1352---100-Trying--->yyy.yyy.yyy.yyy:5060
zzz.zzz.zzz.zzz:1352---183-Session-Progress--->yyy.yyy.yyy.yyy:5060

yyy.yyy.yyy.yyy:5060---PRACK--->zzz.zzz.zzz.zzz:5060
zzz.zzz.zzz.zzz:5060---481-No-Such-Response--->yyy.yyy.yyy.yyy:5060
yyy.yyy.yyy.yyy:5060---CANCEL--->zzz.zzz.zzz.zzz:1352
yyy.yyy.yyy.yyy:5060---481-Call/Transaction-Does-Not-Exist--->xxx.xxx.xxx.xxx:5060
xxx.xxx.xxx.xxx:56956---ACK--->yyy.yyy.yyy.yyy:5060
zzz.zzz.zzz.zzz:1352---200-OK--->yyy.yyy.yyy.yyy:5060
zzz.zzz.zzz.zzz:1352---487-Request-Terminated--->yyy.yyy.yyy.yyy:5060
yyy.yyy.yyy.yyy:5060---ACK--->zzz.zzz.zzz.zzz:1352

Cheers,
Viktor






Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] Multi FS behind same NAT, PRACK goes to wrong port

[x y]

Hy!

There are two different FS behind the same NAT, and there were Reigstration 
Failures about one or to times a day. The gateway status turned down, then I 
got 503 error codes. Then I set up the ext-ip to STUN, as the wiki requests it.
Now I facing the next problem:
Start the call, all goes right, INVITE goes to port 1352, then after 183 
Session progress from port 1352, the PRACK package goes to 5060 instead of 
1352, wich messes up the call procedure. Is there anyway to force PRACK to the 
port to the INVITE has been sent before?

Cheers,
Viktor






Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

Updated with "make current" to rev 10479, but sill not good :(
Failure log sent to pastebin:
http://pastebin.freeswitch.org/6224

Cheers,
Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

I use "make current" for updates, I made just one a few minutes ago.

Cheers,
Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

I've refreshed to latest freeswitch (revision 10478), and tried 
again.
Unfortunately, its still not working. I put the log from failure into pastebin.
http://pastebin.freeswitch.org/6223

Cheers,
Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

I've done as you said.

http://pastebin.freeswitch.org/6220

I'll try to stay on IRC more.

Thx,
Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

Updated, tested, still the same. Need newer logs?

Cheers,
Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

I've reposted them:
http://pastebin.freeswitch.org/6197 
http://pastebin.freeswitch.org/6198

Failure has been added 3 times accedantlysorry.


can you re-post the failure example, it expired from the pastebin






Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

Hi!

I have tasted att_xfer with dtmf 0, 3 way transfer. Works pretty fine, there 
wasn't any failures. I put a log from that too into pastebin, in case you would 
need it.

http://pastebin.freeswitch.org/6194

Cheers,
Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

I tought u thinked of these:

http://pastebin.freeswitch.org/6179
http://pastebin.freeswitch.org/6178

If not, please correct me .

Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

And btw, sorry for my english... thinked and tought, in one 
sentenceif only my english teacher would saw this :)





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

Devices: IP-300 with 1007 registered
 IP-300 with 1009 registered
 X-Lite with 1011 registered

Situation where att_xfer success:
-Dial IP-300 with 6691007 from X-Lite (669 is just a prefix, assuring to 
execute my extension)
-1007 ringing, get the phone
-press *1 on 1007, then dial 1009 when the dialing wav is starting to play, 
while X-Lite plays hold music
-1009 (the other IP-300) ringing, get the phone
-1009 and 1007 bridged together, they can talk, then I hang up 1007
-1009 and 1011 bridged together, they can talk
-few secs later, I hang up 1009
-1011 hangs up too

Situation where att_xfer fails:
-Dial IP-300 with 6691007 from X-Lite (669 is just a prefix, assuring to 
execute my extension)

-1007 ringing, get the phone

-press *1 on 1007, then dial 1009 when the dialing wav is starting to play, 
while X-Lite plays hold music

-1009 (the other IP-300) ringing, get the phone

-1009 and 1007 bridged together, they can talk, then I hang up 1007 
different from 
here-

-1009 silent, but not on hold, nor on hung up, 1011 stops playing waiting 
music, being silent too

-few secs later, I hang up 1009

-1011 doesnt react, so I hang up 1011 too



When the att_xfer went failure, I didnt make any noise. When it went ok, I 
spoke into all phones.
I dont know if it counts.

Logst added to pastebin.

Cheers,
Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Unstable att_xfer

[x y]

Datavox Ip-300, and X-Lite softphone for testing.





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] Unstable att_xfer

[x y]

Hy!

I have tested several times so far the att_xfer function of the freeswitch, and 
I've found it unstable. I'm
using a similar code to the example at the freeswitch wiki, and sometimes a 
call isnt transfered.
The att_xfer works fine in the cases as I figured out from the log, when the 
switch_ivr_bridge comes
back with a "ending bridge by request from read function" message. When it 
comes back with a 
"request from write function", the two channels dont get bridged together: they 
cant hear each other 
form the other side, and they dont even realize that the other hangs up.
I have pasted some logs about this a week ago into paste bin, but if you wish, 
I could paste new logs
with the newest fs.
Could it has something with silence, because I think there are more failures 
when I dont talk into
the phones?
Im using latest fs, and sip phones without built in transfer.

Cheers,
Viktor






Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

Thx for the advices. 
Unfortunately, using transfer with hold wont work for me. For example, with a 
sip phone, wich dont have built in support for transfer.
The whole situaion is that I'm using softphone, wich has a client freeswitch 
running in the background, and there is another freeswitch on the server, witch 
is responsible for the main part. I would like to process the att_xfer 
procedure in the client freeswitch, then just send the numbers to the server's 
dialplan, and let to the server's freeswitch decide what to do with it. 
I tried the regular transfer, and it worked fine. With att_xfer, the problem is 
that the client's freeswitch doesnt know where will be the call transfered, it 
just knows the dialnumber of it. Thats why I have tried to use loopback with 
att_xfer. 
Is there another way to that?
Would it worth to create a similar att_xfer application, wich accepts 
dialnumbers like transfer? I checked the source both of transfer and att_xfer, 
hoping they are similar, but they werent for the first look, and I dont really 
know how much work would it cost.






Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

The dtmf 0 works fine with sofia/ and user/, but with loopback/ I 
dont even have time to make a conference like transfer, because the caller A 
hangs up as soon as C asnwers the call of B, instead of being hold (in a 
A-call->B-att_xfer->C situation), and there will be nobody to transfer 
to. I'm trying to do that, the two calls is bridged together when B hangs up, 
or A gets B back when C hangs up, depends on the situation.





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

I've sent both of them.





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

I sent the log about the failed att_xfer + loopback channel. Are u 
interested on failed attempts with sofia/ and user/ channel url in att_xfer 
too? I wrote that some mails before, sometimes it fails to connect the two 
channels.





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

Thx for the answer, but i dont really understand what did u mean with transfer. 
Maybe I didn explain clearly enough what I would like to achive. I would like 
to use att_xfer, and somehow feed back the number I dial (wich I want to 
transfer to) back to the dialplan, and let the dialplan decide what to do with 
the number.
I know that the regular transfer works like this, but the transfer and the 
att_xfer is different, and I would like to use both of them that way. With 
regular transfer, there was no problem.
If I just misunderstood your advice, sorry. In that case, could you give me 
some more details? Big thanx.

Cheers,
Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

Hi!

I see, I have no luck on this att_xfer with loopback channel. So, if it's not 
possible, is there another way to do the trick? I would like to use att_xfer 
just like transfer, i mean feed a phone number as a param to it, not a channel 
url. I tought using loopback channel will do it, but couldn't make it so far.
If anybody interested, I could send a log from a failed att_xfer. I would 
really appreciate any help on this matter.
Btw, would it be difficult to modify the att_xfer function to accept dialnumber 
as a param instead of channel url? I don't have any idea about this, so just 
asking.

Cheers,
Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

Wich is working with loopback, is transfer:


  


  




  

  


(I know there is no need on that loopback in the loop extension, it's just an 
experimental config about loopback and transfers.)

Attended transfer is still a mistery for me. Btw, keeping up experimenting with 
att_xfer i have found the sofia method a little instable too. It sometimes acts 
like the user channel i said before.
Here is the code that I use:


  

  


Viktor





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

Wrong code has been sent before, here is the right one for sofia:


  


  







Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

Because I just wantet to try that the att_xfer is working with user channel. 
The 668 and 669 perfixes are only made to be assured about that my extensions 
will be processed instead of the default config's, and because of that, I could 
transfer the call to any dialable number in the config later on. The usage of 
the user channel wont help me in this matter, cos its very similar to the 
original sofia method in the att_xfer example on freeswitch wiki. Basicly, i 
just want to know that there is any option to use loopback channel in att_xfer.

Cheers,
Viktor
Why aren't you prefixing the $1 with 668 in the first example?






Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] att_xfer+loopback

[x y]

Hi!

Thanks for the fast answer. Of course, there is user/[EMAIL PROTECTED] in the 
config, i was just lazy or tired, so i forgot the @${domain}. So, here is the 
part of the config.

<extension name="668([0-9]{4})">

  <condition field="destination_number" expression="^668([0-9]{4})$">

<action application="bind_meta_app" data="2 b s 
execute_extension::xfer XML default"/>

<action application="bind_meta_app" data="1 b s 
execute_extension::att_xfer XML default"/>

<action application="bridge" data="user/[EMAIL PROTECTED]"/>

  </condition>

</extension>



<extension name="att_xfer">

  <condition field="destination_number" expression="^att_xfer$">

<action application="read" data="1 15
/opt/freeswitch/sounds/en/us/callie/misc/8000/transfer1.wav
callednumber 4000 #"/>

<action application="att_xfer" data="user/[EMAIL PROTECTED]"/>

  </condition>

</extension>




or for the loopback:



<extension name="att_xfer">
  <condition field="destination_number" expression="^att_xfer$">
<action application="read" data="1 15 
/opt/freeswitch/sounds/en/us/callie/misc/8000/transfer1.wav callednumber 4000 
#"/>
<action application="att_xfer" 
data="loopback/669${callednumber}"/>
  </condition>
</extension>


<extension name="connect_extension">
  <condition field="destination_number" expression="^669([0-9]{4})$">
<action application="bridge" data="user/[EMAIL PROTECTED]"/>
  </condition>
</extension>

The connect_extension just simulates that, we execute something in the dialplan.
The rest is the defult config.
Btw, the xfer is a transfer with loopback channel, and it works fine.

Cheers,
Viktor
-- Eredeti üzenet --
Feladó: Brian West <[EMAIL PROTECTED]>Címzett: [EMAIL PROTECTED]: 
Elküldve: 2008.11.06  17:58Téma: Re: [Freeswitch-users] att_xfer+loopback

Viktor, For the user channel its user/[EMAIL PROTECTED]  ... as for the rest 
can you show me your entire config?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
/b
On Nov 6, 2008, at 9:45 AM, x y wrote:Hi!

I'm new to freeswitch, and I'm trying to make an att_xfer in a dialplan, but 
instead of giving a sofia/${domain}/${called_number} as <channel url>, i 
would like to use a loopback/${called_number}, because i would like to transfer 
the call not just to different extensions. Is there any way to achive this? 
When i'm trying to do this like that:

<action application="att_xfer" data="loopback/${callednumber}"/>

in a A-call->B-att_xfer->C situation, A gets hanged up, as soon as the 
bridge has been estabilished between B and C.

Btw, i have tryed out att_xfer by giving user/${legal_user} as <channel 
url>. I've found att_xfer this way a kind of instable, sometimes it worked 
perfectly, sometimes not: A and C did not hang up, but there weren't 
succesfully connected (1 time from 10). The log printed the same at both cases.

Cheers:
Viktor

___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 





Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] att_xfer+loopback

[x y]

Hi!

I'm new to freeswitch, and I'm trying to make an att_xfer in a dialplan, but 
instead of giving a sofia/${domain}/${called_number} as , i 
would like to use a loopback/${called_number}, because i would like to transfer 
the call not just to different extensions. Is there any way to achive this? 
When i'm trying to do this like that:



in a A-call->B-att_xfer->C situation, A gets hanged up, as soon as the 
bridge has been estabilished between B and C.

Btw, i have tryed out att_xfer by giving user/${legal_user} as . I've found att_xfer this way a kind of instable, sometimes it worked 
perfectly, sometimes not: A and C did not hang up, but there weren't 
succesfully connected (1 time from 10). The log printed the same at both cases.

Cheers:
Viktor






Hirdetés (x)  

Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a 
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org