external call forwarding issue

[mathieu66330]

Hi guys,

i'm having a hard time to understand why this simple function is not working properly.
I have an extension A
In the Inbound routes, a direct number can be dialed, then it goes thru "Times conditions" then to extension A

If A leaves the office, A want to call forward his number to his mobile phone.
This doesn't work.

if extension B call A, it works, i can receive the call on mobile phone.
but if i call the landline, i see call into logs but its stops

I tried to modify those lines from http://wiki.fusionpbx.com/index.php?title=Call_Forward but the results are the same

In the log file
The caller is the number 0468621111
the landline called is 0434269216
the extension A is 206
the call forward is 0666173482

Can someone put me in the right direction please ?

Kind regards

 

[mathieu66330]

INVITE

INVITE sip:gw+16b97920-6d8b-4a39-8da9-b2a047028a3e@XXXXXXX:5080;transport=udp;gw=16b97920-6d8b-4a39-8da9-b2a047028a3e SIP/2.0
Call-ID: 15035-DW-c45ff022-4e24a1ed2@sbc5.fr.sip.ovh
Contact: <sip:XXXXXXX:5060>
INVITE (SDP) x
Content-Type: application/sdp
CSeq: 1278624343 INVITE
From: "0468621111" <sip:0468621111@sbc5.fr.sip.ovh;user=phone>;tag=15035-DN-c45ff023-4a246c524
Max-Forwards: 28
Record-Route: <sip:XXXXXXX:5060;lr>;session=27429
To: <sip:0434269216@10.7.1.65;user=phone>
Via: SIP/2.0/UDP XXXXXX:5060;branch=z9hG4bK-WATK-56d36f62-6e85ce4b
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
x x xUser-Agent: Cirpack/v4.76 (gw_sip)
x x xContent-Length: 274
x x x
x x xv=0
x x xo=anonymous 164880026033 164880026033 IN IP4 91.121.129.132
x x xs=SIP Call
x x xc=IN IP4 XXXXXXXX
x x xt=0 0
x x xm=audio 35414 RTP/AVP 8 0 101
x x xb=AS:82
x x xa=rtpmap:8 PCMA/8000/1
x x xa=rtpmap:0 PCMU/8000/1
x x xa=rtpmap:101 telephone-event/8000
x x xa=fmtp:101 0-15
x x xa=ptime:20
x x xa=sendrecv


TRY

SIP/2.0 100 Trying
SIP/2.0/UDP XXXXXXXX:5060;branch=z9hG4bK-JKXI-56d60219-0e833b1f
"0468621111" <sip:0468621111@sbc5.fr.sip.ovh;user=phone>;tag=24306-MZ-c464e700-4edbe6860
To: <sip:0434269216@10.7.1.65;user=phone>
xCall-ID: 24306-MH-c464e6ff-5a9befd26@sbc5.fr.sip.ovh
CSeq: 1278916459 INVITE
User-Agent: FreeSWITCH
Content-Length: 0


480 temporarily unavailable


SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP XXXXXX:5060;branch=z9hG4bK-JKXI-56d60219-0e833b1f
Max-Forwards: 25
From: "0468621111" <sip:0468621111@sbc5.fr.sip.ovh;user=phone>;tag=24306-MZ-c464e700-4edbe6860
To: <sip:0434269216@10.7.1.65;user=phone>;tag=8F6c7979Q191e
Call-ID: 24306-MH-c464e6ff-5a9befd26@sbc5.fr.sip.ovh
CSeq: 1278916459 INVITE
User-Agent: FreeSWITCH
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
Remote-Party-ID: "0033411937745" <sip:0033411937745@10.7.1.65>;party=calling;privacy=off;screen=no


ACK


ACK sip:gw+16b97920-6d8b-4a39-8da9-b2a047028a3e@XXXXXXXXXX:5080;transport=udp;gw=16b97920-6d8b-4a39-8da9-b2a047028a3e SIP/2.0
Call-ID: 24306-MH-c464e6ff-5a9befd26@sbc5.fr.sip.ovh
CSeq: 1278916459 ACK
From: "0468621111" <sip:0468621111@sbc5.fr.sip.ovh;user=phone>;tag=24306-MZ-c464e700-4edbe6860
Max-Forwards: 28
To: <sip:0434269216@10.7.1.65;user=phone>;tag=8F6c7979Q191e
Via: SIP/2.0/UDP XXXXXXX:5060;branch=z9hG4bK-JKXI-56d60219-0e833b1f
Content-Length: 0

 

[mathieu66330]

Hello everyone,

anyone has any idea ?
@DigitalDaz any clue?

Kind Regards

 

[hfoster]

Have you got an outbound route which matches that 06 number? I see the call just dropping through the domain's dialplan until it reaches the end and hangs up.

 

[mathieu66330]

Have you got an outbound route which matches that 06 number? I see the call just dropping through the domain's dialplan until it reaches the end and hangs up.

Yes, i have. If i make a call to that number it works.
If i a do call forward on 206 extension to that number, then with an other extension 207, i call 206, it works too
But when the call come from outside from a landline ( customer call hotline=>trunk=>Extension 206=> call forward 06) This doesn't work
I tried with Follow me as well, same results

 

[gflow]

I had the same issue and what fixed it was setting the callerID field in the call routing/follow me page. This was on 4.4 (havent tested it on 4.5) but the issue sounds similar. It would work for internal calls because internal didnt care about callerID but was getting rejected by my upstream as the callerID was being set incorrectly. You may need to modify the permissions in FusionPBX to see the CallerID field for that section.

 

[mathieu66330]

No, i ran few test and it doesnt change anything @gflow i made some change in my outbound routes, and now the call is transferred but i don't have audio both ways.


I ran into this topics https://www.pbxforums.com/threads/no-audio-in-inbound-calls.3636/
it looks similar i made the change into SIP profiles and variables. no change

I made a capture on 5060 and 5080

Could anyone have look please ?

Kind Regards

 

[Ttaylor]

Following this as I am having the same issue

 

[mathieu66330]

For me, i think it was a problem from provider side. I tried with an other trunk from an other provider and it worked directly.

 

[Ttaylor]

For me, i think it was a problem from provider side. I tried with an other trunk from an other provider and it worked directly.

I still cant get it. I am using Twilio. My inbounds work fine. But when it tries to send any calls outward it doesnt work.

IDK what to do..

 

[mathieu66330]

I was using OVH, i don't understand either. It worked from internal call forward mobilephone but when it was in inbound routes, or IVR or Ring group, it didn't worked.
Cannot help more sorry mate

 

[mcs3ss2]

enable call forward and follow me in ring group for external call forwarding to work

 

[Ttaylor]

I tried that. It didnt work.

im not sure how to pull a log.

 

[rkbsimpsipper]

Twilio uses 5060 so your internal profile will send your internal cid by default. You have to put the twilio number in the outbound and external cid of each extension +1-234-567-8910 before it will originate the call. I really wish it just used 5080, then this would work perfectly without editing those fields. i am a noob, so I am sure there is a better way to apply this to all outgoing calls that use the twilio gateway, without editing each individual extension. the twilio webui is superb though, i likes

 

[rkbsimpsipper]

yeah so if you hardcode the external number into the outbound route that sources the gateway it works without editing the extensions

effective_caller_id_number=+12345678910