isdn gateway callerid problem

chrischii's picture
Posted December 23rd, 2009 by chrischii (tadpole)

hello all,

i have finally made my configuration lancom isdn router and freepbx working (see post http://www.freepbx.org/forum/freepbx/users/solved-freepbx-with-lancom-ro... )

one problem remains. the callerid of external callers gets lost somewhere. the lancom router is connected with a sip-gateway line to the asterisk and autheticates itself with user 1823.

the following trace of the lancom shows that the callerid is correctly determined and transferred.

i have replaced my callerid with 0172-my-phone-number
---snip
[Callmanager] 2009/12/23 10:13:03,390 [ISDN-LINE] : - info : convert called MSN '140519912' to called-id '12'
[Callmanager] 2009/12/23 10:13:03,390 [ISDN-LINE] : -----[ CONNECT, call-id=301836, plci=0XE3C
[Callmanager] 2009/12/23 10:13:03,390 [VCM] : -----[ CALL INDICATION, call-id=301836
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : Src CallId=1648049598@00a057123f5c
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : Source : 0172-my-phone-number@isdn
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : Destination : 12@isdn
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : - info : number is complete
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : - info : parse call routing table for active entries
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : - info : using routing entry in row # 2
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : - info : first/single way destination 12@192.168.1.66 via ASTERISK-GTW
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : - info : proceeding call
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : -----[ INITIATE CALLS, call-id=301836
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : - info : initiate call to 12@192.168.1.66
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : - info : outgoing line is ASTERISK-GTW
[Callmanager] 2009/12/23 10:13:03,410 [VCM] : - info : called number is complete
[Callmanager] 2009/12/23 10:13:03,430 [VCM] : -----[ INITIATE CALL, call-id=301836
[Callmanager] 2009/12/23 10:13:03,430 [VCM] : -----[ CALL PROCEEDING INDICATION, call-id=301836
[Callmanager] 2009/12/23 10:13:03,430 [VCM] : - info : Src CallId=1648049598@00a057123f5c
[Callmanager] 2009/12/23 10:13:03,430 [ISDN-LINE] : -----[ CALL PROCEEDING REQUEST, call-id=301836, plci=0XE3C
[Callmanager] 2009/12/23 10:13:03,430 [ISDN-LINE] : - info : plci=0XE3C
[Callmanager] 2009/12/23 10:13:03,430 [SIP-Provider] : -----[ INITIATE CALL, call-id=301836
[Callmanager] 2009/12/23 10:13:03,430 [SIP-Provider] : Source : 0172-my-phone-number@isdn
[Callmanager] 2009/12/23 10:13:03,430 [SIP-Provider] : Destination : 12@192.168.1.66
[Callmanager] 2009/12/23 10:13:03,430 [SIP-Provider] : - info : line 'ASTERISK-GTW' operates is gateway mode
[Callmanager] 2009/12/23 10:13:03,430 [SIP-Provider] : - info : convert dst-number '12' -> '12'
[Callmanager] 2009/12/23 10:13:03,430 [SIP-Provider] : - info : complete
[Callmanager] 2009/12/23 10:13:03,430 [VCM] : -----[ INFO IS COMPLETE, call-id=301836
[Callmanager] 2009/12/23 10:13:03,430 [ISDN-LINE] : -----[ INFO COMPLETE REQUEST, call-id=301836, plci=0XE3C
[Callmanager] 2009/12/23 10:13:03,430 [VCM] : - info : called address is complete
[Callmanager] 2009/12/23 10:13:03,430 [VCM] : - info : Line: 'ASTERISK-GTW'
[Callmanager] 2009/12/23 10:13:03,430 [VCM] : - info : Source : '0172-my-phone-number@isdn'
[Callmanager] 2009/12/23 10:13:03,430 [VCM] : - info : Destination : '12@192.168.1.66'
[Callmanager] 2009/12/23 10:13:03,440 [SIP-Provider] : -----[ GENERATE INVITE, call-id=301836
[SIP-Packet] 2009/12/23 10:13:03,440 [PACKET] :
Sending datagram with length 956 from 192.168.1.200:20597 to 192.168.1.66:5060
INVITE sip:12@192.168.1.66 SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.1.200:20597;branch=z9hG4bK-67536460-7571fb0e\r\n
From: sip:1823@192.168.1.66;user=phone;tag=535113160--1966406035\r\n
To: sip:12@192.168.1.66;user=phone\r\n
Call-ID: 824024799@00a057123f5c\r\n
CSeq: 1 INVITE\r\n
Max-Forwards: 70\r\n
Server: lancom\r\n
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS\r\n
Contact: sip:1823@192.168.1.200:20597\r\n
Remote-Party-ID: sip:0172-my-phone-number@isdn;user=phone;screen=yes;party=calling;privacy=off\r\n
Content-Type: application/sdp\r\n
Content-Length: 409\r\n
---snip

my problem is, that always 1823 is shown as callerid. does freepbx not interprete the "Remote-Party-ID: sip:0172-my-phone-number@isdn;user=phone;screen=yes;"??
Any ideas how i can fix this?is it possible to use the rpid as callerid?if so, how can i do it?

regards
christian


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what i've found is the

chrischii's picture
On December 28th, 2009 chrischii (tadpole) said:

what i've found is the following

exten => _+.,1,Set(CID(number)=${CID(number):1})

does anyone know the variable name of the rpid? i guess that
Set(CID(number)=${whatever_rpid_is:1}) should work


push

chrischii's picture
On January 17th, 2010 chrischii (tadpole) said:

any ideas???


I have the same problem with

goulven's picture
On February 3rd, 2010 goulven (tadpole) said:

I have the same problem with Lancom 1723. I tried adding trustrpid=yes but the problem is still there. Did you found a solution?

I will perhaps try to migrate from Asterisk 1.4 to 1.6. What version of Asterisk do you use?


Solved!

goulven's picture
On February 9th, 2010 goulven (tadpole) said:

I have found the solution... the configuration proposed by Lancom was half-right. If you're interrested I can give you the How-To