Hanseman, Todd
2006-Jun-12 16:42 UTC
[Asterisk-Users] ICLID or CNAM calling name and number through a cisco isdn gateway
All,
I need to run this by everyone and see if someone has any idea's. I have a
asterisk server setup and currently am receiving the inbound calling number
where the name should be. My setup is....
One pri terminating into a Cisco 2431 router
Sip messages from the Cisco get sent to a asterisk server
linksys ata's a each remote end.
I can receive the calling name if the call originates from another extension on
the asterisk server, I also can "make" the Cisco send out a generic
name to the asterisk sip server and I see the name I statically assign in the
Cisco appears on the terminating end (linksys ata)
I use the command in the Cisco under sip-ua
calling-info pstn-to-sip from name set name
timers buffer-invite 5000
I have also tried to add the commands...
remote party-id
-voice service voip,sip
ds0-num
Basically I need to take the field Remote-party id and place it in the sip
message "From"
Here is some debug from the sip messages in the Cisco... this is an example of
"no caller id Name"
Sent:
INFO sip:5132017005@65.23.9.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 66.148.165.xxx:5060;x-ds0num="ISDN 1/0:23 1/0:DS1
1:DS0";bran
From: <sip:5136161824@66.148.165.xxx>;tag=4B0025C-131
To: <sip:5132017005@65.23.9.xxx>;tag=as57cef7cb
Date: Fri, 01 Mar 2002 21:50:43 GMT
Call-ID: 36F777D2-2C9511D6-8065D52A-B58C0139@66.148.165.122
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1015019445
CSeq: 102 INFO
Contact: <sip:5136161824@66.148.165.xxx:5060>
Remote-Party-ID: "WIRELESS CALLER"
<sip:66.148.165.xxx>;party=called;screen=no;
Content-Length: 0
________________________
here is an example of a call with a name and number ( this is forced out by the
cisco, for all inbound calls to the ata's )
*Mar 2 03:56:13.617: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INFO sip:5132017005@65.23.9.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 66.148.165.xxx:5060;x-ds0num="ISDN 1/0:23 1/0:DS1
1:DS0";branch=z
From: "Wrking_on_name"
<sip:5136161824@66.148.165.xxx>;tag=5FE9C70-BA
To: <sip:5132017005@65.23.9.xxx>;tag=as416d0081
Date: Sat, 02 Mar 2002 03:56:12 GMT
Call-ID: 4559EB3B-2CC811D6-8110D52A-B58C0139@66.148.165.xxx
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1015041373
CSeq: 102 INFO
Contact: <sip:5136161824@66.148.165.xxx:5060>
Remote-Party-ID: "WIRELESS CALLER"
<sip:66.148.165.xxx>;party=called;screen=no;priv
Content-Length: 0
if you look at the two messages the difference is the FROM message
From: <sip:5136161824@66.148.165.xxx>;tag=4B0025C-131 (only name)
From: "Wrking_on_name"
<sip:5136161824@66.148.165.xxx>;tag=5FE9C70-BA ( name and number ( this is
forced out by the cisco, for all inbound calls to the ata's )
I need to find a way to take the info in the remote party-id and place it in the
From field... Any ideas?
below is my cisco config...
the firmware verson is....
System image file is "flash:c2430-is-mz.123-11.T8.bin"
Cisco IAD2431 (R527x) processor (revision 4.0) with 119808K/11264K bytes
Processor board ID FHK0908F2MD
R527x CPU at 225MHz, Implementation 40, Rev 3.1
1 On-Board Eight FXS Analog Voice Module
1 FastEthernet interface
10 Serial interfaces
1 Channelized T1/PRI port
DRAM configuration is 64 bits wide with parity disabled.
63K bytes of non-volatile configuration memory.
System fpga version is 250025
System readonly fpga version is 240024
cisco config....
hostname 2431_dtci_2_9
!
boot-start-marker
boot-end-marker
!
card type t1 1
enable secret 5 $1$K7c.$PvwV1AKdDLSzj.KZ/QpN8/
!
network-clock-participate T1 1/0
no aaa new-model
ip subnet-zero
!
!
!
!
isdn switch-type primary-ni
!
voice-card 0
!
!
!
voice service voip
fax protocol pass-through g711ulaw
modem passthrough nse codec g711ulaw
sip
ds0-num
header-passing
registrar server expires max 600 min 60
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g723ar53
codec preference 4 g723ar63
codec preference 5 g723r53
codec preference 6 g728
codec preference 7 g729r8
!
!
!
!
!
!
!
!
!
username nvxcpe privilege 15 secret 5 $1$oAvO$zHweQ8lbhZjzROdYw4psM/
!
!
controller T1 1/0
framing esf
linecode b8zs
cablelength short 133
pri-group timeslots 1-8,24
description WAN
!
!
!
interface FastEthernet0/0
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/0
no ip address
encapsulation frame-relay IETF
no ip mroute-cache
no fair-queue
service-module t1 timeslots 1-24
frame-relay lmi-type ansi
!
interface Serial0/0.1 point-to-point
ip address 66.xxx.xxx.xxx255.255.255.252
frame-relay interface-dlci 100
!
interface Serial1/0:23
no ip address
no logging event link-status
isdn switch-type primary-ni
isdn incoming-voice voice
no isdn outgoing display-ie
no cdp enable
!
ip http server
!
ip classless
ip route 0.0.0.0 0.0.0.0 66.xxx.xxx.xxx
!
!
access-list 52 permit 65.xx.xx.xxx
snmp-server community 3Z8pbXn3 RO 52
!
!
!
control-plane
!
!
call application voice app_transfer flash:current.tcl
call application voice app_transfer max-fwd-cnt 2
call application voice app_transfer language 1 en
call application voice app_transfer set-location en 0 flash:/prompt
!
voice-port 1/0:23
description dtci 2x9
!
voice-port 2/0
idle-voltage low
!
voice-port 2/1
!
voice-port 2/2
!
voice-port 2/3
!
voice-port 2/4
!
voice-port 2/5
!
voice-port 2/6
!
voice-port 2/7
!
!
!
dial-peer voice 1 pots
destination-pattern .T
fax rate voice
direct-inward-dial
port 1/0:23
!
dial-peer voice 2017005 voip
application app_transfer
destination-pattern 5132017005
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 2017006 voip
application app_transfer
destination-pattern 5132017006
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8427005 voip
application app_transfer
destination-pattern 5138427005
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 2019293 voip
description Brad_line_1
application app_transfer
destination-pattern 5132019293
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429964 voip
description Brad_line_2
application app_transfer
destination-pattern 5138429964
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429971 voip
description Scott_line_1
application app_transfer
destination-pattern 5138429972
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429972 voip
description Scott_line_2
application app_transfer
destination-pattern 5138429972
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429973 voip
description Kyle_line_1
application app_transfer
destination-pattern 5138429973
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429974 voip
description Kyle_line_2
application app_transfer
destination-pattern 5138429974
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429975 voip
application app_transfer
destination-pattern 5138429975
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429976 voip
application app_transfer
destination-pattern 5138429976
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429977 voip
description Linksys_ATA_Comm_room_line_1
application app_transfer
destination-pattern 5138429977
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429978 voip
description Linksys_ATA_Comm_room_line_2
application app_transfer
destination-pattern 5138429978
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429979 voip
description NON NAT ATA
application app_transfer
destination-pattern 5138429979
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
dial-peer voice 8429980 voip
description NAT ATA in comm Room 10.194.41.97
application app_transfer
destination-pattern 5138429980
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
!
sip-ua
timers buffer-invite 5000
sip-server ipv4:65.xxx.xxx.xxx
!
!
line con 0
exec-timeout 0 0
Any Idea's, I think this is going to be a Cisco problem more than a Asterisk
problem, but I thought I would start here.
Thanks,
This email and any attachments ("Message") may contain legally
privileged and/or confidential information. If you are not the addressee, or if
this Message has been addressed to you in error, you are not authorized to read,
copy, or distribute it, and we ask that you please delete it (including all
copies) and notify the sender by return email. Delivery of this Message to any
person other than the intended recipient(s) shall not be deemed a waiver of
confidentiality and/or a privilege.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060612/4f3567af/attachment.htm
Reasonably Related Threads
Wisdom of the Ancients
