[asterisk-bugs] [Asterisk 0018469]: 1.4.38 does not write external callerID number into SIP From: header

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Dec 14 07:56:46 CST 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18469 
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Reported By:                aragon
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18469
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.38 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-12-13 14:16 CST
Last Modified:              2010-12-14 07:56 CST
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Summary:                    1.4.38 does not write external callerID number into
SIP From: header
Description: 
Outgoing number in CALLERID(all) field is not sent to ITSP
Asterisk 1.4.38
====================================================================== 

---------------------------------------------------------------------- 
 (0129593) aragon (reporter) - 2010-12-14 07:56
 https://issues.asterisk.org/view.php?id=18469#c129593 
---------------------------------------------------------------------- 
[general]
context         =  default-incoming-guest
callevents      =  yes
alwaysauthreject =  yes
jbenable        =  no
t38pt_udptl     =  yes
progressinband  =  never
externip        =  216.235.14.204
localnet        =  192.168.0.0/255.255.0.0
localnet        =  10.0.0.0/255.0.0.0
localnet        =  172.16.0.0/12
localnet        =  169.254.0.0/255.255.0.0
bindport        =  5060
bindaddr        =  0.0.0.0
rtpkeepalive    =  0
limitonpeers    =  yes
notifyringing   =  yes
notifyhold      =  yes
realm           =  asterisk
useragent       =  Asterisk PBX
maxexpirey      =  3600
defaultexpirey  =  120
recordhistory   =  no
autocreatepeers =  no
srvlookup       =  yes
videosupport    =  yes
directrtpsetup  =  no
disallow        =  all
allow           =  ulaw
allow           =  alaw
allow           =  g722
allow           =  g726
allow           =  g723
allow           =  gsm
allow           =  g729
allow           =  slin
allow           =  ilbc
allow           =  lpc10
allow           =  speex
allow           =  adpcm
allow           =  h261
allow           =  h263
allow           =  h263p
allow           =  h264
tos_sip         =  CS0
tos_audio       =  ef
tos_video       =  CS0
pedantic        =  no
allowexternaldomains =  no
allowexternalinvites =  no
autodomain      =  no
relaxdtmf       =  no
trustrpid       =  no
sendrpid        =  yes
promiscredir    =  no
usereqphone     =  yes
compactheaders  =  no
#include "sip-register.conf"
#include "default/sip.conf"
#include "default/sip-extras.conf"
#include "test/sip.conf"
#include "test/sip-extras.conf"
#include "sip-extras.conf"

I don't think this is a support issue since the name and number is
correctly set on an external PRI circuit.  The outgoing number only fails
on SIP trunk calls. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-14 07:56 aragon         Note Added: 0129593                          
======================================================================




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