[asterisk-bugs] [Asterisk 0018973]: Followme not giving the callerid
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Mar 31 12:58:13 CDT 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18973
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Reported By: hamlet505a
Assigned To:
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Project: Asterisk
Issue ID: 18973
Category: Applications/app_followme
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.8.3
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: 2011-03-14 18:07 CDT
Last Modified: 2011-03-31 12:58 CDT
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Summary: Followme not giving the callerid
Description:
I have tested this on 1.8.2 and 1.8.3 Asterisk. I have recently upgraded
from 1.4 to 1.8. I have copied the configurations from 1.4 to 1.8. I have
changed the configs slightly as to account for some syntax. But this was a
working configuration on Asterisk 1.4 as it successfully passed the
callerid through followme.
Here is the code followed by a simple trace without followme that
successfully shows the callerid using Asterisk 1.8:
exten => 2970,1,NoOp(${CALLERID(name)})
exten => 2970,2,set(CALLERID(num)=${prefix}${CALLERID(num)})
exten => 2970,3,Dial(SIP/${icell}@cme,15,t)
== Using SIP RTP CoS mark 5
-- Executing [2970 at incoming-mobile-extensions:1]
NoOp("SIP/4831-00000029", "4831") in new stack
-- Executing [2970 at incoming-mobile-extensions:2]
Set("SIP/4831-00000029", "CALLERID(num)=prefix4831") in new stack
-- Executing [2970 at incoming-mobile-extensions:3]
Dial("SIP/4831-00000029", "SIP/xxxxxxxxxx at cme,15,t") in new stack
== Using SIP RTP CoS mark 5
-- Called xxxxxxxxxx at cme
-- SIP/cme-0000002a is making progress passing it to
SIP/4831-00000029
-- SIP/cme-0000002a is making progress passing it to
SIP/4831-00000029
== Spawn extension (incoming-mobile-extensions, 2970, 3) exited non-zero
on 'SIP/4831-00000029'
Above, icell=xxxxxxxxxx a ten digit phone number (blocked it out). CME is
Cisco call manager express. Prefix is the 1st 6 digits and CallerID(num)
is a 4 digit extension. The above worked.
Now we add followme in:
exten => 2970,1,NoOp(${CALLERID(name)})
exten => 2970,2,set(CALLERID(num)=${prefix}${CALLERID(num)})
exten => 2970,3,Goto(2970-followme,s,1)
[2970-followme]
exten => s,1,FollowMe(2970)
exten => s,n,VoiceMail(2970}@internal)
exten => s,n,Wait(2)
exten => s,n,Hangup()
== Using SIP RTP CoS mark 5
-- Executing [2970 at incoming-mobile-extensions:1]
NoOp("SIP/4831-00000037", "4831") in new stack
-- Executing [2970 at incoming-mobile-extensions:2]
Set("SIP/4831-00000037", "CALLERID(num)=prefix4831") in new stack
-- Executing [2970 at incoming-mobile-extensions:3]
Goto("SIP/4831-00000037", "2970-followme,s,1") in new stack
-- Goto (2970-followme,s,1)
-- Executing [s at 2970-followme:1] FollowMe("SIP/4831-00000037", "2970")
in new stack
-- <SIP/4831-00000037> Playing 'followme/pls-hold-while-try.ulaw'
(language 'en')
-- Music class default requested but no musiconhold loaded.
-- calling xxxxxxxxxx at outgoing
-- Executing [xxxxxxxxxx at outgoing:1]
Dial("Local/xxxxxxxxxx at outgoing-ace0;2", "SIP/xxxxxxxxxx at cme") in new
stack
== Using SIP RTP CoS mark 5
-- Called xxxxxxxxxx at cme
-- SIP/cme-00000038 is making progress passing it to
Local/xxxxxxxxxx at outgoing-ace0;2
-- Local/xxxxxxxxxx at outgoing-ace0;1 is making progress passing it to
SIP/4831-00000037
-- SIP/cme-00000038 is making progress passing it to
Local/xxxxxxxxxx at outgoing-ace0;2
-- Local/xxxxxxxxxx at outgoing-ace0;1 is making progress passing it to
SIP/4831-00000037
== Spawn extension (outgoing, xxxxxxxxxx, 1) exited non-zero on
'Local/xxxxxxxxxxx at outgoing-ace0;2'
Of course, same as above, prefix is the 1st 6 digits. xxxxxxxxxx is the
same as icell. The caller id is not passed to the mobile phone.
--------------------------------------------------
I just performed a sip debug. I see that the "from" line is anonymous
when followme is used. Without followme, the caller ID is correct in the
from line of SIP.
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----------------------------------------------------------------------
(0133217) lmadsen (administrator) - 2011-03-31 12:58
https://issues.asterisk.org/view.php?id=18973#c133217
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In sip.conf do you have trustrpid=yes and sendrpid=yes enabled?
Issue History
Date Modified Username Field Change
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2011-03-31 12:58 lmadsen Note Added: 0133217
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