[asterisk-bugs] [Asterisk 0012464]: Unable to receive caller id from Coral SDBX PBX to Asterisk

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12464 
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Reported By:                ramaseshi
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12464
Category:                   Channels/chan_zap
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.16 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             04-17-2008 01:47 CDT
Last Modified:              04-21-2008 16:36 CDT
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Summary:                    Unable to receive caller id from Coral SDBX PBX to
Asterisk
Description: 

I am using asterisk 1.4 latest build. I have one 100XP FXO card inserted
in the Asterisk box.

I integrated appliance with PBX. It is SDBX coral telecom product. One
extension line 302 is plugged to FXO card. When i dial from 307 which is a
PBX extension (analog phone) to 302 call is coming to asterisk and ivr is
played and acceptign DTNF digits also and call is going to SIP phone. But
it displays as call from asterisk.

I applied this patch 0006683 it working fine when i use POT

########## PLease see below log messages for reference
###################


 --- (0 headers 1 lines) ---
[Apr 17 02:28:24] DEBUG[4273]: chan_zap.c:7281 do_monitor: Monitor
doohicky got event Ring Begin on channel 1
[Apr 17 02:28:24] DEBUG[4273]: chan_zap.c:7281 do_monitor: Monitor
doohicky got event Ring/Answered on channel 1
[Apr 17 02:28:24] DEBUG[4273]: dsp.c:1663 ast_dsp_set_busy_pattern: dsp
busy pattern set to 0,0
[Apr 17 02:28:24] DEBUG[4273]: chan_zap.c:5611 zt_new:
http://bugs.digium.com/view.php?id=372 If (core)
Needs a rethink i = 81c2010, i->rdnis = 81c2d48
    -- Starting simple switch on 'Zap/1-1'
[Apr 17 02:28:26] DEBUG[4273]: chan_zap.c:6771 ss_thread: CallerID number:
04030529261, name: (null), flags=4
[Apr 17 02:28:26] DEBUG[4273]: pbx.c:1699 pbx_extension_helper: Launching
'Wait'
    -- Executing [s at from-pstn:1] Wait("Zap/1-1", "1") in new stack
[Apr 17 02:28:26] DEBUG[4273]: chan_zap.c:4774 __zt_exception: Exception
on 13, channel 1
[Apr 17 02:28:26] DEBUG[4273]: chan_zap.c:3864 zt_handle_event: Got event
Ring Begin(18) on channel 1 (index 0)
[Apr 17 02:28:27] DEBUG[4273]: pbx.c:1699 pbx_extension_helper: Launching
'Answer'
    -- Executing [s at from-pstn:2] Answer("Zap/1-1", "") in new stack
[Apr 17 02:28:27] DEBUG[4273]: chan_zap.c:2976 zt_answer: Took Zap/1-1 off
hook
[Apr 17 02:28:27] DEBUG[4273]: chan_zap.c:1558 zt_enable_ec: Enabled echo
cancellation on channel 1
[Apr 17 02:28:27] DEBUG[4273]: chan_zap.c:1577 zt_train_ec: Engaged echo
training on channel 1
[Apr 17 02:28:27] DEBUG[4273]: pbx.c:1543
pbx_substitute_variables_helper_full: Function result is '04030529261'
[Apr 17 02:28:27] DEBUG[4273]: pbx.c:1699 pbx_extension_helper: Launching
'NoOp'
    -- Executing [s at from-pstn:3] NoOp("Zap/1-1", "04030529261") in new
stack
[Apr 17 02:28:27] DEBUG[4273]: pbx.c:1699 pbx_extension_helper: Launching
'Set'
    -- Executing [s at from-pstn:4] Set("Zap/1-1", "TIMEOUT(digit)=5") in new
stack
    -- Digit timeout set to 5
[Apr 17 02:28:27] DEBUG[4273]: pbx.c:1699 pbx_extension_helper: Launching
'Set'
    -- Executing [s at from-pstn:5] Set("Zap/1-1", "TIMEOUT(response)=10") in
new stack
    -- Response timeout set to 10
[Apr 17 02:28:27] DEBUG[4273]: pbx.c:1699 pbx_extension_helper: Launching
'Playback'
    -- Executing [s at from-pstn:6] Playback("Zap/1-1",
"enter-ext-of-person") in new stack
[Apr 17 02:28:27] DEBUG[4273]: channel.c:2787 set_format: Set channel
Zap/1-1 to write format gsm
[Apr 17 02:28:27] DEBUG[4273]: file.c:411 ast_filehelper: From function
ast_filehelper ACTION
[Apr 17 02:28:27] DEBUG[4273]: channel.c:1934 ast_settimeout: Scheduling
timer at 160 sample intervals
    -- <Zap/1-1> Playing 'enter-ext-of-person' (language 'en')
 


====================================================================== 

---------------------------------------------------------------------- 
 qwell - 04-21-08 16:36  
---------------------------------------------------------------------- 
Is it just the name that is being set to "asterisk"?  Analog does not
support callerid name.  You'll need to set it before sending the call to
the SIP channel. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
04-21-08 16:36  qwell          Note Added: 0085775                          
======================================================================