[asterisk-users] Page() paging application problem

Michael mazarian at gmail.com
Thu Sep 14 09:14:35 MST 2006


Hi all,

I am using Asterisk 1.2.12.1. The problem I'm about to describe was also
apparent in 1.2.10, so I know it has nothing to do with the fact that I
upgraded.

I am using the Page() along with Page.agi (AGI used to find SIP channels
that are available and not in use using hints). I have setup 2 extensions to
do an office intercom, *0 and *00.

*0 : Office Intercom in Half Duplex mode (Meaning Page() is being used
without the 'd' option)

*00 : Office Intercom in Full Duplex mode (Meaning Page() is being used with
the 'd' option)

My problem is this: When users dial *0, all the available SIP phones are
added to a MeetMe, as intended by the Page application. When the page is
initiated, users hear the "beep" and their Linksys SPA942's pickup. After
this beep, however, they hear no sound.

I did a test. I added a channel playing MusicOnHold to the MeetMe, and users
can only hear a buzzing sound in the background. Without the MusicOnHold,
users hear nothing.

When users dial *00, sound is actually transmitted, in both directions.
However, due to the fact that the phones are near each other, there is an
extreme echo. After 4 seconds, however, the audio transmission stops. Even
though the audio stops, the MeetMe is still in progress until the user who
initiated the page hangs up.

When using *00, whether I added the extension that plays MusicOnHold to the
page group, or not, audio stops being transmitted after 4 seconds. However,
when looking on the CLI, I notice the MusicOnHold stops right at that
4-second mark.

If anyone has any similar experience, or knows how to help, I could really
use it. Below are snippets of my config files, CLI, and various commands
I've run to diagnose the problem (ztcfg -vvvv, zttest, lsmod |grep zap).

Thanks in advance for any help.

-Michael Azarian

Extensions.conf (Note: Local/555 at features is the MusicOnHold channel
 *Code:*
;Office Intercom
exten => *0,1,Set(TIMEOUT(absolute) = 25)
exten => *0,n,AGI(page.agi)
exten => *0,n,Set(CALLERID(name)="PAGE: ${CALLERIDNAME}")
exten => *0,n,Set(CALLERID(number)=${CALLERIDNUM})
exten => *0,n,Set(_ALERT_INFO="Ring Answer")
exten => *0,n,SIPAddHeader(Call-Info:<sip:${IPADDR}>\;answer-after=0)
exten => *0,n,NoOp(PAGE GROUP IS: ${PAGE_GROUP})
exten => *0,n,Page(${PAGE_GROUP}&Local/555 at features)
exten => *0,n,Hangup()

exten => *00,1,Set(TIMEOUT(absolute) = 25)
exten => *00,n,AGI(page.agi)
exten => *00,n,Set(CALLERID(name)="PAGE: ${CALLERIDNAME}")
exten => *00,n,Set(CALLERID(number)=${CALLERIDNUM})
exten => *00,n,Set(_ALERT_INFO="Ring Answer")
exten => *00,n,SIPAddHeader(Call-Info:<sip:${IPADDR}>\;answer-after=0)
exten => *00,n,NoOp(PAGE GROUP IS: ${PAGE_GROUP})
exten => *00,n,Page(${PAGE_GROUP}&Local/555 at features,d)
exten => *00,n,Hangup()


CLI Output when *0 is dialed (Note: *0 Dialed from SIP/30)
 *Code:*
pbx*CLI>
    -- Executing Set("SIP/30-08fd96b8", "TIMEOUT(absolute) = 25") in new
stack
    -- Executing AGI("SIP/30-08fd96b8", "page.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/page.agi
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- AGI Script page.agi completed, returning 0
    -- Executing Set("SIP/30-08fd96b8", "CALLERID(name)="PAGE: Michael"") in
new stack
    -- Executing Set("SIP/30-08fd96b8", "CALLERID(number)=30") in new stack
    -- Executing Set("SIP/30-08fd96b8", "_ALERT_INFO="Ring Answer"") in new
stack
    -- Executing SIPAddHeader("SIP/30-08fd96b8", "Call-Info: <sip:
192.168.16.50>;answer-after=0") in new stack
    -- Executing NoOp("SIP/30-08fd96b8", "PAGE GROUP IS:
SIP/22&SIP/18&SIP/17&SIP/16&SIP/13&SIP/12&SIP/11") in new stack
    -- Executing Page("SIP/30-08fd96b8", "
SIP/22&SIP/18&SIP/17&SIP/16&SIP/13&SIP/12&SIP/11&Local/555 at features") in new
stack
    -- Playing 'beep' (language 'en')
    -- Executing Answer("Local/555 at features-3efc,2", "") in new stack
    -- Executing Wait("Local/555 at features-3efc,2", "1") in new stack
    -- Launching MeetMe(964896525d|mqxdw(5)) on Local/555 at features-3efc,1
    -- Created MeetMe conference 1023 for conference '964896525d'
    -- Launching MeetMe(964896525d|mqxdw(5)) on SIP/17-08ff8cb8
    -- Launching MeetMe(964896525d|mqxdw(5)) on SIP/18-08ff3778
    -- Launching MeetMe(964896525d|mqxdw(5)) on SIP/16-08ffea90
    -- Launching MeetMe(964896525d|mqxdw(5)) on SIP/22-08f81540
    -- Launching MeetMe(964896525d|mqxdw(5)) on SIP/13-08f9d718
    -- Launching MeetMe(964896525d|mqxdw(5)) on SIP/11-08fa9ad8
    -- Launching MeetMe(964896525d|mqxdw(5)) on SIP/12-08fa38f8
    -- Executing MusicOnHold("Local/555 at features-3efc,2", "default") in new
stack
    -- Started music on hold, class 'default', on Local/555 at features-3efc,2
    [b]-- Stopped music on hold on Local/555 at features-3efc,2[/b]
  [b]== Spawn extension (features, 555, 3) exited non-zero on '
Local/555 at features-3efc,2'[/b]
    -- Hungup 'Zap/pseudo-531090772'
  == Spawn extension (internal-admin, *0, 8) exited non-zero on
'SIP/30-08fd96b8'
    -- Executing Hangup("SIP/30-08fd96b8", "") in new stack
  == Spawn extension (internal-admin, h, 1) exited non-zero on
'SIP/30-08fd96b8'
pbx*CLI>


Notice: The part I bolded is when the MusicOnHold stopped playing
automatically.

CLI output when dialing *00
 *Code:*
pbx*CLI>
    -- Executing Set("SIP/30-08f7ec20", "TIMEOUT(absolute) = 25") in new
stack
    -- Executing AGI("SIP/30-08f7ec20", "page.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/page.agi
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- AGI Script page.agi completed, returning 0
    -- Executing Set("SIP/30-08f7ec20", "CALLERID(name)="PAGE: Michael"") in
new stack
    -- Executing Set("SIP/30-08f7ec20", "CALLERID(number)=30") in new stack
    -- Executing Set("SIP/30-08f7ec20", "_ALERT_INFO="Ring Answer"") in new
stack
    -- Executing SIPAddHeader("SIP/30-08f7ec20", "Call-Info: <sip:
192.168.16.50>;answer-after=0") in new stack
    -- Executing NoOp("SIP/30-08f7ec20", "PAGE GROUP IS:
SIP/22&SIP/18&SIP/17&SIP/16&SIP/13&SIP/12&SIP/11") in new stack
    -- Executing Page("SIP/30-08f7ec20", "
SIP/22&SIP/18&SIP/17&SIP/16&SIP/13&SIP/12&SIP/11&Local/555 at features|d") in
new stack
    -- Playing 'beep' (language 'en')
    -- Executing Answer("Local/555 at features-2254,2", "") in new stack
    -- Executing Wait("Local/555 at features-2254,2", "1") in new stack
    -- Launching MeetMe(1591753623d|qxdw(5)) on Local/555 at features-2254,1
    -- Created MeetMe conference 1023 for conference '1591753623d'
    -- Launching MeetMe(1591753623d|qxdw(5)) on SIP/12-08ff6058
    -- Launching MeetMe(1591753623d|qxdw(5)) on SIP/18-08fa5020
    -- Launching MeetMe(1591753623d|qxdw(5)) on SIP/13-08ff0b18
    -- Launching MeetMe(1591753623d|qxdw(5)) on SIP/22-08f8b688
    -- Launching MeetMe(1591753623d|qxdw(5)) on SIP/11-08ffbe30
    -- Launching MeetMe(1591753623d|qxdw(5)) on SIP/17-08faa560
    -- Launching MeetMe(1591753623d|qxdw(5)) on SIP/16-08feb5d8
    -- Executing MusicOnHold("Local/555 at features-2254,2", "default") in new
stack
    -- Started music on hold, class 'default', on Local/555 at features-2254,2
  [b]  -- Stopped music on hold on Local/555 at features-2254,2
  == Spawn extension (features, 555, 3) exited non-zero on '
Local/555 at features-2254,2'
    -- Hungup 'Zap/pseudo-480533203'[/b]
  == Spawn extension (internal-admin, *00, 8) exited non-zero on
'SIP/30-08f7ec20'
    -- Executing Hangup("SIP/30-08f7ec20", "") in new stack
  == Spawn extension (internal-admin, h, 1) exited non-zero on
'SIP/30-08f7ec20'
pbx*CLI>


Notice: The part I bolded is when the MusicOnHold stopped on its own (4
seconds). At this point, all audio transmission stops.


[root at pbx zaptel-1.2.9.1]# ./zttest
Opened pseudo zap interface, measuring accuracy...
100.000000% 100.000000% 100.000000% 100.000000% 99.987793% 100.000000%
100.000000%
100.000000% 100.000000% 99.987793% 100.000000% 100.000000% 100.000000%
100.000000% 99.987793%
100.000000% 100.000000% 100.000000%
--- Results after 18 passes ---
Best: 100.000000 -- Worst: 99.987793 -- Average: 99.997965


[root at pbx]# ztcfg -vvvv

Zaptel Configuration
======================


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Kewlstart (Default) (Slaves: 04)
Channel 05: FXS Kewlstart (Default) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Slaves: 08)
Channel 09: FXS Kewlstart (Default) (Slaves: 09)
Channel 10: FXS Kewlstart (Default) (Slaves: 10)
Channel 11: FXS Kewlstart (Default) (Slaves: 11)
Channel 12: FXS Kewlstart (Default) (Slaves: 12)
Channel 13: FXS Kewlstart (Default) (Slaves: 13)
Channel 14: FXS Kewlstart (Default) (Slaves: 14)
Channel 15: FXS Kewlstart (Default) (Slaves: 15)
Channel 16: FXS Kewlstart (Default) (Slaves: 16)

16 channels configured.


If I run lsmod |grep zap, here is what I get

[root at pbx /]# lsmod |grep zap
zaptel 208388 31 wctdm24xxp
crc_ccitt 6337 1 zaptel
[root at pbx /#
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060914/c8b669b3/attachment.htm


More information about the asterisk-users mailing list