<span class="postbody">Hi all,
<br>
<br>
I am using Asterisk <a href="http://1.2.12.1">1.2.12.1</a>. 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.
<br>
<br>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.
<br>
<br>
*0 : Office Intercom in Half Duplex mode (Meaning Page() is being used without the 'd' option)
<br>
<br>
*00 : Office Intercom in Full Duplex mode (Meaning Page() is being used with the 'd' option)
<br>
<br>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.
<br>
<br>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.
<br>
<br>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.
<br>
<br>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.
<br>
<br>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).<br><br>Thanks in advance for any help.<br><br>-Michael Azarian<br>
<br>
<span style="font-weight: bold;"><span style="text-decoration: underline;">Extensions.conf (Note: Local/555@features is the MusicOnHold channel</span></span>
<br>
</span><table align="center" border="0" cellpadding="3" cellspacing="1" width="90%"><tbody><tr> <td><span class="genmed"><b>Code:</b></span></td> </tr> <tr> <td class="code">
<br>
;Office Intercom
<br>
exten => *0,1,Set(TIMEOUT(absolute) = 25)
<br>
exten => *0,n,AGI(page.agi)
<br>
exten => *0,n,Set(CALLERID(name)="PAGE: ${CALLERIDNAME}")
<br>
exten => *0,n,Set(CALLERID(number)=${CALLERIDNUM})
<br>
exten => *0,n,Set(_ALERT_INFO="Ring Answer")
<br>
exten => *0,n,SIPAddHeader(Call-Info:<sip:${IPADDR}>\;answer-after=0)
<br>
exten => *0,n,NoOp(PAGE GROUP IS: ${PAGE_GROUP})
<br>
exten => *0,n,Page(${PAGE_GROUP}&Local/555@features)
<br>
exten => *0,n,Hangup()
<br>
<br>
exten => *00,1,Set(TIMEOUT(absolute) = 25)
<br>
exten => *00,n,AGI(page.agi)
<br>
exten => *00,n,Set(CALLERID(name)="PAGE: ${CALLERIDNAME}")
<br>
exten => *00,n,Set(CALLERID(number)=${CALLERIDNUM})
<br>
exten => *00,n,Set(_ALERT_INFO="Ring Answer")
<br>
exten => *00,n,SIPAddHeader(Call-Info:<sip:${IPADDR}>\;answer-after=0)
<br>
exten => *00,n,NoOp(PAGE GROUP IS: ${PAGE_GROUP})
<br>
exten => *00,n,Page(${PAGE_GROUP}&Local/555@features,d)
<br>
exten => *00,n,Hangup()
<br>
</td> </tr></tbody></table><span class="postbody">
<br>
<br>
<span style="text-decoration: underline;"><span style="font-weight: bold;">CLI Output when *0 is dialed (Note: *0 Dialed from SIP/30)</span></span>
<br>
</span><table align="center" border="0" cellpadding="3" cellspacing="1" width="90%"><tbody><tr> <td><span class="genmed"><b>Code:</b></span></td> </tr> <tr> <td class="code">
<br>
pbx*CLI>
<br>
-- Executing Set("SIP/30-08fd96b8", "TIMEOUT(absolute) = 25") in new stack
<br>
-- Executing AGI("SIP/30-08fd96b8", "page.agi") in new stack
<br>
-- Launched AGI Script /var/lib/asterisk/agi-bin/page.agi
<br>
-- Remote UNIX connection
<br>
-- Remote UNIX connection disconnected
<br>
-- AGI Script page.agi completed, returning 0
<br>
-- Executing Set("SIP/30-08fd96b8", "CALLERID(name)="PAGE: Michael"") in new stack
<br>
-- Executing Set("SIP/30-08fd96b8", "CALLERID(number)=30") in new stack
<br>
-- Executing Set("SIP/30-08fd96b8", "_ALERT_INFO="Ring Answer"") in new stack
<br>
-- Executing SIPAddHeader("SIP/30-08fd96b8", "Call-Info: <sip:<a href="http://192.168.16.50">192.168.16.50</a>>;answer-after=0") in new stack
<br> -- 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
<br> -- Executing Page("SIP/30-08fd96b8",
"SIP/22&SIP/18&SIP/17&SIP/16&SIP/13&SIP/12&SIP/11&Local/555@features")
in new stack
<br>
-- Playing 'beep' (language 'en')
<br>
-- Executing Answer("Local/555@features-3efc,2", "") in new stack
<br>
-- Executing Wait("Local/555@features-3efc,2", "1") in new stack
<br>
-- Launching MeetMe(964896525d|mqxdw(5)) on Local/555@features-3efc,1
<br>
-- Created MeetMe conference 1023 for conference '964896525d'
<br>
-- Launching MeetMe(964896525d|mqxdw(5)) on SIP/17-08ff8cb8
<br>
-- Launching MeetMe(964896525d|mqxdw(5)) on SIP/18-08ff3778
<br>
-- Launching MeetMe(964896525d|mqxdw(5)) on SIP/16-08ffea90
<br>
-- Launching MeetMe(964896525d|mqxdw(5)) on SIP/22-08f81540
<br>
-- Launching MeetMe(964896525d|mqxdw(5)) on SIP/13-08f9d718
<br>
-- Launching MeetMe(964896525d|mqxdw(5)) on SIP/11-08fa9ad8
<br>
-- Launching MeetMe(964896525d|mqxdw(5)) on SIP/12-08fa38f8
<br>
-- Executing MusicOnHold("Local/555@features-3efc,2", "default") in new stack
<br>
-- Started music on hold, class 'default', on Local/555@features-3efc,2
<br>
[b]-- Stopped music on hold on Local/555@features-3efc,2[/b]
<br>
[b]== Spawn extension (features, 555, 3) exited non-zero on 'Local/555@features-3efc,2'[/b]
<br>
-- Hungup 'Zap/pseudo-531090772'
<br>
== Spawn extension (internal-admin, *0, 8) exited non-zero on 'SIP/30-08fd96b8'
<br>
-- Executing Hangup("SIP/30-08fd96b8", "") in new stack
<br>
== Spawn extension (internal-admin, h, 1) exited non-zero on 'SIP/30-08fd96b8'
<br>
pbx*CLI>
<br>
</td> </tr></tbody></table><span class="postbody">
<br>
<br>
Notice: The part I bolded is when the MusicOnHold stopped playing automatically.
<br>
<br>
<span style="text-decoration: underline;"><span style="font-weight: bold;">CLI output when dialing *00</span></span>
<br>
</span><table align="center" border="0" cellpadding="3" cellspacing="1" width="90%"><tbody><tr> <td><span class="genmed"><b>Code:</b></span></td> </tr> <tr> <td class="code">
<br>
pbx*CLI>
<br>
-- Executing Set("SIP/30-08f7ec20", "TIMEOUT(absolute) = 25") in new stack
<br>
-- Executing AGI("SIP/30-08f7ec20", "page.agi") in new stack
<br>
-- Launched AGI Script /var/lib/asterisk/agi-bin/page.agi
<br>
-- Remote UNIX connection
<br>
-- Remote UNIX connection disconnected
<br>
-- AGI Script page.agi completed, returning 0
<br>
-- Executing Set("SIP/30-08f7ec20", "CALLERID(name)="PAGE: Michael"") in new stack
<br>
-- Executing Set("SIP/30-08f7ec20", "CALLERID(number)=30") in new stack
<br>
-- Executing Set("SIP/30-08f7ec20", "_ALERT_INFO="Ring Answer"") in new stack
<br>
-- Executing SIPAddHeader("SIP/30-08f7ec20", "Call-Info: <sip:<a href="http://192.168.16.50">192.168.16.50</a>>;answer-after=0") in new stack
<br> -- 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
<br> -- Executing Page("SIP/30-08f7ec20",
"SIP/22&SIP/18&SIP/17&SIP/16&SIP/13&SIP/12&SIP/11&Local/555@features|d")
in new stack
<br>
-- Playing 'beep' (language 'en')
<br>
-- Executing Answer("Local/555@features-2254,2", "") in new stack
<br>
-- Executing Wait("Local/555@features-2254,2", "1") in new stack
<br>
-- Launching MeetMe(1591753623d|qxdw(5)) on Local/555@features-2254,1
<br>
-- Created MeetMe conference 1023 for conference '1591753623d'
<br>
-- Launching MeetMe(1591753623d|qxdw(5)) on SIP/12-08ff6058
<br>
-- Launching MeetMe(1591753623d|qxdw(5)) on SIP/18-08fa5020
<br>
-- Launching MeetMe(1591753623d|qxdw(5)) on SIP/13-08ff0b18
<br>
-- Launching MeetMe(1591753623d|qxdw(5)) on SIP/22-08f8b688
<br>
-- Launching MeetMe(1591753623d|qxdw(5)) on SIP/11-08ffbe30
<br>
-- Launching MeetMe(1591753623d|qxdw(5)) on SIP/17-08faa560
<br>
-- Launching MeetMe(1591753623d|qxdw(5)) on SIP/16-08feb5d8
<br>
-- Executing MusicOnHold("Local/555@features-2254,2", "default") in new stack
<br>
-- Started music on hold, class 'default', on Local/555@features-2254,2
<br>
[b] -- Stopped music on hold on Local/555@features-2254,2
<br>
== Spawn extension (features, 555, 3) exited non-zero on 'Local/555@features-2254,2'
<br>
-- Hungup 'Zap/pseudo-480533203'[/b]
<br>
== Spawn extension (internal-admin, *00, 8) exited non-zero on 'SIP/30-08f7ec20'
<br>
-- Executing Hangup("SIP/30-08f7ec20", "") in new stack
<br>
== Spawn extension (internal-admin, h, 1) exited non-zero on 'SIP/30-08f7ec20'
<br>
pbx*CLI>
<br>
</td> </tr></tbody></table><span class="postbody">
<br>
<br>
Notice: The part I bolded is when the MusicOnHold stopped on its own (4 seconds). At this point, all audio transmission stops.<br><br><br></span><span class="postbody"><span style="font-weight: bold;">[root@pbx zaptel-1.2.9.1
]# ./zttest
</span><br>
Opened pseudo zap interface, measuring accuracy...
<br>
100.000000% 100.000000% 100.000000% 100.000000% 99.987793% 100.000000% 100.000000%
<br>
100.000000% 100.000000% 99.987793% 100.000000% 100.000000% 100.000000% 100.000000% 99.987793%
<br>
100.000000% 100.000000% 100.000000%
<br>
--- Results after 18 passes ---
<br>
Best: 100.000000 -- Worst: 99.987793 -- Average: 99.997965
<br>
<br>
<br><span style="font-weight: bold;">
[root@pbx]# ztcfg -vvvv
</span><br>
<br>
Zaptel Configuration
<br>
======================
<br>
<br>
<br>
Channel map:
<br>
<br>
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
<br>
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
<br>
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
<br>
Channel 04: FXO Kewlstart (Default) (Slaves: 04)
<br>
Channel 05: FXS Kewlstart (Default) (Slaves: 05)
<br>
Channel 06: FXS Kewlstart (Default) (Slaves: 06)
<br>
Channel 07: FXS Kewlstart (Default) (Slaves: 07)
<br>
Channel 08: FXS Kewlstart (Default) (Slaves: 08)
<br>
Channel 09: FXS Kewlstart (Default) (Slaves: 09)
<br>
Channel 10: FXS Kewlstart (Default) (Slaves: 10)
<br>
Channel 11: FXS Kewlstart (Default) (Slaves: 11)
<br>
Channel 12: FXS Kewlstart (Default) (Slaves: 12)
<br>
Channel 13: FXS Kewlstart (Default) (Slaves: 13)
<br>
Channel 14: FXS Kewlstart (Default) (Slaves: 14)
<br>
Channel 15: FXS Kewlstart (Default) (Slaves: 15)
<br>
Channel 16: FXS Kewlstart (Default) (Slaves: 16)
<br>
<br>
16 channels configured.
<br>
<br>
<br style="font-weight: bold;"><span style="font-weight: bold;">
If I run lsmod |grep zap, here is what I get</span>
<br>
<br><span style="font-weight: bold;">
[root@pbx /]# lsmod |grep zap
</span><br>
zaptel 208388 31 wctdm24xxp
<br>
crc_ccitt 6337 1 zaptel
<br>
[root@pbx /#</span><br>