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nhadie at tbgi.net.ph Guest
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Posted: Tue Jan 08, 2008 12:16 am Post subject: [asterisk-users] conferencing help |
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Hi Matt,
it seems i don't have that command.
*CLI> zap show channels
No such command 'zap' (type 'help' for help)
*CLI>
! abort add ael agent agi
cdr database debug dnsmgr dont dump
dundi
extensions feature group help iax2 include
indication init load local logger meetme
mgcp
mixmonitor moh no realtime reload remove
restart rtp set show sip skinny
soft
stop unload
*CLI> show channeltypes
Type Description Devicestate Indications
Transfer
---------- ----------- ----------- -----------
--------
Feature Feature Proxy Channel Driver no yes no
Agent Call Agent Proxy Channel yes yes no
Local Local Proxy Channel Driver no yes no
Skinny Skinny Client Control Protocol no yes no
Phone Standard Linux Telephony API D no no no
SIP Session Initiation Protocol (S yes yes yes
IAX2 Inter Asterisk eXchange Driver yes yes yes
MGCP Media Gateway Control Protocol no yes no
*CLI> show channeltypes
Type Description Devicestate Indications
Transfer
---------- ----------- ----------- -----------
--------
Feature Feature Proxy Channel Driver no yes no
Agent Call Agent Proxy Channel yes yes no
Local Local Proxy Channel Driver no yes no
Skinny Skinny Client Control Protocol no yes no
Phone Standard Linux Telephony API D no no no
SIP Session Initiation Protocol (S yes yes yes
IAX2 Inter Asterisk eXchange Driver yes yes yes
MGCP Media Gateway Control Protocol no yes no
-- Executing NoOp("SIP/104-58ae", "Using CallerID "104" <104>") in
new stack
-- Executing Set("SIP/104-58ae", "MEETME_ROOMNUM=6000") in new stack
-- Executing GotoIf("SIP/104-58ae", "0?USER") in new stack
-- Executing Answer("SIP/104-58ae", "") in new stack
-- Executing Wait("SIP/104-58ae", "1") in new stack
-- Executing Set("SIP/104-58ae", "MEETME_OPTS=") in new stack
-- Executing Goto("SIP/104-58ae", "STARTMEETME|1") in new stack
-- Goto (from-internal,STARTMEETME,1)
-- Executing MeetMe("SIP/104-58ae", "6000||") in new stack
Matt Riddell wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Nhadie wrote:
Quote: | hi shane,
thanks for your reply. i actually tried 3 phones dialled to the
conference, but cant here anything from those phones. i also enabled the
usercount so i can hear something at least. but still no sound.
i'm using ztdummy, as i dont have a card yet.
|
Can you do a "zap show channels" in the Asterisk console (without the ")
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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matt at venturevoip.com Guest
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Posted: Tue Jan 08, 2008 12:45 am Post subject: [asterisk-users] conferencing help |
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Nhadie wrote:
Quote: | Hi Matt,
it seems i don't have that command.
|
You'll need to make sure that:
1. You have zaptel compiled
2. You compile Asterisk *after* zaptel is compiled and installed
3. You have either modprobed zaptel + ztdummy or made the service and
started it.
You didn't say, is this a straight Asterisk machine or trixbox/freepbx?
If those are done and it still doesn't work then you can report the
errors you get when you type (in the console):
module load chan_zap.so
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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pdhales at optusnet.co... Guest
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Posted: Tue Jan 08, 2008 12:47 am Post subject: [asterisk-users] conferencing help |
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Then it's time to build zaptel, then rebuild asterisk....
later,
PaulH
On Tue, 2008-01-08 at 13:16 +0800, Nhadie wrote:
Quote: | Hi Matt,
it seems i don't have that command.
*CLI> zap show channels
No such command 'zap' (type 'help' for help)
*CLI>
! abort add ael agent agi
cdr database debug dnsmgr dont dump
dundi
extensions feature group help iax2 include
indication init load local logger meetme
mgcp
mixmonitor moh no realtime reload remove
restart rtp set show sip skinny
soft
stop unload
*CLI> show channeltypes
Type Description Devicestate Indications
Transfer
---------- ----------- ----------- -----------
--------
Feature Feature Proxy Channel Driver no yes no
Agent Call Agent Proxy Channel yes yes no
Local Local Proxy Channel Driver no yes no
Skinny Skinny Client Control Protocol no yes no
Phone Standard Linux Telephony API D no no no
SIP Session Initiation Protocol (S yes yes yes
IAX2 Inter Asterisk eXchange Driver yes yes yes
MGCP Media Gateway Control Protocol no yes no
*CLI> show channeltypes
Type Description Devicestate Indications
Transfer
---------- ----------- ----------- -----------
--------
Feature Feature Proxy Channel Driver no yes no
Agent Call Agent Proxy Channel yes yes no
Local Local Proxy Channel Driver no yes no
Skinny Skinny Client Control Protocol no yes no
Phone Standard Linux Telephony API D no no no
SIP Session Initiation Protocol (S yes yes yes
IAX2 Inter Asterisk eXchange Driver yes yes yes
MGCP Media Gateway Control Protocol no yes no
-- Executing NoOp("SIP/104-58ae", "Using CallerID "104" <104>") in
new stack
-- Executing Set("SIP/104-58ae", "MEETME_ROOMNUM=6000") in new stack
-- Executing GotoIf("SIP/104-58ae", "0?USER") in new stack
-- Executing Answer("SIP/104-58ae", "") in new stack
-- Executing Wait("SIP/104-58ae", "1") in new stack
-- Executing Set("SIP/104-58ae", "MEETME_OPTS=") in new stack
-- Executing Goto("SIP/104-58ae", "STARTMEETME|1") in new stack
-- Goto (from-internal,STARTMEETME,1)
-- Executing MeetMe("SIP/104-58ae", "6000||") in new stack
Matt Riddell wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Nhadie wrote:
Quote: | hi shane,
thanks for your reply. i actually tried 3 phones dialled to the
conference, but cant here anything from those phones. i also enabled the
usercount so i can hear something at least. but still no sound.
i'm using ztdummy, as i dont have a card yet.
|
Can you do a "zap show channels" in the Asterisk console (without the ")
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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_______________________________________________
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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tzafrir.cohen at xorco... Guest
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Posted: Tue Jan 08, 2008 2:42 am Post subject: [asterisk-users] conferencing help |
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On Tue, Jan 08, 2008 at 06:45:14PM +1300, Matt Riddell wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Nhadie wrote:
Quote: | Hi Matt,
it seems i don't have that command.
|
You'll need to make sure that:
1. You have zaptel compiled
2. You compile Asterisk *after* zaptel is compiled and installed
3. You have either modprobed zaptel + ztdummy or made the service and
started it.
|
In other words, what is the output of the following command from the
Asteris CLI:
module load chan_zap.so
(This tests both cases right away: gives different error messages in the
different cases)
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
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matt at venturevoip.com Guest
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Posted: Tue Jan 08, 2008 2:56 am Post subject: [asterisk-users] conferencing help |
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Tzafrir Cohen wrote:
Quote: | (This tests both cases right away: gives different error messages in the
different cases)
|
Sweet
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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nhadie at tbgi.net.ph Guest
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Posted: Tue Jan 08, 2008 11:25 pm Post subject: [asterisk-users] conferencing help |
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Hi Steve,
I see. I have this now,
*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo default en
*CLI> load chan_zap.so
Unable to load module chan_zap.so <-- on the log file it says, it as
already loaded that's why it's unable to load.
i tried my calling to my conf 6000
-- Executing Set("SIP/100-081825b0", "MEETME_OPTS=iM") in new stack
-- Executing Goto("SIP/100-081825b0", "STARTMEETME|1") in new stack
-- Goto (from-internal,STARTMEETME,1)
-- Executing MeetMe("SIP/100-081825b0", "6000|iM|") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
-- Created MeetMe conference 1023 for conference '6000'
-- Recording
-- Playing 'vm-rec-name' (language 'en')
-- Executing Set("SIP/100-081825b0", "MEETME_OPTS=iM") in new stack
-- Executing Goto("SIP/100-081825b0", "STARTMEETME|1") in new stack
-- Goto (from-internal,STARTMEETME,1)
-- Executing MeetMe("SIP/100-081825b0", "6000|iM|") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
-- Created MeetMe conference 1023 for conference '6000'
-- Recording
-- Playing 'vm-rec-name' (language 'en')
it's trying to play something 'vm-rec-name' but i cannot hear anything
on the phone. i'm using g711. i'm not using trixbox, i just installed
asterisk, freepbx, zaptel, etc on a debian box. i'm using all the latest
version i downloaded from the website (i used asterisk 1.2).
/usr/include# modprobe -l | grep ztdum
/lib/modules/2.6.18-5-686/misc/ztdummy.ko
/usr/include# modprobe -l | grep zap
/lib/modules/2.6.18-5-686/misc/zaptel.ko
how do i know if my ztdummy is working properly? thanks again!
regards,
nhadie
Steve Edwards wrote:
Quote: | Quote: | dave cantera wrote:
Quote: | nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when
using meetme conferencing.
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|
On Wed, 9 Jan 2008, Nhadie wrote:
Quote: | hi dave thank you for the reply. i have loaded zap and using only
ztdummy but still can't hear anything when i dial ti my conference, i
think this explains it already. will a sangoma card do?
|
I use ztdummy with meetme conferencing and it works fine on CentOS 4.5.
Ztdummy is not an issue until you get xx callers in xx conferences.
I think (but have no empirical data to back it up) that a card yields
better sound quality at higher call levels.
Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
_______________________________________________
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matt at venturevoip.com Guest
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Posted: Wed Jan 09, 2008 12:00 am Post subject: [asterisk-users] conferencing help |
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Nhadie wrote:
Quote: | Hi Steve,
I see. I have this now,
*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo default en
|
That means the zap channel should be ok.
One thing you could do is go to the place you downloaded Zaptel and type:
./zttest -v
Do you get numbers (i.e. something close or closish to 100%)?
Also, if you just have the extensions:
exten => 555,1,Answer()
exten => 555,n,Background(demo-echotest)
exten => 555,n,Echo()
Do you get an answer?
You don't really need the brackets on answer and echo but I usually type
that way and then add options.
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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gary at intele-com.com Guest
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Posted: Wed Jan 09, 2008 12:01 am Post subject: [asterisk-users] conferencing help |
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I will be out of the office on Wednesday, January 9, 2008. If this is an emergency, please call Customer Service at (877) 791-7700. Thank you. |
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nhadie at tbgi.net.ph Guest
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Posted: Wed Jan 09, 2008 2:36 am Post subject: [asterisk-users] conferencing help |
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Hi Matt,
I tried
/usr/local/src/zaptel-1.2.22.1# ./zttest -v
and it just freezes at this.
Opened pseudo zap interface, measuring accuracy...
no more outputs, when i cancelled this is what i got.
--- Results after 0 passes ---
Best: 0.000000 -- Worst: 100.000000 -- Average: 100.000000
does that mean my zaptel is bad?
Matt Riddell wrote:
Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Nhadie wrote:
Quote: | Hi Steve,
I see. I have this now,
*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo default en
|
That means the zap channel should be ok.
One thing you could do is go to the place you downloaded Zaptel and type:
./zttest -v
Do you get numbers (i.e. something close or closish to 100%)?
Also, if you just have the extensions:
exten => 555,1,Answer()
exten => 555,n,Background(demo-echotest)
exten => 555,n,Echo()
Do you get an answer?
You don't really need the brackets on answer and echo but I usually type
that way and then add options.
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
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=fHxG
-----END PGP SIGNATURE-----
_______________________________________________
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matt at venturevoip.com Guest
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Posted: Wed Jan 09, 2008 3:43 am Post subject: [asterisk-users] conferencing help |
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Nhadie wrote:
Quote: | Hi Matt,
I tried
/usr/local/src/zaptel-1.2.22.1# ./zttest -v
and it just freezes at this.
Opened pseudo zap interface, measuring accuracy...
no more outputs, when i cancelled this is what i got.
--- Results after 0 passes ---
Best: 0.000000 -- Worst: 100.000000 -- Average: 100.000000
|
Yeah that's what I thought. Am just trying to remember what caused it
though. Maybe Tzafrir will chime in
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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tzafrir.cohen at xorco... Guest
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Posted: Wed Jan 09, 2008 4:25 am Post subject: [asterisk-users] conferencing help |
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On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote:
Quote: | Hi Matt,
I tried
/usr/local/src/zaptel-1.2.22.1# ./zttest -v
and it just freezes at this.
Opened pseudo zap interface, measuring accuracy...
no more outputs, when i cancelled this is what i got.
--- Results after 0 passes ---
Best: 0.000000 -- Worst: 100.000000 -- Average: 100.000000
does that mean my zaptel is bad?
|
Well, yes.
Is ztdummy loaded?
cat /proc/zaptel/*
What kernel version do you use? What version of Zaptel? What Linux
distribution?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
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nhadie at tbgi.net.ph Guest
|
Posted: Wed Jan 09, 2008 5:26 am Post subject: [asterisk-users] conferencing help |
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Hi Tzafrir,
cat /proc/zaptel/*
Span 1: ZTDUMMY/1 "ZTDUMMY/1 1"
Kernel: 2.6.18-5-686 #1 SMP
Zaptel: zaptel-1.2.20.1
OS: Debian GNU/Linux 4.0
i downgraded my zaptel from 1.2.22.1 to 1.2.20.1 but still the same.
thanks again
regards,
nhadie
Tzafrir Cohen wrote:
Quote: | On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote:
Quote: | Hi Matt,
I tried
/usr/local/src/zaptel-1.2.22.1# ./zttest -v
and it just freezes at this.
Opened pseudo zap interface, measuring accuracy...
no more outputs, when i cancelled this is what i got.
--- Results after 0 passes ---
Best: 0.000000 -- Worst: 100.000000 -- Average: 100.000000
does that mean my zaptel is bad?
|
Well, yes.
Is ztdummy loaded?
cat /proc/zaptel/*
What kernel version do you use? What version of Zaptel? What Linux
distribution?
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