Re: [asterisk-users] Problem with Portech MV-372
Pascal Bruno ha scritto: Hi, I am experiencing a weird issue with my MV-372. Mobile1 Mobile2 are both registered to my asterisk server, I am able to use them for outgoing call with no problem, but when I call the sims in my gateway, they are routed to the right context/extension/priority, but as soon as I hangup, the sim unregistered from asterisk and tries to register with my the callerid of the last incoming call as follows: Registration from 'mv372 sip:+17546542...@77.29.9.16 mailto:sip%3a%2b17546542...@77.29.9.16' failed for '97.26.196.2' - No matching peer found and the registration fails since I dont have a peer created for +17546542334 Anyone have an idea on how to go about fixing this? I am using the MV-372 (in and out) and dont have this problem. First: check if the device has the LATEST firmware, if not, upgrade. Second: send an email to the portech service. :-) In the past there was a lot of bug in the firmware of the MV372, and also buggy hardware release, but not now... so check also the hardware version (in the web interface - firmware update - top on the page). I think this is not asterisk issue. Bye. attachment: massimo.vcf signature.asc Description: OpenPGP digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASTERISK and SNMP
Hi all, I am currently not able to configure SNMP for asterisk, but I am not able to acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/) Does somebody has an example of smnpd.conf file wich is working ? regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN30 Timing Sources (Jon Morgan)
Quoth Jon Morgan jon.mor...@motors.co.uk We have a 2 port Digium TE220P card, one span is configured to connect to our ISDN30 provider (British Telecom), the other span connects to our internal PBX. Here's the zaptel.conf snip: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 FWIW, I (also BT ISDN30 on span 1 with a PBX on the second port of a TE205P) have the following zaptel.conf. span=1,1,1,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,1,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 I do all my call recording in asterisk so can't comment on that but the PBX users are not complaining about the quality. -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
Hello Mickael
Here You have the snmpd.conf file
cat /etc/snmp/snmpd.conf
rocommunity your_community
master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS=${SNMPD_FLAGS} -x /var/agentx/master
mibs +ASTERISK-MIB
and also you need create file /etc/snmp/snmp.conf with following entry
mibs +ASTERISK-MIB
cat /etc/snmp/snmp.conf
mibs +ASTERISK-MIB
Next use command snmpwalk -c your_community -v 1 localhost asterisk
to check is everything correct.
Michał
2009/11/27 mickael ropars mrop...@gmail.com:
Hi all,
I am currently not able to configure SNMP for asterisk, but I am not able to
acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/)
Does somebody has an example of smnpd.conf file wich is working ?
regards
Mickael
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Re: [asterisk-users] ASTERISK and SNMP
Hi Michal,
thanks a lot for you quick answer I appreciate.
I run your commands and I have the following answer
[localhost snmp]# snmpwalk -c local -v 1 localhost asterisk
no answer
[localhost snmp]# snmpwalk -c local -v 2c localhost asterisk
ASTERISK-MIB::asterisk = No Such Object available on this agent at this OID
since I don't know well snmp what's going wrong ?
regards
Mickael
2009/11/27 michal kalinowski michal.kalinow...@interia.pl
Hello Mickael
Here You have the snmpd.conf file
cat /etc/snmp/snmpd.conf
rocommunity your_community
master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS=${SNMPD_FLAGS} -x /var/agentx/master
mibs +ASTERISK-MIB
and also you need create file /etc/snmp/snmp.conf with following entry
mibs +ASTERISK-MIB
cat /etc/snmp/snmp.conf
mibs +ASTERISK-MIB
Next use command snmpwalk -c your_community -v 1 localhost asterisk
to check is everything correct.
Michał
2009/11/27 mickael ropars mrop...@gmail.com:
Hi all,
I am currently not able to configure SNMP for asterisk, but I am not able
to
acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/)
Does somebody has an example of smnpd.conf file wich is working ?
regards
Mickael
___
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Re: [asterisk-users] ASTERISK and SNMP
What operating system do You have ? What asterisk version You compile ?
After install net-snmp do You recompile asterisk with res_snmp module ?
I'm used instruction from here
http://voxilla.com/2009/02/03/configuring-asterisk-snmp-support-1131
and everything work correctly.
BR,
Michał
W dniu 27 listopada 2009 11:18 użytkownik mickael ropars
mrop...@gmail.com napisał:
Hi Michal,
thanks a lot for you quick answer I appreciate.
I run your commands and I have the following answer
[localhost snmp]# snmpwalk -c local -v 1 localhost asterisk
no answer
[localhost snmp]# snmpwalk -c local -v 2c localhost asterisk
ASTERISK-MIB::asterisk = No Such Object available on this agent at this OID
since I don't know well snmp what's going wrong ?
regards
Mickael
2009/11/27 michal kalinowski michal.kalinow...@interia.pl
Hello Mickael
Here You have the snmpd.conf file
cat /etc/snmp/snmpd.conf
rocommunity your_community
master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS=${SNMPD_FLAGS} -x /var/agentx/master
mibs +ASTERISK-MIB
and also you need create file /etc/snmp/snmp.conf with following entry
mibs +ASTERISK-MIB
cat /etc/snmp/snmp.conf
mibs +ASTERISK-MIB
Next use command snmpwalk -c your_community -v 1 localhost asterisk
to check is everything correct.
Michał
2009/11/27 mickael ropars mrop...@gmail.com:
Hi all,
I am currently not able to configure SNMP for asterisk, but I am not
able to
acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/)
Does somebody has an example of smnpd.conf file wich is working ?
regards
Mickael
___
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Re: [asterisk-users] ASTERISK and SNMP
I am running on Trixbox so my OS is Cent0S 5.4 and the Asterisk version is
1.4.22-4
on asterisk side Snmp module is running:
module load res_snmp.so
== Parsing '/etc/asterisk/res_snmp.conf': Found
Loading [Sub]Agent Module
Loaded res_snmp.so = (SNMP [Sub]Agent for Asterisk)
see below my snmpd.conf file (I remove commented line for an easy reading)
regards
Mickael
###
# Access Control
###
# First, map the community name (COMMUNITY) into a security name
# (local and mynetwork, depending on where the request is coming
# from):
# sec.name source community
com2sec local localhost COMMUNITY
com2sec mynetwork NETWORK/24 COMMUNITY
rwcommunity local
rocommunity local
# Second, map the security names into group names:
# sec.model sec.name
group MyRWGroup v1 local
group MyRWGroup v2clocal
group MyRWGroup usmlocal
group MyROGroup v1 mynetwork
group MyROGroup v2cmynetwork
group MyROGroup usmmynetwork
# Third, create a view for us to let the groups have rights to:
# incl/excl subtree mask
view allincluded .1 80
# Finally, grant the 2 groups access to the 1 view with different
# write permissions:
#context sec.model sec.level match read write notif
access MyROGroup any noauthexact allnone none
access MyRWGroup any noauthexact allallnone
###
# System contact information
#
syslocation Right here, right now.
syscontact Me m...@somewhere.org
###
# Process checks.
#
# Make sure mountd is running
proc mountd
# Make sure there are no more than 4 ntalkds running, but 0 is ok too.
proc ntalkd 4
# Make sure at least one sendmail, but less than or equal to 10 are
running.
proc sendmail 10 1
###
# Executables/scripts
#
# a simple hello world
exec echotest /bin/echo hello world
###
# disk checks
#
disk / 1
###
# load average checks
#
# Check for loads:
load 12 14 14
###
# Extensible sections.
#
###
# Pass through control.
#
###
# Subagent control
#
master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS=${SNMPD_FLAGS} -x /var/agentx/master
mibs +ASTERISK-MIB
###
# Further Information
2009/11/27 michal kalinowski michal.kalinow...@interia.pl
What operating system do You have ? What asterisk version You compile ?
After install net-snmp do You recompile asterisk with res_snmp module ?
I'm used instruction from here
http://voxilla.com/2009/02/03/configuring-asterisk-snmp-support-1131
and everything work correctly.
BR,
Michał
W dniu 27 listopada 2009 11:18 użytkownik mickael ropars
mrop...@gmail.com napisał:
Hi Michal,
thanks a lot for you quick answer I appreciate.
I run your commands and I have the following answer
[localhost snmp]# snmpwalk -c local -v 1 localhost asterisk
no answer
[localhost snmp]# snmpwalk -c local -v 2c localhost asterisk
ASTERISK-MIB::asterisk = No Such Object available on this agent at this
OID
since I don't know well snmp what's going wrong ?
regards
Mickael
2009/11/27 michal kalinowski michal.kalinow...@interia.pl
Hello Mickael
Here You have the snmpd.conf file
cat /etc/snmp/snmpd.conf
rocommunity your_community
master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS=${SNMPD_FLAGS} -x /var/agentx/master
mibs +ASTERISK-MIB
and also you need create file /etc/snmp/snmp.conf with following entry
mibs +ASTERISK-MIB
cat /etc/snmp/snmp.conf
mibs +ASTERISK-MIB
Next use command snmpwalk -c your_community -v 1 localhost asterisk
to check is everything correct.
Michał
2009/11/27 mickael ropars mrop...@gmail.com:
Hi all,
I am currently not able to configure SNMP for asterisk, but I am not
able to
acess to the asterisk MIB (the asterisk MIB is in
/usr/share/snmp/mibs/)
Does somebody has an example of smnpd.conf file wich is working ?
regards
Mickael
___
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Re: [asterisk-users] Unable to open sound file error
List. How can I resolve this problem? I've searched on the web but, can't really find a solution. Please help. --- On Wed, 11/25/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: [asterisk-users] Unable to open sound file error To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 25, 2009, 7:45 PM Hello. I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to? I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw but asterisk is telling me it doesn't. Here's what I get when I try to dial the extension for test: [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File good does not exist in any format [Nov 25 20:44:41] WARNING[4334]: file.c:933 ast_streamfile: Unable to open good (format 0x8 (alaw)): No such file or directory [Nov 25 20:44:41] WARNING[4334]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-09b52260 for good -- Executing [...@default:12] BackGround(SIP/102-09b52260, evening ) in new stack [Nov 25 20:44:41] WARNING[4334]: file.c:650 ast_openstream_full: File evening does not exist in any format [Nov 25 20:44:41] WARNING[4334]: file.c:933 ast_streamfile: Unable to open evening (format 0x8 (alaw)): No such file or directory [Nov 25 20:44:41] WARNING[4334]: pbx.c:8251 pbx_builtin_background: ast_streamfile failed on SIP/102-09b52260 for evening -- Executing [...@default:13] Hangup(SIP/102-09b52260, ) in new stack Any suggestions? Thanks in advanced for your help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
Michal
please wait I found some issues in my con file
2009/11/27 mickael ropars mrop...@gmail.com
I am running on Trixbox so my OS is Cent0S 5.4 and the Asterisk version is
1.4.22-4
on asterisk side Snmp module is running:
module load res_snmp.so
== Parsing '/etc/asterisk/res_snmp.conf': Found
Loading [Sub]Agent Module
Loaded res_snmp.so = (SNMP [Sub]Agent for Asterisk)
see below my snmpd.conf file (I remove commented line for an easy reading)
regards
Mickael
###
# Access Control
###
# First, map the community name (COMMUNITY) into a security name
# (local and mynetwork, depending on where the request is coming
# from):
# sec.name source community
com2sec local localhost COMMUNITY
com2sec mynetwork NETWORK/24 COMMUNITY
rwcommunity local
rocommunity local
# Second, map the security names into group names:
# sec.model sec.name
group MyRWGroup v1 local
group MyRWGroup v2clocal
group MyRWGroup usmlocal
group MyROGroup v1 mynetwork
group MyROGroup v2cmynetwork
group MyROGroup usmmynetwork
# Third, create a view for us to let the groups have rights to:
# incl/excl subtree mask
view allincluded .1 80
# Finally, grant the 2 groups access to the 1 view with different
# write permissions:
#context sec.model sec.level match read write notif
access MyROGroup any noauthexact allnone none
access MyRWGroup any noauthexact allallnone
###
# System contact information
#
syslocation Right here, right now.
syscontact Me m...@somewhere.org
###
# Process checks.
#
# Make sure mountd is running
proc mountd
# Make sure there are no more than 4 ntalkds running, but 0 is ok too.
proc ntalkd 4
# Make sure at least one sendmail, but less than or equal to 10 are
running.
proc sendmail 10 1
###
# Executables/scripts
#
# a simple hello world
exec echotest /bin/echo hello world
###
# disk checks
#
disk / 1
###
# load average checks
#
# Check for loads:
load 12 14 14
###
# Extensible sections.
#
###
# Pass through control.
#
###
# Subagent control
#
master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS=${SNMPD_FLAGS} -x /var/agentx/master
mibs +ASTERISK-MIB
###
# Further Information
2009/11/27 michal kalinowski michal.kalinow...@interia.pl
What operating system do You have ? What asterisk version You compile ?
After install net-snmp do You recompile asterisk with res_snmp module ?
I'm used instruction from here
http://voxilla.com/2009/02/03/configuring-asterisk-snmp-support-1131
and everything work correctly.
BR,
Michał
W dniu 27 listopada 2009 11:18 użytkownik mickael ropars
mrop...@gmail.com napisał:
Hi Michal,
thanks a lot for you quick answer I appreciate.
I run your commands and I have the following answer
[localhost snmp]# snmpwalk -c local -v 1 localhost asterisk
no answer
[localhost snmp]# snmpwalk -c local -v 2c localhost asterisk
ASTERISK-MIB::asterisk = No Such Object available on this agent at this
OID
since I don't know well snmp what's going wrong ?
regards
Mickael
2009/11/27 michal kalinowski michal.kalinow...@interia.pl
Hello Mickael
Here You have the snmpd.conf file
cat /etc/snmp/snmpd.conf
rocommunity your_community
master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS=${SNMPD_FLAGS} -x /var/agentx/master
mibs +ASTERISK-MIB
and also you need create file /etc/snmp/snmp.conf with following entry
mibs +ASTERISK-MIB
cat /etc/snmp/snmp.conf
mibs +ASTERISK-MIB
Next use command snmpwalk -c your_community -v 1 localhost asterisk
to check is everything correct.
Michał
2009/11/27 mickael ropars mrop...@gmail.com:
Hi all,
I am currently not able to configure SNMP for asterisk, but I am not
able to
acess to the asterisk MIB (the asterisk MIB is in
[asterisk-users] Virtual Phone for CDR Logging
Hi, I am new to the list, so I hope my questions aren't too stupid. I am using Asterisk 1.4.21.2 and already set it up to use realtime, so a CDR for an incoming SIP call is written in my mysql database. This works fine. The problem is that I don't want to have my phone ringing all the time. I just need a CDR of everyone how is calling me and to read out the CDR from my PHP script. I tried to replace the Dial(SIP/6000|30) command in the extensions table by Ringing(),Wait(5),Busy() but now no CDR entry is created. Same with Ringing(),Wait(5),Hangup(). Looks like I need a Dial() command for CDR. How can I create a virtual phone of some kind, so I get a CDR entry without actually accepting the call. Thanks in advance! Greetings Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime SIP Register
Hi, I would like to have my register directives from sip.conf in my mysql database: register = user[:secret[:authuse...@host[:port][/extension] I already have the sip users and the other config in the DB but how to get the register in there, too? In an old mail (Mon Oct 3 00:49:15 MST 2005) Olle E. Johansson said the [general] section can only be static. Has there anything changed in the last 4 years? Thanks! Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
I use CentOS, and it works fairly well. But I had to piece together info from
several places. I've tried it several different wants and this way worked, as
long as asterisk is run as root.
Copy asterisk-mib.txt and digium-mib.txt from asterisk_source/doc to
/usr/share/snmp/mibs/
mkdir /var/agentx
touch /var/agentx/master
My /etc/asterisk/res_snmp.conf
;
; Configuration file for res_snmp
;
[general]
; We run as a subagent per default -- to run as a full agent
; we must run as root (to be able to bind to port 161)
;subagent = yes
; SNMP must be explicitly enabled to be active
enabled = yes
My snmp.conf
rwcommunity private 127.0.0.1
rocommunity public
disk /
master agentx
agentXperms 0660 0550 root root
restart snmp and the /var/agentx/master should look like srw-rw 1 root root
0 Nov 25 11:31 /var/agentx/master
restart asterisk manually and you see a net-snmp connect.
export MIBS=+ASTERISK-MIB
You should be able to to do a snmpwalk -v 2c -c public localhost asterisk
Regards
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mickael ropars
Sent: 27 November 2009 11:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ASTERISK and SNMP
Michal
please wait I found some issues in my con file
2009/11/27 mickael ropars mrop...@gmail.com
I am running on Trixbox so my OS is Cent0S 5.4 and the Asterisk version is
1.4.22-4
on asterisk side Snmp module is running:
module load res_snmp.so
== Parsing '/etc/asterisk/res_snmp.conf': Found
Loading [Sub]Agent Module
Loaded res_snmp.so = (SNMP [Sub]Agent for Asterisk)
see below my snmpd.conf file (I remove commented line for an easy reading)
regards
Mickael
###
# Access Control
###
# First, map the community name (COMMUNITY) into a security name
# (local and mynetwork, depending on where the request is coming
# from):
# sec.name source community
com2sec local localhost COMMUNITY
com2sec mynetwork NETWORK/24 COMMUNITY
rwcommunity local
rocommunity local
# Second, map the security names into group names:
# sec.model sec.name
group MyRWGroup v1 local
group MyRWGroup v2clocal
group MyRWGroup usmlocal
group MyROGroup v1 mynetwork
group MyROGroup v2cmynetwork
group MyROGroup usmmynetwork
# Third, create a view for us to let the groups have rights to:
# incl/excl subtree mask
view allincluded .1 80
# Finally, grant the 2 groups access to the 1 view with different
# write permissions:
#context sec.model sec.level match read write notif
access MyROGroup any noauthexact allnone none
access MyRWGroup any noauthexact allallnone
###
# System contact information
#
syslocation Right here, right now.
syscontact Me m...@somewhere.org
###
# Process checks.
#
# Make sure mountd is running
proc mountd
# Make sure there are no more than 4 ntalkds running, but 0 is ok too.
proc ntalkd 4
# Make sure at least one sendmail, but less than or equal to 10 are running.
proc sendmail 10 1
###
# Executables/scripts
#
# a simple hello world
exec echotest /bin/echo hello world
###
# disk checks
#
disk / 1
###
# load average checks
#
# Check for loads:
load 12 14 14
###
# Extensible sections.
#
###
# Pass through control.
#
###
# Subagent control
#
master agentx
agentXperms 0660 0550 nobody asterisk
SNMPD_FLAGS=${SNMPD_FLAGS} -x /var/agentx/master
mibs +ASTERISK-MIB
###
# Further Information
2009/11/27 michal kalinowski michal.kalinow...@interia.pl
What operating system do You have ? What asterisk version You compile ?
After install net-snmp do You recompile asterisk with res_snmp module ?
I'm used instruction from here
http://voxilla.com/2009/02/03/configuring-asterisk-snmp-support-1131
and everything work correctly.
BR,
Michał
W dniu 27 listopada 2009 11:18 użytkownik mickael ropars
[asterisk-users] 1800 DID Provider - Suggestion
Hello All, Do you guys suggest any 1800 DID Provider in the US ? I'm having a hard time to find one. Thanks, Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
thanks all for your help, I really appreciate. now it's working My problem was due to Nov 27 12:56:28 trixbox1 snmpd[5743]: /etc/snmp/snmpd.conf: line 61: Error: example config COMMUNITY not properly configured Nov 27 12:56:28 trixbox1 snmpd[5743]: /etc/snmp/snmpd.conf: line 62: Error: example config NETWORK not properly configured with the link michal gave to me I succeed in getting asterisk working regards Mickael 2009/11/27 Lee Archer lee.arc...@thebigword.com I use CentOS, and it works fairly well. But I had to piece together info from several places. I've tried it several different wants and this way worked, as long as asterisk is run as root. Copy asterisk-mib.txt and digium-mib.txt from asterisk_source/doc to /usr/share/snmp/mibs/ mkdir /var/agentx touch /var/agentx/master My /etc/asterisk/res_snmp.conf ; ; Configuration file for res_snmp ; [general] ; We run as a subagent per default -- to run as a full agent ; we must run as root (to be able to bind to port 161) ;subagent = yes ; SNMP must be explicitly enabled to be active enabled = yes My snmp.conf rwcommunity private 127.0.0.1 rocommunity public disk / master agentx agentXperms 0660 0550 root root restart snmp and the /var/agentx/master should look like srw-rw 1 root root 0 Nov 25 11:31 /var/agentx/master restart asterisk manually and you see a net-snmp connect. export MIBS=+ASTERISK-MIB You should be able to to do a snmpwalk -v 2c -c public localhost asterisk Regards Lee *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *mickael ropars *Sent:* 27 November 2009 11:58 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] ASTERISK and SNMP Michal please wait I found some issues in my con file 2009/11/27 mickael ropars mrop...@gmail.com I am running on Trixbox so my OS is Cent0S 5.4 and the Asterisk version is 1.4.22-4 on asterisk side Snmp module is running: module load res_snmp.so == Parsing '/etc/asterisk/res_snmp.conf': Found Loading [Sub]Agent Module Loaded res_snmp.so = (SNMP [Sub]Agent for Asterisk) see below my snmpd.conf file (I remove commented line for an easy reading) regards Mickael ### # Access Control ### # First, map the community name (COMMUNITY) into a security name # (local and mynetwork, depending on where the request is coming # from): # sec.name source community com2sec local localhost COMMUNITY com2sec mynetwork NETWORK/24 COMMUNITY rwcommunity local rocommunity local # Second, map the security names into group names: # sec.model sec.name group MyRWGroup v1 local group MyRWGroup v2clocal group MyRWGroup usmlocal group MyROGroup v1 mynetwork group MyROGroup v2cmynetwork group MyROGroup usmmynetwork # Third, create a view for us to let the groups have rights to: # incl/excl subtree mask view allincluded .1 80 # Finally, grant the 2 groups access to the 1 view with different # write permissions: #context sec.model sec.level match read write notif access MyROGroup any noauthexact allnone none access MyRWGroup any noauthexact allallnone ### # System contact information # syslocation Right here, right now. syscontact Me m...@somewhere.org ### # Process checks. # # Make sure mountd is running proc mountd # Make sure there are no more than 4 ntalkds running, but 0 is ok too. proc ntalkd 4 # Make sure at least one sendmail, but less than or equal to 10 are running. proc sendmail 10 1 ### # Executables/scripts # # a simple hello world exec echotest /bin/echo hello world ### # disk checks # disk / 1 ### # load average checks # # Check for loads: load 12 14 14 ### # Extensible sections. # ### # Pass through control. # ### # Subagent control # master agentx agentXperms 0660 0550 nobody
Re: [asterisk-users] 1800 DID Provider - Suggestion
On Fri, Nov 27, 2009 at 1:54 PM, Marco Cordeiro marco.corde...@globalstar.com.br wrote: Do you guys suggest any 1800 DID Provider in the US ? We like OnSip.com / Junction Networks stable and various service levels from none of hosted pbx. You should post this to the -biz list. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_read does not seem to work with SIP early media (it answers the channel)
I am trying to come up with a way to read a digit *before* the call is answered. My Asterisk version is 1.6.2.0-rc6 SIP early media works fine (I can receive and transmit audio before the call is answered), but as soon as I start the read application, Asterisk answers the call which is not what I want. Read the application help. It's really as simple as that. Thanks! I must have missed the part about the option n. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help with this conf
Hello, I would appreciate if someone can give some help on what I want: When someone call my box (from outside), to a certain ZAP port, it will put him on hold, and immediately the box calls to outside SIP trunk to a preconfigured certain number, then when the other party picks up the phone, both calls connected and CDR starts counting. Any idea? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
original message- From: mickael ropars mrop...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 27 Nov 2009 11:18:30 +0100 - Hi Michal, thanks a lot for you quick answer I appreciate. I run your commands and I have the following answer [localhost snmp]# snmpwalk -c local -v 1 localhost asterisk no answer [localhost snmp]# snmpwalk -c local -v 2c localhost asterisk ASTERISK-MIB::asterisk = No Such Object available on this agent at this OID you may need to do export MIBS=+ASTERISK-MIB snmpwalk ... -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good quality replacement for Linksys SPA-3102 recommendation.
Can anybody recommend good quality replacement for Linksys SPA-3102 ATA? I have to original Sipura 3K for over 4-years that are still working fine but the Linksys 3102 I purchase are very poor quality (not to mention the echo on PSTN line). One unit quit working 2-weeks after arrival (needed to be replaced) Second unit quit working all together after about a year. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which IP Phone and the codecs
Hello All; Anyone can advise for the good phone (Polycom, Linksys, ... etc) that is a stable and support the codecs: g723, g729, and speex? Actually I would like to have the speex codec because it have the ability to compress to very high compression so we can work with the low bandwidth (for speed about 3 or 4 kbps). I tried Grandstream but really it is a bad device and not worthy to buy it or deal with it. The one I got was having a problem in its handset (there is a noise sound), also it capabilities are very weak. Any one can advise for a good phone? What about Linksys? Does it support speex codec? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't call through voip provider
It is not that easy to give the answer. There are lots of itsp typical ways of registration and you haven't provide the info needed to help you out. You need a register line in the general part of sip.conf. It should look something like (mine looks like this register = DID:SECRET:username@ipness.net:6060 And you need a sip entry in sip.conf. For me it looks something like [DID] type=friend host=ipness.net fromuser=DID fromdomain=ipness.net username=username secret=secret insecure=very context=inbound port=6060 qualify=2000 canreinvite=no disallow=all ;allow=ulaw allow=alaw But your provider might need other settings. So ask your provider. If you are on public IP and not behind NAT you should use nat=no From the sip message I make up that the You didn't provide debug info but copied and paste a sip message. If you would like people to help you, you have to provide proper info. CLI output, sip.conf (without passwords and IP adress info) and the sip messages will be helpful. Are you aware of the fact that you need to open UDP ports and not TCP. Your provider should be able to tell you how to configure such an account on an asterisk box, or at least help you to figure it out. A serious ITSP must have customers using Asterisk. If you have no idea what you are doing my advice is to start reading Asterisk: The future of telephony, freely available on http://www.asteriskdocs.org/ . VERY SERIOUS WARNING: Don't put the credentials of a sip account in a mail to a mailing list. People might use your account to call satelite lines for EUR 7,50 per minute. This kind of mistakes might bankcrupt you :-( I hope this helps. Erik On 19 nov 2009, at 22:36, Landy Landy wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 7:51 AM Ok. I do NOT have ports 1-2 opened in. I guess I I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. I ran this test and there was no difference. I still can't get through. --- Retransmitting #5 (NAT) to 190.80.153.193:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: 102 sip:77...@190.80.153.193;tag=as23e02274 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.153.193 Call-ID: 034bf0572cffb96f621211a8439aa...@190.80.153.193 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 12:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 752676658 752676658 IN IP4 190.80.153.193 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.153.193 t=0 0 m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug? Here's my sip.conf file again maybe you can catch an error or something I'm missing. [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=route insucure=port,invite allow=all careinvite=yes Please helppp. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1800 DID Provider - Suggestion
Try IPComms. j On Fri, 27 Nov 2009, Marco Cordeiro wrote: Hello All, Do you guys suggest any 1800 DID Provider in the US ? I'm having a hard time to find one. Thanks, Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Portech MV-372
I finally saw why it was doing it: In Mobile - Settings - SIP From field there is 4 options: Tel/User (Standard) User/User (Standard) Tel/Tel/ (Not Reg) User/Tel (Not Reg) when I choose any of the first two, I dont have this problem but when I use the last two I have this problem. At the same time, if I use the first two, I am not getting the caller id of the person who called the sim, but in the cdr I see the name of the extensions the gateway was registered too. So what I had to do, is to set a fixed IP to the gateway and instead of having host=dynamic I set host=ip_of_gateway. This way the gateway does not have to register, and I can keep the settings that passes the right caller id. Another way would be to have asterisk read another field for the caller id, because the number of the caller is somewhere on the sip invite. 2009/11/27 Massimo Nuvoli mass...@archivio.it Pascal Bruno ha scritto: Hi, I am experiencing a weird issue with my MV-372. Mobile1 Mobile2 are both registered to my asterisk server, I am able to use them for outgoing call with no problem, but when I call the sims in my gateway, they are routed to the right context/extension/priority, but as soon as I hangup, the sim unregistered from asterisk and tries to register with my the callerid of the last incoming call as follows: Registration from 'mv372 sip:+17546542...@77.29.9.16sip%3a%2b17546542...@77.29.9.16 mailto:sip%3a%2b17546542...@77.29.9.16sip%253a%252b17546542...@77.29.9.16' failed for '97.26.196.2' - No matching peer found and the registration fails since I dont have a peer created for +17546542334 Anyone have an idea on how to go about fixing this? I am using the MV-372 (in and out) and dont have this problem. First: check if the device has the LATEST firmware, if not, upgrade. Second: send an email to the portech service. :-) In the past there was a lot of bug in the firmware of the MV372, and also buggy hardware release, but not now... so check also the hardware version (in the web interface - firmware update - top on the page). I think this is not asterisk issue. Bye. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Pascal B. http://www.kameleonlabs.com/ Ted Turner http://www.brainyquote.com/quotes/authors/t/ted_turner.html - Sports is like a war without the killing. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
Everuthing is working fine, but I have another question to SNMP users: There is no hardware info in the MIB. How can you do to send alarm (when one interface is down for exemple), is there no way to check its status? NB: I am using a Digium card regards Mickael 2009/11/27 mickael ropars mrop...@gmail.com Hi all, I am currently not able to configure SNMP for asterisk, but I am not able to acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/) Does somebody has an example of smnpd.conf file wich is working ? regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Portech MV-372
Pascal Bruno ha scritto:
This way the gateway does not have to register, and I can keep the
settings that passes the right caller id. Another way would be to have
asterisk read another field for the caller id, because the number of the
caller is somewhere on the sip invite.
ouch :-) sorry
this is the workaround:
Set(CALLERID(ALL)=${CALLERID(name)})
Bye.
attachment: massimo.vcf
signature.asc
Description: OpenPGP digital signature
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Re: [asterisk-users] can't call through voip provider
Erik. I already solved this problem and posted it. I was reloading all the setting but, it wasn't changing the provider's ip info. After doing a restart now everything worked. Thanks any ways for your help. --- On Fri, 11/27/09, meetmecall i...@meetmecall.nl wrote: From: meetmecall i...@meetmecall.nl Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, November 27, 2009, 9:51 AM It is not that easy to give the answer. There are lots of itsp typical ways of registration and you haven't provide the info needed to help you out. You need a register line in the general part of sip.conf. It should look something like (mine looks like this register = DID:SECRET:username@ipness.net:6060 And you need a sip entry in sip.conf. For me it looks something like [DID] type=friend host=ipness.net fromuser=DID fromdomain=ipness.net username=username secret=secret insecure=very context=inbound port=6060 qualify=2000 canreinvite=no disallow=all ;allow=ulaw allow=alaw But your provider might need other settings. So ask your provider. If you are on public IP and not behind NAT you should use nat=no From the sip message I make up that the You didn't provide debug info but copied and paste a sip message. If you would like people to help you, you have to provide proper info. CLI output, sip.conf (without passwords and IP adress info) and the sip messages will be helpful. Are you aware of the fact that you need to open UDP ports and not TCP. Your provider should be able to tell you how to configure such an account on an asterisk box, or at least help you to figure it out. A serious ITSP must have customers using Asterisk. If you have no idea what you are doing my advice is to start reading Asterisk: The future of telephony, freely available on http://www.asteriskdocs.org/ . VERY SERIOUS WARNING: Don't put the credentials of a sip account in a mail to a mailing list. People might use your account to call satelite lines for EUR 7,50 per minute. This kind of mistakes might bankcrupt you :-( I hope this helps. Erik On 19 nov 2009, at 22:36, Landy Landy wrote: Can someone please share with me a sip configuration to connect an asterisk server to a voip provider since my configuration isn't working for me. thanks. --- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote: From: Landy Landy landysacco...@yahoo.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, November 19, 2009, 7:51 AM Ok. I do NOT have ports 1-2 opened in. I guess I I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. I ran this test and there was no difference. I still can't get through. --- Retransmitting #5 (NAT) to 190.80.153.193:5060: INVITE sip:18292574...@optimumwireless.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 190.80.153.193:5060;branch=z9hG4bK727987ef Max-Forwards: 70 From: 102 sip:77...@190.80.153.193;tag=as23e02274 To: sip:18292574...@optimumwireless.myvnc.com Contact: sip:77...@190.80.153.193 Call-ID: 034bf0572cffb96f621211a8439aa...@190.80.153.193 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 19 Nov 2009 12:50:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 475 v=0 o=root 752676658 752676658 IN IP4 190.80.153.193 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.153.193 t=0 0 m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv I don't know why I don't see my provider's ip address. Isn't supposed to show in this debug? Here's my sip.conf file again maybe you can catch an error or something I'm missing. [voipprovider] type=peer host=208.78.163.3 username=77000 fromuser=77000 secret=77000 port=5060 dtmfmode=rfc2833 nat=route insucure=port,invite allow=all careinvite=yes Please helppp. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation
Re: [asterisk-users] Questions about static
We have swapped out the phone multiple times for the user. Only one user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cb Sent: Wednesday, November 25, 2009 11:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about static On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote: Would be a cause of static for inbound/outbound and ext to ext calls? Its voip both in and out. We swapped, phones, cordes, switches etc... Typically a reboot of the phone resolves the problem.person also swears there is nothing on or near their desk to cause interference (microwave, cell phone is purse). Only one user? Did you check to see if it is a bad handset cord? -chris www.mythtech.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about static
It’s a single user and we have swapped everything. The phone is an Aastra 6731i and its PoE. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michael Wyres *Sent:* Wednesday, November 25, 2009 6:27 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Questions about static Is it a single user? Or every single phone? If it’s a single user, and you can get hold of a UPS with power conditioning on it, try plugging the various devices into it – there might be some dirty power coming along. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dovey Forman *Sent:* Thursday, 26 November 2009 07:08 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Questions about static Using an Asterisk system running 1.2 with Aastra phones. Would be a cause of static for inbound/outbound and ext to ext calls? Its voip both in and out. We swapped, phones, cordes, switches etc….. Typically a reboot of the phone resolves the problem…person also swears there is nothing on or near their desk to cause interference (microwave, cell phone is purse). Strange…… Thanks --Dovey IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/SIP hard phones
Hi Blaz - Do you maybe know for a fairly good quality IAX2/SIP hard phones in up to 40 USD? I don't think there are any IAX hardphone in production anymore. You might be able to find a used Atcom 320, but probably not for anywhere close to $40. It looks like voipsupply.com has some old Cisco 7910s for $40. http://www.voipsupply.com/cisco-cp-7910g That's about the lowest price you're going to find for a hardware IP phone. You should be able to get an Aastra M9116 or a Grandstream BT201 for around $50. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom retrieve call from hold
Hi Mike - I've got a Polycom 501 that's been working with Asterisk for some time. However, I don't seem to be able to put a call on hold and get it back. It goes on hold just fine. But when I press the resume button, nothing happends. Anyone seen this befor? Any ideas on where to start to fix it? Nope, never seen that one, and I've worked with a LOT of Polycoms. Which SIP/bootrom versions? What asterisk version? Maybe the resume soft button is programmed to do something else other than take the call off hold? What happens when you press the physical hold button (to take the call off hold)? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about static
We have swapped out the phone multiple times for the user. Only one user. Bad PoE port on the switch? How about local interference that the user cannot control? Does the same phone experience static when moved elsewhere? Do you have a power brick for the phone so you can try it as non-PoE? Is the static consistent or intermittent? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about static
We swapped PoE switches, phones, cable and switch ports multiple times. What do you mean by local interference? Cell phone? The person swears nothing is near the phone. Its very strange. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah Miller Sent: Friday, November 27, 2009 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about static We have swapped out the phone multiple times for the user. Only one user. Bad PoE port on the switch? How about local interference that the user cannot control? Does the same phone experience static when moved elsewhere? Do you have a power brick for the phone so you can try it as non-PoE? Is the static consistent or intermittent? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue hangup
hi there, How can we track that the calls within queue has been hang up or disposed within extension.conf ? I am trying to run agi script once the call within queue has been finished. Please advice. amir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about static
Could the static be in the user's hearing aid? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman Sent: Friday, November 27, 2009 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about static We swapped PoE switches, phones, cable and switch ports multiple times. What do you mean by local interference? Cell phone? The person swears nothing is near the phone. Its very strange. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah Miller Sent: Friday, November 27, 2009 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions about static We have swapped out the phone multiple times for the user. Only one user. Bad PoE port on the switch? How about local interference that the user cannot control? Does the same phone experience static when moved elsewhere? Do you have a power brick for the phone so you can try it as non-PoE? Is the static consistent or intermittent? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about static
We swapped PoE switches, phones, cable and switch ports multiple times. What do you mean by local interference? Cell phone? The person swears nothing is near the phone. There are lots of things that can cause interference. Radios, elevators, bad electrical wiring, you name it. Is the static still there when you move the identical phone elsewhere? If not, then the static is most probably caused by some local interference where the user is. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restricting transfers between SIP phones
So, does anyone know of a way to detect whether a call from a SIP phone is the first step of an attended transfer or an original call? It could probably work if you put a SIP proxy in between (ref. Kamilio). Another way might be to set up a special transfer extension that all users use to perform transfers. To do a transfer, all users would first transfer to that special transfer extension. The transfer extension could then read the intended destination and compare the source and destination in a series of GotoIf statements. The GotoIf statements would check the source and destination of the transfer, and if it's ok, use the transfer() app. If not, playback a message that the transfer is not allowed. It means a lot of very specific dialplan logic, and a change of procedures for the users, but it's one way to do it. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
Your Digium card is for linux standard interface like eth0 (ethernet), check IF-MIB.txt and OID from there. BR, Michał 2009/11/27 mickael ropars mrop...@gmail.com: Everuthing is working fine, but I have another question to SNMP users: There is no hardware info in the MIB. How can you do to send alarm (when one interface is down for exemple), is there no way to check its status? NB: I am using a Digium card regards Mickael 2009/11/27 mickael ropars mrop...@gmail.com Hi all, I am currently not able to configure SNMP for asterisk, but I am not able to acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/) Does somebody has an example of smnpd.conf file wich is working ? regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
Hello Micha ( all) , On Fri, 27 Nov 2009, michal kalinowski wrote: Your Digium card is for linux standard interface like eth0 (ethernet), check IF-MIB.txt and OID from there. BR, Micha? When doing a snmpwalk of the IF-MIB having a (*) installed there is no mention of an interface associated with this card . Now it is quite possible that Digium in there wisdom has added the necessary components to their drivers that inserts the necessary components into the IF tables thus allowing snmp's IF-MIB to see a known interface . If this is the case where in the driver (or code base) might I find this revelation . I'd sure like to have statistics traps being dumped for this card . (*) 01:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Tia , JimL 2009/11/27 mickael ropars mrop...@gmail.com: Everuthing is working fine, but I have another question to SNMP users: There is no hardware info in the MIB. How can you do to send alarm (when one interface is down for exemple), is there no way to check its status? NB: I am using a Digium card regards Mickael 2009/11/27 mickael ropars mrop...@gmail.com Hi all, I am currently not able to configure SNMP for asterisk, but I am not able to acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/) Does somebody has an example of smnpd.conf file wich is working ? regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.0-rc6 + Teliax = First Part Of Audio File Playback Cut Off
Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an IVR/voicemail, the first part of every audio file that is played gets cut off. This happens regardless of encoding of the file (ulaw/gsm) and regardless of the incoming codec. However when using Echo() both tones voice are flawlessly echoed back to me, as are the Packet2Packet bridging calls connected to remote phones. I tested this issue with 3 other providers (Link2VoIP/Babytel/Junction Networks) and I'm not experiencing this issue with them, despite having identical peer configurations across for all 4. Though with Teliax I'm using SIP, I did try to use IAX2 for the heck of it and the same problem seems to exists, so it's not specific to SIP. Additionally, I tried changing Teliax proxies just for the heck of it and that made no difference. --- Example of what I see and then hear... --- -- SIP/teliax- Playing 'vm-login.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-password.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-youhave.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-no.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-messages.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-opts.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-helpexit.ulaw' (language 'en') In this case, I'd hear gin essages. The 'password', 'youhave', and 'no' prompts are actually so short you don't hear them at all. http://help.teliax.com/discussions/support/1924-asterisk-1620-rc6 --- I've contacted Teliax about this, but I suspect they're short handed due to the holiday weekend. Has anyone experienced this with 1.6.x Teliax? And if so, what did you do to solve it (if anything)? I'd hate to revert, I spent a lot of time redoing my configs. :) Thanks in advance! Jeff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
Check this command snmpwalk -c your_community -v 1 localhost interfaces in my system it's looks like that: IF-MIB::ifNumber.0 = INTEGER: 4 IF-MIB::ifIndex.1 = INTEGER: 1 IF-MIB::ifIndex.2 = INTEGER: 2 IF-MIB::ifIndex.3 = INTEGER: 3 IF-MIB::ifIndex.4 = INTEGER: 4 IF-MIB::ifDescr.1 = STRING: lo IF-MIB::ifDescr.2 = STRING: eth0 IF-MIB::ifDescr.3 = STRING: eth1 IF-MIB::ifDescr.4 = STRING: sit0 IF-MIB::ifType.1 = INTEGER: softwareLoopback(24) IF-MIB::ifType.2 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.3 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.4 = INTEGER: tunnel(131) IF-MIB::ifMtu.1 = INTEGER: 16436 IF-MIB::ifMtu.2 = INTEGER: 1500 IF-MIB::ifMtu.3 = INTEGER: 1500 IF-MIB::ifMtu.4 = INTEGER: 1480 IF-MIB::ifSpeed.1 = Gauge32: 1000 IF-MIB::ifSpeed.2 = Gauge32: 1000 IF-MIB::ifSpeed.3 = Gauge32: 10 IF-MIB::ifSpeed.4 = Gauge32: 0 IF-MIB::ifPhysAddress.1 = STRING: IF-MIB::ifPhysAddress.2 = STRING: 0:14:5e:32:15:70 IF-MIB::ifPhysAddress.3 = STRING: 0:14:5e:32:15:71 IF-MIB::ifPhysAddress.4 = STRING: IF-MIB::ifAdminStatus.1 = INTEGER: up(1) IF-MIB::ifAdminStatus.2 = INTEGER: down(2) IF-MIB::ifAdminStatus.3 = INTEGER: up(1) IF-MIB::ifAdminStatus.4 = INTEGER: down(2) IF-MIB::ifOperStatus.1 = INTEGER: up(1) IF-MIB::ifOperStatus.2 = INTEGER: down(2) IF-MIB::ifOperStatus.3 = INTEGER: up(1) IF-MIB::ifOperStatus.4 = INTEGER: down(2) IF-MIB::ifLastChange.1 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.2 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.3 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.4 = Timeticks: (0) 0:00:00.00 IF-MIB::ifInOctets.1 = Counter32: 37919437 IF-MIB::ifInOctets.2 = Counter32: 0 IF-MIB::ifInOctets.3 = Counter32: 1491657594 IF-MIB::ifInOctets.4 = Counter32: 0 IF-MIB::ifInUcastPkts.1 = Counter32: 335932 IF-MIB::ifInUcastPkts.2 = Counter32: 0 IF-MIB::ifInUcastPkts.3 = Counter32: 162961409 IF-MIB::ifInUcastPkts.4 = Counter32: 0 IF-MIB::ifInNUcastPkts.1 = Counter32: 0 IF-MIB::ifInNUcastPkts.2 = Counter32: 0 IF-MIB::ifInNUcastPkts.3 = Counter32: 131166 IF-MIB::ifInNUcastPkts.4 = Counter32: 0 IF-MIB::ifInDiscards.1 = Counter32: 0 IF-MIB::ifInDiscards.2 = Counter32: 0 IF-MIB::ifInDiscards.3 = Counter32: 0 IF-MIB::ifInDiscards.4 = Counter32: 0 IF-MIB::ifInErrors.1 = Counter32: 0 IF-MIB::ifInErrors.2 = Counter32: 0 IF-MIB::ifInErrors.3 = Counter32: 0 IF-MIB::ifInErrors.4 = Counter32: 0 IF-MIB::ifInUnknownProtos.1 = Counter32: 0 IF-MIB::ifInUnknownProtos.2 = Counter32: 0 IF-MIB::ifInUnknownProtos.3 = Counter32: 0 IF-MIB::ifInUnknownProtos.4 = Counter32: 0 IF-MIB::ifOutOctets.1 = Counter32: 37919437 IF-MIB::ifOutOctets.2 = Counter32: 0 IF-MIB::ifOutOctets.3 = Counter32: 3525337520 IF-MIB::ifOutOctets.4 = Counter32: 0 IF-MIB::ifOutUcastPkts.1 = Counter32: 335932 IF-MIB::ifOutUcastPkts.2 = Counter32: 0 IF-MIB::ifOutUcastPkts.3 = Counter32: 38811075 IF-MIB::ifOutUcastPkts.4 = Counter32: 0 IF-MIB::ifOutNUcastPkts.1 = Counter32: 0 IF-MIB::ifOutNUcastPkts.2 = Counter32: 0 IF-MIB::ifOutNUcastPkts.3 = Counter32: 0 IF-MIB::ifOutNUcastPkts.4 = Counter32: 0 IF-MIB::ifOutDiscards.1 = Counter32: 0 IF-MIB::ifOutDiscards.2 = Counter32: 0 IF-MIB::ifOutDiscards.3 = Counter32: 0 IF-MIB::ifOutDiscards.4 = Counter32: 0 IF-MIB::ifOutErrors.1 = Counter32: 0 IF-MIB::ifOutErrors.2 = Counter32: 0 IF-MIB::ifOutErrors.3 = Counter32: 0 IF-MIB::ifOutErrors.4 = Counter32: 0 IF-MIB::ifOutQLen.1 = Gauge32: 0 IF-MIB::ifOutQLen.2 = Gauge32: 0 IF-MIB::ifOutQLen.3 = Gauge32: 0 IF-MIB::ifOutQLen.4 = Gauge32: 0 IF-MIB::ifSpecific.1 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.2 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.3 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.4 = OID: SNMPv2-SMI::zeroDotZero here You have information about interface descryptions, status, speed, type, etc. BR, Michał 2009/11/27 Mr. James W. Laferriere bab...@baby-dragons.com: Hello Micha ( all) , On Fri, 27 Nov 2009, michal kalinowski wrote: Your Digium card is for linux standard interface like eth0 (ethernet), check IF-MIB.txt and OID from there. BR, Micha? When doing a snmpwalk of the IF-MIB having a (*) installed there is no mention of an interface associated with this card . Now it is quite possible that Digium in there wisdom has added the necessary components to their drivers that inserts the necessary components into the IF tables thus allowing snmp's IF-MIB to see a known interface . If this is the case where in the driver (or code base) might I find this revelation . I'd sure like to have statistics traps being dumped for this card . (*) 01:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Tia , JimL 2009/11/27 mickael ropars mrop...@gmail.com: Everuthing is working fine, but I have another question to SNMP users: There is no hardware info in the MIB. How can you do to send alarm (when one interface is down for exemple), is there no way to check its status? NB: I am using a Digium card regards Mickael
Re: [asterisk-users] ASTERISK and SNMP
Michal, in the IF-MIB you only have 4 interfaces lo,eth0,eth1,sit0 l0 which is the loopback interface eth0, eth1 : ethernet interface sit0 : use for PTP tunneling (use for IPv6) so no information on the digium interface. my IF MIB has also those interfaces I found one the solution to get status of the cards, and all snmp data. the solution is argus : http://argus.tcp4me.com/ with this tools you can have a complete view of your system. regards Mickael 2009/11/27 michal kalinowski michal.kalinow...@interia.pl Check this command snmpwalk -c your_community -v 1 localhost interfaces in my system it's looks like that: IF-MIB::ifNumber.0 = INTEGER: 4 IF-MIB::ifIndex.1 = INTEGER: 1 IF-MIB::ifIndex.2 = INTEGER: 2 IF-MIB::ifIndex.3 = INTEGER: 3 IF-MIB::ifIndex.4 = INTEGER: 4 IF-MIB::ifDescr.1 = STRING: lo IF-MIB::ifDescr.2 = STRING: eth0 IF-MIB::ifDescr.3 = STRING: eth1 IF-MIB::ifDescr.4 = STRING: sit0 IF-MIB::ifType.1 = INTEGER: softwareLoopback(24) IF-MIB::ifType.2 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.3 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.4 = INTEGER: tunnel(131) IF-MIB::ifMtu.1 = INTEGER: 16436 IF-MIB::ifMtu.2 = INTEGER: 1500 IF-MIB::ifMtu.3 = INTEGER: 1500 IF-MIB::ifMtu.4 = INTEGER: 1480 IF-MIB::ifSpeed.1 = Gauge32: 1000 IF-MIB::ifSpeed.2 = Gauge32: 1000 IF-MIB::ifSpeed.3 = Gauge32: 10 IF-MIB::ifSpeed.4 = Gauge32: 0 IF-MIB::ifPhysAddress.1 = STRING: IF-MIB::ifPhysAddress.2 = STRING: 0:14:5e:32:15:70 IF-MIB::ifPhysAddress.3 = STRING: 0:14:5e:32:15:71 IF-MIB::ifPhysAddress.4 = STRING: IF-MIB::ifAdminStatus.1 = INTEGER: up(1) IF-MIB::ifAdminStatus.2 = INTEGER: down(2) IF-MIB::ifAdminStatus.3 = INTEGER: up(1) IF-MIB::ifAdminStatus.4 = INTEGER: down(2) IF-MIB::ifOperStatus.1 = INTEGER: up(1) IF-MIB::ifOperStatus.2 = INTEGER: down(2) IF-MIB::ifOperStatus.3 = INTEGER: up(1) IF-MIB::ifOperStatus.4 = INTEGER: down(2) IF-MIB::ifLastChange.1 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.2 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.3 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.4 = Timeticks: (0) 0:00:00.00 IF-MIB::ifInOctets.1 = Counter32: 37919437 IF-MIB::ifInOctets.2 = Counter32: 0 IF-MIB::ifInOctets.3 = Counter32: 1491657594 IF-MIB::ifInOctets.4 = Counter32: 0 IF-MIB::ifInUcastPkts.1 = Counter32: 335932 IF-MIB::ifInUcastPkts.2 = Counter32: 0 IF-MIB::ifInUcastPkts.3 = Counter32: 162961409 IF-MIB::ifInUcastPkts.4 = Counter32: 0 IF-MIB::ifInNUcastPkts.1 = Counter32: 0 IF-MIB::ifInNUcastPkts.2 = Counter32: 0 IF-MIB::ifInNUcastPkts.3 = Counter32: 131166 IF-MIB::ifInNUcastPkts.4 = Counter32: 0 IF-MIB::ifInDiscards.1 = Counter32: 0 IF-MIB::ifInDiscards.2 = Counter32: 0 IF-MIB::ifInDiscards.3 = Counter32: 0 IF-MIB::ifInDiscards.4 = Counter32: 0 IF-MIB::ifInErrors.1 = Counter32: 0 IF-MIB::ifInErrors.2 = Counter32: 0 IF-MIB::ifInErrors.3 = Counter32: 0 IF-MIB::ifInErrors.4 = Counter32: 0 IF-MIB::ifInUnknownProtos.1 = Counter32: 0 IF-MIB::ifInUnknownProtos.2 = Counter32: 0 IF-MIB::ifInUnknownProtos.3 = Counter32: 0 IF-MIB::ifInUnknownProtos.4 = Counter32: 0 IF-MIB::ifOutOctets.1 = Counter32: 37919437 IF-MIB::ifOutOctets.2 = Counter32: 0 IF-MIB::ifOutOctets.3 = Counter32: 3525337520 IF-MIB::ifOutOctets.4 = Counter32: 0 IF-MIB::ifOutUcastPkts.1 = Counter32: 335932 IF-MIB::ifOutUcastPkts.2 = Counter32: 0 IF-MIB::ifOutUcastPkts.3 = Counter32: 38811075 IF-MIB::ifOutUcastPkts.4 = Counter32: 0 IF-MIB::ifOutNUcastPkts.1 = Counter32: 0 IF-MIB::ifOutNUcastPkts.2 = Counter32: 0 IF-MIB::ifOutNUcastPkts.3 = Counter32: 0 IF-MIB::ifOutNUcastPkts.4 = Counter32: 0 IF-MIB::ifOutDiscards.1 = Counter32: 0 IF-MIB::ifOutDiscards.2 = Counter32: 0 IF-MIB::ifOutDiscards.3 = Counter32: 0 IF-MIB::ifOutDiscards.4 = Counter32: 0 IF-MIB::ifOutErrors.1 = Counter32: 0 IF-MIB::ifOutErrors.2 = Counter32: 0 IF-MIB::ifOutErrors.3 = Counter32: 0 IF-MIB::ifOutErrors.4 = Counter32: 0 IF-MIB::ifOutQLen.1 = Gauge32: 0 IF-MIB::ifOutQLen.2 = Gauge32: 0 IF-MIB::ifOutQLen.3 = Gauge32: 0 IF-MIB::ifOutQLen.4 = Gauge32: 0 IF-MIB::ifSpecific.1 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.2 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.3 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.4 = OID: SNMPv2-SMI::zeroDotZero here You have information about interface descryptions, status, speed, type, etc. BR, Michał 2009/11/27 Mr. James W. Laferriere bab...@baby-dragons.com: Hello Micha ( all) , On Fri, 27 Nov 2009, michal kalinowski wrote: Your Digium card is for linux standard interface like eth0 (ethernet), check IF-MIB.txt and OID from there. BR, Micha? When doing a snmpwalk of the IF-MIB having a (*) installed there is no mention of an interface associated with this card . Now it is quite possible that Digium in there wisdom has added the necessary components to their drivers that inserts the necessary components into the IF tables thus allowing snmp's IF-MIB to see a
Re: [asterisk-users] ASTERISK and SNMP
Yes I know about that :) at this moment i have only machine with lo,eth0,eth1,sit0. On monday I will check that command on the server with e1 card. BR, Michał W dniu 27 listopada 2009 23:51 użytkownik mickael ropars mrop...@gmail.com napisał: Michal, in the IF-MIB you only have 4 interfaces lo,eth0,eth1,sit0 l0 which is the loopback interface eth0, eth1 : ethernet interface sit0 : use for PTP tunneling (use for IPv6) so no information on the digium interface. my IF MIB has also those interfaces I found one the solution to get status of the cards, and all snmp data. the solution is argus : http://argus.tcp4me.com/ with this tools you can have a complete view of your system. regards Mickael 2009/11/27 michal kalinowski michal.kalinow...@interia.pl Check this command snmpwalk -c your_community -v 1 localhost interfaces in my system it's looks like that: IF-MIB::ifNumber.0 = INTEGER: 4 IF-MIB::ifIndex.1 = INTEGER: 1 IF-MIB::ifIndex.2 = INTEGER: 2 IF-MIB::ifIndex.3 = INTEGER: 3 IF-MIB::ifIndex.4 = INTEGER: 4 IF-MIB::ifDescr.1 = STRING: lo IF-MIB::ifDescr.2 = STRING: eth0 IF-MIB::ifDescr.3 = STRING: eth1 IF-MIB::ifDescr.4 = STRING: sit0 IF-MIB::ifType.1 = INTEGER: softwareLoopback(24) IF-MIB::ifType.2 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.3 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.4 = INTEGER: tunnel(131) IF-MIB::ifMtu.1 = INTEGER: 16436 IF-MIB::ifMtu.2 = INTEGER: 1500 IF-MIB::ifMtu.3 = INTEGER: 1500 IF-MIB::ifMtu.4 = INTEGER: 1480 IF-MIB::ifSpeed.1 = Gauge32: 1000 IF-MIB::ifSpeed.2 = Gauge32: 1000 IF-MIB::ifSpeed.3 = Gauge32: 10 IF-MIB::ifSpeed.4 = Gauge32: 0 IF-MIB::ifPhysAddress.1 = STRING: IF-MIB::ifPhysAddress.2 = STRING: 0:14:5e:32:15:70 IF-MIB::ifPhysAddress.3 = STRING: 0:14:5e:32:15:71 IF-MIB::ifPhysAddress.4 = STRING: IF-MIB::ifAdminStatus.1 = INTEGER: up(1) IF-MIB::ifAdminStatus.2 = INTEGER: down(2) IF-MIB::ifAdminStatus.3 = INTEGER: up(1) IF-MIB::ifAdminStatus.4 = INTEGER: down(2) IF-MIB::ifOperStatus.1 = INTEGER: up(1) IF-MIB::ifOperStatus.2 = INTEGER: down(2) IF-MIB::ifOperStatus.3 = INTEGER: up(1) IF-MIB::ifOperStatus.4 = INTEGER: down(2) IF-MIB::ifLastChange.1 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.2 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.3 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.4 = Timeticks: (0) 0:00:00.00 IF-MIB::ifInOctets.1 = Counter32: 37919437 IF-MIB::ifInOctets.2 = Counter32: 0 IF-MIB::ifInOctets.3 = Counter32: 1491657594 IF-MIB::ifInOctets.4 = Counter32: 0 IF-MIB::ifInUcastPkts.1 = Counter32: 335932 IF-MIB::ifInUcastPkts.2 = Counter32: 0 IF-MIB::ifInUcastPkts.3 = Counter32: 162961409 IF-MIB::ifInUcastPkts.4 = Counter32: 0 IF-MIB::ifInNUcastPkts.1 = Counter32: 0 IF-MIB::ifInNUcastPkts.2 = Counter32: 0 IF-MIB::ifInNUcastPkts.3 = Counter32: 131166 IF-MIB::ifInNUcastPkts.4 = Counter32: 0 IF-MIB::ifInDiscards.1 = Counter32: 0 IF-MIB::ifInDiscards.2 = Counter32: 0 IF-MIB::ifInDiscards.3 = Counter32: 0 IF-MIB::ifInDiscards.4 = Counter32: 0 IF-MIB::ifInErrors.1 = Counter32: 0 IF-MIB::ifInErrors.2 = Counter32: 0 IF-MIB::ifInErrors.3 = Counter32: 0 IF-MIB::ifInErrors.4 = Counter32: 0 IF-MIB::ifInUnknownProtos.1 = Counter32: 0 IF-MIB::ifInUnknownProtos.2 = Counter32: 0 IF-MIB::ifInUnknownProtos.3 = Counter32: 0 IF-MIB::ifInUnknownProtos.4 = Counter32: 0 IF-MIB::ifOutOctets.1 = Counter32: 37919437 IF-MIB::ifOutOctets.2 = Counter32: 0 IF-MIB::ifOutOctets.3 = Counter32: 3525337520 IF-MIB::ifOutOctets.4 = Counter32: 0 IF-MIB::ifOutUcastPkts.1 = Counter32: 335932 IF-MIB::ifOutUcastPkts.2 = Counter32: 0 IF-MIB::ifOutUcastPkts.3 = Counter32: 38811075 IF-MIB::ifOutUcastPkts.4 = Counter32: 0 IF-MIB::ifOutNUcastPkts.1 = Counter32: 0 IF-MIB::ifOutNUcastPkts.2 = Counter32: 0 IF-MIB::ifOutNUcastPkts.3 = Counter32: 0 IF-MIB::ifOutNUcastPkts.4 = Counter32: 0 IF-MIB::ifOutDiscards.1 = Counter32: 0 IF-MIB::ifOutDiscards.2 = Counter32: 0 IF-MIB::ifOutDiscards.3 = Counter32: 0 IF-MIB::ifOutDiscards.4 = Counter32: 0 IF-MIB::ifOutErrors.1 = Counter32: 0 IF-MIB::ifOutErrors.2 = Counter32: 0 IF-MIB::ifOutErrors.3 = Counter32: 0 IF-MIB::ifOutErrors.4 = Counter32: 0 IF-MIB::ifOutQLen.1 = Gauge32: 0 IF-MIB::ifOutQLen.2 = Gauge32: 0 IF-MIB::ifOutQLen.3 = Gauge32: 0 IF-MIB::ifOutQLen.4 = Gauge32: 0 IF-MIB::ifSpecific.1 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.2 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.3 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.4 = OID: SNMPv2-SMI::zeroDotZero here You have information about interface descryptions, status, speed, type, etc. BR, Michał 2009/11/27 Mr. James W. Laferriere bab...@baby-dragons.com: Hello Micha ( all) , On Fri, 27 Nov 2009, michal kalinowski wrote: Your Digium card is for linux standard interface like eth0 (ethernet), check IF-MIB.txt and OID from there. BR, Micha? When doing a snmpwalk of the IF-MIB having a (*) installed there is no
Re: [asterisk-users] ASTERISK and SNMP
It will be the same, I already have 4 E1 interfaces. but no information in the MIB 2009/11/28 michal kalinowski michal.kalinow...@interia.pl Yes I know about that :) at this moment i have only machine with lo,eth0,eth1,sit0. On monday I will check that command on the server with e1 card. BR, Michał W dniu 27 listopada 2009 23:51 użytkownik mickael ropars mrop...@gmail.com napisał: Michal, in the IF-MIB you only have 4 interfaces lo,eth0,eth1,sit0 l0 which is the loopback interface eth0, eth1 : ethernet interface sit0 : use for PTP tunneling (use for IPv6) so no information on the digium interface. my IF MIB has also those interfaces I found one the solution to get status of the cards, and all snmp data. the solution is argus : http://argus.tcp4me.com/ with this tools you can have a complete view of your system. regards Mickael 2009/11/27 michal kalinowski michal.kalinow...@interia.pl Check this command snmpwalk -c your_community -v 1 localhost interfaces in my system it's looks like that: IF-MIB::ifNumber.0 = INTEGER: 4 IF-MIB::ifIndex.1 = INTEGER: 1 IF-MIB::ifIndex.2 = INTEGER: 2 IF-MIB::ifIndex.3 = INTEGER: 3 IF-MIB::ifIndex.4 = INTEGER: 4 IF-MIB::ifDescr.1 = STRING: lo IF-MIB::ifDescr.2 = STRING: eth0 IF-MIB::ifDescr.3 = STRING: eth1 IF-MIB::ifDescr.4 = STRING: sit0 IF-MIB::ifType.1 = INTEGER: softwareLoopback(24) IF-MIB::ifType.2 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.3 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.4 = INTEGER: tunnel(131) IF-MIB::ifMtu.1 = INTEGER: 16436 IF-MIB::ifMtu.2 = INTEGER: 1500 IF-MIB::ifMtu.3 = INTEGER: 1500 IF-MIB::ifMtu.4 = INTEGER: 1480 IF-MIB::ifSpeed.1 = Gauge32: 1000 IF-MIB::ifSpeed.2 = Gauge32: 1000 IF-MIB::ifSpeed.3 = Gauge32: 10 IF-MIB::ifSpeed.4 = Gauge32: 0 IF-MIB::ifPhysAddress.1 = STRING: IF-MIB::ifPhysAddress.2 = STRING: 0:14:5e:32:15:70 IF-MIB::ifPhysAddress.3 = STRING: 0:14:5e:32:15:71 IF-MIB::ifPhysAddress.4 = STRING: IF-MIB::ifAdminStatus.1 = INTEGER: up(1) IF-MIB::ifAdminStatus.2 = INTEGER: down(2) IF-MIB::ifAdminStatus.3 = INTEGER: up(1) IF-MIB::ifAdminStatus.4 = INTEGER: down(2) IF-MIB::ifOperStatus.1 = INTEGER: up(1) IF-MIB::ifOperStatus.2 = INTEGER: down(2) IF-MIB::ifOperStatus.3 = INTEGER: up(1) IF-MIB::ifOperStatus.4 = INTEGER: down(2) IF-MIB::ifLastChange.1 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.2 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.3 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.4 = Timeticks: (0) 0:00:00.00 IF-MIB::ifInOctets.1 = Counter32: 37919437 IF-MIB::ifInOctets.2 = Counter32: 0 IF-MIB::ifInOctets.3 = Counter32: 1491657594 IF-MIB::ifInOctets.4 = Counter32: 0 IF-MIB::ifInUcastPkts.1 = Counter32: 335932 IF-MIB::ifInUcastPkts.2 = Counter32: 0 IF-MIB::ifInUcastPkts.3 = Counter32: 162961409 IF-MIB::ifInUcastPkts.4 = Counter32: 0 IF-MIB::ifInNUcastPkts.1 = Counter32: 0 IF-MIB::ifInNUcastPkts.2 = Counter32: 0 IF-MIB::ifInNUcastPkts.3 = Counter32: 131166 IF-MIB::ifInNUcastPkts.4 = Counter32: 0 IF-MIB::ifInDiscards.1 = Counter32: 0 IF-MIB::ifInDiscards.2 = Counter32: 0 IF-MIB::ifInDiscards.3 = Counter32: 0 IF-MIB::ifInDiscards.4 = Counter32: 0 IF-MIB::ifInErrors.1 = Counter32: 0 IF-MIB::ifInErrors.2 = Counter32: 0 IF-MIB::ifInErrors.3 = Counter32: 0 IF-MIB::ifInErrors.4 = Counter32: 0 IF-MIB::ifInUnknownProtos.1 = Counter32: 0 IF-MIB::ifInUnknownProtos.2 = Counter32: 0 IF-MIB::ifInUnknownProtos.3 = Counter32: 0 IF-MIB::ifInUnknownProtos.4 = Counter32: 0 IF-MIB::ifOutOctets.1 = Counter32: 37919437 IF-MIB::ifOutOctets.2 = Counter32: 0 IF-MIB::ifOutOctets.3 = Counter32: 3525337520 IF-MIB::ifOutOctets.4 = Counter32: 0 IF-MIB::ifOutUcastPkts.1 = Counter32: 335932 IF-MIB::ifOutUcastPkts.2 = Counter32: 0 IF-MIB::ifOutUcastPkts.3 = Counter32: 38811075 IF-MIB::ifOutUcastPkts.4 = Counter32: 0 IF-MIB::ifOutNUcastPkts.1 = Counter32: 0 IF-MIB::ifOutNUcastPkts.2 = Counter32: 0 IF-MIB::ifOutNUcastPkts.3 = Counter32: 0 IF-MIB::ifOutNUcastPkts.4 = Counter32: 0 IF-MIB::ifOutDiscards.1 = Counter32: 0 IF-MIB::ifOutDiscards.2 = Counter32: 0 IF-MIB::ifOutDiscards.3 = Counter32: 0 IF-MIB::ifOutDiscards.4 = Counter32: 0 IF-MIB::ifOutErrors.1 = Counter32: 0 IF-MIB::ifOutErrors.2 = Counter32: 0 IF-MIB::ifOutErrors.3 = Counter32: 0 IF-MIB::ifOutErrors.4 = Counter32: 0 IF-MIB::ifOutQLen.1 = Gauge32: 0 IF-MIB::ifOutQLen.2 = Gauge32: 0 IF-MIB::ifOutQLen.3 = Gauge32: 0 IF-MIB::ifOutQLen.4 = Gauge32: 0 IF-MIB::ifSpecific.1 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.2 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.3 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.4 = OID: SNMPv2-SMI::zeroDotZero here You have information about interface descryptions, status, speed, type, etc. BR, Michał 2009/11/27 Mr. James W. Laferriere bab...@baby-dragons.com: Hello Micha (
Re: [asterisk-users] ASTERISK and SNMP
What do You have in ifconfig ? BR, Michał W dniu 28 listopada 2009 00:11 użytkownik mickael ropars mrop...@gmail.com napisał: It will be the same, I already have 4 E1 interfaces. but no information in the MIB 2009/11/28 michal kalinowski michal.kalinow...@interia.pl Yes I know about that :) at this moment i have only machine with lo,eth0,eth1,sit0. On monday I will check that command on the server with e1 card. BR, Michał W dniu 27 listopada 2009 23:51 użytkownik mickael ropars mrop...@gmail.com napisał: Michal, in the IF-MIB you only have 4 interfaces lo,eth0,eth1,sit0 l0 which is the loopback interface eth0, eth1 : ethernet interface sit0 : use for PTP tunneling (use for IPv6) so no information on the digium interface. my IF MIB has also those interfaces I found one the solution to get status of the cards, and all snmp data. the solution is argus : http://argus.tcp4me.com/ with this tools you can have a complete view of your system. regards Mickael 2009/11/27 michal kalinowski michal.kalinow...@interia.pl Check this command snmpwalk -c your_community -v 1 localhost interfaces in my system it's looks like that: IF-MIB::ifNumber.0 = INTEGER: 4 IF-MIB::ifIndex.1 = INTEGER: 1 IF-MIB::ifIndex.2 = INTEGER: 2 IF-MIB::ifIndex.3 = INTEGER: 3 IF-MIB::ifIndex.4 = INTEGER: 4 IF-MIB::ifDescr.1 = STRING: lo IF-MIB::ifDescr.2 = STRING: eth0 IF-MIB::ifDescr.3 = STRING: eth1 IF-MIB::ifDescr.4 = STRING: sit0 IF-MIB::ifType.1 = INTEGER: softwareLoopback(24) IF-MIB::ifType.2 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.3 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.4 = INTEGER: tunnel(131) IF-MIB::ifMtu.1 = INTEGER: 16436 IF-MIB::ifMtu.2 = INTEGER: 1500 IF-MIB::ifMtu.3 = INTEGER: 1500 IF-MIB::ifMtu.4 = INTEGER: 1480 IF-MIB::ifSpeed.1 = Gauge32: 1000 IF-MIB::ifSpeed.2 = Gauge32: 1000 IF-MIB::ifSpeed.3 = Gauge32: 10 IF-MIB::ifSpeed.4 = Gauge32: 0 IF-MIB::ifPhysAddress.1 = STRING: IF-MIB::ifPhysAddress.2 = STRING: 0:14:5e:32:15:70 IF-MIB::ifPhysAddress.3 = STRING: 0:14:5e:32:15:71 IF-MIB::ifPhysAddress.4 = STRING: IF-MIB::ifAdminStatus.1 = INTEGER: up(1) IF-MIB::ifAdminStatus.2 = INTEGER: down(2) IF-MIB::ifAdminStatus.3 = INTEGER: up(1) IF-MIB::ifAdminStatus.4 = INTEGER: down(2) IF-MIB::ifOperStatus.1 = INTEGER: up(1) IF-MIB::ifOperStatus.2 = INTEGER: down(2) IF-MIB::ifOperStatus.3 = INTEGER: up(1) IF-MIB::ifOperStatus.4 = INTEGER: down(2) IF-MIB::ifLastChange.1 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.2 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.3 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.4 = Timeticks: (0) 0:00:00.00 IF-MIB::ifInOctets.1 = Counter32: 37919437 IF-MIB::ifInOctets.2 = Counter32: 0 IF-MIB::ifInOctets.3 = Counter32: 1491657594 IF-MIB::ifInOctets.4 = Counter32: 0 IF-MIB::ifInUcastPkts.1 = Counter32: 335932 IF-MIB::ifInUcastPkts.2 = Counter32: 0 IF-MIB::ifInUcastPkts.3 = Counter32: 162961409 IF-MIB::ifInUcastPkts.4 = Counter32: 0 IF-MIB::ifInNUcastPkts.1 = Counter32: 0 IF-MIB::ifInNUcastPkts.2 = Counter32: 0 IF-MIB::ifInNUcastPkts.3 = Counter32: 131166 IF-MIB::ifInNUcastPkts.4 = Counter32: 0 IF-MIB::ifInDiscards.1 = Counter32: 0 IF-MIB::ifInDiscards.2 = Counter32: 0 IF-MIB::ifInDiscards.3 = Counter32: 0 IF-MIB::ifInDiscards.4 = Counter32: 0 IF-MIB::ifInErrors.1 = Counter32: 0 IF-MIB::ifInErrors.2 = Counter32: 0 IF-MIB::ifInErrors.3 = Counter32: 0 IF-MIB::ifInErrors.4 = Counter32: 0 IF-MIB::ifInUnknownProtos.1 = Counter32: 0 IF-MIB::ifInUnknownProtos.2 = Counter32: 0 IF-MIB::ifInUnknownProtos.3 = Counter32: 0 IF-MIB::ifInUnknownProtos.4 = Counter32: 0 IF-MIB::ifOutOctets.1 = Counter32: 37919437 IF-MIB::ifOutOctets.2 = Counter32: 0 IF-MIB::ifOutOctets.3 = Counter32: 3525337520 IF-MIB::ifOutOctets.4 = Counter32: 0 IF-MIB::ifOutUcastPkts.1 = Counter32: 335932 IF-MIB::ifOutUcastPkts.2 = Counter32: 0 IF-MIB::ifOutUcastPkts.3 = Counter32: 38811075 IF-MIB::ifOutUcastPkts.4 = Counter32: 0 IF-MIB::ifOutNUcastPkts.1 = Counter32: 0 IF-MIB::ifOutNUcastPkts.2 = Counter32: 0 IF-MIB::ifOutNUcastPkts.3 = Counter32: 0 IF-MIB::ifOutNUcastPkts.4 = Counter32: 0 IF-MIB::ifOutDiscards.1 = Counter32: 0 IF-MIB::ifOutDiscards.2 = Counter32: 0 IF-MIB::ifOutDiscards.3 = Counter32: 0 IF-MIB::ifOutDiscards.4 = Counter32: 0 IF-MIB::ifOutErrors.1 = Counter32: 0 IF-MIB::ifOutErrors.2 = Counter32: 0 IF-MIB::ifOutErrors.3 = Counter32: 0 IF-MIB::ifOutErrors.4 = Counter32: 0 IF-MIB::ifOutQLen.1 = Gauge32: 0 IF-MIB::ifOutQLen.2 = Gauge32: 0 IF-MIB::ifOutQLen.3 = Gauge32: 0 IF-MIB::ifOutQLen.4 = Gauge32: 0 IF-MIB::ifSpecific.1 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.2 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.3 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.4 = OID: SNMPv2-SMI::zeroDotZero here You have information about interface
Re: [asterisk-users] Asterisk 1.6.2.0-rc6 + Teliax = First Part Of Audio File Playback Cut Off
Thanks to a tip from someone who replied to me off list, I tried using the 'den.teliax.net' proxy and that solved my issue. I'll have to follow up with Teliax to see what the difference is. Go figure. And thanks to Darrick for the info! Jeff On 11/27/2009 05:27 PM, Jeff Iddings wrote: Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an IVR/voicemail, the first part of every audio file that is played gets cut off. This happens regardless of encoding of the file (ulaw/gsm) and regardless of the incoming codec. However when using Echo() both tones voice are flawlessly echoed back to me, as are the Packet2Packet bridging calls connected to remote phones. I tested this issue with 3 other providers (Link2VoIP/Babytel/Junction Networks) and I'm not experiencing this issue with them, despite having identical peer configurations across for all 4. Though with Teliax I'm using SIP, I did try to use IAX2 for the heck of it and the same problem seems to exists, so it's not specific to SIP. Additionally, I tried changing Teliax proxies just for the heck of it and that made no difference. --- Example of what I see and then hear... --- -- SIP/teliax- Playing 'vm-login.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-password.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-youhave.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-no.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-messages.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-opts.ulaw' (language 'en') -- SIP/teliax- Playing 'vm-helpexit.ulaw' (language 'en') In this case, I'd hear gin essages. The 'password', 'youhave', and 'no' prompts are actually so short you don't hear them at all. http://help.teliax.com/discussions/support/1924-asterisk-1620-rc6 --- I've contacted Teliax about this, but I suspect they're short handed due to the holiday weekend. Has anyone experienced this with 1.6.x Teliax? And if so, what did you do to solve it (if anything)? I'd hate to revert, I spent a lot of time redoing my configs. :) Thanks in advance! Jeff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
Hello Mickael , On Fri, 27 Nov 2009, mickael ropars wrote: Michal, in the IF-MIB you only have 4 interfaces lo,eth0,eth1,sit0 l0 which is the loopback interface eth0, eth1 : ethernet interface sit0 : use for PTP tunneling (use for IPv6) so no information on the digium interface. my IF MIB has also those interfaces I found one the solution to get status of the cards, and all snmp data. the solution is argus : http://argus.tcp4me.com/ with this tools you can have a complete view of your system. regards Mickael While Argus is quite good at monitoring systems and is rather easy to manage . In the case of Asterisk monitoring it uses the Asterisk Management Interface (ie: AMI) not snmp . I was( and still am) hoping that the same information available to the administrator thru the AMI can/will be made available thru snmp polling traps . It should not be too difficult to make net-snmp's daemon make those connections to AMI locally on the asterisk server then report that data back to the snmp client . But everytime I've tried to expand snmpd's functionality I've hit nothing but failures . Twyl , JimL 2009/11/27 michal kalinowski michal.kalinow...@interia.pl Check this command snmpwalk -c your_community -v 1 localhost interfaces in my system it's looks like that: IF-MIB::ifNumber.0 = INTEGER: 4 IF-MIB::ifIndex.1 = INTEGER: 1 IF-MIB::ifIndex.2 = INTEGER: 2 IF-MIB::ifIndex.3 = INTEGER: 3 IF-MIB::ifIndex.4 = INTEGER: 4 IF-MIB::ifDescr.1 = STRING: lo IF-MIB::ifDescr.2 = STRING: eth0 IF-MIB::ifDescr.3 = STRING: eth1 IF-MIB::ifDescr.4 = STRING: sit0 IF-MIB::ifType.1 = INTEGER: softwareLoopback(24) IF-MIB::ifType.2 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.3 = INTEGER: ethernetCsmacd(6) IF-MIB::ifType.4 = INTEGER: tunnel(131) IF-MIB::ifMtu.1 = INTEGER: 16436 IF-MIB::ifMtu.2 = INTEGER: 1500 IF-MIB::ifMtu.3 = INTEGER: 1500 IF-MIB::ifMtu.4 = INTEGER: 1480 IF-MIB::ifSpeed.1 = Gauge32: 1000 IF-MIB::ifSpeed.2 = Gauge32: 1000 IF-MIB::ifSpeed.3 = Gauge32: 10 IF-MIB::ifSpeed.4 = Gauge32: 0 IF-MIB::ifPhysAddress.1 = STRING: IF-MIB::ifPhysAddress.2 = STRING: 0:14:5e:32:15:70 IF-MIB::ifPhysAddress.3 = STRING: 0:14:5e:32:15:71 IF-MIB::ifPhysAddress.4 = STRING: IF-MIB::ifAdminStatus.1 = INTEGER: up(1) IF-MIB::ifAdminStatus.2 = INTEGER: down(2) IF-MIB::ifAdminStatus.3 = INTEGER: up(1) IF-MIB::ifAdminStatus.4 = INTEGER: down(2) IF-MIB::ifOperStatus.1 = INTEGER: up(1) IF-MIB::ifOperStatus.2 = INTEGER: down(2) IF-MIB::ifOperStatus.3 = INTEGER: up(1) IF-MIB::ifOperStatus.4 = INTEGER: down(2) IF-MIB::ifLastChange.1 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.2 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.3 = Timeticks: (0) 0:00:00.00 IF-MIB::ifLastChange.4 = Timeticks: (0) 0:00:00.00 IF-MIB::ifInOctets.1 = Counter32: 37919437 IF-MIB::ifInOctets.2 = Counter32: 0 IF-MIB::ifInOctets.3 = Counter32: 1491657594 IF-MIB::ifInOctets.4 = Counter32: 0 IF-MIB::ifInUcastPkts.1 = Counter32: 335932 IF-MIB::ifInUcastPkts.2 = Counter32: 0 IF-MIB::ifInUcastPkts.3 = Counter32: 162961409 IF-MIB::ifInUcastPkts.4 = Counter32: 0 IF-MIB::ifInNUcastPkts.1 = Counter32: 0 IF-MIB::ifInNUcastPkts.2 = Counter32: 0 IF-MIB::ifInNUcastPkts.3 = Counter32: 131166 IF-MIB::ifInNUcastPkts.4 = Counter32: 0 IF-MIB::ifInDiscards.1 = Counter32: 0 IF-MIB::ifInDiscards.2 = Counter32: 0 IF-MIB::ifInDiscards.3 = Counter32: 0 IF-MIB::ifInDiscards.4 = Counter32: 0 IF-MIB::ifInErrors.1 = Counter32: 0 IF-MIB::ifInErrors.2 = Counter32: 0 IF-MIB::ifInErrors.3 = Counter32: 0 IF-MIB::ifInErrors.4 = Counter32: 0 IF-MIB::ifInUnknownProtos.1 = Counter32: 0 IF-MIB::ifInUnknownProtos.2 = Counter32: 0 IF-MIB::ifInUnknownProtos.3 = Counter32: 0 IF-MIB::ifInUnknownProtos.4 = Counter32: 0 IF-MIB::ifOutOctets.1 = Counter32: 37919437 IF-MIB::ifOutOctets.2 = Counter32: 0 IF-MIB::ifOutOctets.3 = Counter32: 3525337520 IF-MIB::ifOutOctets.4 = Counter32: 0 IF-MIB::ifOutUcastPkts.1 = Counter32: 335932 IF-MIB::ifOutUcastPkts.2 = Counter32: 0 IF-MIB::ifOutUcastPkts.3 = Counter32: 38811075 IF-MIB::ifOutUcastPkts.4 = Counter32: 0 IF-MIB::ifOutNUcastPkts.1 = Counter32: 0 IF-MIB::ifOutNUcastPkts.2 = Counter32: 0 IF-MIB::ifOutNUcastPkts.3 = Counter32: 0 IF-MIB::ifOutNUcastPkts.4 = Counter32: 0 IF-MIB::ifOutDiscards.1 = Counter32: 0 IF-MIB::ifOutDiscards.2 = Counter32: 0 IF-MIB::ifOutDiscards.3 = Counter32: 0 IF-MIB::ifOutDiscards.4 = Counter32: 0 IF-MIB::ifOutErrors.1 = Counter32: 0 IF-MIB::ifOutErrors.2 = Counter32: 0 IF-MIB::ifOutErrors.3 = Counter32: 0 IF-MIB::ifOutErrors.4 = Counter32: 0 IF-MIB::ifOutQLen.1 = Gauge32: 0 IF-MIB::ifOutQLen.2 = Gauge32: 0 IF-MIB::ifOutQLen.3 = Gauge32: 0 IF-MIB::ifOutQLen.4 = Gauge32: 0 IF-MIB::ifSpecific.1 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.2 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.3 = OID: SNMPv2-SMI::zeroDotZero IF-MIB::ifSpecific.4 = OID:
[asterisk-users] Free Polycom Provisioning Tool
In 2007, I released a Polycom Provisioning Tool. I retired the package earlier this year, and have had so many requests for it, I have revived the concept, new, improved, and still FREE. It now lives here: http://www.phoneprovisioning.com/ Provision any Polycom phone from the web, and you can even use our servers to host the files for you. It currently uses the newest version of the SIP application as well as the newest bootROM. Enjoy! Michael Munger, dCAP, MCPS, MCNPS, MBSS High Powered Help, Inc. Microsoft Certified Professional Microsoft Certified Small Business Specialist Digium Certified Asterisk Professional mailto:mich...@highpoweredhelp.com mich...@highpoweredhelp.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
