RE: [Asterisk-Users] Polycom IP500 - Phone TIme
And my dreamthat one day Polycom phones will support Australian Daylight savings... But it's only a dream. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Baranowski Sent: Friday, 29 April 2005 3:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme We set ours through the web interface on the phone Here is what we use for Phoenix. tcpIpApp.sntp.daylightSavings.enable=0 tcpIpApp.sntp.gmtOffset=-25200 tcpIpApp.sntp.address=207.46.130.100 Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin Sent: Thursday, April 28, 2005 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin Sent: Thursday, April 28, 2005 11:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; Subquestion to this ( although I much prefer setting the offset in the ipmid.cfg file myself ): How do you specify a negative offset when you are using the dhcp server that comes with windows server? Sean You need to use the hex value. Go to http://www.cisco.com/warp/public/109/calculate_hexadecimal_dhcp.html and at the bottom of the page there is a chart with the offsets and the hex value. Dan BUT, the hex values on the cisco site have periods in them...don't include those, just the 8 characters. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codec introducing huge latency
On 04/15/05 16:22 chawki hammoud said the following: --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: communications. ulaw is about 80kbps, and gsm about 28-30kbps. I monitored the download and upload data rate during my call using mandrake linux and it gave me 9.3 kb/s using ulaw and 3.1 kb/s for gsm. I think i had a chawki, your measurements seem to be in kiloBYTES per second, while andrew was giving your rates in kiloBITS per second. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Broadcasts
Is there a limit to how many voicemail boxes you can copy a voicemail to? I have a group that has about 40 members and it only copies to voicemail to 20 of them. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call a peer over the asterisk manager with a php script
wau thank you it works!! but,
first it says that e loop is detected,
and secondary what must I do to hand over the new workingchannel to my x-lite to use it???
DENGENSDana Olson [EMAIL PROTECTED] wrote:
On 4/27/05, Guy Boehm <[EMAIL PROTECTED]>wrote: Hello, I want to call a peer over the Asterisk Manager with this php-script: $socket = fsockopen("192.168.204.44","5038", $errno, $errstr, $timeout); fputs($socket, "Action: Login\r\n"); fputs($socket, "UserName: test\r\n"); fputs($socket, "Secret: test\r\n\r\n"); //fputs($socket, "Action: ListCommands\r\n\r\n"); fputs($socket, "Action: Originate\r\n"); fputs($socket, "Channel: 6159bfb47b9\r\n\r\n"); fputs($socket, "Exten: 1009\r\n\r\n"); fputs($socket, "Context: test\r\n\r\n"); fputs($socket, "Priority: 1\r\n\r\n"); fputs($socket, "Action: Logoff\r\n\r\n"); while (!feof($socket)) { $wrets .= fread($socket
,
8192); } fclose($socket); echo
Re: [Asterisk-Users] call a peer over the asterisk manager with a php script
wau thank you it works!! but, first it says that e loop is detected, and secondary what must I do to hand over the new workingchannel to my x-lite to use it??? DENGENSRichard Lyman [EMAIL PROTECTED] wrote: Guy Boehm wrote: fputs($socket, "Channel: 6159bfb47b9\r\n\r\n");Response: ErrorMessage: Invalid channel the Channel: var needs to be in the form of type/dev/numbertocalllike Channel: IAX2/user:[EMAIL PROTECTED]/14085551212___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Gesendet von Yahoo! Mail - Jetzt mit 250MB kostenlosem Speicher___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pattern Matching
We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in use/configured. I have been successful at setting up pattern matching as a catch all for 98 or so numbers not in use yet and I have been successful setting up the 2 numbers I want to make use of for now. Problem is, I can't use both at the same time! If I turn on the pattern matching then my greeting plays for the configured number, then the message plays for the invalid number (basically executing the extension with the pattern matching). I have read about sorting with pattern matching by using an include, I did this but it's not really helping. I have set a response timeout after the first extension plays it's greeting, I would think it should wait until it times out but it doesn't, it just immediately moves to the pattern matched extension. I must be missing something big here.. Any help is appreciated.. -- Private Label Wholesale Internet Access! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Barge In With Queues
Hi ! I wanted to use Barge IN with queues. ACtually what I want to do is a SIP user comes in a queue and then goes to a SIP agent. I want any application that allows me to listen to the conversation between them. I can be a supervisor extension or anything. I have used Flash Operator Panel but it works only if two asterisk SIP extensions are calling eachother. It doesnot work in the case if one of the call comes within from a queue. Any tweaking in extesnions.conf that could help me figure this out Any useful help , comments are appreciated ... thanks. Usman. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Prefix to CALLING Number ?
In article [EMAIL PROTECTED],
Josiah Bryan [EMAIL PROTECTED] wrote:
On Thursday 28 April 2005 11:07 am, barney wrote:
Hi there,
I`m trying to add some prefix before my local extensions, when my calls are
routed to ZAP trunk.
(i.e.: my local extension is , and i would like to send request to my
telco provider with source phone number 55)
Is there any way to do this ? I just know to add prefix (via prefix
application) to the called number (but not calling).
Thread on this 2 days ago.
Serach the archives. See footer on every message in this list.
For those who dont want to google archives, here ya go:
exten = ,1,Dial(Zap/g1/5${EXTEN}/);
Just put the number to add before the number to dial:
That's not the question he asked. He wants to prefix the caller-id.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] first few seconds of call is lost
I'm testing this strange behavior using livevoip, teliax, and voicepulse connect. I'm calling our office phone which picks up after two rings and plays a greeting. With livevoip and teliax I hear 3-4 rings and when the line answers I find myself a few seconds into the initial greeting. With voicepulse I hear two rings and then hear the complete greeting, which is the same as if I call using a pots line. Doesn't seem to make a difference whether I use iax or sip. This has happened consistantly and since day one of using teliax and livevoip, while voicepulse has never had this problem. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Daniel Salama wrote: This is great information. I have the following questions based on a hypothetical scenario and some assumptions: Based on the price of these configurations, I wouldn't even mind putting two servers each with 2 T1s just so that I could get all calls recorded and distribute the risk of failure. Now, I don't know if it would make a difference or not, but here it goes: Assuming the cost of the systems is of no importance for a moment (actually looking for the most scalable and reliable solution), which would be a better approach to solve the issue of activating 4 T1s which will be constantly taxed with load and be able to record all conversations: Scenario 1: 4 T1s into Asterisk (A1) where all SIP agents register. Call recording in A1. PSTN --4xT1-- A1 SIP_Agents Scenario 2: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where all SIP agents register (IAX to SIP transcoding). Call recording in A1 or A2. PSTN --4xT1-- A1 A2 SIP_Agents Scenario 3: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where half of SIP agents register to, and the other half would register in A1. Call recording in A1 and/or A2. PSTN --4xT1-- A1 SIP_Agents A1 --IAX-- A2 SIP_Agents Scenario 4: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisk (A2) will connect to A1 and A3 via IAX. All SIP Agents register at A2 (IAX to SIP transcoding). Call recording in [A1 and A3] or A2. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A2 SIP_Agents Scenario 5: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisks (A2 and A4) will connect to A1 and A3 respectively via IAX. Half SIP Agents register in A2 and other half in A4 (IAX to SIP transcoding). Call recording in [A1 and A3] or [A2 and A4]. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A4 SIP_Agents Hopefully you're all able to understand my 5 scenarios. I guess, my questions would be: 1) Is there a load limiting factor in terms of whether you do the Monitoring of the calls when you're doing TDM-IAX transcoding or IAX-SIP transcoding? 2) Will a single CPU machine handle the 4 T1s to do TDM-IAX transcoding, if another machine is doing the actual recording (IAX-SIP transconding) (Scenarios 2,3,4,5). Basically, just setup cheap Asterisk boxes to act as VoIP gateways and the distribute the load and/or intelligence on other Asterisk boxes to connect SIP agents and all dialing rules, etc? I haven't seen this before--can an agent log into a queue on a remote (i.e. over IAX) Asterisk server? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Prefix to CALLING Number ?
In article [EMAIL PROTECTED],
barney [EMAIL PROTECTED] wrote:
I`m trying to add some prefix before my local extensions, when my calls are
routed to ZAP trunk.
(i.e.: my local extension is , and i would like to send request to my
telco provider
with source phone number 55)
Is there any way to do this ? I just know to add prefix (via prefix
application) to the
called number (but not calling).
I haven't tried this, but the first thing I would try is this (replace
with the extension pattern you are using):
exten = ,1,SetCIDNum(${PREFIX}${CALLERIDNUM})
exten = ,2,Dial(.)
where PREFIX is a global variable containing the prefix you want to prepend.
See http://www.voip-info.org/wiki-Asterisk+cmd+SetCIDNum
You may need the 'a' flag to SetCIDNum too, depending on your application.
PS: sorry for my poor english
It's much better than my non-existent Slovakian!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Some * scripts: Pull asterisk config from LDAP and authenticate() against voicemail passwords
Hello,
Wrote some Python scripts last night to scratch an itch I was having
with Asterisk.
http://www.beerandspeech.org/cgi-bin/blosxom.cgi/tech/linux/050429a.html
http://www.beerandspeech.org/images/050429/asterisk-config.py.txt
http://www.beerandspeech.org/images/050429/asterisk-passwd.py.txt
asterisk-config.py lets me store asterisk config data in LDAP and generate
config files from it
head -n 32 asterisk-config.py
#!/usr/bin/python
# This script connects to an LDAP server looking for attributes that it can use
# to build some asterisk config files
#
# You need to setup your LDAP server (this was written in mind to connect to AD)
# and populate the correct attributes - it will then go away and build sip.conf,
# extensions.conf, SIP$MAC.cnf, SEP$MAC.cnf and CTLSEP$MAC.cnf
#
# This script builds sip.conf and extensions.conf in 3 parts... it reads a file
# (by default /etc/asterisk/sip.conf.head ) then dynamically builds the
# extensions and then appends the contents of sip.conf.tail
#
# This means you can maintain the static content of sip.conf ( the [general]
# parameters and dynamically build the rest
#
# On my extensions.conf.head the last line is [default] so all the
# dynamically created extensions go into this context
#
# You can also elect to not include certain SIP lines, phones and extensions in
# the autoconfiguration process
#
# It also writes out a Cisco XML phone directory file and a HTML phone list
# style file. On these files you can expand your LDAP search to include
# external non-asterisk users - maybe business contacts or such-like
#
# BUGS: Currently doesn't do a lot of error checking so missing attributes may
# make the script barf.. also doesn't do anything to voicemail.conf - thats
coming
# next
#
# If you have any questions, comments or patches please email me at [EMAIL
PROTECTED]
and secondly asterisk-passwd.py allows me to run the Authenticate() command
using the same password
people have for their voicemail. Basically greps voicemail.conf and dumps their
password into separate file
head -n 18 asterisk-passwd.py
#!/usr/bin/python
# I needed to add a Asterisk Authenticate() command to a dialplan but
# I wanted to use the same passwords as those in voicemail.conf to avoid giving
# users multiple passwords - grepping voicemail.conf seemed a little complicated
# so this script runs every 5 minutes under cron and it refreshs the password
files
# in case a user changes their voicemail password (which happens like.
never!)
#
# To include in the dial plan I used this
# [callforwarding]
# exten = s,1,Authenticate(/etc/asterisk/passwords/${CALLERIDNAME})
#
# So the Authenticate command would read /etc/asterisk/passwords/683 for my
extension
#
# BUGS: Hmm, I guess it depends if your ${CALLEDIDNAME} equals your numeric
extension number - mine did :)
# If not play with the splitting of the line to get the correct index for the
# filename - look at the output of asterisk -rvv to see which filename it's
trying to read
# Any questions or comments welcome to [EMAIL PROTECTED]
Hope someone finds these useful - please let me know if you use them and it
works.
Rgds
~sm
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Re: [Asterisk-Users] Prefix to CALLING Number ?
exten = ,1,Dial(Zap/g1/5${EXTEN}/);
Thank you Josiah, but If i do that, asterisk only add prefix to extension
which i`m dialing. But that is not my goal. I need to add prefix before my
local extension:
IP_PHONE --- ASTERISK PSTN --- TDM_PHONE
ext.No:
234567890
I`m trying to call TDM_PHONE from IP_PHONE, but asterisk is sending as
source number.
My goal is to tell asterisk to send number 55 as an source number of
call.
-b
- Original Message -
From: Josiah Bryan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 28, 2005 8:23 PM
Subject: Re: [Asterisk-Users] Prefix to CALLING Number ?
On Thursday 28 April 2005 11:07 am, barney wrote:
Hi there,
I`m trying to add some prefix before my local extensions, when my calls
are
routed to ZAP trunk.
(i.e.: my local extension is , and i would like to send request to
my
telco provider with source phone number 55)
Is there any way to do this ? I just know to add prefix (via prefix
application) to the called number (but not calling).
Thread on this 2 days ago.
Serach the archives. See footer on every message in this list.
For those who dont want to google archives, here ya go:
exten = ,1,Dial(Zap/g1/5${EXTEN}/);
Just put the number to add before the number to dial:
For example, to dial XXX-XXX and put a '9w' before the number when sending
to
a zap trunk:
exten = _NX,1,Dial(Zap/g1/9w${EXTEN})
Cheers!
-josiah
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Re: [Asterisk-Users] Polycom IP500 - Phone TIme
Paul Hales wrote: And my dreamthat one day Polycom phones will support Australian Daylight savings... But it's only a dream. Unless I am missing something, you don't need to dream about it - set it in ipmid.cfg. Look at the Sip Admim PDF for an explanation of: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1 Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Prefix to CALLING Number ?
I wrote:
I haven't tried this, but the first thing I would try is this (replace
with the extension pattern you are using):
exten = ,1,SetCIDNum(${PREFIX}${CALLERIDNUM})
exten = ,2,Dial(.)
where PREFIX is a global variable containing the prefix you want to prepend.
of course, you could just put the prefix digits in directly if you want:
exten = ,1,SetCIDNum(12345${CALLERIDNUM})
etc
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] (no subject)
Hi friends ! Cvan anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the useragent, as given in the asterisk wiki/asterisk+at+large. I don`t know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman Software Engg. Trainee Trail Ridge Software India Pvt. Ltd. Noida ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on Hold can' t hear it!
I have the current version og mpg running. But i am geeting the same problem even with the ringing tone. It seems to disappear sometimes make[1]: Entering directory `/usr/src/mpg123-0.59r' make[2]: Entering directory `/usr/src/mpg123-0.59r' make[2]: `mpg123' is up to date. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Prefix to CALLING Number ?
Thanks Tony, that is exactly what i was looking for :)
-b
- Original Message -
From: Tony Mountifield [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005 9:41 AM
Subject: [Asterisk-Users] Re: Prefix to CALLING Number ?
In article [EMAIL PROTECTED],
barney [EMAIL PROTECTED] wrote:
I`m trying to add some prefix before my local extensions, when my calls
are routed to ZAP trunk.
(i.e.: my local extension is , and i would like to send request to
my telco provider
with source phone number 55)
Is there any way to do this ? I just know to add prefix (via prefix
application) to the
called number (but not calling).
I haven't tried this, but the first thing I would try is this (replace
with the extension pattern you are using):
exten = ,1,SetCIDNum(${PREFIX}${CALLERIDNUM})
exten = ,2,Dial(.)
where PREFIX is a global variable containing the prefix you want to
prepend.
See http://www.voip-info.org/wiki-Asterisk+cmd+SetCIDNum
You may need the 'a' flag to SetCIDNum too, depending on your application.
PS: sorry for my poor english
It's much better than my non-existent Slovakian!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] how to share asterisk load with ser server
Hi friends ! Cvan anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the useragent, as given in the asterisk wiki/asterisk+at+large. I don`t know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman Software Engg. Trainee Trail Ridge Software India Pvt. Ltd. Noida ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Major problems with TDM400 and specifictelephones: suggestions?
I did the modification that Rich explains in his email on March 23rd below. I believe it works for me, because before this mod I was getting Ouch, part reset... errors at least once a week, rendering * unsuitable for production systems. After this mod, the system is running flawlessly for almost a month now. The closest capacitor value I was able to find was 100nF though, but it seems ok. And I had empty module slots, so I did not have to solder anything, I just inserted the pins firmly to the slot (capacitors usually have long legs). Very simple. Now I am quite happy with TDM400, and I recommend Rich's mod to everyone having such problems. Thanks Rich... - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Wednesday, March 23, 2005 1:32 AM Subject: Re: [Asterisk-Users] Major problems with TDM400 and specifictelephones: suggestions? I've improved the stability of my card by adding a capacitor on the reset line. Hasn't taken a hit in over two weeks. Is this the E/F or revised H card? Where and what cap did you install? My card reports as E/F; only have one, so not sure what the differences are between the various revisions. Adding the capacitor seems to have corrected the my TDM card goes out to lunch about every two weeks, and the only way to correct it is to reload the drivers or reboot the machine problem. Trying the following will likely void any digium warranty. Remove the TDM card from the PC and look closely at the pins associated with the four fxs/fxo modules. The pins are labeled on the modules as 1, 2, 19, and 20. The reset line is pin #2 while ground is pin #20. Carefully solder a .02 ufd capacitor between pin #2 and #20 on one module. Solder it onto a single module; no need to add one for each module. Install the card and boot up. Nothing more to it. Also, when the driver is loaded, my system reports an E/F card, but the board clearly says H. Anyone know for sure which is correct. Best guess is the driver reports the E/F based on the pci controller ID, which may or may not have been changed when revisions to the physical card were made. (That guess can be verified by checking the code; I remember seeing it, but don't remember which file.) As illustrated by the problems this card has with PCI slots. Even some motherboards which clearly are PCI 2.2 can't see the card in ANY slot. Yes, but the flacky pci bus issue is a motherboard problem that really has nothing to do with the digium card design. (The pci bus issue is fairly well understood by those involved with heavy audio apps. It just so happens to impact how the TDM card is used as well.) The FXS module is configured as a ground start device to provide dial tone to an EM switch, as well as an inward path. Multiple FXS modules would allow multiple connections, and GS is normally used to prevent GLARE, or head on collisions on the outside chance that several calls in and out occur at the same time. Digium support person #1 has stated that GS does not work on this module, and support person #2 says it should work In fact it does provide a GS trunk that works well for outgoing calls. On incoming calls, the module does not behave properly, in that before ringing begins, Tip should be grounded, and stay that way throughout ringing and answer. In fact, Tip seems to float somewhere during the ring cycle, and providing an external ground causes ring voltage to cease but not trip ringing. That's kind of weird as the Ground Start pin on the chipset isn't wired to anything whatsoever. The SI chipset apparently supports GS, but the circuit board traces don't. Guess it might be possible to _emulate_ GS through software, but its obvious the emulation isn't the same as the real thing. Also dial pulse detection seems very narrow, and different dials that work fine with much other equipment is not so with this card. The dial pulse sensing would be something done in the drivers, so sounds like that routine has the same narrow operating margins the echo canceller has. Trying to follow the code path for a functional TDM card is not to be taken lightly. Code is scattered across multiple drivers and buried in asterisk modules. Even those that consider themselves good asterisk developers stay way from this one. That doesn't bode well for any corrections, does it. Nope. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bri error
Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] bri error
Did you put your card in TE mode ? To it seems you have configured your card to act like a NT but if you are connected to bri telco lines, it should be in TE mode check in your zaptel.conf : bri te signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoyé : vendredi 29 avril 2005 12:08 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] bri error
if I do a zttool it shows TE mode On Fri, 2005-04-29 at 12:14, David Masure wrote: Did you put your card in TE mode ? To it seems you have configured your card to act like a NT but if you are connected to bri telco lines, it should be in TE mode check in your zaptel.conf : bri te signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:08 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] bri error
The problem may then originate from the NT of your telco -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:21 : David Masure Cc : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] bri error if I do a zttool it shows TE mode On Fri, 2005-04-29 at 12:14, David Masure wrote: Did you put your card in TE mode ? To it seems you have configured your card to act like a NT but if you are connected to bri telco lines, it should be in TE mode check in your zaptel.conf : bri te signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:08 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] bri error
and I have signalling = bri_cpe_ptmp On Fri, 2005-04-29 at 12:14, David Masure wrote: Did you put your card in TE mode ? To it seems you have configured your card to act like a NT but if you are connected to bri telco lines, it should be in TE mode check in your zaptel.conf : bri te signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:08 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPSwitchBoard Version 0.110 Released
Version 0.110 - 29. April 2005. Completely rebuild with .NET Version 2.0 BETA 2. IPS now has MDI (Multiple Documents Interface) meaning that you can have many program open at the same time, they will all update in real-time. Hotel/Call Shop Billing module. Status of Agents can be monitored on the main Panel. Show extensions reachable/unreachable Sort order of extensions/agents/queues on the panel. Translation facilities build into IPS Monitoring of channels Refresh extensions/agents/queues from the menu. Download FREE: http://ipswitchboard.thorben.dk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323
Hi I've successfully installed Asterisk-1.0.7, I've successfully installed Openh323 gatekeeper but not registered to Asterisk and so I've install PWlib v1.5.2 and Openh323 v1.12.2 libraries Now i try to install asterisk-oh323-0.5.10 : -edit Makefile inside the asterisk-oh323-0.5.10 directory and set the paths optionsaccording to system - Then i do make to build the oh323wrap library and the ASTERISK OH323 channel driver but i've this error: -I/usr/src/asterisk-1.0.7/include -I.../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:260: '__use_AST_MUTEX_DEFINE_STATIC_rather_then_AST_MUTEX_INITIALIZER..' :Undeclerad here (not in function) ... make:***[chan_oh323.o] Error 1 make :Leaving Directory /lib/asterisk-oh323-0.5.1/asterisk-driver make :***[Subdirs_all] Error 1 Have suggestions? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Errors from MP108 please help - confs included
Hi Pros,
I`m new to Asterisk Getting following errors on my * :
-- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
-- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
-- SIP/venus-e8ba answered SIP/1000-ee7c
-- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)
My SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729
[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18
[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833
[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001
[internal]
include = local-sip
[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup
exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup
Venus is my SIP provider (sorry u might hav guessed already)
1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.
MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1
please can anyone help me ?
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RE: [Asterisk-Users] Sipura SPA-841 and firewall
Just for future reference, I found the answer - I enabled Symmetric RTP: on the Advanced SIP page. Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Thursday, April 28, 2005 6:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Sipura SPA-841 and firewall I have an asterisk server and 4 Sipura phones behind a Linksys WRT54G router. I have set the DMZ to the Asterisk server's IP so that it can be seen from outside. I have a Sipura SPA-841 phone outside the router and set to proxy to the public IP of the router. The outside phone registers fine, dials fine, and I can hear the person speaking from inside the router, but I cannot be heard. Is there any explanation for this? Surely the DMZ allows all traffic to the PBX? This is driving me nuts. Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX attempt - Segmentation fault
Hi,
Yes, I understand that 'Ouch.. etc' comes
frommpg123. So I'm not loading musiconhold to avoid this problem (It
happened when exiting asterisk, nothing to do with the core
dumped).
And now, IAX is still crashing and turned
asterisk down with Segmentation fault everytime I make an IAX attempt. I don't
know whether to blame this version of asterisk or what. I used this machine with
an older version and IAX run without problems.
Just to mention that I'm using different
versions of IAX Softphones with the samesad result.
Regards,
Victor.
--
Victor Alvarez
wrote: Hello, I can't use IAX with my last
CVS-NHEAD-04/28/05-16:00:04 installation. Every time I try to use an iax channel
or register an iax user, I get a Segmentation fault.
Trace: -- Executing Dial("SIP/25-0368",
"IAX2/25|20|Tt") Segmentation
fault [EMAIL PROTECTED] root]# Ouch ... error while
writing audio data: : Broken pipe Warning,
flexibel rate not heavily tested!This is mpg123 error, not an IAX2
error.--
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[Asterisk-Users] how to configure ser and asterisk together to share the load
Hi friends ! Cvan anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the useragent, as given in the asterisk wiki/asterisk+at+large. I don`t know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman Software Engg. Trainee Trail Ridge Software India Pvt. Ltd. Noida ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] missing first digit when dial extension / dtmf problem ???
I'm using dtmfmode=inband with Sipura=3000 when I dial an internal extension most of the time the first digit is missing and I get an invalid extension message. Could it be dtmf problem or SIP? On the spa3k, I've use dtmf tx method = auto. In sip.conf, no dtmf entries at all (uses default). Been working just fine for many months. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CALEA compliance (was voip connection problems)
If you go to the fcc.gov website and search for CALEA there is around 7 documents that come up for April 27 2005. I believe I remember reading it one of those documents. - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 12:54 AM Subject: Re: [Asterisk-Users] voip connection problems trixter http://www.0xdecafbad.com wrote: a couple weeks ago the FCC (america) ruled that all voip providers that connect to the PSTN (vonage, broadvoice, voicepulse, etc) have to have CALEA support (wiretap equipment for law enforcement). Failure to comply is a $10,000 fine per day. Could you please provide a reference for this assertion? Thx. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to share asterisk load with ser
Hi friends ! Cvan anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the useragent, as given in the asterisk wiki/asterisk+at+large. I don`t know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman Software Engg. Trainee Trail Ridge Software India Pvt. Ltd. Noida ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNID empty on incoming calls
Hi, I see others have had this problem. Is there a solution ? I have a BRI, using zaphfc. If I enable debugging so: bri debug span 1 and then make an incoming call I can see that the DNID info is definitely provided by the PSTN - Here's proof: Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1842' (The '1842' is the part of my telephone number that I'm looking for) But how can I get hold of this info - DNID is empty ? -- Executing NoOp(Zap/1-1, DIALEDPEERNAME = ) in new stack -- Executing NoOp(Zap/1-1, DIALEDPEERNUMBER = ) in new stack -- Executing NoOp(Zap/1-1, DIALEDTIME = ) in new stack -- Executing NoOp(Zap/1-1, DIALSTATUS = ) in new stack -- Executing NoOp(Zap/1-1, DNID = ) in new stack -- Executing NoOp(Zap/1-1, EXTEN= s) in new stack -- Executing NoOp(Zap/1-1, RDNIS= ) in new stack Many thanks Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk-oh323
In article [EMAIL PROTECTED], [EMAIL PROTECTED] wrote: Hi I've successfully installed Asterisk-1.0.7, I've successfully installed Openh323 gatekeeper but not registered to Asterisk and so I've install PWlib v1.5.2 and Openh323 v1.12.2 libraries Now i try to install asterisk-oh323-0.5.10 : -edit Makefile inside the asterisk-oh323-0.5.10 directory and set the paths optionsaccording to system - Then i do make to build the oh323wrap library and the ASTERISK OH323 channel driver but i've this error: -I/usr/src/asterisk-1.0.7/include -I.../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:260: '__use_AST_MUTEX_DEFINE_STATIC_rather_then_AST_MUTEX_INITIALIZER..' :Undeclerad here (not in function) ... make:***[chan_oh323.o] Error 1 make :Leaving Directory /lib/asterisk-oh323-0.5.1/asterisk-driver make :***[Subdirs_all] Error 1 Have suggestions? Yes, use asterisk-oh323 version 0.6.5, and the Janus-patch4 versions of PWLib (1.6.6.3) and OpenH323 (1.13.5.3), and all the above problems will magically disappear! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with TDM400P card
Guys I have a problem getting a TDM400P card to go. It has 4 FXS ports (green modules) and I get this error: [EMAIL PROTECTED] root]# ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Slaves: 06) 6 channels configured. ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? My zaptel.conf reads: [EMAIL PROTECTED] root]# more /etc/zaptel.conf fxsks=1 fxsks=2 fxoks=3-6 loadzone=us defaultzone=us And my rc.local loads: /sbin/modprobe zaptel /sbin/modprobe wcfxo /sbin/modprobe wctdm The 2 100p cards load perfectly but the TDM is not. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cost field in Call Detail Records (cdr)
Hi there I don't know if this utility is available anywhere at the moment but I thought id ask you guys if you know of one What I would like is a way of adding a field to my cdr records (either the Master.csv or a destination mysql table) for cost ! based on some sort of config file (or table) which has a listing of all the tariffs for particular prefixes, ie in the UK, the 0870 prefix is national rate at £0.04p per minute (don't know if that is exact btw !) so I would like a way of adding a field that determines , for example that extension 7201 made a 4 minute call to an 0870 number and therefore that call has cost £0.16p. I can then get my reporting software to pull this additional field into the report Is there anything around that does this sort of thing, open source or otherwise ?? Any help would be greatly appreciated Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] recomended phone
Hi all. I am starting to develop a voip solution with asterisk and sip and 100 telephones with SIP . I need the following features. calls group blind transfer attended transfer ( supervised ) pickup groups voicemail record call the attended transfer feature is very important, cause is an enterprise solution. then the question... :( a think that a PIV 3Gz with 2G ram and UW SCSI 200G is enough, but i need a phone that cover all features in an easy mode and as cheap as i can. which Phone should i use.? Help Please. Best Regards. César García. Director de Sistemas, IdecNet S.A. Centro de Gestión de Red. Edificio IdecNet. C/Juan XXIII 44. E-35004, Las Palmas de Gran Canaria, Islas Canarias - España. Tfn: +34 828 111 000 Ext: 340 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Traffic Testing
The homepage http://sipsak.org contains some examples. If you need help with special cases drop me a line. Regards Nils Ohlmeier On Friday 29 April 2005 02:54, Anton Krall wrote: Can you send some command line examples on how to use it? Thx! |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |[EMAIL PROTECTED] |Sent: Jueves, 28 de Abril de 2005 07:05 p.m. |To: asterisk-users@lists.digium.com |Subject: RE: [Asterisk-Users] Traffic Testing | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] Behalf Of Anton | Krall | Sent: Thursday, April 28, 2005 6:07 PM | To: 'Asterisk Users Mailing List - Non-Commercial Discussion' | Subject: [Asterisk-Users] Traffic Testing | | | Guys, is there any way to generate simulated traffic via sip or IAX2 | for testing cpu load and asterisk? (sip client simulation, etc)? | |yes, use sipsak utility | |-- |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] first few seconds of call is lost
I'm testing this strange behavior using livevoip, teliax, and voicepulse connect. I'm calling our office phone which picks up after two rings and plays a greeting. With livevoip and teliax I hear 3-4 rings and when the line answers I find myself a few seconds into the initial greeting. With voicepulse I hear two rings and then hear the complete greeting, which is the same as if I call using a pots line. Doesn't seem to make a difference whether I use iax or sip. This has happened consistantly and since day one of using teliax and livevoip, while voicepulse has never had this problem. I'm using: [bus-ivr-main] exten = s,1,Wait,1 exten = s,2,Answer exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,15 exten = s,5,Background(npi-greeting) ; Thanks for calling press 1 for for both livevoip.com and teliax.com (both with iax), no problems. If you want to listen to it, call 913-440- and listen for the number of rings before the ivr audio. If you're getting something different (using the same sort of config as above), it might be related to exactly where your did's originate. (In other words, I'd have to guess that of the many area codes and CO's that source their did's, some equipment at specific geographical locations might not be configured exactly the same as other sources. But, that's purely a guess knowing both companies rely heavily on level-3 infrastructure, and level-3's primary business is not that of a local exchange carrier.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Major problems with TDM400 and specifictelephones: suggestions?
For the record archives, there are apparently about eight different revisions of the TDM card (observed via pci id revisions in driver code), and this mod only impacts one of apparently multiple problems. Since digium is very tight lipped about problems, there is a high probability that other issues abound on early versions of the card. It should also be noted the digium TDM card has a two year warranty, therefore take advantage of that warranty to address the design problems if problems continue to impact your system. The warranty will expire on the initially shipped TDM cards within about twelve months or so. Rich I did the modification that Rich explains in his email on March 23rd below. I believe it works for me, because before this mod I was getting Ouch, part reset... errors at least once a week, rendering * unsuitable for production systems. After this mod, the system is running flawlessly for almost a month now. The closest capacitor value I was able to find was 100nF though, but it seems ok. And I had empty module slots, so I did not have to solder anything, I just inserted the pins firmly to the slot (capacitors usually have long legs). Very simple. Now I am quite happy with TDM400, and I recommend Rich's mod to everyone having such problems. Thanks Rich... - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Wednesday, March 23, 2005 1:32 AM Subject: Re: [Asterisk-Users] Major problems with TDM400 and specifictelephones: suggestions? I've improved the stability of my card by adding a capacitor on the reset line. Hasn't taken a hit in over two weeks. Is this the E/F or revised H card? Where and what cap did you install? My card reports as E/F; only have one, so not sure what the differences are between the various revisions. Adding the capacitor seems to have corrected the my TDM card goes out to lunch about every two weeks, and the only way to correct it is to reload the drivers or reboot the machine problem. Trying the following will likely void any digium warranty. Remove the TDM card from the PC and look closely at the pins associated with the four fxs/fxo modules. The pins are labeled on the modules as 1, 2, 19, and 20. The reset line is pin #2 while ground is pin #20. Carefully solder a .02 ufd capacitor between pin #2 and #20 on one module. Solder it onto a single module; no need to add one for each module. Install the card and boot up. Nothing more to it. Also, when the driver is loaded, my system reports an E/F card, but the board clearly says H. Anyone know for sure which is correct. Best guess is the driver reports the E/F based on the pci controller ID, which may or may not have been changed when revisions to the physical card were made. (That guess can be verified by checking the code; I remember seeing it, but don't remember which file.) As illustrated by the problems this card has with PCI slots. Even some motherboards which clearly are PCI 2.2 can't see the card in ANY slot. Yes, but the flacky pci bus issue is a motherboard problem that really has nothing to do with the digium card design. (The pci bus issue is fairly well understood by those involved with heavy audio apps. It just so happens to impact how the TDM card is used as well.) The FXS module is configured as a ground start device to provide dial tone to an EM switch, as well as an inward path. Multiple FXS modules would allow multiple connections, and GS is normally used to prevent GLARE, or head on collisions on the outside chance that several calls in and out occur at the same time. Digium support person #1 has stated that GS does not work on this module, and support person #2 says it should work In fact it does provide a GS trunk that works well for outgoing calls. On incoming calls, the module does not behave properly, in that before ringing begins, Tip should be grounded, and stay that way throughout ringing and answer. In fact, Tip seems to float somewhere during the ring cycle, and providing an external ground causes ring voltage to cease but not trip ringing. That's kind of weird as the Ground Start pin on the chipset isn't wired to anything whatsoever. The SI chipset apparently supports GS, but the circuit board traces don't. Guess it might be possible to _emulate_ GS through software, but its obvious the emulation isn't the same as the real thing. Also dial pulse detection seems very narrow, and different dials that work fine with much other equipment is not so with this card. The dial pulse sensing would be something done in the drivers, so sounds like that routine has the same narrow operating margins the echo canceller has. Trying to follow the code path for a functional
[Asterisk-Users] Queue Monitor Filename Problem
Hi ! I am using queues with MOnitor Application but the thing is that Iwant to save the files starting with the Answering agent name. I have tried a lot of things but nothing seems to work. If i put Monitor application on top of dialing the agent then as soon as agent picks up the recording hangs up without recording anyhting. And if I put the Monitor application on top of Queue command then I have to specify the saving filename before I know that to which agent the call is going. ANy comments , suggestions appreciated. Thanks, Usman. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime feature
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, Does someone knows if the next release of Asterisk (1.0.8?) will have Realtime support and when we will have the next Asterisk release with Realtime features? Thanks in advance. - -- Rodrigo P. Telles [EMAIL PROTECTED] IVOZ # 1009 Diretor de Tecnologia Devel-IT - http://www.devel.it Grupo Bestcom -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCciwBiLK8unYgEMQRAqFLAJ449p8tLjyglG+Mt40wUllfDBTyQQCeIAlM 8Q+wdor3HoczTGFxG7Fzdi4= =UKW2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with TDM400P card
Anton Krall wrote: Guys I have a problem getting a TDM400P card to go. It has 4 FXS ports (green modules) and I get this error: [EMAIL PROTECTED] root]# ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Slaves: 06) 6 channels configured. ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? My zaptel.conf reads: [EMAIL PROTECTED] root]# more /etc/zaptel.conf fxsks=1 fxsks=2 fxoks=3-6 Why do you configure 6 channels if you only have 4 FXS? Try fxsks=1-4 regards, klaus loadzone=us defaultzone=us And my rc.local loads: /sbin/modprobe zaptel /sbin/modprobe wcfxo /sbin/modprobe wctdm The 2 100p cards load perfectly but the TDM is not. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 Technology and VoIP Gateway Primer
Why would you use gateways and PRI's when several of the major carriers (ATT, Global Crossing, etc.) also have products that can interface directly with SIP for the same per minute cost? We have a multisite Asterisk call center application and are routing all calls over private VPN to one central Asterisk location from where we have multiple point-to-point T1's going straight into Global Crossing. They are accepting the traffic as SIP g.729a and are handling the gateway themselves. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Callum McGillivray Sent: Friday, April 29, 2005 1:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer Hi Matt everyone else, We have also been steering toward using a gateway for our large installation. Ours differs from your significantly in as much as our setup will involve 8 apartment buildings located throughout the CBD. Each apartment building will have as many as 600 extensions (rooms) with an Asterisk Server in the comms room in the basement. Incoming and Outgoing calls are going to be trunked from the Asterisk box along a fiber link back to our core exchange, where the calls will be handed off to a gateway machine (Cisco?) which will have an impressively large number of PRI's plugged into the back of it. My (very vague) examination so far tells me that I can use something along the lines of a Cisco AS5400 (a couple of which I have kicking around here in the office). Has anyone had experience in handing off / receiving calls from a Cisco AS5400 with Asterisk ? How is it done ? Matt, is this similar to the idea that you have for your project ? What Cisco hardware have you looked at so far ? How many E1/T1 lines are you going to have terminating on your setup ? Cheers, Callum Matt Roth wrote: Michael, Have you decided which PSTN-VoIP gateway you'll use? Not yet, but our preference is a Cisco gateway. Lucent, Quintum, and AudioCodes also make TDM-VoIP gateways. Prior to purchasing any hardware, our entire layout will be posted to this list in detail for review. Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Deb ian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with TDM400P card
I have a problem getting a TDM400P card to go. It has 4 FXS ports (green modules) and I get this error: [EMAIL PROTECTED] root]# ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Slaves: 06) 6 channels configured. ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? My zaptel.conf reads: [EMAIL PROTECTED] root]# more /etc/zaptel.conf fxsks=1 fxsks=2 fxoks=3-6 loadzone=us defaultzone=us And my rc.local loads: /sbin/modprobe zaptel /sbin/modprobe wcfxo /sbin/modprobe wctdm The 2 100p cards load perfectly but the TDM is not. Any ideas? What does zttool indicate? Have you tried moving the cards around in different slots? Any shared interrupt issues? Try loading wctdm before wcfxo. Try removing the x100p's and loading the TDM card only. Any issues? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need help
I am having an issue with the asterisk system not responding to dialed numbers during an active call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone config? and worse I don't even know what Keywords to search for. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Monitor Filename Problem
On 4/29/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi ! I am using queues with MOnitor Application but the thing is that Iwant to save the files starting with the Answering agent name. I have tried a lot of things but nothing seems to work. If i put Monitor application on top of dialing the agent then as soon as agent picks up the recording hangs up without recording anyhting. And if I put the Monitor application on top of Queue command then I have to specify the saving filename before I know that to which agent the call is going. ANy comments , suggestions appreciated. Thanks, Usman. You could start the recording with a manager command remotely via telnet... I've been working on this, but my problem is that I don't know socks and PHP too well. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EuroISDN bearer capability pass thru from (fax) a/b adapter on OctoBRI to TE410P
Hi @all, I have following hardware setup: standalone analog fax-machine - DeTeWe TA33clip a/b adapter - OctoBRI S0 NT-Mode - Digium TE410P - german PSTN Software: Debian unstable binary packages ii asterisk 1.0.7.dfsg.1-2 open source Private Branch Exchange (PBX) ii libpri1 1.0.7-1 Primary Rate ISDN specification library Problem: The a/b adapter signals on an outgoing call going into the OctoBRI the bearer capabilty 3.1khz audio (PRI_TRANS_CAP_3_1K_AUDIO) but the outbound call initiated to the EuroISDN PSTN via an Digium TE410P signals the bearer capability speech (PRI_TRANS_CAP_SPEECH). The result is that my fax calls are rejected by some ISDN destinations because of mismatching bearer capabilities. What can I do to have the bearer capabilites get passed through asterisk correctly ? I tried to patch /usr/src/asterisk-1.0.7.dfsg.1/channels/chap_zap.c (replace PRI_TRANS_CAP_SPEECH with (PRI_TRANS_CAP_3_1K_AUDIO). But then I do a make/make install I get an asterisk version without the BRIstuff compiled in. So I'm still looking how to corretcly compile the debian source myself with the BRI stuff in, as the original binary packages are.. Any hints on this topic are also greatly appreciated. TIA, Bruno -- bruno.voigt -at- ic3s.de ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with TDM400P card
Guys I have a problem getting a TDM400P card to go. It has 4 FXS ports (green modules) and I get this error: [EMAIL PROTECTED] root]# ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Slaves: 06) 6 channels configured. ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? My zaptel.conf reads: [EMAIL PROTECTED] root]# more /etc/zaptel.conf fxsks=1 fxsks=2 fxoks=3-6 loadzone=us defaultzone=us And my rc.local loads: /sbin/modprobe zaptel /sbin/modprobe wcfxo /sbin/modprobe wctdm The 2 100p cards load perfectly but the TDM is not. Any ideas? Could you post the contents of dmesg that are relavant when you load the modules?? Just want to make sure that things are actually loading in the order you have your zapatel.conf set for. It sounds like the cards are not loading in the same order you have the channels configed for. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over solutions
The disk array would be the only expensive add on, more than a normal asterisk system. It all depends on how important voicemail is in your application, although there are cheaper alternatives (NFS for example, but then your NFS server becomes a single point of failure, depending on the disk array that same issue could be true there as well). If you are on a budget, I would suggest to look at a drbd+heartbeat combination. DRBD is a block device which is designed to build high availability clusters. This is done by mirroring a whole block device via (a dedicated) network. You could see it as a network raid-1. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with TDM400P card
Zttool shows nothing inside thebox. I tried removing the x100 cards, moving the tdm card around, disabled all usb and unnecessary stuff still, kudzu when booting up shows the card and the card shows up on /etc/sysconfig/hwconf but I wonder why it shows 2 of these and I only have 1 tdm400p card with 1 module class: MODEM bus: PCI detached: 1 driver: hisax desc: Tiger Jet Network Inc.|Intel 537 vendorId: e159 deviceId: 0001 subVendorId: 8086 subDeviceId: 0003 pciType: 1 - class: MODEM bus: PCI detached: 1 driver: hisax desc: Tiger Jet Network Inc.|Intel 537 vendorId: e159 deviceId: 0001 subVendorId: 8086 subDeviceId: 0003 pciType: 1 Still, interrupts doesn't show the card [EMAIL PROTECTED] root]# cat /proc/interrupts CPU0 0:3994353 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 10: 95510 XT-PIC eth0 14: 129871 XT-PIC ide0 NMI: 0 ERR: 0 And when trying to load the drvier [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobe wctdm /lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wctdm.o: insmod /lib/modules/2.4.20-8/misc/wctdm.o failed /lib/modules/2.4.20-8/misc/wctdm.o: insmod wctdm failed You have new mail in /var/spool/mail/root I tried using diff. modules with no luck.,. Could be the mobo itself? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Viernes, 29 de Abril de 2005 08:59 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Problems with TDM400P card | | I have a problem getting a TDM400P card to go. | | It has 4 FXS ports (green modules) and I get this error: | | [EMAIL PROTECTED] root]# ztcfg -v | | Zaptel Configuration | == | | | Channel map: | | Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS | Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) | (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: |04) Channel | 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart | (Default) (Slaves: 06) | | 6 channels configured. | | ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you | forget that FXS interfaces are configured with FXO |signalling and that | FXO interfaces use FXS signalling? | | My zaptel.conf reads: | | [EMAIL PROTECTED] root]# more /etc/zaptel.conf | fxsks=1 | fxsks=2 | fxoks=3-6 | loadzone=us | defaultzone=us | | And my rc.local loads: | | /sbin/modprobe zaptel | /sbin/modprobe wcfxo | /sbin/modprobe wctdm | | The 2 100p cards load perfectly but the TDM is not. | | Any ideas? | |What does zttool indicate? | |Have you tried moving the cards around in different slots? | |Any shared interrupt issues? | |Try loading wctdm before wcfxo. | |Try removing the x100p's and loading the TDM card only. Any issues? | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need help
This is a DTMF issue, You must adjust this on the especific channel conf file. For example, ia sip phone cannot dial any number during an active call, you must see sip.conf and the config in your hardphone or softphone. Ismael. Tim Touhsaent [EMAIL PROTECTED] Enviado por: [EMAIL PROTECTED] 04/29/2005 03:16 PM Por favor, responda a Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Para Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com cc Asunto [Asterisk-Users] need help I am having an issue with the asterisk system not responding to dialed numbers during an active call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone config? and worse I don't even know what Keywords to search for. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need help
Is this a SIP phone? I had to upgrade the firmware on my SIP phones to alleviate this. It seems that the phone would actually disable it's own keypad after dialling. Ian [EMAIL PROTECTED] 29/04/2005 09:16 I am having an issue with the asterisk system not responding to dialed numbers during an active call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone config? and worse I don't even know what Keywords to search for. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with TDM400P card
Zttool shows nothing inside thebox. I tried removing the x100 cards, moving the tdm card around, disabled all usb and unnecessary stuff still, kudzu when booting up shows the card and the card shows up on /etc/sysconfig/hwconf but I wonder why it shows 2 of these and I only have 1 tdm400p card with 1 module If I remember correctly, when I installed [EMAIL PROTECTED] and it did its reboot, the TDM was removed from kudzu as it loaded the linux zaptel and you want to load the zaptel obtained from Digium. Try removing it permanantly from kudzu then try loading your modules. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with TDM400P card
Anton Krall wrote: Zttool shows nothing inside thebox. I have had similar problems with a TDM400 and CERTAIN Motherboards which are PCI 2.2 but the TDM400 is not seen, in my case, AT ALL The one I have reports it as an E/F but the silk-screen clearly says H, Digium contends there is no problem with the card, the reporting of different version numbers is a known bug but doesn't prevent the card from working, and because I can place it in another machine and get it working, they refuse to acknowledge there is any defect in the board. Perhaps a different motherboard? That is Digium's answer. Just keep going through hardware that otherwise meets published specs until you find one that works. I have to conclude that, due to Digiums refusal to acknowledge there are problems with the design, ( and there are more I won't bore you with ) and no willingness to address the issues that have been raised on this list time and time again, that the TDM400 should be considered a card of last resort when absolutely nothing else will work. Seems their IAXy falls into that same classification. Can't say about their T1/E1 cards JMO John Novack I tried removing the x100 cards, moving the tdm card around, disabled all usb and unnecessary stuff still, kudzu when booting up shows the card and the card shows up on /etc/sysconfig/hwconf but I wonder why it shows 2 of these and I only have 1 tdm400p card with 1 module class: MODEM bus: PCI detached: 1 driver: hisax desc: Tiger Jet Network Inc.|Intel 537 vendorId: e159 deviceId: 0001 subVendorId: 8086 subDeviceId: 0003 pciType: 1 - class: MODEM bus: PCI detached: 1 driver: hisax desc: Tiger Jet Network Inc.|Intel 537 vendorId: e159 deviceId: 0001 subVendorId: 8086 subDeviceId: 0003 pciType: 1 Still, interrupts doesn't show the card [EMAIL PROTECTED] root]# cat /proc/interrupts CPU0 0:3994353 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 10: 95510 XT-PIC eth0 14: 129871 XT-PIC ide0 NMI: 0 ERR: 0 And when trying to load the drvier [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobe wctdm /lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wctdm.o: insmod /lib/modules/2.4.20-8/misc/wctdm.o failed /lib/modules/2.4.20-8/misc/wctdm.o: insmod wctdm failed You have new mail in /var/spool/mail/root I tried using diff. modules with no luck.,. Could be the mobo itself? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Viernes, 29 de Abril de 2005 08:59 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Problems with TDM400P card | | I have a problem getting a TDM400P card to go. | | It has 4 FXS ports (green modules) and I get this error: | | [EMAIL PROTECTED] root]# ztcfg -v | | Zaptel Configuration | == | | | Channel map: | | Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS | Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) | (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: |04) Channel | 05: FXO Kewlstart (Default) (Slaves: 05) Channel 06: FXO Kewlstart | (Default) (Slaves: 06) | | 6 channels configured. | | ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you | forget that FXS interfaces are configured with FXO |signalling and that | FXO interfaces use FXS signalling? | | My zaptel.conf reads: | | [EMAIL PROTECTED] root]# more /etc/zaptel.conf | fxsks=1 | fxsks=2 | fxoks=3-6 | loadzone=us | defaultzone=us | | And my rc.local loads: | | /sbin/modprobe zaptel | /sbin/modprobe wcfxo | /sbin/modprobe wctdm | | The 2 100p cards load perfectly but the TDM is not. | | Any ideas? | |What does zttool indicate? | |Have you tried moving the cards around in different slots? | |Any shared interrupt issues? | |Try loading wctdm before wcfxo. | |Try removing the x100p's and loading the TDM card only. Any issues? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on VMWare ESX/blade servers
Has anyone had any experience (good or bad) running Asterisk under VMWare ESX server on a blade chassis? This application will (fairly obviously) not include Zap channelsactually, it will be SIP-only. Please feel free to contact me off-list and I'll summarize for the list later. Daryl G. Jurbala NGM Tec, Inc. Tel: 215-862-1160 ext. 235 Fax: 215-862-9880 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Hardware Recommendation
If price would truly not an option just get one of the Signate Telephony 5000 servers(http://www.signate.com/pbx.php) They are about $18,000 and allow you to have upto 5000 SIP streams go through it. You could have that be your gateway and do the SIP-IAX through that machine and scale upto 100 T1s if you want. But that is a bit steep. So on to your choices. I would really say that the setup you choose will depend on what kind of users you have as well as how often you need to change/add users to the system and how the users are using the system at what times. Any of them that you listed could work depending on how they are used, but in some cases you may not want to use some of the scenarios listed because they would either be incapable of meeting your needs or overly complex to manage. The easiest and cheapest one would actually not be listed: Scenario 6: Direct SIP-Zap on two separate servers half SIP users on each server PSTN --2xT1-- A1 SIP_Agents PSTN --2xT1-- A2 SIP_Agents There is really no reason to have another 2 servers running IAX to the T1 servers, and this is simple and easy to set up and involves only 2 machines. The next setup I would recommend would be Scenario 4, although you will have to get a machine with a fast/wide BUS(like an Apple G5) to handle ever increasing numbers of SIP-IAX streams as the system would grow. If you can explain more about what kind of use this system will have I can give a better recommendation. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation This is great information. I have the following questions based on a hypothetical scenario and some assumptions: Based on the price of these configurations, I wouldn't even mind putting two servers each with 2 T1s just so that I could get all calls recorded and distribute the risk of failure. Now, I don't know if it would make a difference or not, but here it goes: Assuming the cost of the systems is of no importance for a moment (actually looking for the most scalable and reliable solution), which would be a better approach to solve the issue of activating 4 T1s which will be constantly taxed with load and be able to record all conversations: Scenario 1: 4 T1s into Asterisk (A1) where all SIP agents register. Call recording in A1. PSTN --4xT1-- A1 SIP_Agents Scenario 2: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where all SIP agents register (IAX to SIP transcoding). Call recording in A1 or A2. PSTN --4xT1-- A1 A2 SIP_Agents Scenario 3: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding. Asterisk (A1) connects to Asterisk (A2) via IAX where half of SIP agents register to, and the other half would register in A1. Call recording in A1 and/or A2. PSTN --4xT1-- A1 SIP_Agents A1 --IAX-- A2 SIP_Agents Scenario 4: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisk (A2) will connect to A1 and A3 via IAX. All SIP Agents register at A2 (IAX to SIP transcoding). Call recording in [A1 and A3] or A2. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A2 SIP_Agents Scenario 5: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX transcoding. Asterisks (A2 and A4) will connect to A1 and A3 respectively via IAX. Half SIP Agents register in A2 and other half in A4 (IAX to SIP transcoding). Call recording in [A1 and A3] or [A2 and A4]. PSTN --2xT1-- A1 A2 SIP_Agents PSTN --2xT1-- A3 A4 SIP_Agents Hopefully you're all able to understand my 5 scenarios. I guess, my questions would be: 1) Is there a load limiting factor in terms of whether you do the Monitoring of the calls when you're doing TDM-IAX transcoding or IAX-SIP transcoding? 2) Will a single CPU machine handle the 4 T1s to do TDM-IAX transcoding, if another machine is doing the actual recording (IAX-SIP transconding) (Scenarios 2,3,4,5). Basically, just setup cheap Asterisk boxes to act as VoIP gateways and the distribute the load and/or intelligence on other Asterisk boxes to connect SIP agents and all dialing rules, etc? Thanks, Daniel On Apr 28, 2005, at 9:17 PM, mattf wrote: You can throw together a single P4 3GHz with 1GB RAM and 2 x 80GB SATA HD for about $600. One of those can easily handle a Sangoma dual T1 card($900) or a Digium quad T1 card($1400). For that you can have a system for about $1500-$2000 that will be able to fully record 2 T1s(48 channels) worth of Zap-SIP conversations. Putting two of those together with a nice big fileserver will give you a lot of flexibility, as well as only a reduction in capacity if one of the servers go down instead of a total outage, for about the same overall price of a single high-end Dual Xeon server. Building your system
Re: [Asterisk-Users] Major problems with TDM400 and specifictelephones: suggestions?
Rich Adamson wrote: For the record archives, there are apparently about eight different revisions of the TDM card (observed via pci id revisions in driver code),and this mod only impacts one of apparently multiple problems. Since digium is very tight lipped about problems, there is a high probability that other issues abound on early versions of the card. And they flat out refuse any exchange of a card that has certain problems, and their answer to a card that can't be seen by a Motherboard that is clearly PCI 2.2 is try it in another machine, and if it works, you're stuck The reporting of revision E/f on a board that is marked rev H physically is not seen as a problem that warrents exchange. Best use other hardware unless the TDM400 is the only solution. John Novack It should also be noted the digium TDM card has a two year warranty, therefore take advantage of that warranty to address the design problems if problems continue to impact your system. The warranty will expire on the initially shipped TDM cards within about twelve months or so. Rich I did the modification that Rich explains in his email on March 23rd below. I believe it works for me, because before this mod I was getting Ouch, part reset... errors at least once a week, rendering * unsuitable for production systems. After this mod, the system is running flawlessly for almost a month now. The closest capacitor value I was able to find was 100nF though, but it seems ok. And I had empty module slots, so I did not have to solder anything, I just inserted the pins firmly to the slot (capacitors usually have long legs). Very simple. Now I am quite happy with TDM400, and I recommend Rich's mod to everyone having such problems. Thanks Rich... - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Wednesday, March 23, 2005 1:32 AM Subject: Re: [Asterisk-Users] Major problems with TDM400 and specifictelephones: suggestions? I've improved the stability of my card by adding a capacitor on the reset line. Hasn't taken a hit in over two weeks. Is this the E/F or revised H card? Where and what cap did you install? My card reports as E/F; only have one, so not sure what the differences are between the various revisions. Adding the capacitor seems to have corrected the my TDM card goes out to lunch about every two weeks, and the only way to correct it is to reload the drivers or reboot the machine problem. Trying the following will likely void any digium warranty. Remove the TDM card from the PC and look closely at the pins associated with the four fxs/fxo modules. The pins are labeled on the modules as 1, 2, 19, and 20. The reset line is pin #2 while ground is pin #20. Carefully solder a .02 ufd capacitor between pin #2 and #20 on one module. Solder it onto a single module; no need to add one for each module. Install the card and boot up. Nothing more to it. Also, when the driver is loaded, my system reports an E/F card, but the board clearly says H. Anyone know for sure which is correct. Best guess is the driver reports the E/F based on the pci controller ID, which may or may not have been changed when revisions to the physical card were made. (That guess can be verified by checking the code; I remember seeing it, but don't remember which file.) As illustrated by the problems this card has with PCI slots. Even some motherboards which clearly are PCI 2.2 can't see the card in ANY slot. Yes, but the flacky pci bus issue is a motherboard problem that really has nothing to do with the digium card design. (The pci bus issue is fairly well understood by those involved with heavy audio apps. It just so happens to impact how the TDM card is used as well.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Experienced Asterisk Consultant in Chicago, IL
Kerry, Thanks for the reply but we are looking for someone in the Chicagoland area. Regards, Jon Dahl SKTY Trading, LLC. From: Kerry Garrison [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Experienced Asterisk Consultant in Chicago, IL Date: Thu, 28 Apr 2005 08:31:40 -0700 We do a good amount of remote work if that isn't a problem for you. We can reconfigure the entire system and have it ready to drop into place. If the job is big enough it might warrant a visit during installation but that isnt always the case. Kerry Garrison Tech Data Pros http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Dahl Sent: Thursday, April 28, 2005 8:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Experienced Asterisk Consultant in Chicago, IL I searched for the mailing list guidelines on google and couldn't find them. I apologize in advance if this is not the appropriate list. My company is moving their office and we have decided to use VoIP for our phone solution. We will be using Cisco 7960 phones powered by a Cisco 3560 switch. The server running Asterisk will be a Dell 2650 Dual Xeon with 2GB of RAM running Linux. We need to set this system up in the next month and I was wondering if there are any Asterisk consultants in the Chicagoland area to assist us in the initial setup and quite possibly on an as needed basis? We are located in the Loop area. Regards, Jon Dahl _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ On the road to retirement? Check out MSN Life Events for advice on how to get there! http://lifeevents.msn.com/category.aspx?cid=Retirement ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need help
Yes, I have a snom 190. I'm gonna check out the dtmf signalling now. thank you for the quick responces. Tim - Original Message - From: Ian Pattison [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 9:44 AM Subject: Re: [Asterisk-Users] need help Is this a SIP phone? I had to upgrade the firmware on my SIP phones to alleviate this. It seems that the phone would actually disable it's own keypad after dialling. Ian [EMAIL PROTECTED] 29/04/2005 09:16 I am having an issue with the asterisk system not responding to dialed numbers during an active call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone config? and worse I don't even know what Keywords to search for. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recording in a call center
I would like to record two months of calls. The call center does not have a huge volume, probably like 60 calls a day and average about 15 min a call. I am using a quad port e1 card from digium. i would like to record the calls on a seperate server than the one running asterisk to avoid any problems. any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firefly Qualify Problem
Dear All, I'm using CVS-HEAD 06/04/05 with Realtime, and at present, its working fine generally. However, I'm facing a problem that I find it strange and would like to seek your kind advise. I'm using Firefly 1.9.8 build 3945 and I realise that when I set qualify to yes, then then Asterisk will qualify me as UNREACHEABLE. However, choosing not to qualify will work properly. Is there anyway I can resolve this? For some reason I cannot use IAX, so thats out. Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 SIP: [EMAIL PROTECTED] E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] vmail.cgi: -rwsr-sr-x as root *still* won't read the files
On Fri, Apr 29, 2005 at 12:23:48AM -0500, Brian Capouch wrote: Drat. Perl screams bloody murder if you try to just set its SUID bit, which of course is dangerous as hell. The perl-suid is *not* simply a version of perl with the suid bit set but rather a helper binary which allows perl to run suid scripts. Try it. mike -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-212-431-9090 (democracy now) tel:+1-646-382-7220 (cell) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with TDM400P card
Zttool shows nothing inside thebox. I tried removing the x100 cards, moving the tdm card around, disabled all usb and unnecessary stuff still, kudzu when booting up shows the card and the card shows up on /etc/sysconfig/hwconf but I wonder why it shows 2 of these and I only have 1 tdm400p card with 1 module class: MODEM bus: PCI detached: 1 driver: hisax desc: Tiger Jet Network Inc.|Intel 537 The above suggests the hisax driver is loaded for that card. According to what I see in man pages; that driver has something to do with isdn. I'd have to guess that because that driver is loaded, the zaptel drivers can't load. Sounds like another response you already received somewhat addresses the problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadbri bristuff ztcfg fail
Please can anyone help me with my quadbri card I am desparate L I compiled the bristuff drivers and then I do -- Modprobe zaptel Insmod qozap.ko Ztcfg The it complains it cant find ZT_SPANCONFIG failed on span 1: No such device or address (6) --- When doing lsmod I can see qozap is loaded with zaptel but no entry in /proc/zaptel/ My zaptel.conf -- loadzone=nl defaultzone=nl span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording in a call center
You need something like this ??
exten = _0.,1,SetVar(CALLFILENAME=${CALLERIDNUM}-${EXTEN}-${TIMESTAMP})
exten = _0.,2,Monitor(wav,${CALLFILENAME},m)
exten = _0.,3,Dial,SIP/[EMAIL PROTECTED]
and mount another server with NFS or SAMBA on /var/spool/asterisk/monitor
That would be the job.
Sjaak
I would like to record two months of calls. The call center does not
have a huge volume, probably like 60 calls a day and average about 15
min a call. I am using a quad port e1 card from digium. i would like
to record the calls on a seperate server than the one running asterisk
to avoid any problems.
any ideas?
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Re: [Asterisk-Users] Recording in a call center
60 calls a day is nothing. I'm sure your Asterisk box can handle it with the standard Monitor command. I've recorded many calls, 8+ hours straight and I'm on a crap old Pentium 3 633MHz system. What exactly do you fear will happen if you record on the Asterisk box? -- Dana On 4/29/05, Steve Totaro [EMAIL PROTECTED] wrote: I would like to record two months of calls. The call center does not have a huge volume, probably like 60 calls a day and average about 15 min a call. I am using a quad port e1 card from digium. i would like to record the calls on a seperate server than the one running asterisk to avoid any problems. any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip connection problems
http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-04-187A1.pdf http://www.wi-fiplanet.com/voip/article.php/3390671 http://www.cybertelecom.org/voip/Fcc.htm (scroll down) and of course: FCC To Require 911 for VoIP http://www.newsfactor.com/story.xhtml?story_id=33733 - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 1:54 AM Subject: Re: [Asterisk-Users] voip connection problems trixter http://www.0xdecafbad.com wrote: a couple weeks ago the FCC (america) ruled that all voip providers that connect to the PSTN (vonage, broadvoice, voicepulse, etc) have to have CALEA support (wiretap equipment for law enforcement). Failure to comply is a $10,000 fine per day. Could you please provide a reference for this assertion? Thx. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadbri bristuff ztcfg fail
smells like udev. Checkout README.udev in the zaptel directory. On 4/29/05, Sander [EMAIL PROTECTED] wrote: Please can anyone help me with my quadbri card I am desparate L I compiled the bristuff drivers and then I do -- Modprobe zaptel Insmod qozap.ko Ztcfg The it complains it can't find ZT_SPANCONFIG failed on span 1: No such device or address (6) --- When doing lsmod I can see qozap is loaded with zaptel but no entry in /proc/zaptel/ My zaptel.conf -- loadzone=nl defaultzone=nl span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki http://www.aefirion.org/ http://www.asterisk.com.pl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_zap graceful failure
I was wondering if anyone is working on graceful failure for chan_zap? Let me explain the situation. We are using a T100P and TDM400P (4 FXS for fax). There was a major power outage and asterisk went down after the UPS (not a graceful shutdown -- my fault, no apcupsd running). As soon as power came back, the server started. However when it loaded wcfxs, port 3 on the card failed the tests (I assume from the module not being unloaded before power off). Because this one port failed the test, chan_zap failed to load and asterisk will not start. An unload and load of modules would not fix it. I had to unload and restart (a clean restart). While the unclean shutdown can be controlled in the future, I have had ports go bad and when they do asterisk will not start until the offending lines are removed from zapata.conf. This is not a very resilient solution (especially if you are not on site). I would much prefer for asterisk to keep running with what it has got. I will be looking into the code (and this might be fixed in cvs-head), but I would like to start a discussion on this first. Thanks, Jeb -- Jeb Campbell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web interface Suggestions
I think you will find AMP is about to implement a multi tenant solution. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Niemantsverdriet Sent: Friday, April 29, 2005 1:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Web interface Suggestions Open Source project I assume. I am interested in this project do you have a webpage about it? Thanks, _ /-\ ndrew On 4/28/05, G.Marshall [EMAIL PROTECTED] wrote: Has anyone come across any software that can control adding/editing SIP extension properties and perhaps dial plan properties on a context basis. What I mean is I would like it so an admin user from Company A can manipulate properties for extensions in his context but not in another Companies. I know AMP does something similar to this but from what I understand it does not allow for different users at different companies to control only things that pertain to them. In my spare time, I am developing a php webfrontend to realtime asterisk database which modifies dialplan, users etc. Should not be too difficult to add a login facility which means the user can see their own context only. Regards, Spencer --- https://www.dalmany.co.uk/dundi/dundi.php https://www.dalmany.co.uk/asterisk/index.php ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager interface, setting global vars
Hi all, Does anyone know of a way to setup global var using the manager interface. Basically I want to be able to have multiple manager clients login, however in a sort of master slave scenario. So the first client that logs in, sets a global variable which tells subsequent clients at least one client is already logged in. The Master would then set additional variables which the slaves would periodically read. Is this possible ? Thanks in advance for any help. Umar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank of E1s? (one E1 input -- 2 x E1 output)
Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadbri bristuff ztcfg fail
No udev installed on my system :( so that does not help me Thanks anyway -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Michael Bielicki Verzonden: vrijdag 29 april 2005 17:18 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] quadbri bristuff ztcfg fail smells like udev. Checkout README.udev in the zaptel directory. On 4/29/05, Sander [EMAIL PROTECTED] wrote: Please can anyone help me with my quadbri card I am desparate L I compiled the bristuff drivers and then I do -- Modprobe zaptel Insmod qozap.ko Ztcfg The it complains it can't find ZT_SPANCONFIG failed on span 1: No such device or address (6) --- When doing lsmod I can see qozap is loaded with zaptel but no entry in /proc/zaptel/ My zaptel.conf -- loadzone=nl defaultzone=nl span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki http://www.aefirion.org/ http://www.asterisk.com.pl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording in a call center
It is a critical system and located overseas with no technical people onsite. Logic dictates that changes be made with a light footprint. - Original Message - From: Dana Olson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 11:12 AM Subject: Re: [Asterisk-Users] Recording in a call center 60 calls a day is nothing. I'm sure your Asterisk box can handle it with the standard Monitor command. I've recorded many calls, 8+ hours straight and I'm on a crap old Pentium 3 633MHz system. What exactly do you fear will happen if you record on the Asterisk box? -- Dana On 4/29/05, Steve Totaro [EMAIL PROTECTED] wrote: I would like to record two months of calls. The call center does not have a huge volume, probably like 60 calls a day and average about 15 min a call. I am using a quad port e1 card from digium. i would like to record the calls on a seperate server than the one running asterisk to avoid any problems. any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] need help
Thank you, For the responces i had dtmfmode=inband when rcf2833 was the proper setting. I feel retarded that i missed that, but it happens. thanks again Tim Touhsaent - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, April 29, 2005 9:38 AM Subject: Re: [Asterisk-Users] need help This is a DTMF issue, You must adjust this on the especific channel conf file. For example, ia sip phone cannot dial any number during an active call, you must see sip.conf and the config in your hardphone or softphone. Ismael. "Tim Touhsaent" [EMAIL PROTECTED] Enviado por: [EMAIL PROTECTED] 04/29/2005 03:16 PM Por favor, responda aAsterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Para "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com cc Asunto [Asterisk-Users] need help I am having an issue with the asterisk system not responding to dialednumbers during an activecall. I'm not even sure where to look, zapata.conf? sip.conf? or the phoneconfig? and worse Idon't even know what Keywords to search for.Tim___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *@home 0.9 and AVM B1 Card
Hi all I have installed version 0.9 against a supplier SIP and have put a AVM B1 like backup. The reception of calls works perfectly, but with himself not to make calls to ISDN card. It is possible that this configuration works whith [EMAIL PROTECTED] ? or I am mistaken. AVM card is CAPI and * @home can create ZAP channels. How I create a CAPI channel? Excuse me for my very poor english :-( thanks in advance ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_zap graceful failure
On April 29, 2005 11:22 am, Jeb Campbell wrote:
As soon as power came back, the server started. However when it loaded
wcfxs, port 3 on the card failed the tests (I assume from the module not
being unloaded before power off). Because this one port failed the
test, chan_zap failed to load and asterisk will not start.
It has nothing to do with not being unloaded; I've seen the wctdm driver fail
to detect modules correctly. Run it again and it works just fine. Some kind
of minor tweak is in order, I believe.
While the unclean shutdown can be controlled in the future, I have had
ports go bad and when they do asterisk will not start until the
offending lines are removed from zapata.conf. This is not a very
resilient solution (especially if you are not on site). I would much
prefer for asterisk to keep running with what it has got.
As an interim solution, your asterisk starup script should try to unload any
modules and reload them upon asterisk failure... preferably in a loop:
while(1) {
unload modules
sleep 1
load modules
start asterisk
sleep 5
}
I imagine at this point in time your startup script either does not loop, or
it doesn't try to unload/load the modules inside the loop.
-A.
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Re: [Asterisk-Users] Realtime feature
Rodrigo P. Telles wrote: Does someone knows if the next release of Asterisk (1.0.8?) will have Realtime support and when we will have the next Asterisk release with Realtime features? Where is your failure? I don't see anything. The next stable release of asterisk will be 1.2 and it will have RealTime. The current CVS-HEAD (aka 1.1) has it now and it is very stable. The eta on 1.2 is unknown. You can help 1.2 along by downloading it and running it to help fix bugs. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on VMWare ESX/blade servers
I am running on usermodelinux Itamar Reis Peixoto +55 (34) 3238 3845 e-mail : [EMAIL PROTECTED] http://vps.ispbrasil.com.br --- servidores linux Has anyone had any experience (good or bad) running Asterisk under VMWare ESX server on a blade chassis? This application will (fairly obviously) not include Zap channelsactually, it will be SIP-only. Please feel free to contact me off-list and I'll summarize for the list later. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank of E1s? (one E1 input -- 2 x E1 output)
yes, some multiplexer allows that, but they're quite expensive compared to another E1 card for asterisk. I think you'll need at least 1k $$$ for a such splitter. Matteo. Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto: Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] the beginning of voice menu is cutted
Hi, when I dial my voicemenu the menu voice is always cutted so that i only hear 'word from password. What do i have to configure so that i hear the full text from the beginning? thanks, Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] the beginning of voice menu is cutted
On Friday 29 April 2005 12:12 pm, Kib Eki wrote:
Hi,
when I dial my voicemenu the menu voice is always cutted so that i only
hear 'word from password.
What do i have to configure so that i hear the full text from the
beginning?
thanks, Kib
You might try inserting a Wait in your menu ...e.g...
exten = s,1,Answer ; answer the channel
exten = s,n,Wait(2) ; give the channel time to initalize (2seconds)
exten = s,n,Background(some-recording)
The 'Wait' supposedly gives the channel time to 'initalize' and get ready to
send audio. If you start dumping audio ('Background') down a channel not
initalized, you wont hear anything until the channel is initalized, even if
the audio has already started.
At least, thats my non-developer-ish understanding of the sequence of events
after having the same problem myself...
HTH,
-josiah
--
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [Asterisk-Users] Recording in a call center
Wouldn't introducing Samba into the mix be even worse?
I would think it would add more processing power and network use to be
constantly writing over the network as opposed to recording on the
same box.
If it's such a critical system, it should have the power to do that,
but that's not the point... If I had such a critical system, I'm not
so sure that I would be saving files in real-time over the network via
Samba.
My question is, what's the difference between writing to the local
disk and over the network? What will happen if the network link goes
down? I've had bad experiences with Samba and NFS both, as far as
connectivity issue handling is concerned.
--
Dana
On 4/29/05, sjaak imap [EMAIL PROTECTED] wrote:
You need something like this ??
exten = _0.,1,SetVar(CALLFILENAME=${CALLERIDNUM}-${EXTEN}-${TIMESTAMP})
exten = _0.,2,Monitor(wav,${CALLFILENAME},m)
exten = _0.,3,Dial,SIP/[EMAIL PROTECTED]
and mount another server with NFS or SAMBA on /var/spool/asterisk/monitor
That would be the job.
Sjaak
I would like to record two months of calls. The call center does not
have a huge volume, probably like 60 calls a day and average about 15
min a call. I am using a quad port e1 card from digium. i would like
to record the calls on a seperate server than the one running asterisk
to avoid any problems.
any ideas?
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Re: [Asterisk-Users] Asterisk Hardware Recommendation
Sure. I setup a small lab on a machine with 4 T1s and 36 agents logged in. The system was configured to Monitor all outbound calls as well as monitor all calls distributed by Queue app (monitor-format setting in queues.conf). When recording to local disk, everything was working fine. Agents were busy 99.5% and there were at least 30 calls waiting in Queue to be distributed. Average call conversation length was about 7.5 minutes. Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E. The moment we pushed the load on the Asterisk machine, everything worked for about 40 seconds. Then call quality started suffering significantly. Chopped audio. Bad audio. No audio. Good audio. You could imagine. So we stopped the test. Then we unmounted the NFS drive and repeated the test again. Everything worked fine again. The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. During all tests, CPU utilization was about 55% on the average (for each CPU). Memory usage was under 1G. I would say I need to try more troubleshooting. Maybe there was congestion on the Fast-E, although preliminary analysis indicates there were no CRC errors, collisions, or packet loss. The NFS machine was completely idle. Last, we repeated the test over a 1 hour period. This time, Monitor was recording on local drives and we were copying files every 15 minutes with a background process (perl script) to NFS mount point. Everything worked fine as well. I don't know if these tests are conclusive yet. However, from the results so far, I would recommend staying away from recording to NFS mounted point. I will continue running simulations to see if anything else can be identified. Thanks, Daniel On Apr 28, 2005, at 7:26 PM, Matt Roth wrote: Daniel, Could you expand upon your experience recording to an NFS mounted drive. We are looking to use a TDM-VoIP gateway to route 16+ spans to a single Asterisk server. We were hoping to Monitor using the following scheme: - Monitor application executed on Asterisk server (no 'm' flag) - Pick a codec that the Monitor application can record in natively so that no transcoding is done on the leg files (can this be done?) - Record the leg files to an NFS mounted drive on a remote machine - Have soxmix take care of mixing and transcoding the leg files into the desired format on the remote machine According to you this now looks like a VERY BAD IDEA. Does anyone out there have any experience using monitor to digitally record large numbers of spans (16+) on a single Asterisk server using a VoIP gateway instead of TDM cards? Is it feasible? We are trying to keep the Asterisk server as slim as possible, but would like to stick to one box so that we can have centralized queuing, configuration, and reporting. Has anyone had any luck using Monitor to record to an NFS mounted drive? Are there any other options to remove the overhead of the disk subsystem when recording calls? Thanks, Matthew Roth http://voip-info.org/tiki-index.php? page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Thanks, Daniel On Apr 28, 2005, at 6:42 PM, mattf wrote: I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA drives, with a nice big archive server that the audio will be copied to several times a day. Also, do not record(Monitor) with the 'm' flag on because this will also lead to more disk read-write while you are already trying to write another 100 or so streams. Offload the -in and -out files to the archive server and let it soxmix them together instead. This is the method that we have settled on for our 12 Asterisk servers and it works rather well for us. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Hardware Recommendation Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full
Re: [Asterisk-Users] chan_zap graceful failure
Andrew Kohlsmith wrote:
It has nothing to do with not being unloaded; I've seen the wctdm driver fail
to detect modules correctly. Run it again and it works just fine. Some kind
of minor tweak is in order, I believe.
As an interim solution, your asterisk starup script should try to unload any
modules and reload them upon asterisk failure... preferably in a loop:
while(1) {
unload modules
sleep 1
load modules
start asterisk
sleep 5
}
I imagine at this point in time your startup script either does not loop, or
it doesn't try to unload/load the modules inside the loop.
While I like the idea (and will look into it -- might need a wait, etc),
as I said in original post, unloading and reloading did not fix the
problem. It took a clean shutdown (unload and restart) to fix the problem.
So regardless of why the card has failed, I would like to discuss making
chan_zap fail gracefully. For example if you have a
Dial(Zap/3/${NUMBER}, and Zap/3 does not exist, asterisk will spit a
warning (not fail to startup). However if you have that channel = 3 in
zapata.conf, chan_zap will fail and prevent asterisk from starting.
I would think that everyone would prefer asterisk to start and have
parts of the dialplan fail, rather than have asterisk not load at all.
As I said, I have not checked the behavior of cvs-head, I just wanted to
discuss making asterisk more resilient.
Thanks for the tip and I will look into it.
Jeb
--
Jeb Campbell
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk Hardware Recommendation
Well, I don't think I'm ready to spend that much money :) I understand your point regarding that load depends on usage. SIP_Agents are simply agents answering calls. Average call length would be about 8 minutes. During some of these calls (maybe 25%), agents will conference the call (PSTN channel) with internal IVR script. I like Scenario 6. Will look into that further. However, if the above information gives you more grounds to make additional comments, please do so :) Thanks, Daniel On Apr 29, 2005, at 10:21 AM, mattf wrote: If price would truly not an option just get one of the Signate Telephony 5000 servers(http://www.signate.com/pbx.php) They are about $18,000 and allow you to have upto 5000 SIP streams go through it. You could have that be your gateway and do the SIP-IAX through that machine and scale upto 100 T1s if you want. But that is a bit steep. So on to your choices. I would really say that the setup you choose will depend on what kind of users you have as well as how often you need to change/add users to the system and how the users are using the system at what times. Any of them that you listed could work depending on how they are used, but in some cases you may not want to use some of the scenarios listed because they would either be incapable of meeting your needs or overly complex to manage. The easiest and cheapest one would actually not be listed: Scenario 6: Direct SIP-Zap on two separate servers half SIP users on each server PSTN --2xT1-- A1 SIP_Agents PSTN --2xT1-- A2 SIP_Agents There is really no reason to have another 2 servers running IAX to the T1 servers, and this is simple and easy to set up and involves only 2 machines. The next setup I would recommend would be Scenario 4, although you will have to get a machine with a fast/wide BUS(like an Apple G5) to handle ever increasing numbers of SIP-IAX streams as the system would grow. If you can explain more about what kind of use this system will have I can give a better recommendation. MATT--- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with MusicOnHold
Greetings, I have two machines. One is a P3 Dell Dimension 4100, the other is a PowerEdge SC420. Both are running Asterisk 1.0.7, the PowerEdge has a TE405P card in it, the Dimension has a Digium X100P present (although not modprobed). Each machine has mpg123 0.59r loaded, and is using the exact same set of MP3s for music on hold (both the distributed ones and some of our own). Neither box is sharing any interrupts. I use the same 7960G to test the Music on Hold. On the Dimension 4100, MusicOnHold works flawlessly. No static, no glitches, nothing. On the PowerEdge SC420, MusicOnHold has a lot of static, pops, crackles, and almost everything you can imagine. I can't think of anything else that is applicable. Basically, the machines seem pretty much identical to me. I expected MoH to work the same as well, but it isn't. If anyone has any ideas, please let me know. Thanks, Nathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip endpoints that support re-invite??
Hi, I am doing some testing with asterisk using Cisco IP Phones 7960's and EyeBeam. I have canreinvite=yes on all my devices but the media still goes through the asterisk box. Does it mean that Cisco and Xten do not support re-invites? If yes can you recommend SIP phones or adapters that support re-invites. Thanks in advance. Hamza Moore. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_zap graceful failure
On April 29, 2005 12:38 pm, Jeb Campbell wrote:
While I like the idea (and will look into it -- might need a wait, etc),
as I said in original post, unloading and reloading did not fix the
problem. It took a clean shutdown (unload and restart) to fix the problem.
Hmm; that is odd...
So regardless of why the card has failed, I would like to discuss making
chan_zap fail gracefully. For example if you have a
Dial(Zap/3/${NUMBER}, and Zap/3 does not exist, asterisk will spit a
warning (not fail to startup). However if you have that channel = 3 in
zapata.conf, chan_zap will fail and prevent asterisk from starting.
I would think that everyone would prefer asterisk to start and have
parts of the dialplan fail, rather than have asterisk not load at all.
No; if the driver didn't load that's a major problem. Remember that if the
channel doesn't exist all the subsequent channels move up... serious
potential security issues.
I'd rather have the system as it is, where it fails out with an error that is
easy to understand so I can fix the problem.
-A.
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[Asterisk-Users] IAX2 one way audio
Upgraded one of my asterisk servers to the latest cvs head version last nigh now I get one way audio on IAX2 channels when calling other asterisk servers. Anyone seeing the some problems? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] User events - a dumb question
Ok, this is probably stupid question of the week. I have exten = 888,1,whatever exten = 888,n,UserEvent(Event|Data) exten = 888,n,Hangup If I asterisk -r, when I dial the 888, I see Userevent appearing in the console. However, if I telnet to the * manager using a name and password that has the user option, that telnet session sees everything but the user event. What am I missing ? manager.conf: [event] secret=event read=system,user write=call,command,agent Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Manager interface, setting global vars
There isn't a specific command in the manager API itself to do it. However there is a CLI command and you can use the manager command action to get the information. Below is an example, you will need to parse the response part to see who is connected. Action: Command Command: show manager connected Response: Follows Username IP Address something127.0.0.1 As far as I know, there isn't a way to modify or look at the global variables directly. You could make a kludge that would call to a special extension that runs NoOp or something that can be seen from an Event, but thats not going to be fun. --johann Umar Sear wrote: Hi all, Does anyone know of a way to setup global var using the manager interface. Basically I want to be able to have multiple manager clients login, however in a sort of master slave scenario. So the first client that logs in, sets a global variable which tells subsequent clients at least one client is already logged in. The Master would then set additional variables which the slaves would periodically read. Is this possible ? Thanks in advance for any help. Umar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Daniel, Thanks alot for this post. You were right on time with valuable information. Thanks again, Steve - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 12:37 PM Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation Sure. I setup a small lab on a machine with 4 T1s and 36 agents logged in. The system was configured to Monitor all outbound calls as well as monitor all calls distributed by Queue app (monitor-format setting in queues.conf). When recording to local disk, everything was working fine. Agents were busy 99.5% and there were at least 30 calls waiting in Queue to be distributed. Average call conversation length was about 7.5 minutes. Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E. The moment we pushed the load on the Asterisk machine, everything worked for about 40 seconds. Then call quality started suffering significantly. Chopped audio. Bad audio. No audio. Good audio. You could imagine. So we stopped the test. Then we unmounted the NFS drive and repeated the test again. Everything worked fine again. The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. During all tests, CPU utilization was about 55% on the average (for each CPU). Memory usage was under 1G. I would say I need to try more troubleshooting. Maybe there was congestion on the Fast-E, although preliminary analysis indicates there were no CRC errors, collisions, or packet loss. The NFS machine was completely idle. Last, we repeated the test over a 1 hour period. This time, Monitor was recording on local drives and we were copying files every 15 minutes with a background process (perl script) to NFS mount point. Everything worked fine as well. I don't know if these tests are conclusive yet. However, from the results so far, I would recommend staying away from recording to NFS mounted point. I will continue running simulations to see if anything else can be identified. Thanks, Daniel On Apr 28, 2005, at 7:26 PM, Matt Roth wrote: Daniel, Could you expand upon your experience recording to an NFS mounted drive. We are looking to use a TDM-VoIP gateway to route 16+ spans to a single Asterisk server. We were hoping to Monitor using the following scheme: - Monitor application executed on Asterisk server (no 'm' flag) - Pick a codec that the Monitor application can record in natively so that no transcoding is done on the leg files (can this be done?) - Record the leg files to an NFS mounted drive on a remote machine - Have soxmix take care of mixing and transcoding the leg files into the desired format on the remote machine According to you this now looks like a VERY BAD IDEA. Does anyone out there have any experience using monitor to digitally record large numbers of spans (16+) on a single Asterisk server using a VoIP gateway instead of TDM cards? Is it feasible? We are trying to keep the Asterisk server as slim as possible, but would like to stick to one box so that we can have centralized queuing, configuration, and reporting. Has anyone had any luck using Monitor to record to an NFS mounted drive? Are there any other options to remove the overhead of the disk subsystem when recording calls? Thanks, Matthew Roth http://voip-info.org/tiki-index.php? page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Thanks, Daniel On Apr 28, 2005, at 6:42 PM, mattf wrote: I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA drives, with a nice big archive server that the audio will be copied to several times a day. Also, do not record(Monitor) with the 'm' flag on because this will also lead to more disk read-write while you are already trying to write another 100 or so streams. Offload the -in and -out files to the archive server and let it soxmix them together instead. This is the method that we have settled on for our 12 Asterisk servers and it works rather well for us. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 5:56 PM To:
Re: [Asterisk-Users] IAX2 one way audio
Do you get 2-way audio that sometimes drops off to 1-way audio then picks back up as 2-way? (Thats what I see) Not sure if my problem is a lost packet issue as I am sending IAX off net. Duane Cox - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, April 29, 2005 12:03 PM Subject: [Asterisk-Users] IAX2 one way audio Upgraded one of my asterisk servers to the latest cvs head version last nigh now I get one way audio on IAX2 channels when calling other asterisk servers. Anyone seeing the some problems? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI
David Josephson, Not off-base, but you haven't made it all the way home yet. This is another layer of the puzzle, and again we are not talking about apples and apples here. Circuit switched means that there is a (real or virtual) circuit that takes data on an input port and delivers it to an output port somewhere. Packet switched means that each packet of data is examined by each port it passes, to see where it should be sent. Normally this layer of VoIP traffic is handled not in Asterisk, but in a router. You could run the router on the same Linux box that's running Asterisk (and send packets to different Ethernet ports depending on their destination address) but normally this task is handled by a separate router. There is a small computational overhead associated with adding and decoding Ethernet packets but the main routing work is done outside Asterisk, and isn't too intensive. You could read up on TCP/IP routing and understand how this works in more detail. We plan on using a Gb switch with 100 Mbps ports to handle the routing. It's not something you can take a look at in my experience. Some of the Bell System training material that comes up on eBay is good. You need to follow the progress from circuit-switched voice telephony circa 1930 through modern TDM, and then look at the development of TCP/IP switching separately. 75 years of telephony and network technology to cover, eh? Looks like it's going to be a long weekend. ; ) No sound card, no monitor. Recording to the various file formats is possible, as Herman mentioned. This seems like an odd limitation to me. Any idea why it's designed so that you must have a sound card to digitally record calls? They could always be moved to another box in order to listen to them. Your reference picture is fine ... but note that Asterisk can be the TDM/VoIP gateway, particularly when Digium releases their DS3 card (644 voice channels!) working, a lot more cheaply than a standalone box from some hardware vendor. I'm not sure that the DS3000P is in our timeframe. I am interested in knowing how it will perform, considering more than two Digium quad-span cards currently overload the CPU with interrupts. It seems that Monitor cannot handle digitally recording more than ~50 concurrent calls, either. Maybe these limitations are being addressed as we speak. Thank you for sharing your knowledge with me, Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call a peer over the asterisk manager with a php script
Guy Boehm wrote: wau thank you it works!! but, first it says that e loop is detected, and secondary what must I do to hand over the new working channel to my x-lite to use it??? DENGENS Richard Lyman [EMAIL PROTECTED] wrote: Guy Boehm wrote: fputs($socket, Channel: 6159bfb47b9\r\n\r\n); Response: Error Message: Invalid channel the Channel: var needs to be in the form of type/dev/numbertocall like Channel: IAX2/user:[EMAIL PROTECTED]/14085551212 i have no clue what you meant by 'e loop', as for handing over the call.. i think you really need to read the handbook and get a base knowledge of what asterisk is and how it works. without that, you would be in here 5 times aday asking questions and probably getting flamed like crazy. fire up a brower and goto www.digium.com click the documentation link on the left side. there is a getting started section, read the FAQ there is a reference doc section, read the asterisk project handbook, version 2 there is also get http://www.digium.com/handbook-draft.pdf good luck ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Traffic Testing
I'm using sip-tester you should try it gnuws:~# apt-cache search sip-tester sip-tester - a performance testing tool for the SIP protocol gnuws:~# On Fri, 2005-04-29 at 14:14 +0200, Nils Ohlmeier wrote: The homepage http://sipsak.org contains some examples. If you need help with special cases drop me a line. Regards Nils Ohlmeier On Friday 29 April 2005 02:54, Anton Krall wrote: Can you send some command line examples on how to use it? Thx! |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |[EMAIL PROTECTED] |Sent: Jueves, 28 de Abril de 2005 07:05 p.m. |To: asterisk-users@lists.digium.com |Subject: RE: [Asterisk-Users] Traffic Testing | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] Behalf Of Anton | Krall | Sent: Thursday, April 28, 2005 6:07 PM | To: 'Asterisk Users Mailing List - Non-Commercial Discussion' | Subject: [Asterisk-Users] Traffic Testing | | | Guys, is there any way to generate simulated traffic via sip or IAX2 | for testing cpu load and asterisk? (sip client simulation, etc)? | |yes, use sipsak utility | |-- |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- René Mayorga Internet Data El Salvador Telecom S.A. de S.V. Tel:(503) 247-7246 (503) 247-7156 Cel:(503) 962-8205 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
