[Freeswitch-users] Call limit in freeswitch?
Hi all, I am testing freeswitch with sipp calls, but freeswitch is accepting 70 to 80 calls only, no more calls are processed at all. Is there any configuration settings behind this, that limits calls? Is there any settings or parameters for call limit in freeswitch, so we can extend the call limits. Thanks msp -- View this message in context: http://www.nabble.com/Call-limit-in-freeswitch--tp19809553p19809553.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call limit in freeswitch?
Check switch.conf.xml there are some limits in there for both concurrent sessions and sessions per second... You'll need to adjust those up... Or you can look at fsctl from the cli and you can adjust them on the fly from there From: shehzad p [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Fri, 3 Oct 2008 23:45:20 -0700 (PDT) To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Call limit in freeswitch? Hi all, I am testing freeswitch with sipp calls, but freeswitch is accepting 70 to 80 calls only, no more calls are processed at all. Is there any configuration settings behind this, that limits calls? Is there any settings or parameters for call limit in freeswitch, so we can extend the call limits. Thanks msp -- View this message in context: http://www.nabble.com/Call-limit-in-freeswitch--tp19809553p19809553.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newbie Questions
Another way would be to export to a Radius server using a MySQL backend. I am not using the feature so I can not report how well it works but I am confident that it should just work and that if it does not, it will be fixed :) Thomas Vito Andolini wrote: Michael, Thanks a lot for the answers, it answers my questions, I guess the next thing would be downloading and trying FS :) Also, as I saw in the wiki, there is no built in support for mysql right? The solution that was suggested was setting up a cron to import the csv files into mysql, is this still what's available or are there any updates on this? Thank, Vito A. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Michael Jerris *Sent:* Friday, October 03, 2008 7:06 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Newbie Questions On Oct 3, 2008, at 4:54 AM, Vito Andolini wrote: Hi All, I am familiar with Asterisk and doing some testing for my next project. I have had some difficulties Asterisk, and now researching FreeSwitch hoping that it has some out-of-the box answers for my questions. Basically I want to implement a Click 2 Call service. Very simple, user types in his/her number on a website, that number is called, after it's answered, the company number is called and they are bridged together. 1) Is there a way to communicate with FreeSwitch programatically and issue commands such as initiate calls etc ? (ver much like manager API in Asterisk) There is an interface that we call the fsapi interface that can be accessed in many ways, including over a socket method similar to a combination of AMI and FAGI: http://wiki.freeswitch.org/wiki/Event_Socket and xmlrpc: http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC 2) If you initiate a call from the software and then once its answered call a 2nd number. How do you bridge them? You can do this all in one command: http://wiki.freeswitch.org/wiki/Mod_commands#originate 3) After the 2 numbers talk and hang up. How does your cdr look like? Do you have 2 cdr's that correspond to both calls or just 1 after both numbers are bridged together? This is one of the problems I can't solve with Asterisk as it generates only 1 cdr after the 2 calss are bridged. The reason for this request is, in case of a Click2Call service, you are charged for both calls by your SIP provider therefore you need to be able to track both calls for invoices/payments etc. We can do either per leg or combined cdr's http://wiki.freeswitch.org/wiki/Channel_Variables#process_cdr We have multiple supported formats for cdr: http://wiki.freeswitch.org/wiki/Mod_xml_cdr http://wiki.freeswitch.org/wiki/Mod_cdr_csv 4) Is there a way to programatically know if a call has been asnwered or not and act based upon that. I understand the cdr contains that information. But what I want is, if the call is not answered maybe I can play a prerecorded message or take them to the voicemail or whatnot. So I need a way to do a flow-control based on if the call has been asnwered or not in the dialplan. Does that exist? If so can you point me to some resources? There are several approaches you could take to this. You could do this all in dialplan if there is not any real forking other than if the call worked. You can use the variables: http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail and just playing the sounds in the dial-plan after a bridge line. Mike No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.173 / Virus Database: 270.7.5/1702 - Release Date: 10/2/2008 9:35 PM ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call limit in freeswitch?
Thank Rice, In switch.conf.xml, default values is 1000 maximum session and 30 session per seconds, I've not changed that. Eventhough, freeswitch is accepting 70 to 80 calls only, after that no more sip calls are processed. Further i stop sipp calls and then start the same sipp test calls, again it happen same as above. Is there any other configuration settings that might affect call limit? Thanks again msp. Ken Rice-2 wrote: Check switch.conf.xml there are some limits in there for both concurrent sessions and sessions per second... You'll need to adjust those up... Or you can look at fsctl from the cli and you can adjust them on the fly from there From: shehzad p [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Fri, 3 Oct 2008 23:45:20 -0700 (PDT) To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Call limit in freeswitch? Hi all, I am testing freeswitch with sipp calls, but freeswitch is accepting 70 to 80 calls only, no more calls are processed at all. Is there any configuration settings behind this, that limits calls? Is there any settings or parameters for call limit in freeswitch, so we can extend the call limits. Thanks msp -- View this message in context: http://www.nabble.com/Call-limit-in-freeswitch--tp19809553p19809553.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Call-limit-in-freeswitch--tp19809553p19810121.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call limit in freeswitch?
There are several things that can affect whats going on... From amount of ram in the machine, to 64 vs 32 bit, hdd access times (assuming you are using the default or close to default configs)... A PentiumD, 3Ghz Machine, w/ 2G ram is capable of doing 200 concurrent calls processing media and much more then that if in no media mode, dual quad core machines are capable of handling thousands of calls From: shehzad p [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Sat, 4 Oct 2008 01:36:31 -0700 (PDT) To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Call limit in freeswitch? Thank Rice, In switch.conf.xml, default values is 1000 maximum session and 30 session per seconds, I've not changed that. Eventhough, freeswitch is accepting 70 to 80 calls only, after that no more sip calls are processed. Further i stop sipp calls and then start the same sipp test calls, again it happen same as above. Is there any other configuration settings that might affect call limit? Thanks again msp. Ken Rice-2 wrote: Check switch.conf.xml there are some limits in there for both concurrent sessions and sessions per second... You'll need to adjust those up... Or you can look at fsctl from the cli and you can adjust them on the fly from there From: shehzad p [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Fri, 3 Oct 2008 23:45:20 -0700 (PDT) To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Call limit in freeswitch? Hi all, I am testing freeswitch with sipp calls, but freeswitch is accepting 70 to 80 calls only, no more calls are processed at all. Is there any configuration settings behind this, that limits calls? Is there any settings or parameters for call limit in freeswitch, so we can extend the call limits. Thanks msp -- View this message in context: http://www.nabble.com/Call-limit-in-freeswitch--tp19809553p19809553.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Call-limit-in-freeswitch--tp19809553p19810121.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load test - performance not even matching Asterisk
Hi all An update on the performance measurements: The measurements I have referred to earlier all involved an Asterisk as the call generator. Somehow this setup leads to extensive rtp bandwidth usage. Each channel used around 500 kbps. If a phone is entered into the loop, this is reduced to the expected 64 kbps. I have not found any reason for this, but it certainly fouls up the test, and I have changed the test setup. Further, and since the earlier tests, the network has been updated to a Gbits network. I have now made two new test: 1) Using WinSIP from Touchstone as a call generator. 2) Using the Asterisk as one component, and setting up a chain of calls which goes forth and back from the Asterisk and the FS. All call are started from a real phone, and after 100 loops, where the calls are answered and sent on by the dial plan, the calls are terminated by an tone (action application=gentones data=%(50,0,400)/) in the FS. The two test show similar top-figures at similar loads. The first test would be my preferable, but it is limited to 50 calls due to the trial licence limitations. Using an external non-FS and non-Asterisk device will eliminate some uncertainties, that's why it would be preferred. The other test has been done with 600, 400 and 200 channels (300, 200 and 100 calls), and the results of the top command are: cpu sy ni id wa hi si total * 600 10 30 0 33 0 2 25 100 FS600 22 33 0 30 0 0 15 100 0 * 400 7 18 0 67 0 1 7 100 FS400 14 17 0 62 0 0 7 100 0 * 200 3 10 0 84 1 0 2 100 FS200 7 8 0 82 1 0 2 100 The results do not show significant differences between the capacity behaviour of the Asterisk (*) and the FS. The also show an expected interrupt load (si) proportional to the square of the call load. Still the FS does not really outperform the Asterisk - which I find disappointing. Any comments are welcome. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load test - performance not even matching Asterisk
Try using something like SIPP for load testing you can load test to much higher numbers http://www.freeswitch.org/eg/load_test.tgz is what we use for testing so you can duplicate the results... Also, look at the configuration you are doing and determine if you really need all the features that are there... Things like presence tracking, certain CDR loggers, and a few other things under high CPS loads can cause more problems then you think... Hint... Mount freeswitch/db as a ram drive in linux this is a big performance booster (since it takes the load of sqlite of the hdd), also turn off presence tracking on all sip profiles that don't need it... Something doesn't sound right on the 500k of rtp... Also remember that asterisk RTP stack doesn't handle async rtp.. It depends on receiving a packet to transmit a packet From: Jon Bruel [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Sat, 4 Oct 2008 11:03:40 +0200 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Load test - performance not even matching Asterisk Hi all An update on the performance measurements: The measurements I have referred to earlier all involved an Asterisk as the call generator. Somehow this setup leads to extensive rtp bandwidth usage. Each channel used around 500 kbps. If a phone is entered into the loop, this is reduced to the expected 64 kbps. I have not found any reason for this, but it certainly fouls up the test, and I have changed the test setup. Further, and since the earlier tests, the network has been updated to a Gbits network. I have now made two new test: 1) Using WinSIP from Touchstone as a call generator. 2) Using the Asterisk as one component, and setting up a chain of calls which goes forth and back from the Asterisk and the FS. All call are started from a real phone, and after 100 loops, where the calls are answered and sent on by the dial plan, the calls are terminated by an tone (action application=gentones data=%(50,0,400)/) in the FS. The two test show similar top-figures at similar loads. The first test would be my preferable, but it is limited to 50 calls due to the trial licence limitations. Using an external non-FS and non-Asterisk device will eliminate some uncertainties, that's why it would be preferred. The other test has been done with 600, 400 and 200 channels (300, 200 and 100 calls), and the results of the top command are: cpu sy ni id wa hi si total * 600 10 30 0 33 0 2 25 100 FS600 22 33 0 30 0 0 15 100 0 * 400 7 18 0 67 0 1 7 100 FS400 14 17 0 62 0 0 7 100 0 * 200 3 10 0 84 1 0 2 100 FS200 7 8 0 82 1 0 2 100 The results do not show significant differences between the capacity behaviour of the Asterisk (*) and the FS. The also show an expected interrupt load (si) proportional to the square of the call load. Still the FS does not really outperform the Asterisk - which I find disappointing. Any comments are welcome. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] fring
Finally an iPhone app that connects with SIP to FreeSWITCH. I tried it out, and it works perfect with my FreeSWITCH installation. Can be downloaded at iTunes App Store. Ivan ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GotoIf
On Sat, Oct 4, 2008 at 5:21 AM, henkoegema [EMAIL PROTECTED]wrote:
I discovered some small errors in previous threads concerning this item.
(transfer i.s.o bridge)
To conclude:
Asterisk:
---
exten = s,n,GotoIf($[${CALLERID(num)}=32476478861]?default,1000,1)
FS:
extension name=Henk GSM
condition field=caller_id_number expression=^32476378861$
action application=bridge
data=sofia/internal/1000%$${domain}/
/condition
/extension
Remember that this analogy is wrong. Asterisk's dialplan runs a goto when
the expression is true, while FS' originates a call and bridge the two
parties.
Henk
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Arnaldo M Pereira
[EMAIL PROTECTED]
http://www.arnaldopereira.com
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Re: [Freeswitch-users] fring
On 10/4/2008 4:47 AM, Ivan C Myrvold wrote: Finally an iPhone app that connects with SIP to FreeSWITCH. I tried it out, and it works perfect with my FreeSWITCH installation. Can be downloaded at iTunes App Store. Ivan Ivan, I've been playing with fring. The only issue I have is that I can't get the registered fring client to hit anything but the 'public' dialplan context. I have: variables variable name=user_context value=default/ /variables set on the userid I'm registering fring with. Have you been able to get it to hit your default dialplan context? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] fring
On 4. okt.. 2008, at 14:14, Rupa Schomaker wrote: On 10/4/2008 4:47 AM, Ivan C Myrvold wrote: Finally an iPhone app that connects with SIP to FreeSWITCH. I tried it out, and it works perfect with my FreeSWITCH installation. Can be downloaded at iTunes App Store. Ivan Ivan, I've been playing with fring. The only issue I have is that I can't get the registered fring client to hit anything but the 'public' dialplan context. I have: variables variable name=user_context value=default/ /variables set on the userid I'm registering fring with. Have you been able to get it to hit your default dialplan context? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Yes, after som more playing with it, I also have problems receiving calls. Originating is OK. I first thought this was an ordinary SIP client running on the iPhone, but I see now it isn't. Ivan ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] get channel status
Hi,
We tried to execute with perl program itself, attached is the perl program
and we can get the output, but in PHP program we cant. Thanks
On Sat, Oct 4, 2008 at 12:51 AM, Michael Collins [EMAIL PROTECTED]wrote:
Before I go any further I need to ask… do you have support for the preg
functions in your PHP version? If not then you'll need it for this to work.
-MC
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Gopal
krishnan
*Sent:* Friday, October 03, 2008 11:07 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] get channel status
Please find the attached PHP file
On Fri, Oct 3, 2008 at 11:29 PM, Michael Collins [EMAIL PROTECTED]
wrote:
You need to handle each response from the server, no? Can you post your PHP
code here?
In Perl I would do something like this.
# $data contains CHANNEL_ANSWER event stuff
if ( $data =~ m/Answered-State: (\w+)/m ) {
my $state = $1;
print Channel state is $state\n;
if ( $state eq 'answered' ) {
# do whatever U need to do on an answer event
}
}
-MC
P.S. – I tested the regex with the data file you posted and it worked
perfectly for me in Perl 5. Since PHP is a Perl derivative I think you
should be able to do this without too much hassle as long as you have a
means of reading the socket data reliably and acting accordingly.
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Gopal
krishnan
*Sent:* Friday, October 03, 2008 10:33 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] get channel status
Hi,
I am trying to get thru a PHP file but i get the output as ,
*authMessage sent, got response:*
*Content-Type: command/reply Reply-Text: +OK accepted*
*command is api originate sofia/default/[EMAIL PROTECTED] 1001
command sent, got response:*
*Content-Type: api/response Content-Length: 41*
*command sent, got response:*
*+OK c5df1f4c-02ae-4353-b709-ad791ca332a1*
*command is bgapi originate sofia/default/[EMAIL PROTECTED] 1001 park
command sent, got response:*
*Content-Type: command/reply Reply-Text: +OK Job-UUID:
d5270c3d-872e-46b6-b556-74bc373b1fe4 Job-UUID:
d5270c3d-872e-46b6-b556-74bc373b1fe4*
*command is event channel_answer command sent, got response:*
*Content-Type: command/reply Reply-Text: +OK event listener enabled plain
*
So I need to run the event channel_answer as a separate program with
autorefreshing?
On Fri, Oct 3, 2008 at 10:49 PM, Brian West [EMAIL PROTECTED] wrote:
Yes if you parse the event using something like perl, ruby, php and get
it...
/b
On Oct 3, 2008, at 12:10 PM, Gopal krishnan wrote:
File attached
On Fri, Oct 3, 2008 at 10:36 PM, Gopal krishnan [EMAIL PROTECTED] wrote:
Hi,
By giving event channel_answer in telnet console I get lots of variables,
I am attaching it as a text file with this email. And my query is for
example If I want to pickup only Answer state from that output, is that
possible?
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Thank you with regards,
Gopal,
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Thank you with regards,
Gopal,
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Thank you with regards,
Gopal,
event.php
Description: Binary data
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Re: [Freeswitch-users] fring
On Oct 4, 2008, at 8:39 AM, Ivan C Myrvold wrote: Yes, after som more playing with it, I also have problems receiving calls. Originating is OK. I first thought this was an ordinary SIP client running on the iPhone, but I see now it isn't. Like all iphone apps, its only running while its on the screen. If you put the phone to sleep or go back to the menu or any other app its going to not be able to receive a call. I have not tested this myself, but will do so today. MIke ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Process_cdr question
On Oct 4, 2008, at 8:22 AM, Vito Andolini wrote: Let's say I am programatically initiating two calls and then bridging them together. If I have the dialplan as originate sofia/example/[EMAIL PROTECTED] bridge(sofia/example/[EMAIL PROTECTED]) and have the process_cdr set to true which is the default. I'd like to know how both cdr would look like... Obviously the b leg will be logged starting right after 400 have answered the call... Which is fine Now would the a leg be logged starting right after its bridged to the b OR right after it picks up the call (even though b is being called or ringing at that moment) The answer is important, because I am using VOIP for both calls, and the voip starts charging me soon as a picks up the phone, so i am wondering if I am able to get the same record into my cdr to process it or do I have to run some sort of magic? Thanks, Vito A. Why don't you look at the cdr and see? There are multiple fields in the cdr to represent when the session was created (INVITE), when it was answered (200) and when it hangs up (BYE). You get this timetable for each leg. Mike___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Routing problem
I'm using http://www.voxbone.com for incoming virtual PSTN calls to
[EMAIL PROTECTED]
oegema.com is my (dyndns)domain and
voxbone is the extension in my domain (my FS server)
Based on the callerid number to route incoming PSTN calls, I've been using
following in Asterisk:
[default]
exten = voxbone,1,Goto(incoming,s,1)
[incoming]
exten = s,1,NoOP(Time=${STRFTIME(${EPOCH},,%H)}.${STRFTIME(${EPOCH},,%M)})
exten = s,n,GotoIf($[${CALLERID(num)}=32476378xxx]?default,2004,1)
exten = s,n,GotoIf($[${CALLERID(num)}=32486632xxx]?default,2005,1)
exten = s,n,GotoIf($[${CALLERID(num)}=0476378xxx]?default,2000,1)
.
.
exten = s,n,Goto(default,1000,1)
So far so good.
===
Now I want to do the same in FS.
If I don't have an extension voxbone and only use the following two
extensios:
extension name=Henk GSM
condition field=caller_id_number expression=32476378xxx
action application=bridge
data=sofia/internal/2004%$${domain}/
/condition
/extension
extension name=Tobie GSM
condition field=caller_id_number expression=^32486632xxx$
action application=bridge data=sofia/internal/2005%$${domain}/
/condition
/extension
then calls from 32476378xxx and from 32486632xxx are bridged correctly.
All other PSTN numbers are not routed anywhere. Which is correct.(because
voxbone don't exist)
However, when I add extension voxbone
extension name=voxbone
condition field=destination_number expression=^voxbone$
action application=bridge data=sofia/internal/2000%$${domain}/
action application=hangup/
/condition
/extension
then ALL calls are routed to 2000.([EMAIL PROTECTED] exist)
So the extensions extension name=Henk GSM and extension name=Tobie
GSM
are ignored.
It is however not possible to put multiple condition fields for
caller_id_number inside the voxbone extension
condition field=caller_id_number expression=32476378xxx and
condition field=caller_id_number expression=^32486632xxx$
(like in Asterisk [incoming] exten =s, )
See also discussion thread: http://www.nabble.com/GotoIf-td19793170.html
So I can't route my calls based on caller-id anymore.
So I'm still faced with the problem how to a route calls based on caller-id,
and if the called-id doesn't match, the calls should be routed to extension
voxbone.
My apologies if the story is too long, but I didn't find another wy to
explain.
Henk
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Re: [Freeswitch-users] GotoIf
we also have the cond fsapi
action application=transfer data=${cond(${caller_id_number}==1234 ?
${ext1} : ${ext2})}/
On Sat, Oct 4, 2008 at 5:45 AM, Arnaldo de Moraes Pereira
[EMAIL PROTECTED] wrote:
On Sat, Oct 4, 2008 at 5:21 AM, henkoegema [EMAIL PROTECTED]wrote:
I discovered some small errors in previous threads concerning this item.
(transfer i.s.o bridge)
To conclude:
Asterisk:
---
exten = s,n,GotoIf($[${CALLERID(num)}=32476478861]?default,1000,1)
FS:
extension name=Henk GSM
condition field=caller_id_number expression=^32476378861$
action application=bridge
data=sofia/internal/1000%$${domain}/
/condition
/extension
Remember that this analogy is wrong. Asterisk's dialplan runs a goto when
the expression is true, while FS' originates a call and bridge the two
parties.
Henk
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ClueCon http://www.cluecon.com/
AIM: anthm
MSN:[EMAIL PROTECTED] [EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED]
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
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googletalk:[EMAIL PROTECTED][EMAIL PROTECTED]
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Re: [Freeswitch-users] GotoIf
On Oct 4, 2008, at 3:21 AM, henkoegema wrote:
I discovered some small errors in previous threads concerning this
item.
(transfer i.s.o bridge)
To conclude:
Asterisk:
---
exten = s,n,GotoIf($[${CALLERID(num)}=32476478861]?default,1000,1)
You wanted an example that does this. The example I provided does
exactly what gotoif does. It sends to call to extension 1000 in
context default.
Its not an error but a matter of choice on how you want to handle the
situation.
FS:
extension name=Henk GSM
condition field=caller_id_number
expression=^32476378861$
action application=bridge
data=sofia/internal/1000%$${domain}/
/condition
/extension
You can bridge or transfer. If you wanted the exact behavior of
gotoif then transfer was it. Since you wanted to send it to extension
1000 in context default. Using bridge doesn't do that. If you notice
in the default config their are things like voicemail, and various
other things that get set when you call 1000 but when you bridge you
bypass all that and the extension now has NO voicemail or any of the
features setup by calling 1000 directly in the dialplan.
/b
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Re: [Freeswitch-users] fring
fring also only makes every other call I try. It also can't call the register phone. /b On Oct 4, 2008, at 11:27 AM, Martin Joseph wrote: There is also Siphon which is available to jailbroken iphones. This appears tome to be more of a real SIP client... FYI, Marty ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] fring
Does this work on iphone firmware 2.x? /b On Oct 4, 2008, at 11:27 AM, Martin Joseph wrote: There is also Siphon which is available to jailbroken iphones. This appears tome to be more of a real SIP client... FYI, Marty ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] fring
On Oct 4, 2008, at 9:31 AM, Brian West wrote: Does this work on iphone firmware 2.x? Huh, I can't seem to find any info other then support for 1.1x firmware, so I don't know. Didn't realize that. Thanks for the heads up, Marty /b On Oct 4, 2008, at 11:27 AM, Martin Joseph wrote: There is also Siphon which is available to jailbroken iphones. This appears tome to be more of a real SIP client... FYI, Marty ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Process_cdr question
Let's say I am programatically initiating two calls and then bridging them together. If I have the dialplan as originate sofia/example/[EMAIL PROTECTED] blocked::mailto:sofia/example/[EMAIL PROTECTED] bridge( blocked::mailto:sofia/example/[EMAIL PROTECTED] sofia/example/[EMAIL PROTECTED]) and have the process_cdr set to true which is the default. I'd like to know how both cdr would look like... Obviously the b leg will be logged starting right after 400 have answered the call... Which is fine Now would the a leg be logged starting right after its bridged to the b OR right after it picks up the call (even though b is being called or ringing at that moment) The answer is important, because I am using VOIP for both calls, and the voip starts charging me soon as a picks up the phone, so i am wondering if I am able to get the same record into my cdr to process it or do I have to run some sort of magic? Thanks, Vito A. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] get channel status
I don't know PHP. If no one else here does either then you'll need to
ask this question on a PHP list or IRC channel.
-MC
Sent from my iPhone
On Oct 4, 2008, at 6:15 AM, Gopal krishnan [EMAIL PROTECTED] wrote:
Hi,
We tried to execute with perl program itself, attached is the perl
program and we can get the output, but in PHP program we cant. Thanks
On Sat, Oct 4, 2008 at 12:51 AM, Michael Collins [EMAIL PROTECTED]
wrote:
Before I go any further I need to ask… do you have support for the p
reg functions in your PHP version? If not then you'll need it for th
is to work.
-MC
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Gopal krishnan
Sent: Friday, October 03, 2008 11:07 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] get channel status
Please find the attached PHP file
On Fri, Oct 3, 2008 at 11:29 PM, Michael Collins [EMAIL PROTECTED]
wrote:
You need to handle each response from the server, no? Can you post
your PHP code here?
In Perl I would do something like this.
# $data contains CHANNEL_ANSWER event stuff
if ( $data =~ m/Answered-State: (\w+)/m ) {
my $state = $1;
print Channel state is $state\n;
if ( $state eq 'answered' ) {
# do whatever U need to do on an answer event
}
}
-MC
P.S. – I tested the regex with the data file you posted and it worke
d perfectly for me in Perl 5. Since PHP is a Perl derivative I think
you should be able to do this without too much hassle as long as yo
u have a means of reading the socket data reliably and acting accord
ingly.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Gopal krishnan
Sent: Friday, October 03, 2008 10:33 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] get channel status
Hi,
I am trying to get thru a PHP file but i get the output as ,
authMessage sent, got response:
Content-Type: command/reply Reply-Text: +OK accepted
command is api originate sofia/default/[EMAIL PROTECTED]
1001 command sent, got response:
Content-Type: api/response Content-Length: 41
command sent, got response:
+OK c5df1f4c-02ae-4353-b709-ad791ca332a1
command is bgapi originate sofia/default/[EMAIL PROTECTED] 1001
park command sent, got response:
Content-Type: command/reply Reply-Text: +OK Job-UUID:
d5270c3d-872e-46b6-b556-74bc373b1fe4 Job-UUID: d5270c3d-872e-46b6-
b556-74bc373b1fe4
command is event channel_answer command sent, got response:
Content-Type: command/reply Reply-Text: +OK event listener enabled
plain
So I need to run the event channel_answer as a separate program with
autorefreshing?
On Fri, Oct 3, 2008 at 10:49 PM, Brian West [EMAIL PROTECTED]
wrote:
Yes if you parse the event using something like perl, ruby, php and
get it...
/b
On Oct 3, 2008, at 12:10 PM, Gopal krishnan wrote:
File attached
On Fri, Oct 3, 2008 at 10:36 PM, Gopal krishnan [EMAIL PROTECTED]
wrote:
Hi,
By giving event channel_answer in telnet console I get lots of
variables, I am attaching it as a text file with this email. And my
query is for example If I want to pickup only Answer state from that
output, is that possible?
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Thank you with regards,
Gopal,
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Thank you with regards,
Gopal,
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Thank you with regards,
Gopal,
event.php
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Re: [Freeswitch-users] Process_cdr question
On Oct 4, 2008, at 4:20 PM, Vito Andolini wrote: Let's say I am programatically initiating two calls and then bridging them together. If I have the dialplan as originate sofia/example/[EMAIL PROTECTED] bridge(sofia/example/[EMAIL PROTECTED]) and have the process_cdr set to true which is the default. I'd like to know how both cdr would look like... Obviously the b leg will be logged starting right after 400 have answered the call... Which is fine Now would the a leg be logged starting right after its bridged to the b OR right after it picks up the call (even though b is being called or ringing at that moment) The answer is important, because I am using VOIP for both calls, and the voip starts charging me soon as a picks up the phone, so i am wondering if I am able to get the same record into my cdr to process it or do I have to run some sort of magic? Thanks, Vito A. Posting the same message to the mailing list twice is unlikely to get you different response. Why don't you try my suggestion? Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GotoIf
Brian West-3 wrote: You can bridge or transfer. If you wanted the exact behavior of gotoif then transfer was it. Since you wanted to send it to extension 1000 in context default. Using bridge doesn't do that. If you notice in the default config their are things like voicemail, and various other things that get set when you call 1000 but when you bridge you bypass all that and the extension now has NO voicemail or any of the features setup by calling 1000 directly in the dialplan. This doesn't work: extension name=ext1 condition field=destination_number expression=2020/ condition field=caller_id_number expression=2000 action application=transfer data=999 XML default/ /condition /extension But this does: extension name=ext1 condition field=destination_number expression=2020/ condition field=caller_id_number expression=2000 action application=bridge data=sofia/internal/[EMAIL PROTECTED]/ /condition /extension I've been testing endless with these examples. That's why I came to the (wrong) conclusion that transfer doesn't work and bridge does. Until I saw my mistake: action application=transfer data=999 XML default/ should have been action application=transfer data= XML default/ Sorry for the confusion. You were so right Brian.:clap: -- View this message in context: http://www.nabble.com/GotoIf-tp19793170p19817097.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GotoIf
Anthony Minessale-2 wrote:
we also have the cond fsapi
action application=transfer data=${cond(${caller_id_number}==1234 ?
${ext1} : ${ext2})}/
Thanks for the hint.
Never knew about this one.
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Re: [Freeswitch-users] GotoIf
Now if you really wanna get fancy and you're running SVN trunk you can do this: extension name=ext1 condition field=destination_number expression=2020/ condition field=caller_id_number expression=2000 action application=bridge data=loopback// /condition /extension Now if loopback is used and you come to a point where the underlying channels bridge to each other the loopback channel will bow out and leave the two channels bridged. /b PS: everyone PLEASE test SVN trunk! ;) On Oct 4, 2008, at 3:36 PM, henkoegema wrote: Sorry for the confusion. You were so right Brian.:clap: ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load test - performance not even matching Asterisk
Anthony, your last response is surprising me. I appreciate the work you have done in supporting people doing load testing. But the information about this is rather unorganized and difficult to access for newbies. Further, many threads on this and other topics are written in telegram style language, which increases the learning time. As an example, the first time I heard about the testing tool sipp was yesterday - through the users list - thanks. What a great tool! The documentation though, is not aimed at making the learning curve easy. A these are the 20 steps to do test manual does not exist to my knowledge. Bear in mind that I'm not a programmer but a generalist with all-round knowledge about telecoms and virtual PBXs. I hope you can use this feedback in a productive way in order to improve the overall level of documentation. And it also pinpoints the need for the userlist and repetitive questions. Setting up commercial relations also makes sense when we have reached the proof of concept and a business plan can be made. The proof of concept includes a decision about which switch to use, Asterisk is still an alternative to FS. Asterisk may have some architectural drawbacks, but it has been tested for a longer time and in more setups than FS. Further, Asterisk has a bad reputation. As you have mentioned before, it may be worth while visiting your team soon, and I'm preparing for that. /Jon ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load test - performance not even matching Asterisk
Sorry, We are all very busy. We take time out of our day to answer as many questions as we can in the little spare time we have. I personally do not have the time to educate everyone on sipp. There is a community here working together on all the docs and I am afraid you may have just insulted them with your comments. My point is I do not have time to stop what I am doing and support people trying to do load testing when they clearly have a lot to learn about the whole concept. I have already added 2 features for you for free and answered all of your questions. What elese do you want from us? sent from my phone -stop- On 10/4/08, Jon Bruel [EMAIL PROTECTED] wrote: Anthony, your last response is surprising me. I appreciate the work you have done in supporting people doing load testing. But the information about this is rather unorganized and difficult to access for newbies. Further, many threads on this and other topics are written in telegram style language, which increases the learning time. As an example, the first time I heard about the testing tool sipp was yesterday - through the users list - thanks. What a great tool! The documentation though, is not aimed at making the learning curve easy. A these are the 20 steps to do test manual does not exist to my knowledge. Bear in mind that I'm not a programmer but a generalist with all-round knowledge about telecoms and virtual PBXs. I hope you can use this feedback in a productive way in order to improve the overall level of documentation. And it also pinpoints the need for the userlist and repetitive questions. Setting up commercial relations also makes sense when we have reached the proof of concept and a business plan can be made. The proof of concept includes a decision about which switch to use, Asterisk is still an alternative to FS. Asterisk may have some architectural drawbacks, but it has been tested for a longer time and in more setups than FS. Further, Asterisk has a bad reputation. As you have mentioned before, it may be worth while visiting your team soon, and I'm preparing for that. /Jon ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GotoIf
Brian West-3 wrote: Now if you really wanna get fancy and you're running SVN trunk you can do this: extension name=ext1 condition field=destination_number expression=2020/ condition field=caller_id_number expression=2000 action application=bridge data=loopback// /condition /extension Now if loopback is used and you come to a point where the underlying channels bridge to each other the loopback channel will bow out and leave the two channels bridged. [EMAIL PROTECTED] version FreeSWITCH Version 1.0.trunk (9841) [EMAIL PROTECTED] 2008-10-05 00:16:26 [NOTICE] switch_channel.c:552 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [16089410-9262-11dd-ae2c-771a315e7f60] 2008-10-05 00:16:26 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing 2000-2020 in context default 2008-10-05 00:16:26 [ERR] switch_core_session.c:249 switch_core_session_outgoing_channel() Could not locate channel type loopback 2008-10-05 00:16:26 [ERR] switch_ivr_originate.c:964 switch_ivr_originate() Cannot create outgoing channel of type [loopback] cause: [CHAN_NOT_IMPLEMENTED] 2008-10-05 00:16:26 [INFO] mod_dptools.c:1848 audio_bridge_function() Originate Failed. Cause: CHAN_NOT_IMPLEMENTED -- View this message in context: http://www.nabble.com/GotoIf-tp19793170p19818399.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GotoIf
Make sure you add it to modules.conf in the root of your src folder. endpoints/mod_loopback then make mod_loopback-install its in the default build if you do a fresh checkout. /b On Oct 4, 2008, at 5:19 PM, henkoegema wrote: [EMAIL PROTECTED] version FreeSWITCH Version 1.0.trunk (9841) [EMAIL PROTECTED] 2008-10-05 00:16:26 [NOTICE] switch_channel.c:552 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [16089410-9262-11dd-ae2c-771a315e7f60] 2008-10-05 00:16:26 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing 2000-2020 in context default 2008-10-05 00:16:26 [ERR] switch_core_session.c:249 switch_core_session_outgoing_channel() Could not locate channel type loopback 2008-10-05 00:16:26 [ERR] switch_ivr_originate.c:964 switch_ivr_originate() Cannot create outgoing channel of type [loopback] cause: [CHAN_NOT_IMPLEMENTED] 2008-10-05 00:16:26 [INFO] mod_dptools.c:1848 audio_bridge_function() Originate Failed. Cause: CHAN_NOT_IMPLEMENTED ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Load test - performance not even matching Asterisk
your last response is surprising me. I appreciate the work you have done in supporting people doing load testing. Countless hours at no cost to you or me have been put in by Anthony, Brian, Mike and many others in building documenting and supporting FreeSWITCH. If you truly do appreciate the work then join the community and do things that help make it better. These include helping with documentation as you learn it, helping to support it or contributing to it with money. Load testing has not been done by many in the community. So it is logical that the road less used will have less documentation. As you learn the load testing you should document it on the wiki. There has been enormous man hours put into this documentation already. Hundreds of pages worth of documentation and examples are already available to anyone. A suggestion if you choose to be a part of the FreeSWITCH community. Recognize that some things should help Keep out comments that portray a feeling of entitlement and word your questions and comments with a little gratitude. Then as people voluntarily answer your questions give back to the community in some way. Mark --- On Sat, 10/4/08, Jon Bruel [EMAIL PROTECTED] wrote: From: Jon Bruel [EMAIL PROTECTED] Subject: Re: [Freeswitch-users] Load test - performance not even matching Asterisk To: freeswitch-users@lists.freeswitch.org Date: Saturday, October 4, 2008, 3:22 PM Anthony, your last response is surprising me. I appreciate the work you have done in supporting people doing load testing. But the information about this is rather unorganized and difficult to access for newbies. Further, many threads on this and other topics are written in telegram style language, which increases the learning time. As an example, the first time I heard about the testing tool sipp was yesterday - through the users list - thanks. What a great tool! The documentation though, is not aimed at making the learning curve easy. A these are the 20 steps to do test manual does not exist to my knowledge. Bear in mind that I'm not a programmer but a generalist with all-round knowledge about telecoms and virtual PBXs. I hope you can use this feedback in a productive way in order to improve the overall level of documentation. And it also pinpoints the need for the userlist and repetitive questions. Setting up commercial relations also makes sense when we have reached the proof of concept and a business plan can be made. The proof of concept includes a decision about which switch to use, Asterisk is still an alternative to FS. Asterisk may have some architectural drawbacks, but it has been tested for a longer time and in more setups than FS. Further, Asterisk has a bad reputation. As you have mentioned before, it may be worth while visiting your team soon, and I'm preparing for that. /Jon ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] caller-controls -- Invalid caller control action name 'dial'.
Hi All, I am trying to include dial caller control in my custom profile, but I get this warning and looks like this action is not defined? 2008-10-04 18:42:11 [WARNING] mod_conference.c:4797 conference_new_install_caller_controls_custom() Invalid caller control action name 'dial'. Here is the caller-controls I included as mentioned in wiki control action=dial digits=888 data=sofia/default/[EMAIL PROTECTED] 60 00 FreeSwitch/ Thanks, Sheeju ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] caller-controls -- Invalid caller control action name 'dial'.
the caller controls are only the ones that you can dial with the DTMF. Dial is a FSAPI command which you would execute at the CLI On Sat, Oct 4, 2008 at 8:51 PM, Sheeju Alex [EMAIL PROTECTED] wrote: Hi All, I am trying to include dial caller control in my custom profile, but I get this warning and looks like this action is not defined? 2008-10-04 18:42:11 [WARNING] mod_conference.c:4797 conference_new_install_caller_controls_custom() Invalid caller control action name 'dial'. Here is the caller-controls I included as mentioned in wiki control action=dial digits=888 data=sofia/default/[EMAIL PROTECTED] 60 00 FreeSwitch/ Thanks, Sheeju ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Process_cdr question
On Oct 4, 2008, at 4:45 PM, Vito Andolini wrote: what do you mean by look at the cdr? I checked these 2 wiki pages bu tthey provide only SOME of the fields not all... http://wiki.freeswitch.org/wiki/Mod_xml_cdr http://wiki.freeswitch.org/wiki/Mod_cdr_csv I also checked the API section but couldn't find it... Vito We mean setup FreeSWITCH, make a call, look at the cdr. Mike___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
