Re: [asterisk-users] Alphanumeric DTMF !?
Eric thats really a nice idea to communicate between two or more of our servers. Make the call to the remote system and send the digits in the encoded string, you will need something on the other end to decode the text. But the other end is not our's but could be any solution which requires to feed alphanumeric DTMF that something on the other end could be a propriety solution like CISCO as I mentioned about its alphanumeric relay. SO, I can't ask the other end to change. I'm having a strong feeling that I shouldn't push further into this as Asterisk has its DTMF methods defined and those don't send Alpha-numeric. that's it - end of line. :( On Tue, Feb 28, 2012 at 9:27 PM, Eric Wieling ewiel...@nyigc.com wrote: Just for fun I did something similar at one point. 0-9 A-D and * and # make a character set of 16 characters, perfect for encoding as hex. Take your string, get the ASCII value of each character, convert it to hex, and add it to the encoded string. Just before dialing, replace all e with # and all f with * in your encoded string Make the call to the remote system and send the digits in the encoded string, you will need something on the other end to decode the text. Took about 30 seconds to send Hello World! because of limitations in Asterisk's maximum digits on Read (about 40 digits) and needing to ACK packets of 32 characters each and timeouts, etc. I think I used CRC8 to validate the received packets. Overall it was a cool hack and totally impractical in the real world. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Tuesday, February 28, 2012 2:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Alphanumeric DTMF !? Yeah I know about A-D but can we send more than those !? I've read about h245-alphanum thing but that is definitely not in asterisk, so what other options are there is I've to send more than just A-D ? On Tue, Feb 28, 2012 at 12:42 PM, Matt Darnell mattdarn...@gmail.com wrote: On Mon, Feb 27, 2012 at 8:23 PM, Sammy Govind govoi...@gmail.com wrote: Hi list, What possibilities are there in asterisk to send an alphanumeric DTMF from/to asterisk !? Regards, Sammy Do you mean A-D? You send those just like 0-9*# -Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk RTCP
Hi kevin, I've observed that I've rtcp set debug command (rtcp based commands) available on my asterisk console. Can you please explain about RTCP. I really need RTCPs in my setup, it doesnt matter if the RTCPs are separate for both A-leg and B-leg i.e A-leg===Asterisk and Asterisk===B-leg I can live with RTPs flowing for each leg with asterisk separately. But problem is I dont get any RTCPs for each leg independently as well !! Please suggest. Regards. Sammy On Fri, Feb 17, 2012 at 5:21 PM, Gohar Ahmed gohar.ah...@vopium.com wrote: Hello list, Kevin I agree with you on independent monitored entity for A leg while the outbound leg has separate QoS measures. But after this thread I went to my monitoring tool and saw that for some calls on the same asterisk setup I had no RTP or RTCP while there were calls with both RTP and RTCP captured as well. Since I've a SIP proxy on top of asterisk servers layers, could it be possible that RTP and RTCPs bypass asterisk (media redirect) and that's why I see RTCPs and RTPs logged into monitoring tool while those call who couldn't redirect/bypass media from asterisk don't show any RTCPs!? Sammy can you provide further details of your setup please! Regards, Gohar -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, February 17, 2012 5:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk RTCP On 02/17/2012 12:09 AM, Sammy Govind wrote: Hello, Thanks for taking out tome for my query. Yes I do have an actual problem. I've a monitoring tool to record the VoIP QoS (Asterisk servers port mirrored to it). My end points(soft-phones) are sending RTCP connection strings to asterisk, and Asterisk then forwards their call to their destination choosing any suitable carrier. If I don't get RTCP flowing through asterisk the monitoring tool simply fails to display and call stats. Please advice what should I be doing to cater this. As I said before, you will never get RTCP *flowing through* Asterisk. When your softphone calls Asterisk, that will be a separate call leg from the one from Asterisk to your provider. Your monitoring tool should treat those as separate call legs and produce an analysis for them independently. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk RTCP
Hello, Thanks for taking out tome for my query. Yes I do have an actual problem. I've a monitoring tool to record the VoIP QoS (Asterisk servers port mirrored to it). My end points(soft-phones) are sending RTCP connection strings to asterisk, and Asterisk then forwards their call to their destination choosing any suitable carrier. If I don't get RTCP flowing through asterisk the monitoring tool simply fails to display and call stats. Please advice what should I be doing to cater this. Thanks, Sammy On Thu, Feb 16, 2012 at 10:00 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/16/2012 01:16 AM, Sammy Govind wrote: Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why is this so! or if there is anything I can do to make RTCPs flow through the asterisk server ! I've asterisk 1.6.2.20 in production. It is not mandatory to signal anything related to RTCP in the SDP. RTCP is implicitly handled on the next port up from the port being used for RTP; the signaling in SDP is only needed if the RTCP is *not* going to be on the next port up. RTCP will never *flow through* Asterisk, as Asterisk is terminating both RTP flows and thus is an endpoint for both of them. Do you have an actual problem you are trying to resolve, or are you just asking questions about RTCP? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk perl AGI confusing variables
Thanks for good advice, will definitely keep these in mind while doing coding - starting from now :) On Mon, Feb 13, 2012 at 12:30 PM, Steve Edwards asterisk@sedwards.comwrote: On Mon, 13 Feb 2012, Sammy Govind wrote: Hi again,Just to update I fixed the issue. I read through your reply and the URL in it and tried alot to make things working but in vain- then I took the tough way and started looking at the production AGI from the first line and amended all the warning and unwanted stuff, finally I figured out that the agi-verbose() function just a few lines above the problematic code was having a warning and once that was fixed all the code started working fine. I still wonder what do variable assignments has to do with verbose function warning, but its all working fine now. Thanks for the help. It's a good idea to track down all warnings and errors even when they seem unrelated to the problem at hand. Keep in mind executing an AGI completely external from Asterisk can be a valuable debugging aid. Just create a text file containing all the proper responses and feed it to the AGI's STDIN. I do this frequently with the AGIs I write in C so I can use GDB to step through my code and figure out what's going on. On Sat, Feb 11, 2012 at 11:20 PM, Ron Bergin r...@i.frys.com wrote: Finally, add a couple debugging statements after the get_variable statements to verify/dump the vars. Doing any I/O on STDIN or STDOUT will violate the AGI protocol. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk perl AGI confusing variables
Hi again,
Just to update I fixed the issue. I read through your reply and the URL in
it and tried alot to make things working but in vain- then I took the tough
way and started looking at the production AGI from the first line
and amended all the warning and unwanted stuff, finally I figured out that
the agi-verbose() function just a few lines above the problematic code was
having a warning and once that was fixed all the code started working fine.
I still wonder what do variable assignments has to do with verbose function
warning, but its all working fine now.
Thanks for the help.
Regards,
Sammy
On Sun, Feb 12, 2012 at 10:40 AM, Sammy Govind govoi...@gmail.com wrote:
Hey Ron,
Thanks for taking out time for this weird issue. No this is the only code
thats running and I simply copy pasted it here. I'll go through the artivle
you mentioned and other advices you gave may hopefully resolve this issue.
But in general its beyond my logic to see whats actually going on here.
Simply mind blowing trick for me :)
Just to add here, even changing the arrangement of verbose statement above
or below the Addheader statement changes the variables as well.
Additional Details:
I tested the code without enclosing it in a sub , in a very small agi just
for this and this same code was giving me 100% results. So that means that
the production AGI/perl code has something in it thats causing the issue !?
Regards,
Sammy
On Sat, Feb 11, 2012 at 11:20 PM, Ron Bergin r...@i.frys.com wrote:
Sammy Govind wrote:
Hello all,
I'm struck with a very strange problem today. I've an AGI with some code
subroutine snippet as follows:
sub enable_sbc($) {
my $carrier = shift;
my $tmp = substr($carrier,1);
my $jkh = $tmp;
$server_port = $ast_agi-get_variable(SIPPEER($jkh,port));
$ser_ip = $ast_agi-get_variable(SIPPEER($tmp,ip));
$ast_agi-exec(SIPAddHeader,P-PORT: $server_port);
$ast_agi-exec(SIPAddHeader,P-IPADDRESS: $ser_ip);
return 0;
}
Where $carrier resolves to @my-carrier
Strangely and very weird get variable is returning correct values on
console as given below but the variables containing the values gets lost
or
confused with each other !
SIP/sipproxy3.32-AGI Rx GET VARIABLE
SIPPEER(my-carrier,port)
SIP/sipproxy3.32-AGI Tx 200 result=1 (5060)
SIP/sipproxy3.32-AGI Rx GET VARIABLE SIPPEER(my-carrier,ip)
SIP/sipproxy3.32-AGI Tx 200 result=1 (192.168.2.19)
SIP/sipproxy3.32-AGI Rx EXEC SIPAddHeader P-PORT:
-- AGI Script Executing Application: (SIPAddHeader) Options:
(P-PORT:
)
SIP/sipproxy3.32-AGI Tx 200 result=0
SIP/sipproxy3.32-AGI Rx EXEC SIPAddHeader P-IPADDRESS:
5060
-- AGI Script Executing Application: (SIPAddHeader) Options:
(P-IPADDRESS: 5060)
SIP/sipproxy3.32-AGI Tx 200 result=0
Anyone please help. Am I doing anything wrong ?
Regards,
Sammy.
--
_
Did you copy/paste the code in the email posting, or did you retype it?
Is it possible that you have multiple versions of the script and the wrong
one is being executed?
First step I'd suggest, after checking the above possibility, is to remove
the prototype. It is almost never needed/wanted and can introduce bugs if
not used correctly.
http://www.modernperlbooks.com/mt/2009/08/the-problem-with-prototypes.html
Next, are you using the strict and warnings pragmas? If not, you should
add them and fix the problems that they point out.
Next, declare $server_port and $ser_ip as lexical vars in the sub.
Finally, add a couple debugging statements after the get_variable
statements to verify/dump the vars.
---
Ron Bergin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk perl AGI confusing variables
Hello all,
I'm struck with a very strange problem today. I've an AGI with some code
subroutine snippet as follows:
sub enable_sbc($) {
my $carrier = shift;
my $tmp = substr($carrier,1);
my $jkh = $tmp;
$server_port = $ast_agi-get_variable(SIPPEER($jkh,port));
$ser_ip = $ast_agi-get_variable(SIPPEER($tmp,ip));
$ast_agi-exec(SIPAddHeader,P-PORT: $server_port);
$ast_agi-exec(SIPAddHeader,P-IPADDRESS: $ser_ip);
return 0;
}
Where $carrier resolves to @my-carrier
Strangely and very weird get variable is returning correct values on
console as given below but the variables containing the values gets lost or
confused with each other !
SIP/sipproxy3.32-AGI Rx GET VARIABLE SIPPEER(my-carrier,port)
SIP/sipproxy3.32-AGI Tx 200 result=1 (5060)
SIP/sipproxy3.32-AGI Rx GET VARIABLE SIPPEER(my-carrier,ip)
SIP/sipproxy3.32-AGI Tx 200 result=1 (192.168.2.19)
SIP/sipproxy3.32-AGI Rx EXEC SIPAddHeader P-PORT:
-- AGI Script Executing Application: (SIPAddHeader) Options: (P-PORT: )
SIP/sipproxy3.32-AGI Tx 200 result=0
SIP/sipproxy3.32-AGI Rx EXEC SIPAddHeader P-IPADDRESS: 5060
-- AGI Script Executing Application: (SIPAddHeader) Options:
(P-IPADDRESS: 5060)
SIP/sipproxy3.32-AGI Tx 200 result=0
Anyone please help. Am I doing anything wrong ?
Regards,
Sammy.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?
I'd definitely go with AMI ! On Sat, Feb 11, 2012 at 6:39 PM, asterisk jobs asteriskcod...@gmail.comwrote: Thanks for the input but using spool files or AMI or AGI is way different from each other and that is what I want to get an input on. I do have hands on with all methods like I noted but want to know what the trend is now-a-days and what is more robust and proven out of all three. Best, On Sat, Feb 11, 2012 at 8:12 AM, David Backeberg dbackeb...@gmail.comwrote: On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs asteriskcod...@gmail.com wrote: Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the easiest way to approach the issue. The way to call 5000 numbers is to call one number, really well. Then you put it in a loop. You need to run a lab for long enough that you have the bugs worked out, before you subject real people to problems. With asterisk you can always tell the real-time status of a call, even if you initiate from a call file. Perhaps you would enjoy reading up on Local channels. Some people prefer to initiate calls from AMI. I tried it and didn't like it. But because most of us have been annoyed by an autodialer in our lives, even if we ourselves have made autodialers in the past, this is probably about the limit of the help you're going to get, unless you ask a more specific question that shows you've been trying to learn this hands-on and you've gotten stuck on a particular problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk perl AGI confusing variables
Hey Ron,
Thanks for taking out time for this weird issue. No this is the only code
thats running and I simply copy pasted it here. I'll go through the artivle
you mentioned and other advices you gave may hopefully resolve this issue.
But in general its beyond my logic to see whats actually going on here.
Simply mind blowing trick for me :)
Just to add here, even changing the arrangement of verbose statement above
or below the Addheader statement changes the variables as well.
Additional Details:
I tested the code without enclosing it in a sub , in a very small agi just
for this and this same code was giving me 100% results. So that means that
the production AGI/perl code has something in it thats causing the issue !?
Regards,
Sammy
On Sat, Feb 11, 2012 at 11:20 PM, Ron Bergin r...@i.frys.com wrote:
Sammy Govind wrote:
Hello all,
I'm struck with a very strange problem today. I've an AGI with some code
subroutine snippet as follows:
sub enable_sbc($) {
my $carrier = shift;
my $tmp = substr($carrier,1);
my $jkh = $tmp;
$server_port = $ast_agi-get_variable(SIPPEER($jkh,port));
$ser_ip = $ast_agi-get_variable(SIPPEER($tmp,ip));
$ast_agi-exec(SIPAddHeader,P-PORT: $server_port);
$ast_agi-exec(SIPAddHeader,P-IPADDRESS: $ser_ip);
return 0;
}
Where $carrier resolves to @my-carrier
Strangely and very weird get variable is returning correct values on
console as given below but the variables containing the values gets lost
or
confused with each other !
SIP/sipproxy3.32-AGI Rx GET VARIABLE
SIPPEER(my-carrier,port)
SIP/sipproxy3.32-AGI Tx 200 result=1 (5060)
SIP/sipproxy3.32-AGI Rx GET VARIABLE SIPPEER(my-carrier,ip)
SIP/sipproxy3.32-AGI Tx 200 result=1 (192.168.2.19)
SIP/sipproxy3.32-AGI Rx EXEC SIPAddHeader P-PORT:
-- AGI Script Executing Application: (SIPAddHeader) Options: (P-PORT:
)
SIP/sipproxy3.32-AGI Tx 200 result=0
SIP/sipproxy3.32-AGI Rx EXEC SIPAddHeader P-IPADDRESS:
5060
-- AGI Script Executing Application: (SIPAddHeader) Options:
(P-IPADDRESS: 5060)
SIP/sipproxy3.32-AGI Tx 200 result=0
Anyone please help. Am I doing anything wrong ?
Regards,
Sammy.
--
_
Did you copy/paste the code in the email posting, or did you retype it?
Is it possible that you have multiple versions of the script and the wrong
one is being executed?
First step I'd suggest, after checking the above possibility, is to remove
the prototype. It is almost never needed/wanted and can introduce bugs if
not used correctly.
http://www.modernperlbooks.com/mt/2009/08/the-problem-with-prototypes.html
Next, are you using the strict and warnings pragmas? If not, you should
add them and fix the problems that they point out.
Next, declare $server_port and $ser_ip as lexical vars in the sub.
Finally, add a couple debugging statements after the get_variable
statements to verify/dump the vars.
---
Ron Bergin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?
Yes why not, I made an aut-odialer (the code I can share on my blogpost in couple of days for you.) The basic structure of the script/code was to: 1- Start, connect to DB, fetch campaign data 2- Fetch numbers to dial from campaign, If no numbers goto step 6 3- Feed those number in a loop to AMI using a php-AMI helper script (Async Event, don't wait for reply from Asterisk) 4- Check asterisk if its dialing capacity has reached or not 5a- If Not, goto step 2 5b- If Yes, wait for sometime for calls to finish, goto step 4 6- Close DB,Stop So, I had a context that was connecting to MySQL and on each incoming call trigger it was pushed with primary keys/identifiers of campaign and callednumber. Using those I updated the CDRs/STATUS of that particular number if it failed or successfully answered. That was all. Obviously there are major advanced features in this script which are missing and need time and proper coding expertise to develop..i.e multi-campaign mode, aggressiveness of dialer, retrying of failed numbers etc. Regards, Sammy. On Sat, Feb 11, 2012 at 9:23 PM, Bruce B bruceb...@gmail.com wrote: Sammy, Would you care to elaborate please. Have you had experience doing such a campaign using AMI? Maybe you can share of the code. Most appreciated, On Sat, Feb 11, 2012 at 10:15 AM, Sammy Govind govoi...@gmail.com wrote: I'd definitely go with AMI ! On Sat, Feb 11, 2012 at 6:39 PM, asterisk jobs asteriskcod...@gmail.comwrote: Thanks for the input but using spool files or AMI or AGI is way different from each other and that is what I want to get an input on. I do have hands on with all methods like I noted but want to know what the trend is now-a-days and what is more robust and proven out of all three. Best, On Sat, Feb 11, 2012 at 8:12 AM, David Backeberg dbackeb...@gmail.comwrote: On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs asteriskcod...@gmail.com wrote: Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the easiest way to approach the issue. The way to call 5000 numbers is to call one number, really well. Then you put it in a loop. You need to run a lab for long enough that you have the bugs worked out, before you subject real people to problems. With asterisk you can always tell the real-time status of a call, even if you initiate from a call file. Perhaps you would enjoy reading up on Local channels. Some people prefer to initiate calls from AMI. I tried it and didn't like it. But because most of us have been annoyed by an autodialer in our lives, even if we ourselves have made autodialers in the past, this is probably about the limit of the help you're going to get, unless you ask a more specific question that shows you've been trying to learn this hands-on and you've gotten stuck on a particular problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org
Re: [asterisk-users] Asterisk V/s FreeSwitch
Wow, I bet even asterisk developers wouldn't believe so. What have they done !. No, actually can you tell if server was processing media along with the calls as well !? I once tested without media and really I had some 1000+ CCs on asterisk server on a regular dev machine with choppy audio on an actual call while still under stress. Kindly please confirm your stats. Regards, Sammy On Thu, Feb 9, 2012 at 4:49 PM, Stefan Schmidt s...@sil.at wrote: Am 07.02.12 12:38, schrieb virendra bhati: Hi List, Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... I had done some load tests with asterisk 10 and my highest results was: 1750 calls per seconds up to 13000 concurrent calls done on a intel xeon with dual six core and hyperthreading (= 24 cores) and 12 GB ram. the sysload was around 2.5 during this test. so i am not impressed by 1000 concurrent calls. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback with noanswer in AGI
Hi, Given invites seems fine, can you take a wireshark trace of the call on your eyebeam machine! from that wireshark trace use telephony calls options and hear if you are actually receiving RTPs on your system. If you could hear the played back sound file on your eyembeam machine . this would mean that your eyebeam client is not good enough to play media while its in 183 session progress. Also can you send me the short sample php-agi script you are executing so i actually test this on my virtual machines as well. Regards, Sammy On Tue, Feb 7, 2012 at 1:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer
Re: [asterisk-users] Playback with noanswer in AGI
Exactly that's what I expected. Great - now have fun On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Sammy, Problem is at phones, with a linksys phone it works but with eyebeam and fanvill it doesn't Maybe they don't support early media. I think i will have to stick with ResetCDR and that will be okay now as I've modified the code for that Thank you Regards, Zohair Raza On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer() the call rather put a progress() and soon after that playing back the sound file from playback with noanswer
Re: [asterisk-users] MixMonitor and ChanSpy
Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded ! Which version of asterisk you are using ! can you paste the CLI logs which show a complete call with a failed attempt to Chanspy ? Regards, Sammy On Tue, Feb 7, 2012 at 2:12 PM, Jonas Kellens jonas.kell...@telenet.bewrote: ** On 02/02/2012 11:24 AM, Jonas Kellens wrote: Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Anyone with some feedback ?! I notice that ongoing recordings are temporarily saved in the directory /tmp. How could I look from the dialplan into the /tmp-directory to see if there is an ongoing recording for the channel that one wants to spy on ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy
Oh Come on you are Using Asterisk 1.6.2.22. already. Atleast give it a shot and if this still persists then look for other methods or fixes. On Tue, Feb 7, 2012 at 5:44 PM, Jonas Kellens jonas.kell...@telenet.bewrote: ** On 02/07/2012 01:07 PM, Sammy Govind wrote: Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded ! Which version of asterisk you are using ! can you paste the CLI logs which show a complete call with a failed attempt to Chanspy ? Using Asterisk 1.6.2.22. The fact that ChanSpy can not be used with MixMonitor is something I read on the wiki : Attention - Up to and including Asterisk 1.4.17 ChanSpy can cause a * crash/segfault* if used together with Monitorhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Monitoror MixMonitor http://www.voip-info.org/wiki/view/MixMonitor at the same time. 1.4.18 is supposed to attack this issue by using audiohooks that replaces the current ChanSpy approach. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback with noanswer in AGI
Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer() the call rather put a progress() and soon after that playing back the sound file from playback with noanswer and then I get the file streaming as 183-Session progress file. I do understand that playing any sound file before establishing any audio session between two end point will result in no-adio from playback() BUT the combination of progress() and playback(,noanswer) works fine for me. What I think the issue could be for Zohair is that its requesting/incoming session(carrier) isn't allowing the 183-Session progress. Zohair can you do a SIP trace for this particular call along with the dialplan executing for it!? Regards, Sammy. On Tue, Feb 7, 2012 at 11:55 AM, Zohair Raza engineerzuhairr...@gmail.comwrote: Thanks for this explanation Dany! Regards, Zohair Raza On Mon, Feb 6, 2012 at 10:11 PM, Danny Nicholas da...@debsinc.com wrote: You are mis-understanding the concept – the noanswer option is playing the file as you requested, but since you aren’t answering the call, no channel is established to actually present the sound to you. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zohair Raza *Sent:* Monday, February 06, 2012 12:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Playback with noanswer in AGI ** ** Hi All, ** ** I want to play a file in agi but dont want to answer the call ** ** I am dialing through sip phone and running asterisk 1.8.6, ** ** I tried following with no luck ** ** $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); $agi-hangup(); ** ** When I dial I can't hear the audio but if I answer the call or remove noanswer argument I can hear the audio. ** ** phpAGI's stream_file didn't help either. ** ** I ended up with ResetCDR() before hangup to reset billsec, duration and disposition but don't want to do it this way. ** ** What could be the problem? ** ** From Voip-info.org : *noanswer*: Play the sound file, but don't answer the channel first (if hasn't been answered already). Not all channels support playing messages while still on hook. ** ** Is it because the channel is not supported? ** ** ** ** Regards, Zohair Raza ** ** ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can someone tell me what is this issue ?
Your Server Voipon isn't responding- See if internet is working fine, or your Voipon sip trunk/peer is registered OK? On Fri, Feb 3, 2012 at 5:53 PM, virendra bhati virbh...@gmail.com wrote: Call is not routing from server to destination. app8*CLI console dial 00918885268942 [Feb 3 06:01:15] NOTICE[28124]: console_video.c:133 console_video_start: voice only, console video support not present -- Executing [00918885268942@default:1] Answer(Console/dsp, ) in new stack Console call has been answered -- Executing [00918885268942@default:2] Dial(Console/dsp, SIP/ 00918885268942@voipon) in new stack == Using SIP RTP CoS mark 5 Audio is at 10.30.131.136 port 12556 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 217.14.138.127:5065: INVITE sip:00918885268...@sip.voipon.co.uk:5065;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.30.131.136:5060;branch=z9hG4bK5388007b;rport Max-Forwards: 70 From: asterisk sip:7476...@sip.voipon.co.uk;tag=as2f61c90c To: sip:00918885268...@sip.voipon.co.uk:5065;user=phone Contact: sip:7476849@10.30.131.136 Call-ID: 3cd12da658b42c10186c01ed3a7d2...@sip.voipon.co.uk CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.21 Date: Fri, 03 Feb 2012 06:01:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 313 v=0 o=root 1850926672 1850926672 IN IP4 10.30.131.136 s=Asterisk PBX 1.6.2.21 c=IN IP4 10.30.131.136 t=0 0 m=audio 12556 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 00918885268942@voipon Retransmitting #1 (NAT) to 217.14.138.154:5060: INVITE sip:00918885268...@sip.voipon.co.uk:5065;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.30.131.136:5060;branch=z9hG4bK5388007b;rport Max-Forwards: 70 From: asterisk sip:7476...@sip.voipon.co.uk;tag=as2f61c90c To: sip:00918885268...@sip.voipon.co.uk:5065;user=phone Contact: sip:7476849@10.30.131.136 Call-ID: 3cd12da658b42c10186c01ed3a7d2...@sip.voipon.co.uk CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.21 Date: Fri, 03 Feb 2012 06:01:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 313 Scheduling destruction of SIP dialog ' 3cd12da658b42c10186c01ed3a7d2...@sip.voipon.co.uk' in 32000 ms (Method: INVITE) -- SIP/voipon-0014 is circuit-busy Scheduling destruction of SIP dialog ' 3cd12da658b42c10186c01ed3a7d2...@sip.voipon.co.uk' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/1/0) -- Executing [00918885268942@default:3] NoOp(Console/dsp, **CONGESTION**) in new stack -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound file
You can use a combination of ChanSpy() and a local extension playing the required file to caller/callee. On Thu, Jan 26, 2012 at 2:11 PM, Eyal e...@mcr-m.com wrote: Thanks ** ** But this is not what I am looking for, in this way I can start the sound file from some point in the file but the callers must hear the file until the end. I need something that allows me to start from some place in the file and end it in some other place in the file (say song from time 01:32 until 01:57), Or Like the *controlplayback* doing fast-forward but without having to click any key by caller. ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nasir Iqbal *Sent:* Thursday, January 26, 2012 10:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] play sound file ** ** check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ On Wed, Jan 25, 2012 at 8:29 PM, Eyal e...@mcr-m.com wrote: Hi, How can I play a sound file from the middle and end it after a certain number of seconds? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SDP Issue
:D pretty much true ! On Tue, Jan 24, 2012 at 12:23 PM, Alex Balashov abalas...@evaristesys.comwrote: Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, one of those who rocket-jumps onto the platform and camps with the grenade launcher, trying to stop the reds from capturing the blue flag? I hate how the health and the ammo takes so long to respawn. Is there any way to fix that in deathmatch? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/http://www.evaristesys.com/ On Jan 24, 2012, at 2:10 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: --[ UxBoD ]-- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro vs sub
Yes I've personally experienced issue with nested macros and eventually asterisk failing to process call any further. So I moved onto using GoSUBs and everything worked perfectly. Since then I'm using GoSUBs happily. On Wed, Jan 18, 2012 at 4:54 PM, Jonas Kellens jonas.kell...@telenet.bewrote: ** Can someone confirm that the nesting of macro's or the continuous and simultaneous use of different macro's, can lead to stack-problems and cause an Asterisk spontaneous reboot/restart ? Kind regards, Jonas. On 01/17/2012 03:02 PM, Bryant Zimmerman wrote: Jonas From what I understand they are trying to phase out Macros. We are slowly removing them from our dialplans as time allows for testing. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Jonas Kellens jonas.kell...@telenet.bejonas.kell...@telenet.be *Sent*: Tuesday, January 17, 2012 5:53 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com asterisk-users@lists.digium.com *Subject*: [asterisk-users] Macro vs sub Hello list, can I conclude that it is better to use sub's in stead of macro's ? I read the following in an Asterisk-book : GoSub() works in a different manner from Macro(), though, in that it doesn’t have the stack space requirements, so it nests effectively. Essentially, GoSub() acts like Goto() with a memory of where it came from. Is it then not better to use a method that does not stack ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro vs sub
Hi, why don't you try write two macros only and recursively call both of them incrementing a counter each time you call the inner macro. Also print(NOOP) system stats along with the counter. You'll soon see what happens. The para Matthew quoted is cent percent true. But if you don't need to call macros within macros and do kind of recursove macro calling then you can continue using macros safely. Its not that I never use macros at all, I only use where I know I'll get in macro and safely exit without going any deeper within the dial-plan. On Wed, Jan 18, 2012 at 6:41 PM, Jonas Kellens jonas.kell...@telenet.bewrote: On 01/18/2012 01:51 PM, Matthew Jordan wrote: Anyone else ? Maybe one of the developers can confirm this risk of working with macros ? I don't think you need an Asterisk developer to tell you the risks of using macros in deeply nested situations. Quoting the documentation of Macro: Because of the way Macro is implemented (it executes the priorities contained within it via sub-engine), and a fixed per-thread memory stack allowance, macros are limited to 7 levels of nesting (macro calling macro calling macro, etc.); It may be possible that stack-intensive applications in deeply nested macros could cause asterisk to crash earlier than this limit. It is advised that if you need to deeply nest macro calls, that you use the Gosub application (now allows arguments like a Macro) with explict Return() calls instead. How would I notice that this is really the case here ? I should see the RAM-memory spike on the server ? I do not see this... Webmin says : 3.83 GB total, 375.51 MB used Kind regards, Jonas. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as UAC: How to put call OnHold
Hey, I have never worried about looking at the SIP re-invites or anything when we engage MoH() application in asterisk. You can do a quick test on your test machine for this. Regards, Sammy On Mon, Jan 16, 2012 at 2:57 PM, Johannes Zweng john999...@zweng.at wrote: Hi! Many thanks for this hint. I will try this! :-) A quick question: when doing this with MusicOnHold(): will the SIP server be aware that the call is placed onHold (i.e. will Asterisk send the mentioned re-INVITE)? The point is - if possible - we want the caller to hear the OnHold Music from the SIP server. If not we would have to copy the MoH to our Asterisk (and change it on our side too, when it changes at the SIP-server). Kind regards, John 2012/1/16 Sammy Govind govoi...@gmail.com Hi, yes, please see MusicOnHold() Application. You can call this app in your dialplan. This however will use the default music class and the corresponding music files placed in the asterisk server. If you don't want to stream music from Asterisk server side, try creating a new MusiconHold Class without any proper directory. That way Asterisk would only complain that there is no file to be streamed. Regards, Sammy On Sat, Jan 14, 2012 at 6:25 AM, Johannes Zweng john999...@zweng.at wrote: Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: -- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the dialplan and should place the call on hold after some time, so that the caller only hears the on hold music from my provider (not streamed by my Asterisk). Technically speaking I want asterisk to send a re-INVITE message containing an updated SDP body with the attribute a=sendonly or a=inactive added so that the SIP server of my provider (where Asterisk is registered to as user) will recognize that the call should be placed on hold. A good example of what I want to achieve is presented in Section 2.1 of RFC 5359 (Session Initiation Protocol Service Examples) ( http://tools.ietf.org/html/rfc5359#section-2.1) where Bob would be my Asterisk (as UAC), Alice is the external caller and Proxy is the provider's SIP server. Question: -- Is there any way to perform this from the dialplan or by means of the manager API? Is there an application like Hold? Kind regards and greetings from Austria, John :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer doesn't answer
Paste some SIP traces of the call while Unmonitored. On Mon, Jan 16, 2012 at 4:58 PM, Arlen Nascimento arlen.nascime...@gmail.com wrote: It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arlen Nascimento -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer doesn't answer
I'm only expecting NAT issues if not the latency issues. SIP traces of any such calls will make more sense. On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento arlen.nascime...@gmail.com wrote: the client is aware of the adverse environment and this is the only solution for him On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda flaviormira...@hotmail.com wrote: Unless you are doing test with SIP under adverse environmet, that is not the point, but, if you intend to have Communication, you should worry about this detail. Basic infra-estructure is the first thing to think in any new project. Good luck! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Mon, 16 Jan 2012 07:58:34 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Peer doesn't answer It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arlen Nascimento -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arlen Nascimento -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as UAC: How to put call OnHold
Hi, yes, please see MusicOnHold() Application. You can call this app in your dialplan. This however will use the default music class and the corresponding music files placed in the asterisk server. If you don't want to stream music from Asterisk server side, try creating a new MusiconHold Class without any proper directory. That way Asterisk would only complain that there is no file to be streamed. Regards, Sammy On Sat, Jan 14, 2012 at 6:25 AM, Johannes Zweng john999...@zweng.at wrote: Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: -- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the dialplan and should place the call on hold after some time, so that the caller only hears the on hold music from my provider (not streamed by my Asterisk). Technically speaking I want asterisk to send a re-INVITE message containing an updated SDP body with the attribute a=sendonly or a=inactive added so that the SIP server of my provider (where Asterisk is registered to as user) will recognize that the call should be placed on hold. A good example of what I want to achieve is presented in Section 2.1 of RFC 5359 (Session Initiation Protocol Service Examples) ( http://tools.ietf.org/html/rfc5359#section-2.1) where Bob would be my Asterisk (as UAC), Alice is the external caller and Proxy is the provider's SIP server. Question: -- Is there any way to perform this from the dialplan or by means of the manager API? Is there an application like Hold? Kind regards and greetings from Austria, John :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exceptionally long voice queue length
which version of Asterisk are you using !. AFAIK this issue has been in asterisk for queue calls and I'm not sure if this has ever been resolved fully and stabilized. Not binding to Local channel only, I've seen this on SIP and IAX channels as well ! On Thu, Jan 12, 2012 at 12:56 AM, Vik Killa vipki...@gmail.com wrote: I'm seeing this error thousands of times per minute and it's causing the CPU to sky rocket WARNING[16095]: channel.c:1039 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/*7...etc... Any idea what could be causing this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
Hi, Sorry for late reply. Hope you've already found out something about it. What version of asterisk you are using, that function for choosing inbound/outbound call leg codecs is for newer versions of asterisk. See these pages: http://www.voip-info.org/wiki/view/Asterisk+variables https://issues.asterisk.org/view.php?id=13243 Regards, Sammy On Tue, Jan 3, 2012 at 2:31 PM, Faraj Khasib fkha...@iconnecths.com wrote: thats excatly what I want, can u plz give me the command, I want to choose only ulow From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind [ govoi...@gmail.com] Sent: Tuesday, January 03, 2012 3:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call type in dial plan Hi, For such call you just need to select the outbound codec before the dial() app. choose the audio-only codecs and thus no video codec strings will be exchanged in that call. -- Regards, Sammy On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib fkha...@iconnecths.com mailto:fkha...@iconnecths.com wrote: this is what my SIP Invite message when I make Video call INVITE sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport From: sip:6097@192.168.21.102mailto:sip%3A6097@192.168.21.102 ;tag=1857098215 To: sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102 Contact: sip:6097@192.168.21.193:52933 ;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel Call-ID: b9453704-d76a-b8ce-3247-c999abff7395 CSeq: 324677463 INVITE Content-Type: application/sdp Content-Length: 588 Max-Forwards: 70 Route: sip:192.168.21.102:5060;lr;transport=udp Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel v=0 o=doubango 1983 678901 IN IP4 192.168.21.193 s=- c=IN IP4 192.168.21.193 t=0 0 m=audio 36372 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 m=video 59296 RTP/AVP 125 106 121 103 a=rtpmap:125 VP8/9 a=fmtp:125 CIF=2;QCIF=2;SQCIF=2 a=rtpmap:106 H264/9 a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; max-mbps=11880 a=rtpmap:121 MP4V-ES/9 a=fmtp:121 profile-level-id=3 a=rtpmap:103 H263-1998/9 a=fmtp:103 CIF=2;QCIF=2;SQCIF=2 when I make Audio call requests I dont have the video part but at receiver since two clients can make video call they have Asterisks adds the Video Part in request sent to receiver,I dont want that part added , how I can delete it ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk as a softphone
Hi, one reason for having that robotic voice could be improper codecs others include low CPU processing power, memory not free etc. I once had the same kind of issue with VAD(voice activity detection) turned ON from my service providers equipment so my asterisk was performing poorly with VAD. Asterisk version and its codec play more important role. Regards, Sammy On Tue, Jan 3, 2012 at 6:34 PM, Christian Jaeger chr...@gmail.com wrote: Hello I'm using softphones as my only 'landline' phone service for almost 3 years now (Diamondcard and now voip.ms), so far using SIP (and mostly Twinkle). Also, I'm using Linux (Debian) as my choice of desktop OS. Also, sometimes I'm in networks with badly behaving NAT routers (for some time I used openvpn to solve this unreliably, then I ended up using 3G instead of wifi while in Canada, but now I'm abroad and don't have 3G). I'm now sufficiently fed up with SIP to give IAX2 another try. I want a softphone solution that: * works on Linux (Debian) * works reliably (e.g. remain connected for incoming calls, work with shitty NAT routers) * preferably encrypts both signalling and voice (dunno if voip.ms supports it, I might use a proxy asterisk instance on an own server instead) * properly handles audio with the 8000 samples/second dictated by the POTS systems (ALSA combined with some hardware (like both of my laptops) doesn't do proper lowpass filtering for mic input, so I will have to either use OSS or PulseAudio or rely on Asterisk doing proper downsampling in software). Asterisk seems to fit the first three; I'll happily build a GUI on top if this turns out to be a stable solution. My problems right now: - when I issue console dial without a number, it plays a recording with a woman's voice, and I can understand what is being said, but it sounds very garbled, like modulated with some about 20 Hz signal (a bit like a robot voice). What could be the problem? (Not using pulseaudio; +- default configuration.) One hypothesis I have is that it uses a too small buffer somewhere. - I don't understand how the extensions stuff is working. voip.ms wiki told me to create sections named [voipms], but how do I switch to 'default'? tie*CLI console dial 4443 No such extension '4443' in context 'default' tie*CLI console dial 04443 No such extension '04443' in context 'default' tie*CLI console dial 004443 No such extension '004443' in context 'default' - I haven't found anyone in google who tried to do the same as me, except http://www.junghanns.net/en/asteriskassoftphone.html but that doesn't lead me far (and the patch linked is unavailabe). Has anyone here done what I envision, or seen some docs specifically matching my use case? Thanks Christian. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] From address missing 'sip:', using it anyway
Hi, The server or client application that is sending you sip packets is missing the sip: string in from header. You should have it sorted out because if that header goes to some external equipment the call may fail because of this. Regards, Sammy On Thu, Jan 5, 2012 at 12:44 AM, motty.cruz motty.c...@gmail.com wrote: Hello, I see the following error in the logs [Jan 4 11:37:35] NOTICE[21]: chan_sip.c:15388 check_user_full: From address missing 'sip:', using it anyway Does anybody know how to stop this error? It does not seem to be affecting performance on the Asterisk 1.8.4.1 running on Centos Linux 2.6. I have google it but empty! Thanks, Celso -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT/IPTABLES workarounds
Are you talking about having an SSH tunnel and route your SIP traffic through it !!? On Thu, Jan 5, 2012 at 4:20 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/03/2012 10:03 AM, Patrick Lists wrote: On 03-01-12 16:24, Danny Nicholas wrote: Hello List, I work in an environment where I have to request IPTABLES changes rather than doing them myself. Is there a way to utilize my SSH (port 22) to get a functional (and with good sound) Asterisk installation with multiple channels up without requesting the 5060(SIP) 5061 (TLS) and UDP/RTP (usually 10001-2) IPTABLES allowances? Not with SIP as it needs a port for signaling (usually 5060) and RTP ports for sending the actual voice packets. So for SIP you will always need multiple ports. If you can use IAX then you could use port 22 as IAX only needs one port. The question is how are you going to SSH into the box if you use the SSH port for Asterisk? It is not practical (although not impossible) to run UDP over an SSH tunnel. Since VoIP media is generally transported over UDP, this will be a major obstacle. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
Hi, For such call you just need to select the outbound codec before the dial() app. choose the audio-only codecs and thus no video codec strings will be exchanged in that call. -- Regards, Sammy On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib fkha...@iconnecths.com wrote: this is what my SIP Invite message when I make Video call INVITE sip:6500@192.168.21.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport From: sip:6097@192.168.21.102;tag=1857098215 To: sip:6500@192.168.21.102 Contact: sip:6097@192.168.21.193:52933 ;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel Call-ID: b9453704-d76a-b8ce-3247-c999abff7395 CSeq: 324677463 INVITE Content-Type: application/sdp Content-Length: 588 Max-Forwards: 70 Route: sip:192.168.21.102:5060;lr;transport=udp Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel v=0 o=doubango 1983 678901 IN IP4 192.168.21.193 s=- c=IN IP4 192.168.21.193 t=0 0 m=audio 36372 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 m=video 59296 RTP/AVP 125 106 121 103 a=rtpmap:125 VP8/9 a=fmtp:125 CIF=2;QCIF=2;SQCIF=2 a=rtpmap:106 H264/9 a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; max-mbps=11880 a=rtpmap:121 MP4V-ES/9 a=fmtp:121 profile-level-id=3 a=rtpmap:103 H263-1998/9 a=fmtp:103 CIF=2;QCIF=2;SQCIF=2 when I make Audio call requests I dont have the video part but at receiver since two clients can make video call they have Asterisks adds the Video Part in request sent to receiver,I dont want that part added , how I can delete it ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Testing software to test IVR system
Easy, use Read() to capture the incoming DTMF from Server-B
Server-A Server-B
Initiate-Call - AnswerCall()
SendDTMF(5)-- Read()
Read()-SendDTMF(4)
SendDTMF(3)-- Read()
Read()-SendDTMF(2)
SendDTMF(1)-- Read()
Put proper GOTOIFs after reads if you like.
--
Regards,
Sammy
On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati virbh...@gmail.com wrote:
I originate calls from .call file and 1 channel I have at A server A and
another channel at B server.
*A server code is below:-*
exten = 43689956,1,Answer()
same = n,Wait(5)
same = n,SendDTMF(1)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,wait(2)
same = n,SendDTMF(123456789012345#)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,Hangup()
_ _
| A server | ___DTMF Send_= | B server |
|_| =--- Responce - |_|
*B server code is below:-*
At B server call come to 201 extension which is mention here..
exten = _20[1-6],1,Answer()
same = n,Ringing()
same = n,wait(2)
same = n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
same = n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
$[${EXTEN}=205] ||
$[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
same = n,Hangup()
Now I can send the DTMF from A to B. But How I will get the responce at
server A. I checked all the channels variable but they didn't reply status
of B server channel. All information I will get of server A. Main problem
is that control reach to AGI and then I don't have any rights to do any
update or modification on AGI. So if I can work on request and responce
then it will be the last solution as per my knowledge.
Is this possible with the dialplan or I am just westing time?
On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-12-28 03:25 AM, virendra bhati wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used
it?
like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as
per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same = n,Read(value,**pleasePress1forSupportPress2fo**
rHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(**suppot,1))
same = n,ExecIf($[${value}=2]?Goto(**help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
We have DTMF based tests for the testsuite[1] that you could use.
[1]
http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/http://svn.asterisk.org/svn/testsuite/asterisk/trunk/
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
--
__**__**_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
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Re: [asterisk-users] DTMF Testing software to test IVR system
o in that case you need to observer the call flow in Server-B, i.e what is
the length of sound file playing. what DTMF it requires etc etc and once
you detect the call flow for a successful IVR traversal then mimic the
behaviour of the call from Server-A.
Thats all you can do.
Think of it exactly the same as Answering Machine Detection Algorithm, but
in your case its like Server-B Detection Algorithm :)
--
Regards,
Sammy
On Thu, Dec 29, 2011 at 2:15 PM, virendra bhati virbh...@gmail.com wrote:
In server B if I use SendDTMF then it means I am changing programming at
server B. Actually I don't have right or permission to change programming
in server B.
otherwise your suggestion is best for channel base communication.
On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind govoi...@gmail.com wrote:
Easy, use Read() to capture the incoming DTMF from Server-B
Server-A Server-B
Initiate-Call - AnswerCall()
SendDTMF(5)-- Read()
Read()-SendDTMF(4)
SendDTMF(3)-- Read()
Read()-SendDTMF(2)
SendDTMF(1)-- Read()
Put proper GOTOIFs after reads if you like.
--
Regards,
Sammy
On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati virbh...@gmail.comwrote:
I originate calls from .call file and 1 channel I have at A server A and
another channel at B server.
*A server code is below:-*
exten = 43689956,1,Answer()
same = n,Wait(5)
same = n,SendDTMF(1)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,wait(2)
same = n,SendDTMF(123456789012345#)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,Hangup()
_ _
| A server | ___DTMF Send_= | B server |
|_| =--- Responce - |_|
*B server code is below:-*
At B server call come to 201 extension which is mention here..
exten = _20[1-6],1,Answer()
same = n,Ringing()
same = n,wait(2)
same = n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
same = n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
$[${EXTEN}=205] ||
$[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
same = n,Hangup()
Now I can send the DTMF from A to B. But How I will get the responce at
server A. I checked all the channels variable but they didn't reply status
of B server channel. All information I will get of server A. Main problem
is that control reach to AGI and then I don't have any rights to do any
update or modification on AGI. So if I can work on request and responce
then it will be the last solution as per my knowledge.
Is this possible with the dialplan or I am just westing time?
On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger
pabelan...@digium.comwrote:
On 11-12-28 03:25 AM, virendra bhati wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and
then
dialplan(IVR system) gets DTMF from it. I mean to say that
automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can
used it?
like A script make the call by using localdail, .call file or any
method.
And after landing the call we send dtmf to IVR system automatically as
per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same = n,Read(value,**pleasePress1forSupportPress2fo**
rHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(**suppot,1))
same = n,ExecIf($[${value}=2]?Goto(**help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
We have DTMF based tests for the testsuite[1] that you could use.
[1]
http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/http://svn.asterisk.org/svn/testsuite/asterisk/trunk/
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
--
__**__**
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
Re: [asterisk-users] DTMF Testing software to test IVR system
Hi,
You can use combination of SendDTMF() and wait() in such a way that you
traverse through the IVR tree just as Satish mentioned.
SendDTMF(1)
Wait(3)
SendDTMF(2)
Wait(2)
SendDTMF(5678123490)
See also:
*WaitForNoise()* , WaitForSilence(), AMD()
Regards,
Sammy.
On Wed, Dec 28, 2011 at 2:32 PM, virendra bhati virbh...@gmail.com wrote:
Hi Satish,
Thank you Satish. I did the same before your e-mail i saw. But i got
another issue in such case.
DTMF is passed to that channels but in case I will make the complete IVR
system for calling server end. and which become so complected to do it.
Is there any alternate way by which I get the response and send DTMF only.
So that complete IVR flow willn't be required to implement at originator
server.
On Wed, Dec 28, 2011 at 2:50 PM, Satish Barot
satish4aster...@gmail.comwrote:
Create a callfile with local channel and once first call leg is answered,
use wait() and senddtmf() application on second call leg.
CALLFILE sample:
Channel: LOCAL/1234\@test_ivr
Context: senddtmf
Extension: s
Priority: 1
Extensions.conf sample:
;-- FIRST LEG CALL --;
[test_ivr]
exten = 1234,1,Answer()
same = n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
;--SECOND LEG CALL --;
[senddtmf]
exten = s,1,Noop(# TEST:IVR ##)
; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same = n,Wait(10)
same = n,SendDTMF(1)
--SATISH BAROT
On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati virbh...@gmail.comwrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used
it?
like A script make the call by using localdail, .call file or any
method. And after landing the call we send dtmf to IVR system automatically
as per my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same =
n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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Thanks and regards
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+91-8885268942
Software Engineer
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Re: [asterisk-users] how to used SIPp for sip load testing
Hi, as the Logs say clearly you need to create an extension in default context named service [default] . exten = service,1,NOOP(Incoming call from SIPp) . Regards, Sammy On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati virbh...@gmail.com wrote: Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. haddock8-astrx*CLI -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] execute command just after Dial()
Hi,
Please see the Dial application documents from CLI, i.e core show
application dial. There is an option which will let you continue in the
DIal-plan after the Dial command on hangup.
Regards,
Sammy.
On Fri, Dec 23, 2011 at 5:54 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:
Hello,
I'm using AGI scripting with asterisk and need to execute certain commands
just after Dial(). But once dial command is executed, further
commands/instructions are ignored.
$agi-exec(Dial,SIP/100);
$dialstatus = $agi - get_variable(DIALSTATUS);
if($dialstatus[data]==ANSWER)
{
do something...
}
thanks,
Kamlesh
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Re: [asterisk-users] Asterisk call file size calculation
Hi, STAT function can give you size of a file ( http://www.voip-info.org/wiki/view/Asterisk+func+stat) - Codecs do effect the call file size, you can see the size difference in case of a gsm and a wav recorded call. Regards, Sammy On Wed, Dec 21, 2011 at 6:41 PM, silent sayz silent.s...@gmail.com wrote: Hello, Can i get some figures about the file size of the call that is recorded by asterisk. i.e the exact figures and does the codec effect the file size ? Thanks in advance Good luck -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get start-time of all active calls
Hi, Not sure why you didnt get it, when I did thta command for originator channel it showed me the CDR variables list which included CDR Variables: level 1: dnid= level 1: clid=XXX level 1: src= level 1: dst= level 1: dcontext=SIP-incoming level 1: channel= level 1: dstchannel= level 1: lastapp=Dial level 1: lastdata=SIP/ *level 1: start=2011-12-14 09:15:54* level 1: answer=2011-12-14 09:16:01 level 1: duration=11 level 1: billsec=4 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1323854154.856 level 1: linkedid=1323854154.856 level 1: sequence=1096 Thats valid for an ongoing bridged call-initiator side only. Regards, Sammy On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, 'sip show channel' also does not give this info. sip show channel f600ed29f561d57 localhost*CLI * SIP CallI Curr. trans. direction: Incoming Call-ID:f600ed29f561d57f Owner channel ID: SIP/100- Our Codec Capability: 14 Non-Codec Capability (DTMF): 1 Their Codec Capability: 302 Joint Codec Capability: 14 Format: 0x2 (gsm) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open NAT Support:Always Audio IP: xxx.xxx.xxx.xxx (local) Our Tag:as2a60820a Their Tag: 1b7d6a7d SIP User agent: eyeBeam release 3007n stamp 17816 Username: 10036 Peername: 10036 Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 Caller-ID: 100 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:1...@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive regards, Kamlesh -- Date: Wed, 14 Dec 2011 12:43:14 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get start-time of all active calls
oops, you got it. On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield t...@softins.co.ukwrote: In article CAJUJwthT= mpyxq+omt5hrextl1iqvd0kbs+jhtqlvsqscay...@mail.gmail.com, Sammy Govind govoi...@gmail.com wrote: Hi, Not sure why you didnt get it, when I did thta command for originator channel it showed me the CDR variables list which included That's from show channel, not sip show channel. Cheers Tony CDR Variables: level 1: dnid= level 1: clid=XXX level 1: src= level 1: dst= level 1: dcontext=SIP-incoming level 1: channel= level 1: dstchannel= level 1: lastapp=Dial level 1: lastdata=SIP/ *level 1: start=2011-12-14 09:15:54* level 1: answer=2011-12-14 09:16:01 level 1: duration=11 level 1: billsec=4 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1323854154.856 level 1: linkedid=1323854154.856 level 1: sequence=1096 Thats valid for an ongoing bridged call-initiator side only. Regards, Sammy On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, 'sip show channel' also does not give this info. sip show channel f600ed29f561d57 localhost*CLI * SIP CallI Curr. trans. direction: Incoming Call-ID:f600ed29f561d57f Owner channel ID: SIP/100- Our Codec Capability: 14 Non-Codec Capability (DTMF): 1 Their Codec Capability: 302 Joint Codec Capability: 14 Format: 0x2 (gsm) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:xxx.xxx.xxx.xxx:5060 Received Address: xxx.xxx.xxx.xxx:5060 SIP Transfer mode: open NAT Support:Always Audio IP: xxx.xxx.xxx.xxx (local) Our Tag:as2a60820a Their Tag: 1b7d6a7d SIP User agent: eyeBeam release 3007n stamp 17816 Username: 10036 Peername: 10036 Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 Caller-ID: 100 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:1...@xxx.xxx.xxx.xxx:5060 DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive regards, Kamlesh -- Date: Wed, 14 Dec 2011 12:43:14 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=- -=-=-=-=-=- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=- -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org
Re: [asterisk-users] get start-time of all active calls
Hi, I think you need to use the command sip show channel channel-id Regards, Sammy On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which port should be open for asterisk communication
Hi, That depends on what else your asterisk is doing i.e if an AMI-based code is running then AMI port needs to be open as well. It also depends what other appliactions are running on asterisk-box which require port opening i.e apache or mysql etc. Regards, Sammy On Mon, Dec 12, 2011 at 3:21 PM, virendra bhati virbh...@gmail.com wrote: Hi List, Please tell me which ports should be required open for communication with asterisk. like 5060 for sip calls, 4569 for IAX, 10,000 to 20,000.. Apart from these ports what else is required ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple route failover zaps registration
Hi, I'm only going to rephrase what James said, shorten the registration expiration timer and retry timers. That way phones will retry registrations lets say after 1 min so after 1 min all phones will failover to the secondary SRV record. Regards, Sammy On Mon, Dec 12, 2011 at 10:35 AM, Mike Diehl mdi...@diehlnet.com wrote: Actually, I've configured the phones to use DNS SRV records to find the Asterisk server, and this works very well. The problem is that when the router fails over, the phones IP address changes and this causes them to be unavailable from Asterisk's point of view. On Sunday 11 December 2011 10:02:21 pm Faisal Hanif wrote: Why don't you use FQDN in phone instead of IP of server and configure DNS Server to failover resolve to next IP while set SIP reg expiry same as DNS TTL. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, December 12, 2011 5:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Multiple route failover zaps registration Hi all, I've got a customer who is bringing up a second Internet connection for fail- over. I've configured a WRT54 with 2 LAN ports and arranged for it to fail over when one of the routes is no longer available. That works just fine at the IP level. However, when the router fails over, the phones lose their registration, presumably because their IP address has changed from Asterisk's point of view. The phones happen to be Polycom 335's, and I'm running Asterisk 1.6.2.9. What is the best way to manage this situation so that the phones don't become unavailable during failover? I'm considering using the Tinc VPN solution to prevent the IP address from chaing, but I'm hoping for a more simple solution. Any ideas? -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make app_meetme enable
Install DAHDI then !!? On Thu, Dec 8, 2011 at 12:46 PM, Durgesh Mishra durgesh.mis...@rancoretech.com wrote: In make menuselect =application=XXX app_meetme . I am doing confrence call using sip softphone. I checked It Depends on: dahdi(E) . How I can do app_meetme enable? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Hints in asterisk
Hello, AFAIK Hints are used for looking out for a device state before actually doing anything. Hints can be used from Dial-plan, AMI, or AGI. An example can be to look for state of a SIP user. Read these links for better understanding. http://www.smartvox.co.uk/astfaq_subscribe_hints.htm http://www.voip-info.org/wiki/view/Asterisk+standard+extensions Regards, Sammy. On Tue, Dec 6, 2011 at 5:14 PM, virendra bhati virbh...@gmail.com wrote: Hi All, I did some google and found some documents on that and finally I got some response from asterisk . Below is the CLI output of my google. *haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:IdleWatchers 0 1 hint matching extension 2218 == Using SIP RTP CoS mark 5 -- Executing [222@bhati-test:1] NoOp(SIP/2218-02c3, Call from Gtalk ) in new stack -- Executing [222@bhati-test:2] NoOp(SIP/2218-02c3, Hint for Extension 2218 is ) in new stack -- Executing [222@bhati-test:3] Set(SIP/2218-02c3, CALLERID(name)=From Google Talk) in new stack -- Executing [222@bhati-test:4] Wait(SIP/2218-02c3, 10) in new stack haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:InUse Watchers 0 1 hint matching extension 2218 -- Executing [222@bhati-test:5] Dial(SIP/2218-02c3, SIP/my_sip_phones) in new stack == Using SIP RTP CoS mark 5 [Dec 6 17:39:38] WARNING[6689]: chan_sip.c:5331 create_addr: No such host: my_sip_phones [Dec 6 17:39:38] WARNING[6689]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/2218-02c3' status is 'CHANUNAVAIL' -- Executing [h@bhati-test:1] NoOp(SIP/2218-02c3, hangup the call now) in new stack haddock8-astrx*CLI core show hint 2218 2218@bhati-subscribe : SIP/2218 State:Idle Watchers 0 1 hint matching extension 2218 * *Is this the right way to use HINT of asterisk ?* On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.com wrote: Hi All, I read about the *Hint* in asterisk. I want to implements into my server for testing purpose. How to use it ? please help me... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk registrations by SER proxy
Hi again, Asterisk could be aware of the registrations if the sipusers table is shared with asterisk sip realtime, but then again the issue would remain the same that asterisk want to authenticate the sip peer from scratch..maybe try some Realtime configurations in sip.conf to avoid authentications of clients having active register-expiry timer. Already answered you prev. in other list to define a sip section for your opensips Regards, Sammy. On Mon, Dec 5, 2011 at 7:41 PM, Matt Hamilton mistral9...@hotmail.comwrote: I integrated Opensips with Asterisk Realtime (Asterisk sipusers/peers point to Opensips subscribe table via a view). Opensips handles the registrations. However, when a call comes in (INVITE is routed to Asterisk), it seems like Asterisk doesn't know about the user (or sees the users as not authorized), so can't create the SIP channel. (I use queues and conferencing also.) If I route the REGISTER to Asterisk after authorizing in Opensips, Asterisk does the authorization/registration again from scratch. In that case call goes through, but I end up duplicating the authorization process. I was hoping to take the load of handling registrations from Asterisk. I know this is a very common scenario, but I'm not very clear about the process. Is it possible to make Asterisk be aware of those registrations made by the proxy server? Thanks, Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to program a 100ms delay between the ringing of queued calls w/ ringall
Hi, I dont think that 2 Queue commands would help, also wrapup time is for an putting delay in an agent who just answered the call and hungup. I think for this purpose you may need to open up the source code for queue and put some delay in the relevant code. Regards, Sammy. On Mon, Dec 5, 2011 at 6:56 PM, Scott Gifford sgiff...@suspectclass.comwrote: On Tue, Nov 22, 2011 at 5:34 PM, Douglas Mortensen d...@impalanetworks.com wrote: Hello, ** ** Does anyone have any idea of how I can program a 100ms delay in between the ringing of 2 subsequent calls in a queue configured with a ringall strategy? Does the wrapuptime queue option do what you want? http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf -Scott. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to download sample video file for Asterisk 1.8x playback?
Hi, 1- Are you sure Playback is capable of understanding/playing video files. 2- Make sure you've enabled videosupport in sip [general] and also allowed h264,h263 in the sip users section trying to execute this playback app. Regards, Sammy. On Sat, Dec 3, 2011 at 4:13 AM, asterisk jobs asteriskcod...@gmail.comwrote: Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4 (ulaw)): No such file or directory The file of course exists and it's chowned to asterisk.asterisk. I think it's a file format issue. So, I appreciate it someone can give me a link to a file or maybe point me a universal convertor (open-source or linux based software) that can convert my videos to Asterisk readable format. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS Values
Hi, How are you calling this AGI in your dialplan !!? Regards, Sammy. On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 *SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1* ** Please help me in this. Thanks, Kamlesh ** * * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS Values
Can you also paste the Asterisk Console logs around the part where AGI is dialing and after the dialing part ! make sure AGi debug is enabled as well. On Fri, Dec 2, 2011 at 4:24 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, in /etc/extension.conf [privoip] exten = _00X.,n,AGI(isdcall.php) Regards, Kamlesh -- Date: Fri, 2 Dec 2011 16:16:27 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS Values Hi, How are you calling this AGI in your dialplan !!? Regards, Sammy. On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 *SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1* ** Please help me in this. Thanks, Kamlesh ** * * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk Support SIP Video Call ?
Hey, Did you try google.com for this! I've done this several times now. Video for one-to-one call works if H264 is supported at both end points. All you need to do is enable video in sip.conf and set allow=h264 in the sip peers with video capability. You may need to see if your asterisk has h264 compiled on it. Regards, Sammy On Wed, Nov 16, 2011 at 2:23 PM, Faraj Khasib fkha...@iconnecths.comwrote: Hi all, I tried making a video SIP call using Asterisk But it didnt workonly voice call works? Regards Faraj Khasib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
Yes, Skype was a good thing. R.I.P On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote: Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit monthly calls by context
I'd say try a2billing- thats abit of an overkill for just this functionality but you'll get lot or options to play with there. On Wed, Nov 16, 2011 at 7:02 PM, ad...@3a.hu wrote: Hello Hans, On 11-16-2011 14:46, Hans Goossen wrote: I guess some billing solution can do the trick, but I think it's too much for that little. I don't need any other feature. i would create a macro which calls an agi. The agi searches the CDR table (mine is in sql) and calculates if the call can go through. Then i'd call this macro from every extension in the dial plan just before the dial cmd. I was thinking something like checking the CDR before make the call, I know it may permit some extra minutes to be used, but it really doesn't need to be that exact. A couple of extra minutes won't hurt. It depends on the number of simultaneous calls from within the same context. The agi can return a number of seconds (calculated from sql) which the dial cmd can use as an absolute limit and after that amount of seconds it can hang up the call (see S or L flags). regards adam -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
can I make call without registration to an registered SIP account? -- Yes, you can but first you need to set allowguest=yes in sip.conf (makes ur server insecure) I guess you can put in same user/sip account in all iphones and like (in x-lite) don't let the phones register to server rather set the server IP as outbound proxy. /Sammy On Tue, Nov 15, 2011 at 7:40 PM, Faraj Khasib fkha...@iconnecths.comwrote: btw the call is one direction from clients to Call center My question can be rephrased can I make call without registration to an registered SIP account? From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [ fkha...@iconnecths.com] Sent: Tuesday, November 15, 2011 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple SIP endpoint registrations I have phone system and I am connecting Asterisk to it trunk. Now I want my iphone users (clients ) to call my call center which is in phone system by using the same SIP account the user will call asterik with for example 6000 as account then the asterik will forward the call via trunk to that Phone system. My question is this : Can all my iPhone users which are using the 6000 as an account call the call center ? with asterisk 1.7? From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming [ kpflem...@digium.com] Sent: Tuesday, November 15, 2011 8:25 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP endpoint registrations On 11/15/2011 07:28 AM, Faraj Khasib wrote: Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center) Now My question is about the iPhone user part... Does the Asterisk 1.8 support that all my iPhone users register with the same account(6000@mydomain) and call that extension(dont worry about this extension)? No Asterisk does not support multiple registrations to the same SIP account (AoR), but that is irrelevant in this case, because registrations are not used for placing calls *to* Asterisk, only receiving calls *from* Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More than one route to a destination
Hi, Can I use 2 SIP Trunks from each remote offices to the central site and permit 2 simultaneous calls across the SIP trunk that passes over the smaller line, and permit 10 simultaneous calls across the larger link? Yes. I also wish to have priorities, so that more important calls are sent over the smaller link (but more reliable) and the larger link used for less important calls. 1- find out the criteria for Imp calls and write dialplan to use the reliable link and use other SIP trunk otherwise. Can you do this priority based on the user ID of the caller? Yes. For any outbound call see who is the caller and if CALLERID(num) matches use desired link. If a user with a SIP client starts off in remote office1, and then moves to remote office4, can then keep the same phone number? AFAIK, you need to use DUNDI between the Asterisk Servers on top of SIP trunks. Once DUNDI is setup your users can move between offices and have just one extension. Regards, Sammy On Tue, Nov 15, 2011 at 8:12 PM, James Courtier-Dutton james.dut...@gmail.com wrote: Hi, I have a setup with 5 remote offices, each having a Asterisk PBX. I then have a central office, also with an Asterisk PBX. The remote offices have 2 links to the central office, a large link, and a smaller, but more reliable link. Unfortunately, using IAX is not an option for me. Can I use 2 SIP Trunks from each remote offices to the central site and permit 2 simultaneous calls across the SIP trunk that passes over the smaller line, and permit 10 simultaneous calls across the larger link? I also wish to have priorities, so that more important calls are sent over the smaller link (but more reliable) and the larger link used for less important calls. Can you do this priority based on the user ID of the caller? Another question: If a user with a SIP client starts off in remote office1, and then moves to remote office4, can then keep the same phone number? Kind Regards James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
Hey, I haven't thoroughly read the whole of your reply- just a quick answer to your timers question-generally I think you're right. Those timers are property of UAC so you may need to look into the phone configurations. I'd CISCO 79X0 phones and we wanted those to refresh their registrations at very short intervals of time as well as the INVITES timers was reduced too,...umm..I think that was for DNS-SRV based failovers. Though reducing the default timers from UAC heavily increased SIP traffic but we achieved the target by reducing the SIP timers in all phones. So that was an example. When you are using Asterisk as UAC to register onto another SIP server you can change the registration timeout and retry variables..and yes you can change these SIP timers in Asterisk sip.conf but thats not recommended.(see sip.conf.sample for details too) PS: with a quick look at sip.conf.sample + voip-info.org sip.conf details + google you can find lot more information than what you've collected so far. -- BR, Sammy On Wed, Nov 16, 2011 at 6:11 AM, Douglas Mortensen d...@impalanetworks.comwrote: OK. Thanks everyone for the responses. If I can summarize, I think here’s what’s been discussed: ** ** Asterisk becomes aware of SIP extensions/peers, as soon as they register.* *** ** ** Regarding how asterisk becomes aware of (or determines) that they are unavailable/unreachable, I believe I am hearing two possible scenarios:*** * ** ** **1. **“The Interval of Registration”. So asterisk has a timeout value that it is expecting the phone to reregister within. If the phone does not reregister within the timeout period, then asterisk determines that the extension/peer is no longer available. A few questions I have on this are: **a. **Where does this “timeout” interval come from? Is it a configuration parameter that we configure asterisk with, or is it something that is dynamically determined, or is it something that the phone/peer actually dictates to asterisk? **b. **If it is an asterisk configuration parameter, where does it exist (how do I set it confirm what it is currently set to)? It is a per-extension/peer setting, or is it global? **c. **Is there a command I can issue from the asterisk CLI to query it? **2. **“qualify=yes” can be configured for any given SIP peer in asterisk. This will send a SIP OPTIONS message/packet to the peer every 1 or 2 minutes (depending on the configuration) that probes the peer to confirm it is still online. The keepalives (SIP OPTIONS packets) are actually sent from asterisk to the SIP peer, correct? But then the SIP peer actually has to respond to each one with its own SIP packet, correct? With this scenario, asterisk will still utilize scenario 1 (reregistration) as a means of determining that the peer is available, but additionally will continue to monitor the peer constantly (every 1-2 seconds) via these keepalives? This way asterisk is able to have a much more rapid discovery of peers that become unavailable (because they are literally no longer reachable, as they’re no longer responding to the keepalives), correct? So my next questions are: **a. **Am I wrong with any of the above interpretations of the explanations you guys have given? **b. **Is the “no-reply” timer Sammy mentioned [(max time)x(max retries)] a parameter that can be set within asterisk? If so, what are the corresponding configuration parameters called? If not, what are the “max time” and “max retries” values? **c. **Is the SIP response the peer is supposed to give also an OPTIONS packet or something else? ** ** Thanks a LOT! I really appreciate all of the input insight you guys bring! ** ** - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 ** ** *From:* Sammy Govind [mailto:govoi...@gmail.com] *Sent:* Monday, November 14, 2011 10:36 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How do extensions stay registered ** ** Continuing eherr here, behind the OPTIONS messages(infact all SIP comm) you definitely to look into SIP timers which tell how many time to resend a packet if no response is received and for how long to wait before thinking that the SIP packet got lost(network disconnected or end-point lost) ** ** so, qualify=yes a peer means to send-keep alives and have the NAT mechanism stay active, as soon as the SIP keep-alive packets reach a no-reply (max time)x(max retries) Asterisk marks the peer as UNREACHABLE.* *** ** ** qualify=no wouldn't do all of the above. ** ** Another interesting thing to know is that SIP end-points have registrations time-out and refresh Registration timers as well. So if everything is going well, SIP end-points refresh their registration after some defined time. ** ** On Tue, Nov 15, 2011
Re: [asterisk-users] How do extensions stay registered
Continuing eherr here, behind the OPTIONS messages(infact all SIP comm) you definitely to look into SIP timers which tell how many time to resend a packet if no response is received and for how long to wait before thinking that the SIP packet got lost(network disconnected or end-point lost) so, qualify=yes a peer means to send-keep alives and have the NAT mechanism stay active, as soon as the SIP keep-alive packets reach a no-reply (max time)x(max retries) Asterisk marks the peer as UNREACHABLE. qualify=no wouldn't do all of the above. Another interesting thing to know is that SIP end-points have registrations time-out and refresh Registration timers as well. So if everything is going well, SIP end-points refresh their registration after some defined time. On Tue, Nov 15, 2011 at 3:35 AM, eherr email.eherr9...@gmail.com wrote: I think the wrap up answer is the interval of registration compacted, if used, with the SIP OPTION packet. ** ** I like the SIP OPTION packet because we have scripts to monitor the status and lets us know when a phone is up or down. ** ** --E ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Carlos Alvarez *Sent:* Monday, November 14, 2011 5:30 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How do extensions stay registered ** ** I think the registration part was answered. The de-registration part is different. If the phone is gracefully taken off line it specifically de-registers. If it just can't be reached because it powers off or the router closes NAT, or whatever, then Asterisk won't know this until it times out. ** ** On Mon, Nov 14, 2011 at 3:19 PM, eherr email.eherr9...@gmail.com wrote:* *** I think the question is more along the lines of how does asterisk know immediately when a sip phone becomes on line and when you unplug the phone from the network, how does asterisk essentially know immediately that it status is “UNKNOWN” If I am not mistaken. --E *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Monday, November 14, 2011 5:01 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] How do extensions stay registered “Extensions” do not register – peers do. A peer can register itself or be registered by Asterisk. In most cases the “extension” is equivalent to the “peer” (301 = 301) but it can be quite different (301 = sipuser1) or (301 = d...@impalanetworks.com). *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Douglas Mortensen *Sent:* Monday, November 14, 2011 3:52 PM *To:* 'asterisk-users@lists.digium.com' *Subject:* [asterisk-users] How do extensions stay registered I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? Do the extensions simply register repeatedly as a means of telling asterisk “I’m still here”, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former.* *** But am I oversimplifying it? Is there more to the process? Thanks, - Doug Mortensen Network Consultant *Impala Networks Inc* CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** -- Carlos Alvarez TelEvolve 602-889-3003 ** ** ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] Calling an independent gateway from asterisk
Hey,
Though your requirements are unclear and below may not exactly fit your
specs unless you give some more usage details.
if your gateway requires no authentication, yes you can do this by writing
a dialplan extension like below
exten = calling-togw,1,NOOP(I'll be getting some variables from AMI caller)
same = n,DIAL(SIP/${CALLTHIS}@my-example.com)
Now, in the AMI script you need to do the following.
1- Connect to asterisk,
2- Set the variable CALLTHIS as the destination you want to dial-out
3- use the
Originate-AMIhttp://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originateaction
properly.
Regards,
Sammy
On Tue, Nov 15, 2011 at 11:01 AM, Amar Akshat amar.aks...@gmail.com wrote:
Hello,
I have a testing scenario at hand. I want to make a call from Asterisk
CLI or AMI to an external network gateway. Is this possible.
Let me explain the use case.
Asterisk server (say 192.168.5.10) has few registered endpoints or
softphone.
Now an external gateway (say my-example.com or XXX.XXX.XXX.XXX:5060),
listening for SIP invites, but this gateway is not registered with
Asterisk,
can I send out SIP invites (call) to this external gateway, without
having to register on Asterisk.
--
Thank you...
Amar Akshat
Please excuse any spelling mistakes, as this email was sent from a
not so good mobile device.
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Re: [asterisk-users] Call to Asterisk registered sofphone from an independent unregistered Endpoint
Hi,
The end-point which isn't registered in asterisk will hit the default
context in asterisk. This is the one which you've defined in sip.conf
general section i.e
[general]
...
context=my-context
Also, if your calls are successful from any unregistered endpoint then I
think you've enable allowguest in sip.conf.
So if you need to bridge the call to 1234 extension make sure you've a
dialplan like this in extensions.conf
[my-context]
exten = 1234,1,Dial(SIP/1234)
same = n,Hangup()
OR
exten = _X.,1,Dial(SIP/${EXTEN}) ;== Security Warning, don't use in
production server.
Hope this helps,
--
Regards,
Sammy
On Mon, Nov 14, 2011 at 8:25 AM, Amar Akshat amar.aks...@gmail.com wrote:
Hi,
I have an Endpoint written in C, which simply sends out SIP invite to
another endpoint and also sets up media session after the call is
initiated. Now I am using this endpoint to call to the Asterisk PBX.
And the call is successfull.
Now, I have a softphone registered with asterisk with extension 1234,
and I want to call that softphone from my external endpoint which is
not resgistered with Asterisk. So I am sending an invite to the SIP
URI
sip:1234@host-ip:port, however, this call does not ring the
Softphone with extension, and the call is auto answered by Asterisk.
How can I configure/enable Asterisk to forward that call to the
softphone, rather than answering itself.
--
Thank you...
Amar Akshat
Please excuse any spelling mistakes, as this email was sent from a
not so good mobile device.
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Re: [asterisk-users] Logging Specific Verbose Level To Seperate File
Hello, Reading about the application DumpChan() shows this: [Synopsis] Dump Info About The Calling Channel. [Description] Displays information on channel and listing of all channel variables. If level is specified, output is only displayed when the verbose level is currently set to that number or greater. [Syntax] DumpChan([level]) So in theory its just another Verbose output on CLI, you can separate Verbose logging to another file in logger.conf. Your verbose level is 1001 so whenever you set core set verbose 1001 this DumpChan() application will start dumping output in CLI and then fro there be logged in the Verbose logging file. I don't think this is exactly what you require. -- Regards, Sammy On Mon, Nov 14, 2011 at 3:19 AM, Tristram Cheer trist...@tristramcheer.comwrote: Hi All, Hopefully this is considered on-topic for this list. I'm using DumpChan(1001) in a Macro called debug in order to debug issues within the dialplan, I would like to dump this output to a file specifically for DumpChan output but I'm having issues with figuring out how to do this under logger.conf. Ideally I would like to put DumpChan into SQL using func_ODBC but it seems that you can't do this so runner up is a file. Anyone have any pointers on how to do this? I would like to log DumpChan output and only DumpChan output to a separate file. Cheers! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as SoftSwitch - Hardware
Hey Sunny, I think your initial post on what you're looking for don't really tells much. I think initially you were looking at a different architecture than now i.e Kamailio+RTPproxy, this changes a lot of things. If you dont want transcoding and thinking on using Kam+Rtpproxy then I think asterisk isn't required any more. If that's not the case then for 1200 CCs you'll be required to put in multiple asterisk servers behind Kamailio/RTpproxy Server. Share some more details and I'm expecting that your design is going to change. Regards. Sammy. On Tue, Nov 8, 2011 at 9:31 PM, Sunny no7f...@gmail.com wrote: Jeff, Kamailio + rtpproxy Do you know how to make these configuration work? I know this is not the best place to ask that question. Thanks, Sunny On 3 November 2011 19:09, Jeff Brower jbro...@signalogic.com wrote: Sunny- I was thinking in Kamailio, but this sip proxy handles only the SIP signalling traffic, no media processing. Kamailio + rtpproxy. -Jeff On 3 November 2011 17:07, Nick Khamis sym...@gmail.com wrote: Shouldn't you be using a Proxy? Nick. On Thu, Nov 3, 2011 at 1:04 PM, Sunny no7f...@gmail.com wrote: Hi list, Could anyone tell me what is the recommended hardware to a system for following configuration: SBC -- Asterisk (SS) -- Carrier GW Asterisk should work as a Class 4 SoftSwitch, with following functionalists: - Do the IP Authentication - All communications on RTP/G729 (no transcoding required) - Load of 1200 concurrent call sessions - No call routing required Thanks in advance, Sunny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem when exiting from record file function without pressing the escape digit
Hey,
How are you starting the recording? MixMonitor? or Monitor? or some option
in an application?
If you are using MixMonitor or anything alike then you should
StopMixMonitor when the call hits the h extension.
Paste your dialplan relevant to the recording scenario to suggest you
something better.
--
Regards,
Sammy
On Fri, Nov 4, 2011 at 12:57 PM, Yaprak Ayazoglu
yaprak.ayazo...@gmail.comwrote:
Hi everybody,
I've been working on a project which records the voice of the incoming
call.
I use record_file function of asterisk as described below:
RECORD FILE filename format escape digits timeout [offset samples]
[BEEP] [s=silence]
filename: record1
format: wav
escape digits: #
timeout: -1
offset samples: 0
BEEP: 1
silence: 3000
*Please read the scenario for an incoming call:*
*
*
- The user calls the asterisk
- The user talks on the telephone
- Ends the conversation without pressing the escape digit (#)
In this scenario, unfortunately, record file do not end automatically
but since the silence field is
3 seconds, the record file function waits for 3 seconds to end.
After record file function is ended, I listened the recorded file and I
heard a busy tone sound in this file as
beeep bp beeep continously.
*My question:*
*
*
Why do I need to hit the escape character ('#') to end the record file
function? Is there any way, that
asterisk shall detect that the caller has closed the telephone?
Regards.
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Re: [asterisk-users] problem when exiting from record file function without pressing the escape digit
Hello again,
Hmmm...So you are in the AGI, I'm not much into AGI stuff, but let me
reproduce this in my local env...BTW which asterisk version you are using !
--
Regards,
Sammy
On Fri, Nov 4, 2011 at 1:43 PM, Yaprak Ayazoglu
yaprak.ayazo...@gmail.comwrote:
Hi Sammy,
Sorry for the previous answer. I accidentally pressed the send button.
So, I'm re-sending this mail with
my additional information.
Thank you for the reply.
I'm not using MixMonitor or Monitor. I'm recording the file in the perl
script that I call in my dial plan.
When there is an incoming call, asterisk answers it and plays the
wellcome.wav file. Later on, the
recording.pl script is executed. In this script, I call the
AGI-record_file(...) function and I require that
the client must press the escape character ('#'). If the client presses
the escape character in his/her phone
wrong. If the client presses the end of call button in his/her phone,
record_file(...) function keeps
working 3 more seconds (because silence = 3) and exits this function.
When I listen to the recorded wav file, I hear the busy line sound(beeep
beeep beeep ...)
Isn't it possible for asterisk to understand the call is ended when the
telephone is closed.
My dialplan is as follows:
#-dialplan-
exten = 500,1,Answer()
exten = 500,2,Playback(wellcome) ; play the wellcome message
exten = 500,3,AGI(recording.pl) ; Do the echo test
exten = 500,4,Playback(demo-echodone) ; Let them know it's over
exten = 500,5,Hangup
#-EOF dialplan
The relevant part of the perl script is as follows:
#-recording.pl
$filename = 'recordedSound';
$format = 'wav';
$digits = '#';
$timeout = -1;
$beep = 1;
$offset = 0;
$silence = 3;
$AGI-record_file($filename, $format, $digits, $timeout, $beep, $offset,
$beep, $silence);
#-EOF recording.pl
On Fri, Nov 4, 2011 at 10:05 AM, Sammy Govind govoi...@gmail.com wrote:
Hey,
How are you starting the recording? MixMonitor? or Monitor? or some
option in an application?
If you are using MixMonitor or anything alike then you should
StopMixMonitor when the call hits the h extension.
Paste your dialplan relevant to the recording scenario to suggest you
something better.
--
Regards,
Sammy
On Fri, Nov 4, 2011 at 12:57 PM, Yaprak Ayazoglu
yaprak.ayazo...@gmail.com wrote:
Hi everybody,
I've been working on a project which records the voice of the incoming
call.
I use record_file function of asterisk as described below:
RECORD FILE filename format escape digits timeout [offset
samples] [BEEP] [s=silence]
filename: record1
format: wav
escape digits: #
timeout: -1
offset samples: 0
BEEP: 1
silence: 3000
*Please read the scenario for an incoming call:*
*
*
- The user calls the asterisk
- The user talks on the telephone
- Ends the conversation without pressing the escape digit (#)
In this scenario, unfortunately, record file do not end automatically
but since the silence field is
3 seconds, the record file function waits for 3 seconds to end.
After record file function is ended, I listened the recorded file and I
heard a busy tone sound in this file as
beeep bp beeep continously.
*My question:*
*
*
Why do I need to hit the escape character ('#') to end the record file
function? Is there any way, that
asterisk shall detect that the caller has closed the telephone?
Regards.
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Re: [asterisk-users] custom automated meeting
There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin directory by developing it yourself. On Tue, Nov 1, 2011 at 6:57 PM, Thanasis thana...@asyr.hopto.org wrote: on 11/01/2011 03:25 PM Danny Nicholas wrote the following: One way to do this (there are probably more and better ways). Incoming call to 123456789 launches meetme(1234,b(connecta.agi)) Connecta.agi calls lines B and C and connects them to meetme(1234). Thanks, but could you be more elaborate please? Where can I find connecta.agi ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanasis Sent: Tuesday, November 01, 2011 1:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] custom automated meeting I just want to make two specific sip phone sets to ring together, when someone dials a specific incoming extension. And then, when each of the ringed sets answers, to be placed immediately into meeting session with the caller together with the other phone set. Here is exactly what I mean: Person A dials 123456789. Asterisk routes the incoming call and rings sip phones B and C. Person B answers phone B and starts talking with person A, while phone C keeps ringing. A minute later, and while A and B are still talking together, person C answers phone C, and starts talking with A and B together (that is aromatically all being placed in the same conference session). Is that doable? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] custom automated meeting
Type in asterisk CLIcore show application meetme
or google asterisk cmd meetme simple?
On Tue, Nov 1, 2011 at 10:33 PM, Thanasis thana...@asyr.hopto.org wrote:
on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:
You need simple dialplan of four steps:
same =n,Set(conf_name=conf-${RAND(1,1000)})
same =n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same =n,MeetMe(${conf_name},dFI1xAC)
same =n,Noop(do post conference stuff)
Thanks!
What is the meaning of the options dFI1xAC passed to
app,MeetMe,${conf_name} ?
Where can I find them described please?
2011/10/31 Thanasis thana...@asyr.hopto.org:
I need your help in implementing the following scenario:
A certain extension will ring two sip phones simultaneously and when one
of them answers, the other keeps ringing until it answers too, and then
all three (the caller and the other two) are immediately placed in a
conference room (same room for all three).
Can we do it?
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Re: [asterisk-users] Queue announcements and MOH blanking out on calls from PSTN over IAX2
hmmm so IAX channel is playing with you guys. 1- Cant you guys use SIP, does this happen with SIP trunk as well !? 2- Which version of asterisk are there on both servers. 3- See the output of the command core show file versions in your both asterisk servers. Mainly lookout for IAX channel version. Also try enabling IAX debug and paste the output on console. 2011/10/30 Raj Mathur (राज माथुर) r...@linux-delhi.org On Sunday 30 Oct 2011, Raj Mathur (राज माथुर) wrote: After looking further, the problem seems to be purely in playing recorded messages over IAX2. Looking at the debug logs on the SIP server (which is playing the recorded messages) shows that it stops playing one of the messages at some point in the flow, and then never plays anything again. This seems to be very similar to: https://issues.asterisk.org/view.php?id=17232 except there is no virtualisation involved in the process -- everything is working on native hardware. It /is/ amd64 Debian Squeeze running on Intel, though. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tips best practices for asterisk troubleshooting parsing logs
Hi Douglas, You;re right, that method is useful only for one-to-one call but as soon as the call gets transferred etc etc as you mentioned everything will get mixed and confusing. Any way I this can be done? Can’t a call be passed off from one channel to another, which would leave me with only seeing a part of the logs for the life of the call if I only grep the logs based on one channel id? Yes, is the answer if you want this to implement. You need to do the following in order to achieve an start-end logging of a call. 1- As soon as call Enters Asterisk dialplan save its UNIQUEID (plus any other key i.e timestamp ) in a CALL-IDENTIFY variable. 2- Use the CALL-IDENTIFY variable throughout your dialplan contexts to verbose() useful information.(I saw a log() application in asterisk 1.8.5 to do this in a the log file...i.e print logs of your own) Another interesting thing for this purpose would be CEL, though it maynot be available in your older deployments. I haven't toyed around with CEL myself but so far I've the impression that its a very verbose form of CDRs. So using CEL to keep track of your call in a DB would help as well. Another Idea is to use the SIP-Header Call-ID as your CALL-IDENTIFY variable. This way when you're debugging the issue using asterisk logs alongside taking SIP-traces it'll help you identify which packets belongs to which log lines. Wireshark is a great tool. I take Sip traces, open up in wireshark goto voip calls and you'll see all the calls that were at-least initiated after when u started the trace. Apply filter on your specific call and see only sip traces relevant to one particular call. Thats all I could come up at this time. I hope this would be of some help. -- Regards, Sammy On Fri, Oct 28, 2011 at 11:10 PM, Douglas Mortensen d...@impalanetworks.com wrote: Anton, ** ** Thanks for the input. I wasn’t aware of ngrep. I’ll check it out. A packet analyzer is a good idea. I am accustomed to using a packet analyzer mostly in a “reactive” approach, or during an incident. Are you suggesting that I just setup a capture to be running continuously until we become aware of the problem, and then at that point, review it to see what really happened (regarding what was was not transmitted on the network)? ** ** Also, thanks for the link in the Asterisk Cookbook. I’ll check it out. ** ** From your egrep example here: tail -f /var/log/asterisk/full | egrep --color -w 'chan_sip.*SIP/911|pbx.*SIP/911' ** ** Are you basically using 911 as an example extension that we wanted to see logging for? That seems useful. Thanks. FYI, I grepped for chan_sip, and pbx, but didn’t really get anything from those with any SIP extension logging. Could that be because I’m using asterisk 1.4? FYI, the customer I’m troubleshooting for is using 1.6, so maybe it would give me something on their version….? ** ** Still one of my concerns is the ability to follow an inbound call from the time it hits asterisk, until it is finally gone. I’d like to follow the call through the logging to have a logical view of what happened to the call from the time it rang in (where the call got sent to [time conditions, queues, ring groups, extensions, transfers, etc.], what phones were rung trying to connect the call, etc. ** ** Any way I this can be done? Can’t a call be passed off from one channel to another, which would leave me with only seeing a part of the logs for the life of the call if I only grep the logs based on one channel id? ** ** Thanks, - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 ** ** *From:* Anton Kvashenkin [mailto:anton.juga...@gmail.com] *Sent:* Thursday, October 27, 2011 8:27 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Tips best practices for asterisk troubleshooting parsing logs ** ** Capture pcap with tshark or tcpdump for the future analysis with wireshark. Ngrep is also handy tool for captaring, say, INVITE. You can use grep like this: tail -f /var/log/asterisk/full | egrep --color -w 'chan_sip.*SIP/911|pbx.*SIP/911' Interesting technique from Astresk Cookbook, Debugging dialplan with Verbose() http://ofps.oreilly.com/titles/9781449303822/DialplanFundamentals.html 2011/10/27 Sammy Govind govoi...@gmail.com It was a challenge to read through all the interesting experience you've shared over here. I don't know what others may be using for parsing the logs beautifully and make them usable. What I would recommend you at the very beginning ,since you mentioned using egrep, is figure out the Channel identifier string from the logs for a particular call. That's underlined below for you. ** ** [Oct 26 17:58:01] VERBOSE[14274] logger.c: -- Executing [s@tc-maint:3] System(*Local/s@tc-maint-2496,2*,/var/lib/asterisk/bin/schedtc.php 60 /var/spool/asterisk
Re: [asterisk-users] Queue announcements and MOH blanking out on calls from PSTN over IAX2
Try turning on the Sip debug for the PSTN call as well as RTP debug. Paste the output here. The Dial server is connected to multiple 4-port Redfone devices for handling PSTN incoming and outgoing calls. Outgoing calls always originate from and incoming calls always terminate at the SIP server. SIP and Dial servers are connected over IAX2. Explain the above abit as well..couldnt get the clear picture of what it looks like. Seems to me that you guys are using two servers and call-audio gets lost in between the servers OR in between the Dial-Server and redfone device for Queue Calls. 2011/10/29 Raj Mathur (राज माथुर) r...@linux-delhi.org On Saturday 29 Oct 2011, Raj Mathur (राज माथुर) wrote: [snip] Callers coming in from the PSTN (through the Dial server, over IAX2) can also talk normally after an agent has picked up the call. However, callers from the PSTN get the announcement and/or MOH blanked out after a random period of time, typically 5-10 seconds. This often happens in the middle of the queue position or thank-you announcement. After the blanking out, the call is still alive, queue functions are working, and if an agent picks up the calls s/he can talk normally to the caller. However, blanking out of the MOH/announcement makes the caller think that the call has been dropped, and they hang up before an agent answers. Debug logs show that Asterisk is playing the MOH and announcement files continuously even though the caller cannot hear them. Unable to figure out why the blanking happens ONLY on incoming calls from the PSTN. Any help appreciated. Further simplified the issue to an extension that just does: ... Answer() ... MusicOnHold(default) When called from the PSTN, the musiconhold blanks out after a few seconds, while it plays fine when the extension is called locally. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tips best practices for asterisk troubleshooting parsing logs
It was a challenge to read through all the interesting experience you've shared over here. I don't know what others may be using for parsing the logs beautifully and make them usable. What I would recommend you at the very beginning ,since you mentioned using egrep, is figure out the Channel identifier string from the logs for a particular call. That's underlined below for you. [Oct 26 17:58:01] VERBOSE[14274] logger.c: -- Executing [s@tc-maint:3] System(*Local/s@tc-maint-2496,2*,/var/lib/asterisk/bin/schedtc.php 60 /var/spool/asterisk/outgoing 0) in new stack Once you Figure out this part use egrep tool and you'll end up seeing only the data related to this particular call. More advanced tool or techniques may involve setting up a central logging server where all the other servers deposit their logs and use monitoring tools like swatch, splunk, zabbix etc etc etc to parse the logs for you and generate alerts. I haven't came across any Asterisk-specific log parser utility so far. Honestly, I never needed one. On Thu, Oct 27, 2011 at 5:16 AM, Douglas Mortensen d...@impalanetworks.comwrote: Hello all, I have been running asterisk systems since summer of 2008. I do not claim to be an expert. But I have worked through many issues during this period. I have setup manage 5 systems, which serve 6 companies total (and of course process calls for all of the people they do business with). I have always been happy with asterisk (well, obviously less happy during the problem times... :-). And I continue to prefer to us it. However, if I could name the one largest struggle that I have with asterisk, it is the facilities that it provides for troubleshooting issues parsing logs. I am hoping that someone on this mailing list can help me to realize how ignorant I really am, and how much time I have wasted parsing, grepping lessing logs manually. I am hoping that one of you can help me see the light. If so, I would be most grateful. Specifically, here are the challenges I encounter, which I would desperately appreciate help with: Here's an example scenario: A customer calls me says that a call just came in some of their wireless DECT phones (I know, trouble already :-) didn't ring, while others did. I tell the customer that I'll start looking into the problem immediately. I am using AsteriskNOW with asterisk 1.6. So I SSH into the system cd to /var/log/asterisk start looking at the full log via less. We have configured the bulk of our system via FreePBX 2.9. Inbound calls are routed first to a time condition which checks whether it is after hours. If it is not afterhours, then are then routed to a queue, which rings all phones (4 wireless DECT phones on 1 DECT wireless server that registers the SIP extensions on behalf of its 4 phones, and 4 more wireless DECT phones on their own wireless server configured the same, and an ATA connected to a paging amp that rings a loud speaker). From there, someone typically will answer the call. Often times they then transfer the call to another extension. However, sometimes no one answers the call, and it winds up going to VM. From the logging aspect of asterisk, it has usually felt like I am trudging through a swampy marsh trying to put the bits pieces together. The challenge I've seen is that the above scenario can actually consist of multiple SIP calls w/ different legs. I *think* (but am not 100% sure) that often times a call can be handed off from 1 asterisk process to another. The result is that grepping by the asterisk process ID shown after the VERBOSE (or NOTICE or DEBUG section [see below]), I don't actually get to see the full sequence of events in following all logging that is relevant to that phone call. [Oct 26 17:58:01] VERBOSE[14274] logger.c: -- Executing [s@tc-maint:3] System(Local/s@tc-maint-2496,2,/var/lib/asterisk/bin/schedtc.php 60 /var/spool/asterisk/outgoing 0) in new stack Then on a busy asterisk system, if I filter by the process id, the one process that starts handling the call originally, may wind up immediately taking on another totally unrelated call after handing the initial call off to another process. If I am not extremely careful, I may wind up mistaking the log lines for the 2nd call, as being a part of the 1st call, and then I'm totally barking up the wrong tree :-) Another option I've tried is to enable SIP debugging. Generally, I do like this. And one nice thing is that asterisk seems to usually add the SIP Date: parameter with its SIP invites, etc. The result is that I can grep the asterisk log like this `egrep -v ^\[ full` (SIP debug lines don't have the standard timestamp at the beginning) and then I'm only seeing the SIP debugging, in a pretty clean output. Still, there can be a LOT of SIP traffic going on, when I'm ringing 9 different extensions from a queue. Trying to parse it all can make me go cross-eyed. :-) And doing so can take a LONG
Re: [asterisk-users] Concurrent call monitoring
I wrote my own shell scripts to collect core show calls value from
asterisk and then push the filtered value to an opensource monitoring tool.
That worked perfectly well.
#!/usr/bin/perl -w
use strict;
open(LINE, 'asterisk -rx core show channels|');
my ($chans, $calls, $line)=(0,0,undef);
while ($line = LINE)
{
$calls = $1 if ($line =~ /^(\d+) active call/);
}
close(LINE);
printf $calls;
On Tue, Oct 25, 2011 at 6:40 PM, Danny Nicholas da...@debsinc.com wrote:
The Simplest method of seeing the number of concurrent calls is service
asterisk status. If I understand question two, asterisk -rx core show
channels verbose is probably your best bet.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Tuesday, October 25, 2011 8:29 AM
To: Asterisk Users
Subject: [asterisk-users] Concurrent call monitoring
Hi
What are people using to monitor the concurrent number of calls at any
given
time?
Also, is there any good way of monitoring concurrent inbound and outbound
calls so that we can see the 2 different numbers?
Thanks in advance
Ish
--
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Office: 0161 660 3062
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Re: [asterisk-users] how to know RTP por of a SIP client in
OP may be able to use System through Dial plan but I'm thinking that since
tcpdump don't just give output within seconds or neither do it get
daemonized? so this system() call will hold the call to that priority. This
may even result in call failure. I think this system call should trigger a
shell script that launches an instance of tcpdump and move forward in the
dial plan.
Can anyone tell if we can extract a header value from SDP(for RTP Tx/RX
ports) within the SIP packet using the SIP_HEADER function?
How about using sipgrep: The idea is launch a sipgrep based scripts in the
background which just takes Call-ID and parse RTP port data and save it in
memcached. This memchache Key/value register will just save [Call-ID:RTP
port data] for each call entering into the Server. This script should start
separate instances of tcpdump for each call with separate file names.
On each call hangup call the h extensions will use the SIP_HEADER(call-id)
Key and trigger a stop command for the background tcpdump for this
particular call.
On Mon, Oct 24, 2011 at 4:36 AM, Bruce B bruceb...@gmail.com wrote:
Then you may use system() in dial-plan to run that shell command along with
what I suggested.
-Bruce
On Sun, Oct 23, 2011 at 5:22 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Yes, I need to know to get in in dialplan because I want to capture
traffic per call. I would like to launch $SHELL{tcpdump src port } in
the dialplan or something like this. And I want RTP traffic only of a
certain call.
Thank you!
===
Date: Fri, 21 Oct 2011 09:41:39 -0400
From: Bruce B bruceb...@gmail.com
Subject: Re: [asterisk-users] how to know RTP por of a SIP client in
the dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
CAJyE_uWLRXkhrWQ-6SvNW1ihN-nGA3HFwHt=
pu-tfr6lybi...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Do you need to know to get it in dialplan? If I not, from shell (not
Asterisk CLI) I usually use:
netstata -a | grep asterisk
By default Asterisk settings it should be something between 10k-20k
-Bruce
On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
How can I get the RTP port one SIP client is using for sending/receiving
RTP flow? Can I obtain it in from SIP_HEADER of something like that in
the
dialplan?
Thank you!
** **
Isabel
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Re: [asterisk-users] ${CALLERID(num)} after doing transfer from extension to extension
Set CDR(destination) or whichever field you need to get recorded in CDRs to
get your desired stats.
On Mon, Oct 24, 2011 at 2:19 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
As I am using the ${CALLERID(num)} to be part of the filename that I am
recording it, I am facing the following problem:
If the incoming call (via PSTN) reached for an extension (which is the
reception), and then the extension transferred the call to the proper
person, and we need to do recording for the call at this proper person, the
problem that at this point the ${CALLERID(num)} will represnt the reception
guy extension and not the original caller id of the caller who called from
outside via the PSTN. How can I get this original caller id?
Regards
Bilal
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Re: [asterisk-users] Voicemail: playing a message to give option if need to transfer for operator
Yes, Macro will return to calling context BUT use GoSub instead and your
life will be easy. Forget using Macro whenever you need to get user input in
there.
On Mon, Oct 24, 2011 at 2:35 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
Is it possible to be part of the voicemail to play a wave message as
following:
The person you are calling is not available, press 0 if you need to call
the operator or 1 to leave voice message?
I know that I can do this as part of the extensions.conf, but I am looking
if it possible to be part of the voicemail function it self?
Actually below is the macro that I am using it for the voicemail, but
really I am facing a troubles and it is not working properly. I would like
to ask about somthing: the macro is not considered to be a context? In other
words, if I used the Background function, so it come back to the original
context or it apply the rules in the macro?
[macro-voicemail]
exten = 108,1,Dial(${ARG1},20)
exten = 108,2,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = 108,3,Goto(IncomingPSTN,t,3)
exten = s,1,Dial(${ARG1},20)
exten = s,2,Background(voicemail-opt)
exten = s,102,Background(voicemail-opt)
exten = 1,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = 1,2,Goto(IncomingPSTN,t,3)
exten = 0,1,Macro(voicemail,SIP/108)
exten = i,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = i,2,Hangup()
exten = t,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = t,2,Goto(IncomingPSTN,t,3)
exten = t,3,Hangup()
exten = a,1,VoicemailMain(${MACRO_EXTEN})
;
Regards
Bilal
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Re: [asterisk-users] Storing a variable at a context and using it in another context
Try using variables between macros and contexts without doing anything. It
works fine for me in asterisk 1.6.13+. If not then use _ before variable
name.
On Mon, Oct 24, 2011 at 2:46 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
Is it possible to store a variable at context and using it in another
context or in the MACRO? For example, how I can store the ${CALLERID(num)}
in a variable and use it in another context or in a MACRO?
Regards
Bilal
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Re: [asterisk-users] Monitor does not work well (little cuts in the audio file)
Hi, I've done some similar thing in one of my testing, using MixMonitor and monitor at the same time. Everything worked perfectly well no issues even on Vmware. Can you check if the CPU utilization is normal. Also which version of asterisk you are using? -- Regards, Sammy On Thu, Oct 20, 2011 at 1:12 PM, ISABEL ORDAS ARNAL i...@tid.es wrote: Hi, I have been testing MixMonitor and Monitor to record some calls in Asterisk and I have noticed that MixMonitor works fine whereas in the Monitor files of the 2 separate channels, we can find little cuts of the audio. We are using U law codec and wav files for the recording. Anyone have suffered the same problems. It is that Monitor does not work well. It is another way to record the 2 legs of the call separately by using MixMonitor? Regards Isabel -- Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor does not work well (little cuts in the audio file)
Have you tried changing/upgrading asterisk version.? On Thu, Oct 20, 2011 at 5:34 PM, ISABEL ORDAS ARNAL i...@tid.es wrote: Hi, CPU usage does not change when a call is served by Asterisk. I performed several there was no influence. Version is Asterisk 1.6.2 Regards, Date: Thu, 20 Oct 2011 13:30:20 +0500 From: Sammy Govind govoi...@gmail.com Subject: Re: [asterisk-users] Monitor does not work well (little cuts in the audio file) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: cajujwtg-18j7g1bregqy_bgh+rnd2o3gkymcw-oxlk2rjtj...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi, I've done some similar thing in one of my testing, using MixMonitor and monitor at the same time. Everything worked perfectly well no issues even on Vmware. Can you check if the CPU utilization is normal. Also which version of asterisk you are using? -- Regards, Sammy On Thu, Oct 20, 2011 at 1:12 PM, ISABEL ORDAS ARNAL i...@tid.es wrote: Hi, I have been testing MixMonitor and Monitor to record some calls in Asterisk and I have noticed that MixMonitor works fine whereas in the Monitor files of the 2 separate channels, we can find little cuts of the audio. We are using U law codec and wav files for the recording. Anyone have suffered the same problems. It is that Monitor does not work well. It is another way to record the 2 legs of the call separately by using MixMonitor? Regards Isabel Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems during calls
Hi, Call getting silenced in the middle definitely point to RTP but I think the redialling part should be considered as well. I think that Phones are loosing registrations or like Zeeshan mentioned could be getting blocked by firewall - Asterisk server's firewall as well as any other firewall in front of server should be inspected for sessions/connections limit etc. -- Regards, Sammy On Wed, Oct 19, 2011 at 12:27 AM, Aksel Celasun ak...@abacus-it.no wrote: Thank you for the reply. ** ** ** ** The Asterisk is behind a firewall, but not in a dmz, been thinking of placing it in a dmz soon, maybe that will solve the problem. Or else, I will try your guide with wireshark. ** ** Thank you very much. ** ** ** ** Best regards ** ** Aksel ** ** *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne av* VisionVoIP *Sendt:* 18. oktober 2011 16:31 *Til:* Asterisk Users Mailing List - Non-Commercial Discussion *Emne:* Re: [asterisk-users] Problems during calls ** ** I can only make another guess. If your system is behind a firewall, try adding 'insecure=invite' in your sip.conf's general section. To troubleshoot such cases, do a tcpdump trace like this: 1. Run tcpdump on your server before making a call. Use command tcpdump port 5060 -s0 -w dumpfile.pcap. 2. When you notice the silence problem, hangup, and stop the trace using CTRL+C. 3. Copy the dumpfile.pcap to a computer with Wireshark installed. 4. Open this file in Wireshark and follow my blog at http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/ 5. Given that you know some basics of how VoIP works over SIP, the wireshark graph will tell you if RTP was still flowing when it was silent. It probably is, but to which IP address. My guess is your RTP, i.e. voice date, starts flowing towards some wrong IP address, or stop flowing, or is blocked by the router. A good solution is to put your Asterisk server in DMZ mode. There can be many other guesses, but the above is a good start. -- Zeeshan A Zakaria PBX - visionvoip.com Blog - ilovetovoip.com On 18/10/2011 10:02, Aksel Celasun wrote: Thank you for replying My sip.conf is set to no on canreinvite ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail
1- Are you sure your Asterisk Box is configured with an MTA / email utility to send emails ? 2- Like Ishfaq suggested you should be getting into the voicemail application after 10 seconds of Dial timeout. Are you even recording and saving a voicemail? 3- To recieve an SMS to notify you of voicemails you've multiple choices, a- Configure your asterisk with smsq to send/receive sms which is tough : b- Get an SMS notification utility in your receiving email to trigger an SMS when a particular type of message is recieved in inbox.(We used this one) c- Configure kennel to Send out SMSs and write an integration bash script to be called after the voicemail application. On Tue, Oct 18, 2011 at 8:40 PM, salaheddine elharit salah.elharit...@gmail.com wrote: thanks for your response itry this but i didn't recive any email,also if there is a way to recive a SMS in my mobile 0678XX regards 2011/10/18 Ishfaq Malik i...@pack-net.co.uk On Tue, 2011-10-18 at 12:26 +, salaheddine elharit wrote: hello list i have configured the voicemail in my server asterisk 1.4 i can use it without issue ,i have a question i want to receive an email in my address email when there is no response from 270 after 10 s could you please verify the code below and tell me what is wrong thanks and regards extensions.conf exten = ,1,VoiceMailMain(777@mb_tutorial) exten = 270,1,Dial(SIP/270, 10) exten = 270,n,VoiceMail(777@mb_tutorial) exten = 270,n,PlayBack(vm-goodbye) exten = 270,n,HangUp() voicemail.conf [mb_tutorial] 777 = ,270,salah.elharit...@gmail.com,,|attach=no|review=yes -- In 1.4 the delimiter is | so try exten = 270,1,Dial(SIP/270|10) -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy() not working with group in asterisk 1.4.42
Hey,
I don't think you are doing it right. The memebers/channels you need to spy
should be added in SPYGROUP and not the channel which is spying. i.e your
code maybe something like this.
exten = 4368,1,Answer()
exten = 4368,n,NoOp(${CHANNEL})
exten = 4368,n,Set(SPYGROUP=my-group)
exten = 4368,n,Konference(VADSTR)
exten = 43681156,1,Answer()
exten = 43681156,n,NoOp(***${SPYGROUP})
exten = 43681156,n,ChanSpy(DAHDI,g(my-group))
exten = 43681156,n,Hangup()
On Tue, Oct 18, 2011 at 12:30 PM, virendra bhati virbh...@gmail.com wrote:
Hi list,
I have write down my code on which chanspy not working when I make a group
with name of spy. Please help me where is the issue on that.
a) caller will call this number to join konference and spy group
exten = 4368,1,Answer()
exten = 4368,n,NoOp(${CHANNEL})
exten = 4368,n,Set(GROUP(${CHANNEL})=spy)
exten = 4368,n,Set(a=${GROUP_LIST(spy)})
exten = 4368,n,Set(b=${GROUP_LIST()})
exten = 4368,n,Konference(VADSTR)
b) spy will dial it to spy the channels
exten = 43681156,1,Answer()
exten = 43681156,n,NoOp(***${SPYGROUP})
exten = 43681156,n,Set(SPYGROUP=spy)
exten = 43681156,n,NoOp(***${SPYGROUP})
exten = 43681156,n,ChanSpy(DAHDI,g(spy))
exten = 43681156,n,Hangup()
when I used chanspy without option then It works
like Chanspy(DAHDI)
Any help will be appreciated
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
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Re: [asterisk-users] Conference solution to handle 10, 000 participants - possible at all?
Hi, I'd been thinking about such a huge conferencing system for about last few months. Like Steve suggested, my concept is almost similar but instead of making a central hub conference junction between multiple Conferences I was thinking of making a peer2peer runtime connection between conferences hosted on multiple servers. All the asterisks are load balanced by a super node which will be OpenSIPS/Sip proxy. Any conference participant call will first land on SIP proxy where Prosy will do some required resgiteration of the participant, decide if the required conference server is full or not- If not route the call to previously used server else route the call to newer server and send a trigger to new asterisk server to bridge with the older server's conference. -- Regards, Sammy On Tue, Oct 18, 2011 at 6:08 AM, Steve Edwards asterisk@sedwards.comwrote: On Mon, 17 Oct 2011, VisionVoIP wrote: A client is asking to setup an asterisk based conferencing solution which could handle 10,000 participants (in one single conference or combined in multiple conferences), and later could be scaled to handle up to 50,000 participants. All callers will be over SIP, using g711. If you scour the archives, you'll find discussion about this kind of thing several years ago, and then again sometime in the last 6 months. Googling about a bit should also yield relevant references. The OP built a system where NASCAR fans could call into conferences and listen to the cockpit chatter of the car of their choice. His system handled around 6,000 callers, but could be scaled higher. Think of a tree where 1 system hosts the conference. All 'callers' to this host are the next level of Asterisk systems. Add additional layers to build out to the number of real callers you want on an individual server. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with outbound dialing from remote phone
Hey, Can you enable sip trace for that particular sip extension. This sounds weird that while other INVITES from the phone are reaching but the external extensions are filtered. If there are no invites for external calls only then more chances are that the phone is using some dial pattern(phonebook help) etc like Doug and Eric said. Sometimes in asterisk console I don't see anything in logs if the Sip extensions' context don't contain the number that is being dialled Do you've access to any phone debugging console? Sounds like problem is somewhere around She :p j/k . -- Regards, Sammy. On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins arob...@pharmacentra.comwrote: The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind.I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that. Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then suddenly went bad. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, October 14, 2011 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I am assuming you are using a provisioning server. If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Friday, October 14, 2011 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone I've already done that. Both phones worked fine in a different remote location just prior to shipping. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, October 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone Adam Robins wrote: No change, thanks Well, In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified
Re: [asterisk-users] dahdi show status command not avilable in CLI
Please paste the configurations in the #included files as well. On Fri, Oct 7, 2011 at 7:07 PM, michael k mich...@inapp.com wrote: Hi, This is my /etc/asterisk/chan_dahdi.conf file. [root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ;added by mic 06-oct-20011 #include /etc/asterisk/dahdi-channels.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf Any issues in this ? Michael.k On Fri, Oct 7, 2011 at 7:19 PM, Eric Wieling ewiel...@nyigc.com wrote: It is likely you have an error in your /etc/asterisk/chan_dahdi.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Friday, October 07, 2011 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I am getting this error message while executing the module load chan_dahdi.so. astrisks*CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_general.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_groups.conf': == Found == Parsing '/etc/asterisk/dahdi-channels.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found Thanks, Michael.k On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote: What happens when you do the module load chan_dahdi.so command? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, astrisks*CLI module unload chan_dahdi.so Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues ? Michael.k On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, what if you reload the module chan_dahdi from within the * CLI? It should give some hints. Giorgio On 10/06/2011 05:22 PM, michael k wrote: Hi Giorgio, Thanks for your reply. I will produce some output for your reference. # lsmod | grep dahdi dahdi_echocan_mg2 39688 1 dahdi_transcode42372 1 wctc4xxp dahdi_voicebus 79424 2 wctdm24xxp,wcte12xp dahdi 238384 14 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 35265 2 wctdm24xxp,dahdi # service dahdi status ### Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 FXOFXSKS (SWEC: MG2) (battery) # dahdi_cfg -vv DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Re: [asterisk-users] PSTN connectivity
If DAHDI is not really configured or chan_dahdi isn't loaded the the error mesage would be can not create channel of type DAHDI but here its not the case. Dadhi module is indeed loaded but the DAHDI device is not working properly. On Thu, Oct 6, 2011 at 8:49 PM, Gohar Ahmed gohar.ah...@vopium.com wrote: Hey, How’ve you configured your Outbound trunk ? DAHDI/1/04712527270 : What do you’ve in your dahdi configuration file ! I doubt this “/1” is the culprit or else your DAHDI channel is not really working at all. ** ** Regards, Gohar A. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michael k *Sent:* Thursday, October 06, 2011 8:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] PSTN connectivity ** ** Hi All, I got a busy message like all lines are currently busy and please try again later in call to ZAP trunk. Please help me to resolve this issue == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [904712527270@from-internal:1] Macro(SIP/157-, user-callerid,SKIPTTL,) in new stack -- Executing [s@macro-user-callerid:1] Set(SIP/157-, AMPUSER=157) in new stack -- Executing [s@macro-user-callerid:2] GotoIf(SIP/157-, 0?report) in new stack -- Executing [s@macro-user-callerid:3] ExecIf(SIP/157-, 1?Set(REALCALLERIDNUM=157)) in new stack -- Executing [s@macro-user-callerid:4] Set(SIP/157-, AMPUSER=157) in new stack -- Executing [s@macro-user-callerid:5] Set(SIP/157-, AMPUSERCIDNAME=Rojar S) in new stack -- Executing [s@macro-user-callerid:6] GotoIf(SIP/157-, 0?report) in new stack -- Executing [s@macro-user-callerid:7] Set(SIP/157-, AMPUSERCID=157) in new stack -- Executing [s@macro-user-callerid:8] Set(SIP/157-, CALLERID(all)=Rojar S 157) in new stack -- Executing [s@macro-user-callerid:9] ExecIf(SIP/157-, 0?Set(CHANNEL(language)=)) in new stack -- Executing [s@macro-user-callerid:10] GotoIf(SIP/157-, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set(SIP/157-, CALLERID(number)=157) in new stack -- Executing [s@macro-user-callerid:20] Set(SIP/157-, CALLERID(name)=Rojar S) in new stack -- Executing [s@macro-user-callerid:21] NoOp(SIP/157-, Using CallerID Rojar S 157) in new stack -- Executing [904712527270@from-internal:2] Set(SIP/157-, _NODEST=) in new stack -- Executing [904712527270@from-internal:3] Macro(SIP/157-, record-enable,157,OUT,) in new stack -- Executing [s@macro-record-enable:1] GotoIf(SIP/157-, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf(SIP/157-, 0?MacroExit()) in new stack -- Executing [s@macro-record-enable:5] GotoIf(SIP/157-, 0?Group:OUT) in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf(SIP/157-, 0?IN) in new stack -- Executing [s@macro-record-enable:16] ExecIf(SIP/157-, 1?MacroExit()) in new stack -- Executing [904712527270@from-internal:4] Macro(SIP/157-, dialout-trunk,1,04712527270,,) in new stack -- Executing [s@macro-dialout-trunk:1] Set(SIP/157-, DIAL_TRUNK=1) in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf(SIP/157-, 0?sub-pincheck,s,1) in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf(SIP/157-, 0?disabletrunk,1) in new stack -- Executing [s@macro-dialout-trunk:4] Set(SIP/157-, DIAL_NUMBER=04712527270) in new stack -- Executing [s@macro-dialout-trunk:5] Set(SIP/157-, DIAL_TRUNK_OPTIONS=tr) in new stack -- Executing [s@macro-dialout-trunk:6] Set(SIP/157-, OUTBOUND_GROUP=OUT_1) in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf(SIP/157-, 0?nomax) in new stack -- Executing [s@macro-dialout-trunk:8] GotoIf(SIP/157-, 0?chanfull) in new stack -- Executing [s@macro-dialout-trunk:9] GotoIf(SIP/157-, 0?skipoutcid) in new stack -- Executing [s@macro-dialout-trunk:10] Set(SIP/157-, DIAL_TRUNK_OPTIONS=) in new stack -- Executing [s@macro-dialout-trunk:11] Macro(SIP/157-, outbound-callerid,1) in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf(SIP/157-, 0?Set(CALLERPRES()=)) in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf(SIP/157-, 0?Set(REALCALLERIDNUM=157)) in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf(SIP/157-, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set(SIP/157-, USEROUTCID=)
Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw codec,
so it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in
my head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this
is just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2
characters, not xx/ (nl/fvdb) (feel free to correct my assumption that
language has not been expanded beyond the 2 character limitation)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten = s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
This is my complete CLI logging
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
0) in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
ast_openstream_full: File beep does not exist in any format [Oct 4
16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/service
line/${UNIQUEID) exten = s,n,Record(${A_serviceline_file}.wav,0,60)
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 16:30
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
Usually this message is received because you did something like
playback(beep.gsm) or playback(beep.wav) instead of playback(beep).
It is
(IMO) somewhat confusing because you have to do record(foo.gsm) but
you have to playback using playback(foo).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan
Kroon | Mobillion
Sent: Tuesday, October 04, 2011 9:21 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Beep file with Record
Hi,
I'm using the functionality Record in asterisk 1.8.5.
But when I want to record something I get the following error message:
file.c:644 ast_openstream_full: File beep does not exist in any format
Could anybody tell me where I have to place the beep.gsm file?
I already tried the following directories:
/var/lib/asterisk/sounds/beep.gsm
/var/lib/asterisk/sounds/recordings/beep.gsm
Regards,
Arjan Kroon
Beep is called from
http://svn.asterisk.org/svn/asterisk/trunk/apps/app_record.c and it looks
fine a first glance. Are you using the language prefix?
--
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Re: [asterisk-users] Beep file with Record
Since you've changed the language (sound directory) So just as a test change
the language back to en and if it goes well revert back language after the
recording.
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
CLI::
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec:
ast_streamfile failed on CAPI/ISDN1#02/318647615-37
** **
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
I don’t call the beep file in my dialplan.
** **
** **
*Van:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
*Verzonden:* 05-10-2011 09:04
*Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
*Onderwerp:* Re: [asterisk-users] Beep file with Record
** **
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
** **
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw codec,
so it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in
my head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this
is just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2
characters, not xx/ (nl/fvdb) (feel free to correct my assumption that
language has not been expanded beyond the 2 character limitation)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten = s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
This is my complete CLI logging
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
0) in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
ast_openstream_full: File beep does not exist in any format [Oct 4
16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/service
line/${UNIQUEID) exten = s,n,Record(${A_serviceline_file}.wav,0,60)
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 16:30
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
Usually this message is received because you did something like
playback(beep.gsm) or playback(beep.wav) instead of playback(beep).
It is
(IMO) somewhat confusing because you have to do record(foo.gsm) but
you have to playback using playback(foo).
-Original Message-
From
Re: [asterisk-users] Beep file with Record
hmmm...what i'm saying is this
*exten = s,n,Set(CHANNEL(language)=en))*
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
*exten = s,n,Set(CHANNEL(language)=nl))*
*
*
On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
Yes I already try this (only with language nl)
exten = s,n,Set(CHANNEL(language)=nl))
** **
I also try to place the voicefile in the directory
/var/lib/asterisk/sounds/ and /var/lib/asterisk/sounds/applications/ of but
without any success.
** **
*Van:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
*Verzonden:* 05-10-2011 09:26
*Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
*Onderwerp:* Re: [asterisk-users] Beep file with Record
** **
Since you've changed the language (sound directory) So just as a test
change the language back to en and if it goes well revert back language
after the recording.
** **
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
CLI::
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec:
ast_streamfile failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
I don’t call the beep file in my dialplan.
*Van:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
*Verzonden:* 05-10-2011 09:04
*Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
*Onderwerp:* Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw codec,
so it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in
my head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this
is just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2
characters, not xx/ (nl/fvdb) (feel free to correct my assumption that
language has not been expanded beyond the 2 character limitation)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten = s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
This is my complete CLI logging
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
0) in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
ast_openstream_full: File beep does not exist in any format [Oct 4
16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file
Re: [asterisk-users] Beep file with Record
Sorry:
*exten = s,n,Set(CHANNEL(language)=en)*
and
exten = s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/
recordings/serviceline/${UNIQUEID*}*)
NOT
*exten = s,n,Set(CHANNEL(language)=en))*
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
On Wed, Oct 5, 2011 at 12:31 PM, Sammy Govind govoi...@gmail.com wrote:
hmmm...what i'm saying is this
*exten = s,n,Set(CHANNEL(language)=en))*
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
*exten = s,n,Set(CHANNEL(language)=nl))*
*
*
On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
Yes I already try this (only with language nl)
exten = s,n,Set(CHANNEL(language)=nl))
** **
I also try to place the voicefile in the directory
/var/lib/asterisk/sounds/ and /var/lib/asterisk/sounds/applications/ of but
without any success.
** **
*Van:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
*Verzonden:* 05-10-2011 09:26
*Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
*Onderwerp:* Re: [asterisk-users] Beep file with Record
** **
Since you've changed the language (sound directory) So just as a test
change the language back to en and if it goes well revert back language
after the recording.
** **
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
CLI::
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File
beep does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to
open beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec:
ast_streamfile failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
I don’t call the beep file in my dialplan.
*Van:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
*Verzonden:* 05-10-2011 09:04
*Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
*Onderwerp:* Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw
codec, so it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in
my head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this
is just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2
characters, not xx/ (nl/fvdb) (feel free to correct my assumption that
language has not been expanded beyond the 2 character limitation)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten = s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
This is my complete CLI logging
-- Executing [s
Re: [asterisk-users] Beep file with Record
The alaw extension is bugging me.. can you locate the default beep.gsm /beep.wav file in asterisk sounds directory !? Also check the output of *core show file formats* *core show translation* Also find out the codec of the established call.! On Wed, Oct 5, 2011 at 12:50 PM, Jeroen Eeuwes jeroeneeu...@gmail.comwrote: Hi Arjan, I also try to place the voicefile in the directory /var/lib/asterisk/sounds/ and /var/lib/asterisk/sounds/applications/ of but without any success. Just for double-checking, but what directory is listed as the astdatadir in asterisk.conf? Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold
Give that moh1 directory permissions, I once had similar issue that same
files being placed in default moh directory were played but making a new
call and directory couldn't play anything. So I fixed that by granting
directory permissions.
On Wed, Oct 5, 2011 at 2:25 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hi
yes i have noticed the same result when i play a file like the default i
can hear the music but when i play another file there is no sound
about your question danny :yes i have created a file in
/var/lib/asterisk/moh1
and i configure in musiconhold.conf like below
[default1]
mode=files
directory=/var/lib/asterisk/moh1
2011/10/4 Kevin Oravits korav...@rcolegal.com
I’ve noticed on our system the sound files have to be in an exact format
for Asterisk to play them.
Bit Rate: 128kbps
Audio sample size: 16 bit
Channels: 1(mono)
Audio Sample rate: 8kHz
Audio format: PCM
** **
I actually downloaded a program and remixed the audio files to match these
settings. Before that, I couldn’t get my Asterisk to play any non-standard
music.
** **
*Kevin Oravits *
** **
*From:* Danny Nicholas [mailto:da...@debsinc.com]
*Sent:* Tuesday, October 04, 2011 11:48 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] music on hold
** **
You have files in /var/lib/asterisk/moh1?
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
elharit
*Sent:* Tuesday, October 04, 2011 12:49 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] music on hold
** **
i configure new music on hold like below in order to play music for
outbond calls
i want tp play a music until answer form customer
[default1]
mode=files
directory=/var/lib/asterisk/moh1
exten = 0678XX,1,Set(CALLERID(number)=520XX)
exten =
0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 0678XX,n,Dial(Zap/g1/${EXTEN},,m(default1))
exten = 0678XX,n,Hangup()
when i put the default music i can listen without issue but when i put
another music .wav Or gsm or Mp3
there is no music there is just the ringing
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Re: [asterisk-users] Passive wait in dialplan?
Can you please explain what you are trying to do? What I've perceived from
this thread is that you want to put call on hold (passively as in no
resources usage) and then on the base of some User's input from UI proceed
with the call accordingly !!
On Wed, Oct 5, 2011 at 3:33 PM, Yaroslav Panych panyc...@gmail.com wrote:
I don't know much about queues, but if channel enter into queue it
should not change its state. I.e. not answer, no moh, no interacting
with user input(DTMF). Less I use unknown helpers, better my
configuration is.
Second issue which can appear using queues - its async state. User can
issue 2 serial commands, and I should have synchronisation tools. In
dialplan I using UserEvent application - which issues event in AMI,
with given data headers. Queue - is there any possibility to customise
queue join(or like) AMI event? Without patches(I already have made
some patches to core and wrote additional module to make * work as I
require).
2011/10/5 Nasir Iqbal na...@ictinnovations.com:
What about waiting in queues?
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Wed, Oct 5, 2011 at 1:35 PM, Yaroslav Panych panyc...@gmail.com
wrote:
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same = n,Set(COMMAND=${CMD_NOOP})
same = n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =
n(COMMAND_SWITCH),GoToIf($[${COMMAND}=${CMD_DOSTUFF1}]?LBL_DO_STUFF1)
same = n,GoToIf($[${COMMAND}=${CMD_DOSTUFF2}]?LBL_DO_STUFF2)
same = n,GoToIf($[${COMMAND}=${CMD_DOSTUFF3}]?LBL_DO_STUFF3)
same = n,Wait(0.2)
same = n,GoTo(COMMAND_SWITCH)
same = n,NoOp(--- NOT REACHED ---)
UserEvent sends blah-event via AMI to high-level UI, user makes
decision and issues some command via Action:SetVar, then dialplan
continues to work.
The problem is, in dialplan there is an active wait loop, i.e. waiting
mechanism which rapidly checks some var(consuming processor resources
and flooding logs). Is it possible to create passive waiting loop
within current abilities of Asterisk 1.8?
regards, Yaroslav
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Re: [asterisk-users] Passive wait in dialplan?
So here's what I think about your scenario: CALL-FLOW 1- Call come in to asterisk (channel not answered) 2- Event is triggered and User decides what to do with call 3- On basis of what user decided a variable is set. 4- Asterisk on the base of that variable route the call further. If this is the intended behaviour I'd make the dialplan which would be something like. DIAL-PLAN ALGO 1- Progress() ; Won't Answer the channel and put the call in trying... mode. 2- Generate User Evnt 3- While(USERDECISION == ) 4- Endwhile 5- Execute anything on base of USERDECISION This has some limitation due to progress. GUI user needs to decide fast as progress will time-up and the caller will get NO_ANSWER from the system. Queue can be used to put call on wait until something is decided by GUI user but for that you'll have to use system resources and also answer the channel first. I hope some real expert here guide you in a better direction. On Wed, Oct 5, 2011 at 4:44 PM, Yaroslav Panych panyc...@gmail.com wrote: Yes, something like that, but hold-state should not answer channel. answer command will be given explicitly. or call can be transfered to Dial command, etc. 2011/10/5 Sammy Govind govoi...@gmail.com: Can you please explain what you are trying to do? What I've perceived from this thread is that you want to put call on hold (passively as in no resources usage) and then on the base of some User's input from UI proceed with the call accordingly !! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
