[asterisk-users] Linux/Asterisk on game consoles?
Hello I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux/Asterisk on game consoles?
On Fri, 2009-10-16 at 08:16 +0200, Vincent wrote: Hello I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? Thank you. I did it with PS3 and Asterisk 1.2 about a year ago With Yellow Dog linux running on PS3 Was not using any of co-processors though, just the main cpu. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The City of Amsterdam has been deploying asterisk throughout the city!
Ron Arts wrote: If you're interested, here is the press release: http://www.neonova.nl/nl/content/press/?tid=129735 This list is for non-commercial discussion. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4
What you have described is an attended transfer so the docs that you found should help you. - Original Message - From: Miguel Molina To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 13, 2009 16:22 Subject: Re: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4 We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on attended transfers. What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to Party C, Ext B brings Caller A onto the call and introduces A to C. After the into, ext B then drops off the call while A C continue the call. Any help would be appreciated. Thanks Much, Jeff Johnson This email and any attached files are confidential and intended solely for the intended recipient(s). If you are not the named recipient you should not read, distribute, copy or alter this email. Any views or opinions expressed in this email are those of the author and do not represent those of NeturallySpeaking, LLC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Lee, John (Sydney) escribió: I don't think this can be done. In your scenario, B is effectively the host and if B drops the line, both A and C will be dropped off as well. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Johnson Sent: Monday, 12 October 2009 2:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4The built-in attended transfer asterisk has, works OK and won't drop the entire call when B hangs up to complete the attended transfer (with the * key). On the asterisk attended transfer, B calls C to explain the A issue (while A waits with MoH) and then hangs up (with the * key) to complete the attended transfer leaving A and C connected, but the part where you want all three talking do to the warm introduction cannot be done, unless you use a Meetme conference to put all of them on the same conversation. There may be some applications that do this without the need of a Meetme conference, maybe someone else can enlighten us. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sporadic one-way audio
Hi For the sporadic one way audio, check that the codec list in the snom phones is the same as set by the server. The codec list is in the RTP tab of the identities. Hope that helps Ish Brent Davidson wrote: We have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice, but they can't hear us. If we immediately call them back everything is fine. The problem affects all offices and also happens when making sip to sip calls from one snom 300 to another. In addition we periodically have calls that drop off in the middle of a conversation like the connection was lost. I haven't been able to replicate any of these problems and the people that are having them can't seem to keep track of when they occur so I can go back and look in the logs. I suspect that both problems may be related though. Possibly a registration issue? Any ideas are welcome. Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF failing in some calls
Asterisk had some major issues with rfc2833 in 1.2.X. You should consider upgrading. - Original Message - From: abdelkader To: asterisk-users@lists.digium.com Cc: mettichi ; jetcomm2...@yahoo.fr Sent: Wednesday, October 14, 2009 11:36 Subject: [asterisk-users] DTMF failing in some calls Hello, I am using Asterisk 1.2.33 under Debian ETCH linux. I have the following problem with DTMF: In my callback system, I calls an access DID. My system calls me back to my phone. It asks me for a password to let me dial an international number. If the authentication succeeds, I can dial a number in the system. Sometimes Asterisk catches only some of the digits I have entered. Sometimes, it duplicates some other digits. Sometimes, the system does not catch anything. And finally, sometimes it works properly. I am using rfc2833 as dtmf mode. Please help. Thanks. -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple call
- Original Message - From: Matt Riddell li...@venturevoip.com To: asterisk-users@lists.digium.com Sent: Thursday, October 15, 2009 09:50 Subject: Re: [asterisk-users] multiple call On 15/10/09 4:42 AM, Faheem wrote: Through Asterisk AMI, you can not dial multiple number at the same time. If you are going to implement a concurrent call scenario, then AMI would not be a valid choice. Multiple calls can be implemented with callfile. Totally incorrect. We do hundreds of simultaneous calls at the same time using the Asterisk Manager. -- Cheers, Matt Riddell Director ___ I would agree with Mat. We had more problems with call files then with the AMI. The only issue I had with the AMI (1.4.X) is that after multiple connections our script would connect to the AMI but the calls would not go out. We used astmanproxy and that resolved the issue. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
C. Chad Wallace cwall...@lodgingcompany.com writes:
OK, I decided to write it up in AEL. It's incomplete and untested, but
it probably gets the idea across a little better.
context agentcalls {
_2XX = {
Set(AGENT=${EXTEN}); // Assuming agent ID is extension.
if (${EPOCH}${DB(AgentPaused/${AGENT})}) {
// Let the call through to the cell phone
Dial(...);
if (cell call was rejected) {
// Flag agent as paused for the next 30 seconds.
Set(DB(AgentPaused/${AGENT})=$[${EPOCH}+30]);
};
}
else {
// Agent still paused.
};
};
};
I was going in the same direction at the end of my first mail, but I
hadn't written any code. There is a problem though: The Queue
application will keep sending calls to the Local channel, which have to
be rejected, over and over.
Would it perhaps work to simply Wait(30) if the call is rejected by the
phone? If the Queue assumes that the phone is busy for those 30 seconds,
I have accomplished my goal. It's worth a shot.
/Benny
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[asterisk-users] Check if a variable is set
Hi
Is there any way to check if a variable is set in asterisk? I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been passed with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
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[asterisk-users] Invite after bye?
Hi there noticed a strange thing in asterisk 1.6.2x 1.6.1x after one of the clients sends bye asterisk first sends invite to other side then after 200 ok it sends bye I am not sure but that could be some missconfiguration issue or a bug? so it's like this: side A sends bye to asterisk, asterisk responds with 200 OK to side A, then it sends INVITE to side B, expects 200 OK from side B, and then sends ACK and BYE to side B Thanks, Josip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Check if a variable is set
If you want to check in Console then NOOP can be used .if in case of
function call you can check its length if there exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk? I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been passed with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Shakeel Abbas
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Re: [asterisk-users] Check if a variable is set
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
If you want to check in Console then NOOP can be used .if in case of
function call you can check its length if there exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk? I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been passed with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
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Software Developer
PackNet Ltd
Office: 0161 660 3062
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Shakeel Abbas
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Shakeel Abbas
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Re: [asterisk-users] Check if a variable is set
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428, EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Best Regards
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--
Best Regards
Shakeel Abbas
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[asterisk-users] Soft phone not registering
HI All, I have installed Asterisk 1.4.26.2 on a centOS box on a public IP and trying to connect from softphone behind ADSL router. The softphone is not able to register, we get some SIP messages on the server, which look like below. Please advise where could be the issue. Thnx Rakesh --- Retransmitting #3 (NAT) to x.x.x.x:38155: OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport From: asterisk sip:aster...@x.x.x.x;tag=as7d8aae9d To: sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP Contact: sip:aster...@203.211.60.167 Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 16 Oct 2009 10:47:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- Retransmitting #4 (NAT) to x.x.x.x:38155: OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport From: asterisk sip:aster...@203.211.60.167;tag=as7d8aae9d To: sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP Contact: sip:aster...@x.x.x.x Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 16 Oct 2009 10:47:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 sip.conf [general] context = tutorial bindport = 5060 bindaddr =0.0.0.0 domain = x.x.x.x nat=yes disallow = all allow = alaw keeprtpalive = yes notifyringing = yes canreinvite = no type = peer realm = asterisk qualify = yes [test2] type = peer host = dynamic username = test2 context = tutorial port = 5060 notifyringing = yes nat = yes type = friend canreinvite = no realm = asterisk qualify = yes mailbox=...@mb_tutorial --- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Invite after bye?
Josip Djuricic wrote: so it's like this: side A sends bye to asterisk, asterisk responds with 200 OK to side A, then it sends INVITE to side B, expects 200 OK from side B, and then sends ACK and BYE to side B This occurs often when directmedia (canreinvite) is in use, and the media path has been redirected to be between the endpoints; when endpoint A issues a BYE, Asterisk knows that endpoint B can no longer send media directly to endpoint A, so it reinvites the media path back to Asterisk... only to then hangup the call. It's not optimal, but it shouldn't cause any harm. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Check if a variable is set
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428, EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com mailto:shakeel.abbas@gmail.com
wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Best Regards
Shakeel Abbas
--
Best Regards
Shakeel Abbas
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Office: 0161 660 3062
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Re: [asterisk-users] Check if a variable is set
My first though is using the isnull function.
http://www.voip-info.org/wiki/view/Asterisk+func+isnull
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428, EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Best Regards
Shakeel Abbas
--
Best Regards
Shakeel Abbas
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Re: [asterisk-users] Check if a variable is set
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Shakeel Abbas
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Re: [asterisk-users] Check if a variable is set
Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
___
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Best Regards
Shakeel Abbas
--
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Shakeel Abbas
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Best Regards
Shakeel Abbas
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To
Re: [asterisk-users] Check if a variable is set
This might work:
exten = s,2,ExecIf($[LEN(${ARG3}) 0] ${ARG3}=1|whatever I want
to do
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, October 16, 2009 7:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Check if a variable is set
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Best Regards
Shakeel Abbas
--
Best Regards
Shakeel Abbas
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Ishfaq Malik
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PackNet Ltd
Office: 0161 660 3062
___
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Best Regards
Shakeel Abbas
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PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Check if a variable is set
Something like:
exten = s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Should work from what I read on voip-info.org.
On Fri, Oct 16, 2009 at 10:19 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
___
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--
Best Regards
Shakeel Abbas
--
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Shakeel Abbas
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Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Check if a variable is set
Actually just noticed a typo try:
exten = s,1,ExecIf($[${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Had { instead of [ in the ExecIf.
On Fri, Oct 16, 2009 at 10:26 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Something like:
exten = s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Should work from what I read on voip-info.org.
On Fri, Oct 16, 2009 at 10:19 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
___
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Shakeel Abbas
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Re: [asterisk-users] Soft phone not registering
First suggestion is if this Asterisk server is accessible from the internet put a secret in the peer definition. What you have now is wide open. Second thing is if I understand it you are going: PC(Soft Phone) ADSL Router Internet Asterisk box. Is that correct? If not, can you descibe it better. On Fri, Oct 16, 2009 at 7:56 AM, Rakesh Sabharwal sabharwal_rak...@yahoo.co.uk wrote: HI All, I have installed Asterisk 1.4.26.2 on a centOS box on a public IP and trying to connect from softphone behind ADSL router. The softphone is not able to register, we get some SIP messages on the server, which look like below. Please advise where could be the issue. Thnx Rakesh --- Retransmitting #3 (NAT) to x.x.x.x:38155: OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport From: asterisk sip:aster...@x.x.x.x;tag=as7d8aae9d To: sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP Contact: sip:aster...@203.211.60.167 Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 16 Oct 2009 10:47:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- Retransmitting #4 (NAT) to x.x.x.x:38155: OPTIONS sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport From: asterisk sip:aster...@203.211.60.167;tag=as7d8aae9d To: sip:te...@192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP Contact: sip:aster...@x.x.x.x Call-ID: 3c92389c5e72d3e92fd8d20b70055...@x.x.x.x CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 16 Oct 2009 10:47:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 sip.conf [general] context = tutorial bindport = 5060 bindaddr =0.0.0.0 domain = x.x.x.x nat=yes disallow = all allow = alaw keeprtpalive = yes notifyringing = yes canreinvite = no type = peer realm = asterisk qualify = yes [test2] type = peer host = dynamic username = test2 context = tutorial port = 5060 notifyringing = yes nat = yes type = friend canreinvite = no realm = asterisk qualify = yes mailbox=...@mb_tutorial --- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP and call manager
On Asterisk 1.4, Call doesn't line Channel: AB. You could put the second dialplan snippet into a context and do your callfile like this: [callccm] exten = s,1,Dial(SIP/CCMMAIN,10,KkTt) Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt) -- Channel: SIP/104 CallerID: SIP/104 MaxRetries: 1 WaitTime: 60 retryTime: 5 Context: callccm Extension: s -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, October 15, 2009 7:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP and call manager Here are two ways to address this 1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once 2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt) Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt) CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3 rings) Danny thats good to know for extensions.conf but I am using call files. echo Channel: SIP/CCMMAIN/5551212 /tmp/call echo Context: smvoice-test /tmp/call Can I do the Channel: SIP/CCMMAIN/5551212SIP/CCMSLAVE/5551212 in the Channel for the call file? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_swift issue
On Tue, 25 Aug 2009 23:37:12 -0700, Abbas Shakeel wrote: when i try to execute make command on app_swift-1.6.2 I get the following error [r...@asterisk app_swift-1.6.2]# make gcc -I/opt/swift/include -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -c -o app_swift.o app_swift.c app_swift.c: In function ‘engine’: app_swift.c:402: error: incompatible types in assignment app_swift.c: In function ‘load_module’: app_swift.c:546: error: ‘AST_MODULE’ undeclared (first use in this function) try the patch at http://jeremy.kister.net/code/app_swift-1.6.2.patch -- Jeremy Kister http://jeremy.kister.net./ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Check if a variable is set
Grrr, none of it works, and the ExecIf's default position in the case of
confusion is to execute rather than to not
Darrin Henshaw wrote:
Actually just noticed a typo try:
exten = s,1,ExecIf($[${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Had { instead of [ in the ExecIf.
On Fri, Oct 16, 2009 at 10:26 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Something like:
exten = s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Should work from what I read on voip-info.org.
On Fri, Oct 16, 2009 at 10:19 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk
wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
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Re: [asterisk-users] Check if a variable is set
Hi
Here it is
[macro-extcall]
;Macro created by Ish to handle external national calls
exten = s,1,Set(CALLERID(all)=${ARG2})
exten = s,2,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
exten = s,3,Dial(SIP/44${ar...@carrier,45)
exten = s,4,Hangup
Execution id Macro(extcall|Dialled Number|Caller CLI)
Now to enable monitoring for an outgoing line it would be
Macro(extcall|Dialled Number|Caller CLI|1)
But I have 79 of these in the system already with just the 2 arguments and we
use realtime so they are all in a DB rather than text file so I'd rather not go
through all the existing ones and change them to
Macro(extcall|Dialled Number|Caller CLI|0)
If I don't have to.
Also, I don't like getting bested by machines!
Ish
Darrin Henshaw wrote:
Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1 0${ARG3}=1|
On Fri, Oct 16, 2009 at 3:45 PM, Ishfaq Malik i...@pack-net.co.uk
mailto:i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten = s,2,ExecIf(EXISTS(${ARG3})=1 ${ARG3}=1|whatever I want
to do
I see this in the console
Executing [...@macro-extcall:2] ExecIf(SIP/PACK501-08222428,
EXISTS()=1
=1|whatever I want to do
As I didn't pass a third argument.
Essentially, what I'm trying to do in php terms would be this
if(isset($var) $var==1)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com
mailto:shakeel.abbas@gmail.com wrote:
If you want to check in Console then NOOP can be used .
if in case of function call you can check its length if there
exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik
i...@pack-net.co.uk mailto:i...@pack-net.co.uk
mailto:i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk?
I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been
passed
with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Best Regards
Shakeel Abbas
--
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Re: [asterisk-users] Check if a variable is set
Why not this?
[macro-extcall]
;Macro created by Ish to handle external national calls
exten = s,1,Set(CALLERID(all)=${ARG2})
exten = s,n,Gotoif($[${ARG3} != 1]?dialit)
exten = s,n,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
exten = s,n(dialit),Dial(SIP/44${ar...@carrier,45)
exten = s,n,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, October 16, 2009 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Check if a variable is set
Hi
Here it is
[macro-extcall]
;Macro created by Ish to handle external national calls
exten = s,1,Set(CALLERID(all)=${ARG2})
exten = s,2,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
exten = s,3,Dial(SIP/44${ar...@carrier,45)
exten = s,4,Hangup
Execution id Macro(extcall|Dialled Number|Caller CLI)
Now to enable monitoring for an outgoing line it would be
Macro(extcall|Dialled Number|Caller CLI|1)
But I have 79 of these in the system already with just the 2 arguments and
we use realtime so they are all in a DB rather than text file so I'd rather
not go through all the existing ones and change them to
Macro(extcall|Dialled Number|Caller CLI|0)
If I don't have to.
Also, I don't like getting bested by machines!
Ish
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Re: [asterisk-users] OT wanted old Sipura firmware 2.0.13
Joseph wrote: Does anybody know here I can find old Sipura firmware 2.0.13 for SPA-3000 I have Cisco 3.1.20 but it is not working as it suppose to. http://www.totek.ca/index.php?option=com_contenttask=viewid=151Itemid=39 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Benny Amorsen benny+use...@amorsen.dk writes: Would it perhaps work to simply Wait(30) if the call is rejected by the phone? If the Queue assumes that the phone is busy for those 30 seconds, I have accomplished my goal. It's worth a shot. This works! Actually I tried out Wait(1000), but that worked fine. After 30 seconds (the timeout in the queue) the Local channel was closed, and a short while later a new call attempt was made. Just as I was hoping. It would still be neat to have a min_dial_interval option, so that Queue never overwhelms the server with failing dial attempts. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] inquire if SIP connections are active or not
Is there a way to ask asterisk from a shell script if its connection (SIP) is valid to another system. Lets say for example to cisco call manager? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Check if a variable is set
We have a winner!
Thanks Danny.
My excuse of not thinking of that myself is working wholly in realtime
and mysql where you have to use priorities apart from when I'm writing
Macros. Did I get away with that?
Ish
Danny Nicholas wrote:
Why not this?
[macro-extcall]
;Macro created by Ish to handle external national calls
exten = s,1,Set(CALLERID(all)=${ARG2})
exten = s,n,Gotoif($[${ARG3} != 1]?dialit)
exten = s,n,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
exten = s,n(dialit),Dial(SIP/44${ar...@carrier,45)
exten = s,n,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, October 16, 2009 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Check if a variable is set
Hi
Here it is
[macro-extcall]
;Macro created by Ish to handle external national calls
exten = s,1,Set(CALLERID(all)=${ARG2})
exten = s,2,ExecIf(${ARG3}=1|Monitor|My monitor arguments)
exten = s,3,Dial(SIP/44${ar...@carrier,45)
exten = s,4,Hangup
Execution id Macro(extcall|Dialled Number|Caller CLI)
Now to enable monitoring for an outgoing line it would be
Macro(extcall|Dialled Number|Caller CLI|1)
But I have 79 of these in the system already with just the 2 arguments and
we use realtime so they are all in a DB rather than text file so I'd rather
not go through all the existing ones and change them to
Macro(extcall|Dialled Number|Caller CLI|0)
If I don't have to.
Also, I don't like getting bested by machines!
Ish
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Re: [asterisk-users] inquire if SIP connections are active or not
You could validate whether it has a physical connection I believe. Add qualify=yes in the sip definition and use something like: /usr/sbin/asterisk -rx sip show peer | grep UNREACHABLE | wc -l Where is the name of the sip definition on your system. If the return is 0 then all is well, if the return is 1 then you have a connection issue. Not sure how to do any other type of validation, but no doubt it's possible. On Fri, Oct 16, 2009 at 11:40 AM, Jerry Geis ge...@pagestation.com wrote: Is there a way to ask asterisk from a shell script if its connection (SIP) is valid to another system. Lets say for example to cisco call manager? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with a Cisco AS5300 gateway
destination-pattern .T What does destination-pattern .T mean? I'm not familiar with what .T would match. I would suggest using a more specific pattern that you expect to be coming down the line. One or more characters (up to 31 characters), waiting timeouts inter-digit before sending. http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/dial_peer/dp_plan.html You could be more specific, if you know what is always going to be coming down the line, like 503... if you only have Oregon numbers, and get 10 digits from the provider. T is useful for outbound calls with a trunk number such as 9T because you never know what number those crazy users will try to call. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT wanted old Sipura firmware 2.0.13
On 10/16/09 10:23, Dave Fullerton wrote: Joseph wrote: Does anybody know here I can find old Sipura firmware 2.0.13 for SPA-3000 I have Cisco 3.1.20 but it is not working as it suppose to. http://www.totek.ca/index.php?option=com_contenttask=viewid=151Itemid=39 Thank you, that is a golden link. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Origin of Exceptionally long voice queue length queuing to IAX2/blahblah messages
Hello, I'm using asterisk for a quite long period, i integrated a lot of stuff to make it behave like any carrier class system, so users can: manage Call forward on busy manage Call forward on no answer manage unconditionnal call forward call back missed calls and a lot of such services that can be both configured by menus and dtmf or by web interfaces (PHP+SQL), in fact i recreated services everybody use to get with their classical operator. All of theses functions are achieved throught dialplan commands, first MYSQL commands, conditions on results etc.. It worked as a charm but got wrong with the growing number of users, i began having asterisk network connectivity loss, in fact qualified IAX2 friends became UNREACHABLE for seconds, connected calls were disturbed when not simply cut. I first thought of IAX problems on charge, moved to SIP, but got the same problems, i then moved my MYSQL commands to ODBC functions, optimized my macros, it worked for a short while, i changed dialplan to reduce as much as possible LOCAL channels, it worked for a while but now the problem comes back with its IAX messages: [Oct 16 17:55:40] WARNING[2514]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/blahblah [Oct 16 17:55:40] WARNING[2511]: channel.c:951 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/blahblah [Oct 16 17:57:38] NOTICE[2513]: chan_iax2.c:9186 __iax2_poke_noanswer: Peer 'XXX' is now UNREACHABLE! Time: 9 [Oct 16 17:57:38] NOTICE[2505]: chan_iax2.c:9186 __iax2_poke_noanswer: Peer 'YYY' is now UNREACHABLE! Time: 9 [Oct 16 17:57:55] WARNING[2505]: chan_iax2.c:804 jb_warning_output: Resyncing the jb. last_delay 0, this delay 22334, threshold 1000, new offset -22334 [Oct 16 17:57:57] WARNING[2510]: chan_iax2.c:804 jb_warning_output: Resyncing the jb. last_delay -135, this delay 1529, threshold 1000, new offset -1529 [Oct 16 17:58:05] NOTICE[2507]: chan_iax2.c:8264 socket_process: Peer 'XXX' is now REACHABLE! Time: 4 [Oct 16 17:58:07] NOTICE[2511]: chan_iax2.c:8264 socket_process: Peer 'YYY' is now REACHABLE! Time: 11 On some servers i used less advanced dialplan, with ODBC functions but geeting rid of LOCAL channels, and, until now, i didn't got any problem on them... so i suspect LOCAL channels, without proof of it, but what else? And for my advanced services, it will be hard to get rid of them without scripts so, i'm actually thinking of moving my dialplan commands to AGI script(s), so i may get rid of LOCAL channels and manage the processes more easily, but reading wikis and mailing lists, some of them consider dialplan commands better than scripts for performances... but if so, why are AGI used by so much projects (asterbilling or ccard systems), which seems to support very large amount of users So, if somebody on the list has any idea or advice... please let me know... Regards Daren _ A la recherche de bons plans pour une rentrée pas chère ? Bing ! Trouvez ! http://www.bing.com/search?q=bons+plans+rentr%C3%A9eform=MVDE6___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT Old Sipura OK - Linksys (junk)
I have two old Sipura 3000 original (green) boxes and they are running perfectly faxes going through, no echo. Firmware 2.0.13(GWg), hardware 2.0.1(1813) Does the number in the bracket means anything? In addition I have Linksys 3000 running the same firmware and same hardware (except the number in the bracket). I can not get to work correctly. The unit has a lot of echo, can not get rid of it, faxes will not go through. Newer firmware 3.1.10 is buggy so I'll not try it; I've loaded 3.1.20 and it will not work correctly can not dial out without on default setting (have to set the SPA to PSTN high). So in other words Linksys unit (at least this model) is a piece of junk. Does anybody runs Linksys 3102 with Cisco firmware 5.1.10? How do they run? I have two units Linksys 3102 with firmware 5.1.7(GW) hardware: 1.4.5(a) and I'm not impress with them either, lot of echo as well. Maybe that is why they sold the unit to Cisco as they can not make it to work correctly :-/ Any recommendation where to move from here, which unit to buy for use with Asterisk? -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
I assume you have a 610 entry in users.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Friday, October 16, 2009 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
At 11:23 AM on 16 Oct 2009, Benny Amorsen wrote: I was going in the same direction at the end of my first mail, but I hadn't written any code. There is a problem though: The Queue application will keep sending calls to the Local channel, which have to be rejected, over and over. Would it perhaps work to simply Wait(30) if the call is rejected by the phone? If the Queue assumes that the phone is busy for those 30 seconds, I have accomplished my goal. It's worth a shot. It would only be trying one agent at a time for each waiting queue member... I don't know how expensive it is to open and close a Local channel and do a DB lookup, but I wouldn't expect it to be a real problem. You are at least avoiding multiple calls out to the cellular network. Also, if there is another agent available, the caller would be connected immediately, and it wouldn't have to make any more attempts. With the Wait() solution, that caller would be waiting for 30 seconds regardless of whether there's anyone else available. Of course, I don't know your business case, so you'll have to decide which of the two problems is worse. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
No, probably my ignorance but why would I do that? I set up all the users, extensions, and mailboxes manually by editing the config files in order to have more control than the user.conf gives me (if I understand the user.conf file properly - I've never used it based upon reading the documentation). Thanks - John On Fri, 2009-10-16 at 12:07 -0500, Danny Nicholas wrote: I assume you have a 610 entry in users.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Friday, October 16, 2009 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux/Asterisk on game consoles?
I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? The Xbox is an x86 machine, so running linux and/or asterisk on it should not be too difficult. There's even a not-so-difficult method of adding a USB port, which would allow you to attach Xorcom hardware for PSTN connections. The Xbox360 is a PowerPC machine. I don't know what the status of having it run *nix is, but there's a site dedicated to it here: http://www.free60.org For the wii, there's: http://wiibrew.org/wiki/Wii_Linux - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
C. Chad Wallace cwall...@lodgingcompany.com writes: It would only be trying one agent at a time for each waiting queue member... Would it? Almost all our queues are on a ringall strategy. I don't know how expensive it is to open and close a Local channel and do a DB lookup, but I wouldn't expect it to be a real problem. You are at least avoiding multiple calls out to the cellular network. Not that expensive, but still a bit of a waste when done every couple of seconds. Especially if multiple agents are unavailable. Also, if there is another agent available, the caller would be connected immediately, and it wouldn't have to make any more attempts. With the Wait() solution, that caller would be waiting for 30 seconds regardless of whether there's anyone else available. This bit is solved by the ringall strategy. Of course, I don't know your business case, so you'll have to decide which of the two problems is worse. I'm fairly happy with the Wait(1000) solution for now. We'll see if testing shows any problems with it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to find IMAP storage doc ?
We're also working fine with it but I also do not know what the available imapflags are and what they mean. I have seen notls and novalidatecert. Out of curiosity, I spent the last 20 minutes googling for information on c-client imapflags and didn't find any definitions or even a simple list, either. There is a list of flags in the c-client man page but they seem to be a different set of flags. Let me know what you find as I would like to know what functionality and options they give us. I'd recommend compiling c-client from source. I've never run Lenny before, but I had a number of issues with various pre-compiled versions of c-client. I feel your pain on lack of documentation for compiling from source, though. The magic steps for me on CentOS were: 1. Modify the EXTRACFLAGS line of the uw-imap makefile: EXTRACFLAGS=-DDISABLE_POP_PROXY=1 -DIGNORE_LOCK_EACCES_ERRORS=1 -I/usr/include/openssl -fPIC -fno-strict-aliasing -Wall -Wno-pointer-sign -Wno-parentheses (I think this is all I had to modify, but I can send you my complete working Makefile, if you like). 2. Compile for your platform: For lenny, I think it would be: make ldb 3. For asterisk, manually configure the location of uw-imap: ./configure --with-imap=/path/to/imap - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI for multiple voice mail boxes
Preparing for the lightning bolt here (ready to duck!!), the way I have things set up, tkeeley would have an entry in users.conf as 612 and 610 would have an entry in users.conf as 610. There would be an entry in sip.conf for tkeeley under 612 and no entry for 610 since it's just a mailbox and not a physical extension. Not necessarily best or even correct, just works for me. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Friday, October 16, 2009 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes No, probably my ignorance but why would I do that? I set up all the users, extensions, and mailboxes manually by editing the config files in order to have more control than the user.conf gives me (if I understand the user.conf file properly - I've never used it based upon reading the documentation). Thanks - John On Fri, 2009-10-16 at 12:07 -0500, Danny Nicholas wrote: I assume you have a 610 entry in users.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Friday, October 16, 2009 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can i use Asterisk to send sms to my database users?
Hi. that is the question. I need send periodically some sms messages to my users, stored in a database. We are doing that process with a cell service provider, but i wanna know if i can use my own server to do that. Any sugestion? thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can i use Asterisk to send sms to my databaseusers?
Absolutely. Just set up the phone address (201...@cingular.net) in users.conf and restart. I'm not sure what apps will send besides voicemail, but then that wasn't the question. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kazabe Sent: Friday, October 16, 2009 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] can i use Asterisk to send sms to my databaseusers? Hi. that is the question. I need send periodically some sms messages to my users, stored in a database. We are doing that process with a cell service provider, but i wanna know if i can use my own server to do that. Any sugestion? thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
At 7:35 PM on 16 Oct 2009, Benny Amorsen wrote: C. Chad Wallace cwall...@lodgingcompany.com writes: Also, if there is another agent available, the caller would be connected immediately, and it wouldn't have to make any more attempts. With the Wait() solution, that caller would be waiting for 30 seconds regardless of whether there's anyone else available. This bit is solved by the ringall strategy. Of course, I don't know your business case, so you'll have to decide which of the two problems is worse. I'm fairly happy with the Wait(1000) solution for now. We'll see if testing shows any problems with it. Oh yeah, I hadn't even considered the ringall strategy! With that, your Wait() solution sounds perfect to me. Congrats! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP to IAX to SIP
Hi all, I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs very well. On that machine I have a SIP phone. I have configured a netgear wgt634u with asterisk and a SIP phone and linked the two systems together via IAX. Audio from Ubuntu to netgear is not bad, audio from netgear to ubuntu is unintelligible. Any clues as to whether this will work? Configuration suggestions? Is a 200MHz arm processor just too small? Any help appreciated. sip phone -- wgt634u - iax - ubuntu -- sip phone wireless 200MHz arm 3GHz AMDhardwired 100MB 100MB lan between systems Hope this is clear enough. Cheers George ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX to SIP
George Farris wrote: I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs very well. On that machine I have a SIP phone. I have configured a netgear wgt634u with asterisk and a SIP phone and linked the two systems together via IAX. Audio from Ubuntu to netgear is not bad, audio from netgear to ubuntu is unintelligible. Any clues as to whether this will work? Configuration suggestions? Is a 200MHz arm processor just too small? what codec are you using? -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux/Asterisk on game consoles?
Out of curiosity why would you want to? Hello I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux/Asterisk on game consoles?
At 4:48 PM on 16 Oct 2009, Adam Moffett wrote: Out of curiosity why would you want to? Because he can? or, Because it's there. http://www.askoxford.com/worldofwords/quotations/quotefrom/mallory/ ...but hopefully the OP doesn't end up bricking his console. I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Whither asterisk-addons?
I noticed that asterisk.org got a redesign, quite recently it seems, which is very nice, but the addons package isn't listed for download any longer, nor are releases posted to http://downloads.asterisk.org/pub/telephony/ . That said, looks like it's still available in svn, http://svnview.digium.com/svn/asterisk-addons/tags/1.6.1.1/ . So just wondering if addons will be around for the forseeable future? - Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR
Ladies and Gentlemen, We already have an Asterisk Call center suite installed at our contact center. Now we wish to commence IVR services. We are offering Health Information Services. Can someone help us to develop this Addon / Solution? Best regards. -- السلام عليكم ورحمة الله وبركاته Nazir Ahmed Vaid Cell:+92300-828 eHealth Services (Pvt) Ltd. http://www.ehealth-services.com NexSource Pakistan (Pvt) Ltd. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nehalem Digium Wildcard issues?
Just putting this out there to see if anyone else has seen any issues. May cross-post to asterisk-dev if it's indeed a bug (and not my own stupidity). I've got a Digium TE220 (2xT1 interface w/Echo canceller) that in two separate Nehalem-based (Xeon E5520 Gainestown) boxes (HP ProLiant ML350 G6; HP Z800 Workstation) has caused numerous kernel panics. This is only when the dahdi service is running with a very simple config (I've defined the first span, the bchans and the dchan, and that's about it). If dahdi is stopped, or the card is removed, everything's fine. I instead installed a Digium TE122P in the ML350 and haven't had any issues. I also haven't seen this in a pre-Nehalem Xeon server. I'm using Asterisk 1.6.1.6, Dahdi 2.2.0 and LibPRI 1.4.10.1, running on CentOS 5.3 (2.6.18-164.el5). Has anyone seen anything similar? - Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whither asterisk-addons?
Correction, I did notice it for download at http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.1.1.tar.gz - Chris On 16 Oct, 2009, at 4:06 PM, Chris Brentano wrote: I noticed that asterisk.org got a redesign, quite recently it seems, which is very nice, but the addons package isn't listed for download any longer, nor are releases posted to http://downloads.asterisk.org/pub/telephony/ That said, looks like it's still available in svn, http://svnview.digium.com/svn/asterisk-addons/tags/1.6.1.1/ So just wondering if addons will be around for the forseeable future? - Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
