[asterisk-users] Call status register
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! Some time ago I'm using Asterisk (currently 1.8.10.0) at home to manage the calls. Nothing yet very complex, just something compiled by me using the source code from the official site of the project and configuring the files manually to both Asterisk and DAHDI. For now I'm not using any GUI, but when I have more time, I plan to try something in the future, for example, to make a statistic of the calls. But, thinking about the statistics of the calls, in the last days I was taking a look at the /var/log/asterisk/cdr-csv/Master.csv file, which I understand is where the calls are registered. But all seem to have a ANSWERED state, even those receiving a busy tone. This happens with both internal calls between SIP extension and from SIP to PSTN. A test I did is putting a Grandstream BT200 on DND mode (Do Not Disturb) and call it from a softphone. While the softphone receives the message that the extension is busy, the CDR registered the call as ANSWERED. Not sure if it's something usually due to the way it is configured the dialplan or any other configuration issue. Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAk+LIeAACgkQZpa/GxTmHTcvQwCdEqsEI3Y9ka5Z41CTXlzerPbD qQIAnAwEpac8dcLh5t84XLuDryqFuD40 =R07r -END PGP SIGNATURE- -- _ -- 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] Call status subscriptions on multiple servers
Hello List Is there a way to have hints sent between multiple servers? We are currently implementing a cluster solution for our asterisk servers, and the problem is this. User A registers on Asterisk 1 and user B registers on Asterisk 2. User A subscribes to user B's status, through SIP NOTIFY messages. As user B is registered to Asterisk 2, and not Asterisk 1, the NOTIFY messages are only sent on Asterisk 2, and Asterisk 1 does not know the status of user B. Is there a way to replicate subscription info between asterisk servers? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call status subscriptions on multiple servers
- Jon Schøpzinsky [EMAIL PROTECTED] wrote: Is there a way to replicate subscription info between asterisk servers? Not at this time, no. That will be probably be worked on during the next development cycle. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Call status subscriptions on multiple servers
Can you then inform me on what structures this information is stored in, in the asterisk code? Then ill try to do a quick dirty version of the replication. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Kevin P. Fleming Sendt: 9. juni 2006 16:25 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Call status subscriptions on multiple servers - Jon Schøpzinsky [EMAIL PROTECTED] wrote: Is there a way to replicate subscription info between asterisk servers? Not at this time, no. That will be probably be worked on during the next development cycle. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Call status subscriptions on multiple servers
- Jon Schøpzinsky [EMAIL PROTECTED] wrote: Can you then inform me on what structures this information is stored in, in the asterisk code? Then ill try to do a quick dirty version of the replication. It's in a lot of places... start with ast_extension_state() and work your way backwards. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Status from a IAX trunk to a Zaptel trunk
Hello, I'm actually sending calls from a * box1 to * box2 for termination through a zaptel channel using an IAX2 channel. My problem is that * box1 is always getting an ANSWER on the call even if * box2 cannot successfully terminate the call. The Zaptel channel on * box2 is always ANSWERING before sending the call on its zaptel channel. How can i setup one of the 2 boxes to get DIALSTATUS to ANSWER only if the call is correctly terminated by * box2 ? Thanks for your help. Moussgrd. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call status after Answer
Hi, I setup asterisk as an autoattendant. When I call using IAX I get the autoattendent okay, but when I dial one of the extensions, there is no ringing sound passed back to the caller. It happens when I use my DID number, but I also configured a context so I can get it to happen with Firefly (iax client) as the caller. It seems that once the Answer command is executed in the dialplan, status commands (RINGING, etc) aren't passed back through the IAX channel. My only workaround has been to use music on hold instead of making a ringing sound. Has anyone seen this, or a solution. It seems basic, but I have been working all day on it. I've tried this with 1-0-2, 1-0.5 and the head version, but all behave the same. The IAX trace shows that RINGING is getting sent back to the client. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call status after Answer
Who is your DID provider? On Sun, 6 Feb 2005 15:03:19 -0500, Scott Simpson [EMAIL PROTECTED] wrote: Hi, I setup asterisk as an autoattendant. When I call using IAX I get the autoattendent okay, but when I dial one of the extensions, there is no ringing sound passed back to the caller. It happens when I use my DID number, but I also configured a context so I can get it to happen with Firefly (iax client) as the caller. It seems that once the Answer command is executed in the dialplan, status commands (RINGING, etc) aren't passed back through the IAX channel. My only workaround has been to use music on hold instead of making a ringing sound. Has anyone seen this, or a solution. It seems basic, but I have been working all day on it. I've tried this with 1-0-2, 1-0.5 and the head version, but all behave the same. The IAX trace shows that RINGING is getting sent back to the client. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Status
Shaun Tierney wrote:
When I use the Dial command to connect a call using my Asterisk PBX, it
seems that the PBX says that the call was answered right when the two
channels are bridged together, rather than when the actually callee answers
their phone. I would like to be able to detect the actual call status and
respond to it in the dialplan. ${DIALSTATUS} just seems to tell me whether
or not the channel answered rather than the actual callee, so it equals
ANSWER every time I dial because the bridge is automatic. Is there any
command or variable out there that will allow me to determine the actual
call status?
It's not the Dial command or Asterisk. I'm willing to be you are dialing
an outside line, right?
Well, on analog lines it can sometime happens that from your PBX point
of view the phone line is being answered by a switch or pbx on the other
side which continues to send dialing sounds down the line until the
other side actually answers.
For a simple analog phone it doesn't matter - you simply hear ring
sounds until someone answers. But for PBX equipment it's a problem -
there is no way [1] for Asterisk to know that the line hasn't really
been answered yet.
Gilad
[1] Well, I lied - there is a way, which is the callprogress feature.
it's far from perfect (or in some states, working) though. Look for
callprogress on voip-info.org
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[Asterisk-Users] Call Status
When I use the Dial command to connect a call using my Asterisk PBX, it
seems that the PBX says that the call was answered right when the two
channels are bridged together, rather than when the actually callee answers
their phone. I would like to be able to detect the actual call status and
respond to it in the dialplan. ${DIALSTATUS} just seems to tell me whether
or not the channel answered rather than the actual callee, so it equals
ANSWER every time I dial because the bridge is automatic. Is there any
command or variable out there that will allow me to determine the actual
call status?
Thanks,
Shaun Tierney
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Re: [Asterisk-Users] Call Status
On Wed, 2004-11-17 at 16:17 -0600, Shaun Tierney wrote:
When I use the Dial command to connect a call using my Asterisk PBX, it
seems that the PBX says that the call was answered right when the two
channels are bridged together, rather than when the actually callee answers
their phone. I would like to be able to detect the actual call status and
respond to it in the dialplan. ${DIALSTATUS} just seems to tell me whether
or not the channel answered rather than the actual callee, so it equals
ANSWER every time I dial because the bridge is automatic. Is there any
command or variable out there that will allow me to determine the actual
call status?
It might be a good idea to also include what channels and interfaces you
are using. Sounds like you are dialing back out via a analog line and
should read the archives about callprogress.
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Steven Critchfield [EMAIL PROTECTED]
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