Re: [asterisk-users] Cancel a ringing SIP call when the other party disconnect
On Fri, Feb 21, 2014 at 10:55 AM, Ruddy Gbaguidi plugwo...@micnes.com wrote: Hi, Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged). But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with the qualify. Is there any setup or asterisk configuration I need to enable to have A close its call ? Note: when A is already talking with B, the call is hanged up on rtp timeout. But not during the Ringing phase. I'm not sure it is possible to configure Asterisk to hang up during the ringing phase when a peer/endpoint becomes unreachable. I don't see an option or parameter for that behavior. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call transfer problem.
Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer problem.
I'm sorry, I should have mentioned that he's doing a phone-based transfer, not an asterisk-based transfer. Mike. On Mon, Feb 24, 2014 at 1:30 PM, Don Kelly d...@donkelly.biz wrote: Does he complete the call as a supervised transfer--waits for the called party to answer before completing the transfer? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, February 24, 2014 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call transfer problem. Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer problem.
Does he complete the call as a supervised transfer--waits for the called party to answer before completing the transfer? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, February 24, 2014 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call transfer problem. Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is being initiated by phone A, not phone B! Thus the call transfer fails. I have other users, with other phones, that are able to transfer just fine. What could be different with this particular user? Any ideas? Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Add SIPCALLID of egress leg to CDR
Hey all, I've been fighting with this all morning, and I feel like this should be a relatively simple task, but I just can't get it to work. I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip trunk to my carrier. What I'm trying to accomplish is simply adding the asterisk generated SIPCALLID of the leg between asterisk and the carrier to the CDR when it's written. (This is in addition to the SIPCALLID of the original call leg, which I've already added.) I've been able to print the SIPCALLID for the carrier leg to the CLI if I jump into a Macro or Gosub from the Dial command (using the M or b options respectively), but that's about it. From what I can tell, this should be easily accomplished with variable inheritance, but based on what I'm seeing during test that seems to be a one way relationship, as updates in a child channel don't seem to reach the parent. Am I missing something? Is there a better way to go about this? The few questions I've found on forums and mailing list archives on this subject are all from 2007 and earlier, so I'm hoping something new has come along in the last few years and I just haven't found it anywhere. Can anyone help? TIA, Ricky -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add SIPCALLID of egress leg to CDR
On Mon, Feb 24, 2014 at 1:38 PM, Ricky Hartmann rhartma...@gmail.com wrote: Hey all, I've been fighting with this all morning, and I feel like this should be a relatively simple task, but I just can't get it to work. I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip trunk to my carrier. What I'm trying to accomplish is simply adding the asterisk generated SIPCALLID of the leg between asterisk and the carrier to the CDR when it's written. (This is in addition to the SIPCALLID of the original call leg, which I've already added.) I've been able to print the SIPCALLID for the carrier leg to the CLI if I jump into a Macro or Gosub from the Dial command (using the M or b options respectively), but that's about it. From what I can tell, this should be easily accomplished with variable inheritance, but based on what I'm seeing during test that seems to be a one way relationship, as updates in a child channel don't seem to reach the parent. Am I missing something? Is there a better way to go about this? The few questions I've found on forums and mailing list archives on this subject are all from 2007 and earlier, so I'm hoping something new has come along in the last few years and I just haven't found it anywhere. Can anyone help? You're correct - variable inheritance only goes in one direction, from the parent to the child. Since the child channel is going to be Party B in the CDR, variables set on it are unlikely to show up. I can think of a few options: 1) Use the userfield on the outbound channel to store the SIPCALLID. A little known fact of the userfield is that, on two channels in a bridge together, the userfields are concatenated together using a ';' as a delimiter. 2) Use the MASTER_CHANNEL function to reach back to the parent channel and set the CDR variable there. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add SIPCALLID of egress leg to CDR
Matt, Both of those suggestions worked perfectly. Thanks! Ricky Hartmann | PO Box 1354 Jackson, NJ 08527 Phone - (732) 580-5853 | eFax - (716) 809-3073 On Mon, Feb 24, 2014 at 3:51 PM, Matthew Jordan mjor...@digium.com wrote: On Mon, Feb 24, 2014 at 1:38 PM, Ricky Hartmann rhartma...@gmail.com wrote: Hey all, I've been fighting with this all morning, and I feel like this should be a relatively simple task, but I just can't get it to work. I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip trunk to my carrier. What I'm trying to accomplish is simply adding the asterisk generated SIPCALLID of the leg between asterisk and the carrier to the CDR when it's written. (This is in addition to the SIPCALLID of the original call leg, which I've already added.) I've been able to print the SIPCALLID for the carrier leg to the CLI if I jump into a Macro or Gosub from the Dial command (using the M or b options respectively), but that's about it. From what I can tell, this should be easily accomplished with variable inheritance, but based on what I'm seeing during test that seems to be a one way relationship, as updates in a child channel don't seem to reach the parent. Am I missing something? Is there a better way to go about this? The few questions I've found on forums and mailing list archives on this subject are all from 2007 and earlier, so I'm hoping something new has come along in the last few years and I just haven't found it anywhere. Can anyone help? You're correct - variable inheritance only goes in one direction, from the parent to the child. Since the child channel is going to be Party B in the CDR, variables set on it are unlikely to show up. I can think of a few options: 1) Use the userfield on the outbound channel to store the SIPCALLID. A little known fact of the userfield is that, on two channels in a bridge together, the userfields are concatenated together using a ';' as a delimiter. 2) Use the MASTER_CHANNEL function to reach back to the parent channel and set the CDR variable there. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI can not receive events unless keyboard input
Hi, This is evenily. My one Asterisk server and event listen App was working well for several month. Yesterday My event listen app stop work suddenly. I telnet to AMI via localhost 5038 to check the events, and I find Asterisk does not push the events until I press the Enter key. When I press Enter key, all events was displayed, then stop display new events until I press Enter key Again. BTW, My others Asterisk box is not the same problem. Asterisk version: 1.8.7.0 Help please! -- _ -- 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
