Re: [asterisk-users] Queue and Interface time out
Okay, that makes sense. I wasn't thinking about the SIP driver needing to be told to track the peer's status. I assumed it just did that. So now there's a new problem. The Queue application doesn't always clear the member interface's status after completing a call. The SIP peer no longer has an active channel but the queue will still show the member 'In use'. The occurrence of this is erratic and I have been unable to determine any commonalities among the callers or members other than that it happens to all members. Connecting to the peer outside of the queue will clear the status. Any ideas? Thanks, James Watkins, Bradley wrote: What it actually does is tell the SIP channel driver to track whether or not any given peer has a call to it. It can then subsequently inform the Queue application so that another call will not be given to that user. If you did not have the ringinuse=no in your queue definition, you would then be able to receive up to 5 simultaneous calls (after five, then the SIP channel driver would return busy and Queue wouldn't be able to dial that peer). Regards, - Brad From: [EMAIL PROTECTED] on behalf of James Fromm Sent: Fri 1/19/2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out That worked. I don't understand what call-limit has to do with this. I set it to 5. Why does that keep the member interface from getting a second call from the Queue application? I would think it would allow the member interface to get up to 5 calls. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 ___ --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 ___ --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] Queue and Interface time out
Okay, that makes sense. I wasn't thinking about the SIP driver needing to be told to track the peer's status. I assumed it just did that. So now there's a new problem. The Queue application doesn't always clear the member interface's status after completing a call. The SIP peer no longer has an active channel but the queue will still show the member 'In use'. The occurrence of this is erratic and I have been unable to determine any commonalities among the callers or members other than that it happens to all members. Connecting to the peer outside of the queue will clear the status. Any ideas? Thanks, James Watkins, Bradley wrote: What it actually does is tell the SIP channel driver to track whether or not any given peer has a call to it. It can then subsequently inform the Queue application so that another call will not be given to that user. If you did not have the ringinuse=no in your queue definition, you would then be able to receive up to 5 simultaneous calls (after five, then the SIP channel driver would return busy and Queue wouldn't be able to dial that peer). Regards, - Brad From: [EMAIL PROTECTED] on behalf of James Fromm Sent: Fri 1/19/2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out That worked. I don't understand what call-limit has to do with this. I set it to 5. Why does that keep the member interface from getting a second call from the Queue application? I would think it would allow the member interface to get up to 5 calls. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 ___ --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] Queue and Interface time out
What it actually does is tell the SIP channel driver to track whether or not any given peer has a call to it. It can then subsequently inform the Queue application so that another call will not be given to that user. If you did not have the ringinuse=no in your queue definition, you would then be able to receive up to 5 simultaneous calls (after five, then the SIP channel driver would return busy and Queue wouldn't be able to dial that peer). Regards, - Brad From: [EMAIL PROTECTED] on behalf of James Fromm Sent: Fri 1/19/2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out That worked. I don't understand what call-limit has to do with this. I set it to 5. Why does that keep the member interface from getting a second call from the Queue application? I would think it would allow the member interface to get up to 5 calls. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. winmail.dat___ --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] Queue and Interface time out
Does anyone have ringinuse=no and autopause=yes working together in queues.conf? We assign members to our customer service queue from an application based on actions the agents take on their PCs. No static agents are defined in agents.conf and no members are specified in queues.conf. All member interfaces are SIP with only the basics configured in sip.conf. Even with 'ringinuse=no' configured, the Queue application continues to send callers to busy members causing them to get paused when their SIP device returns that it's busy. Does the Queue application need hints for member interfaces to determine their status? Thanks, James James Fromm wrote: No, call-limit is not being used. Do you have ringinuse=no working? Has anyone seen it work? Each SIP device has a very minimal config in sip.conf. Here's a show sip peer: * Name : 3207 Secret : Set MD5Secret: Not set Context : outbound Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : Sam 3207 MaxCallBR: 384 kbps Expire : 40 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 216.239.128.189 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 3207 SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (ulaw:20) Auto-Framing: No Status : OK (14 ms) Useragent: PolycomSoundPointIP-SPIP_650-UA/2.0.3.0131 Reg. Contact : sip:[EMAIL PROTECTED] Watkins, Bradley wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Thursday, January 18, 2007 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out I guess I'm missing something else. 'ringinuse = no' doesn't change anything. While on a call, the queue still sends another call and proceeds to set the member paused after receiving 'Busy Here' back from the SIP device. My queues.conf is: [general] persistentmembers = no [customerservice] persistentmembers = no musiconhold = default reportholdtime = no strategy = leastrecent timeout = 20 retry = 5 wrapuptime = 30 ;allow agents 30 seconds to wrap up work maxlen = 0 ;unlimited callers on hold servicelevel = 60 ;calls must be answered within 60 seconds announce-holdtime = no autopause = yes ringinuse = no joinempty = yes leavewhenempty = no Am I missing something obvious? What do your SIP peers look like? Are you using the call-limit feature? - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 ___ --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 ___ --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] Queue and Interface time out
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Friday, January 19, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out Does anyone have ringinuse=no and autopause=yes working together in queues.conf? We assign members to our customer service queue from an application based on actions the agents take on their PCs. No static agents are defined in agents.conf and no members are specified in queues.conf. All member interfaces are SIP with only the basics configured in sip.conf. Even with 'ringinuse=no' configured, the Queue application continues to send callers to busy members causing them to get paused when their SIP device returns that it's busy. Does the Queue application need hints for member interfaces to determine their status? Thanks, James Queue does not need hints, but it does need the channel driver (in your case SIP) to inform it whether or not the member interface is in use. That is actually why I asked about call-limit. Can you try adding a call-limit (even if it's 10 or 20 or whatever) and see if that solves your problem? Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Queue and Interface time out
That worked. I don't understand what call-limit has to do with this. I set it to 5. Why does that keep the member interface from getting a second call from the Queue application? I would think it would allow the member interface to get up to 5 calls. Watkins, Bradley wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Friday, January 19, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out Does anyone have ringinuse=no and autopause=yes working together in queues.conf? We assign members to our customer service queue from an application based on actions the agents take on their PCs. No static agents are defined in agents.conf and no members are specified in queues.conf. All member interfaces are SIP with only the basics configured in sip.conf. Even with 'ringinuse=no' configured, the Queue application continues to send callers to busy members causing them to get paused when their SIP device returns that it's busy. Does the Queue application need hints for member interfaces to determine their status? Thanks, James Queue does not need hints, but it does need the channel driver (in your case SIP) to inform it whether or not the member interface is in use. That is actually why I asked about call-limit. Can you try adding a call-limit (even if it's 10 or 20 or whatever) and see if that solves your problem? Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 ___ --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] Queue and Interface time out
I guess I'm missing something else. 'ringinuse = no' doesn't change anything. While on a call, the queue still sends another call and proceeds to set the member paused after receiving 'Busy Here' back from the SIP device. My queues.conf is: [general] persistentmembers = no [customerservice] persistentmembers = no musiconhold = default reportholdtime = no strategy = leastrecent timeout = 20 retry = 5 wrapuptime = 30 ;allow agents 30 seconds to wrap up work maxlen = 0 ;unlimited callers on hold servicelevel = 60 ;calls must be answered within 60 seconds announce-holdtime = no autopause = yes ringinuse = no joinempty = yes leavewhenempty = no Am I missing something obvious? Thanks, James James Fromm wrote: DoH! I missed that ringinuse. Thanks! Julian Lyndon-Smith wrote: James Fromm wrote: Hmm, the use of autopause in queues.conf introduces a new issue. When a queue member is on a call, the queue continues to try to send calls to the member's interface. Getting the 'Busy Here' response from the SIP device causes the caller to continue holding. The new issue is that autopause appears to pause the member interface even when they're on another call. Am I missing something or is this the expected behavior? queues.conf: ; Autopause will pause a queue member if they fail to answer a call ; ;autopause=yes I didn't expect the Queue application to try member interfaces that are busy. queues.conf: ; If you want the queue to avoid sending calls to members whose devices are ; known to be 'in use' (via the channel driver supporting that device state) ; uncomment this option. (Note: only the SIP channel driver currently is able ; to report 'in use'.) ; ; ringinuse = no Julian Thanks, James James Fromm wrote: NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf ___ --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 ___ --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 ___ --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] Queue and Interface time out
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Thursday, January 18, 2007 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out I guess I'm missing something else. 'ringinuse = no' doesn't change anything. While on a call, the queue still sends another call and proceeds to set the member paused after receiving 'Busy Here' back from the SIP device. My queues.conf is: [general] persistentmembers = no [customerservice] persistentmembers = no musiconhold = default reportholdtime = no strategy = leastrecent timeout = 20 retry = 5 wrapuptime = 30 ;allow agents 30 seconds to wrap up work maxlen = 0 ;unlimited callers on hold servicelevel = 60 ;calls must be answered within 60 seconds announce-holdtime = no autopause = yes ringinuse = no joinempty = yes leavewhenempty = no Am I missing something obvious? What do your SIP peers look like? Are you using the call-limit feature? - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Queue and Interface time out
No, call-limit is not being used. Do you have ringinuse=no working? Has anyone seen it work? Each SIP device has a very minimal config in sip.conf. Here's a show sip peer: * Name : 3207 Secret : Set MD5Secret: Not set Context : outbound Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : Sam 3207 MaxCallBR: 384 kbps Expire : 40 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 216.239.128.189 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 3207 SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (ulaw:20) Auto-Framing: No Status : OK (14 ms) Useragent: PolycomSoundPointIP-SPIP_650-UA/2.0.3.0131 Reg. Contact : sip:[EMAIL PROTECTED] Watkins, Bradley wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Thursday, January 18, 2007 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out I guess I'm missing something else. 'ringinuse = no' doesn't change anything. While on a call, the queue still sends another call and proceeds to set the member paused after receiving 'Busy Here' back from the SIP device. My queues.conf is: [general] persistentmembers = no [customerservice] persistentmembers = no musiconhold = default reportholdtime = no strategy = leastrecent timeout = 20 retry = 5 wrapuptime = 30 ;allow agents 30 seconds to wrap up work maxlen = 0 ;unlimited callers on hold servicelevel = 60 ;calls must be answered within 60 seconds announce-holdtime = no autopause = yes ringinuse = no joinempty = yes leavewhenempty = no Am I missing something obvious? What do your SIP peers look like? Are you using the call-limit feature? - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 ___ --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] Queue and Interface time out
Hmm, the use of autopause in queues.conf introduces a new issue. When a queue member is on a call, the queue continues to try to send calls to the member's interface. Getting the 'Busy Here' response from the SIP device causes the caller to continue holding. The new issue is that autopause appears to pause the member interface even when they're on another call. Am I missing something or is this the expected behavior? I didn't expect the Queue application to try member interfaces that are busy. Thanks, James James Fromm wrote: NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf James Fromm wrote: We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James ___ --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 ___ --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 ___ --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 ___ --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] Queue and Interface time out
James Fromm wrote: Hmm, the use of autopause in queues.conf introduces a new issue. When a queue member is on a call, the queue continues to try to send calls to the member's interface. Getting the 'Busy Here' response from the SIP device causes the caller to continue holding. The new issue is that autopause appears to pause the member interface even when they're on another call. Am I missing something or is this the expected behavior? queues.conf: ; Autopause will pause a queue member if they fail to answer a call ; ;autopause=yes I didn't expect the Queue application to try member interfaces that are busy. queues.conf: ; If you want the queue to avoid sending calls to members whose devices are ; known to be 'in use' (via the channel driver supporting that device state) ; uncomment this option. (Note: only the SIP channel driver currently is able ; to report 'in use'.) ; ; ringinuse = no Julian Thanks, James James Fromm wrote: NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf ___ --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] Queue and Interface time out
DoH! I missed that ringinuse. Thanks! Julian Lyndon-Smith wrote: James Fromm wrote: Hmm, the use of autopause in queues.conf introduces a new issue. When a queue member is on a call, the queue continues to try to send calls to the member's interface. Getting the 'Busy Here' response from the SIP device causes the caller to continue holding. The new issue is that autopause appears to pause the member interface even when they're on another call. Am I missing something or is this the expected behavior? queues.conf: ; Autopause will pause a queue member if they fail to answer a call ; ;autopause=yes I didn't expect the Queue application to try member interfaces that are busy. queues.conf: ; If you want the queue to avoid sending calls to members whose devices are ; known to be 'in use' (via the channel driver supporting that device state) ; uncomment this option. (Note: only the SIP channel driver currently is able ; to report 'in use'.) ; ; ringinuse = no Julian Thanks, James James Fromm wrote: NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf ___ --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 ___ --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] Queue and Interface time out
I guess I'm missing something else. 'ringinuse = no' doesn't change anything. While on a call, the queue still sends another call and proceeds to set the member paused after receiving 'Busy Here' back from the SIP device. My queues.conf is: [general] persistentmembers = no [customerservice] persistentmembers = no musiconhold = default reportholdtime = no strategy = leastrecent timeout = 20 retry = 5 wrapuptime = 30 ;allow agents 30 seconds to wrap up work maxlen = 0 ;unlimited callers on hold servicelevel = 60 ;calls must be answered within 60 seconds announce-holdtime = no autopause = yes ringinuse = no joinempty = yes leavewhenempty = no I'm I missing something obvious? Thanks, James James Fromm wrote: DoH! I missed that ringinuse. Thanks! Julian Lyndon-Smith wrote: James Fromm wrote: Hmm, the use of autopause in queues.conf introduces a new issue. When a queue member is on a call, the queue continues to try to send calls to the member's interface. Getting the 'Busy Here' response from the SIP device causes the caller to continue holding. The new issue is that autopause appears to pause the member interface even when they're on another call. Am I missing something or is this the expected behavior? queues.conf: ; Autopause will pause a queue member if they fail to answer a call ; ;autopause=yes I didn't expect the Queue application to try member interfaces that are busy. queues.conf: ; If you want the queue to avoid sending calls to members whose devices are ; known to be 'in use' (via the channel driver supporting that device state) ; uncomment this option. (Note: only the SIP channel driver currently is able ; to report 'in use'.) ; ; ringinuse = no Julian Thanks, James James Fromm wrote: NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf ___ --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 ___ --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 ___ --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] Queue and Interface time out
We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James ___ --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] Queue and Interface time out
try autopause in queues.conf James Fromm wrote: We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James ___ --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 ___ --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] Queue and Interface time out
NICE! That did the trick. Thanks! Julian Lyndon-Smith wrote: try autopause in queues.conf James Fromm wrote: We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James ___ --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 ___ --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 ___ --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
