Re: [asterisk-users] Asterisk 1.4 Polycom buddy status
Olle E Johansson wrote: 24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling: James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. Again, a relation to call transfer. I think the bug is that we don't handle call-limits properly during a call transfer. That needs to be verified and fixed. There may be, but transfers are not the cause of the issue I describe. SIP interfaces that are members of a Queue, will erratically not be released from 'in-use' when a call is completed. I have tested with both caller terminated and agent terminated calls and both will cause this behavior. It happens on approximately 20% of all calls the queue members receive. Dialing the SIP device with another device will immediately free the status. I wonder if this only happens on calls sent to the SIP device by the Queue application. I will test that today. ___ --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] Asterisk 1.4 Polycom buddy status
26 jan 2007 kl. 16.31 skrev James Fromm: Olle E Johansson wrote: 24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling: James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in- use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. Again, a relation to call transfer. I think the bug is that we don't handle call-limits properly during a call transfer. That needs to be verified and fixed. There may be, but transfers are not the cause of the issue I describe. SIP interfaces that are members of a Queue, will erratically not be released from 'in-use' when a call is completed. I have tested with both caller terminated and agent terminated calls and both will cause this behavior. It happens on approximately 20% of all calls the queue members receive. Dialing the SIP device with another device will immediately free the status. I wonder if this only happens on calls sent to the SIP device by the Queue application. I will test that today. If you are using chan_agent as a proxy channel, check if that changes things. /O ___ --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] Asterisk 1.4 Polycom buddy status
Olle E Johansson wrote: 26 jan 2007 kl. 16.31 skrev James Fromm: Olle E Johansson wrote: 24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling: James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. Again, a relation to call transfer. I think the bug is that we don't handle call-limits properly during a call transfer. That needs to be verified and fixed. There may be, but transfers are not the cause of the issue I describe. SIP interfaces that are members of a Queue, will erratically not be released from 'in-use' when a call is completed. I have tested with both caller terminated and agent terminated calls and both will cause this behavior. It happens on approximately 20% of all calls the queue members receive. Dialing the SIP device with another device will immediately free the status. I wonder if this only happens on calls sent to the SIP device by the Queue application. I will test that today. If you are using chan_agent as a proxy channel, check if that changes things. We don't have agents defined so I don't think chan_agent applies. The Queue's members are assigned through the management port from an application running on the the agent's PC. I think the Queue application loses sync to the SIP channel driver's information containing the state of the SIP interfaces. ___ --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] Asterisk 1.4 Polycom buddy status
Hi there, We traced this issue to transfers (see http://bugs.digium.com/ view.php?id=8848). On Monday, I will be attaching the various debugs to the case as requested. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 26-Jan-07, at 5:16 PM, James Fromm wrote: Olle E Johansson wrote: 26 jan 2007 kl. 16.31 skrev James Fromm: Olle E Johansson wrote: 24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling: James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. Again, a relation to call transfer. I think the bug is that we don't handle call-limits properly during a call transfer. That needs to be verified and fixed. There may be, but transfers are not the cause of the issue I describe. SIP interfaces that are members of a Queue, will erratically not be released from 'in-use' when a call is completed. I have tested with both caller terminated and agent terminated calls and both will cause this behavior. It happens on approximately 20% of all calls the queue members receive. Dialing the SIP device with another device will immediately free the status. I wonder if this only happens on calls sent to the SIP device by the Queue application. I will test that today. If you are using chan_agent as a proxy channel, check if that changes things. We don't have agents defined so I don't think chan_agent applies. The Queue's members are assigned through the management port from an application running on the the agent's PC. I think the Queue application loses sync to the SIP channel driver's information containing the state of the SIP interfaces. ___ --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: Polycom Firmware -- Was: [asterisk-users] Asterisk 1.4 Polycom buddy status
I second that request On 1/25/07, Kenneth Padgett [EMAIL PROTECTED] wrote: I ran into this problem with an early batch of IP650s. Polycom's firmware version 2.0.3b made this issue go away. Speaking of Polycom firmware, anyone have an up to date source for the stuff? The site I ordered from took down their FTP site that had it. :( -Kenneth ___ --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] Asterisk 1.4 Polycom buddy status
24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling: James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in- use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. Again, a relation to call transfer. I think the bug is that we don't handle call-limits properly during a call transfer. That needs to be verified and fixed. /O ___ --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] Asterisk 1.4 Polycom buddy status
I ran into this problem with an early batch of IP650s. Polycom's firmware version 2.0.3b made this issue go away. Thanks, Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 23, 2007, at 10:09 AM, Chris Bullock wrote: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris ___ --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] Asterisk 1.4 Polycom buddy status
We also use Polycom IP650 phones. They are assigned to our customer service department. Each SIP interface is a member of our customer service Queue in Asterisk. The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. When this happens there is no SIP channel and the SIP peer appears normal. I have been unable to isolate a procedure to duplicate the problem. It happens erratically to all member interfaces throughout the day. I know that removing the call-limit option from the device's config will stop the problem. This will also remove the ability for the SIP channel driver to track the device's state so we can't remove it permanently. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. James Andrewartha wrote: Olle E Johansson wrote: 23 jan 2007 kl. 16.09 skrev Chris Bullock: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. I've seen reports on it, but haven't been able to repeat this. I need to know a way to force this to happen, repeatably. If I can get that, I can propably trace it and fix it. It can also happen if you have packet loss in the network, of course. I've seen it happen when asterisk restarts (or possibly even just reloads SIP) without the phone being restarted - it's generally accompanied by -- Incoming call: Got SIP response 500 Internal Server Error back from 10.0.0.51 on the console. I think the status gets stuck as available most of the time, but you don't notice it because that's the default. ___ --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] Asterisk 1.4 Polycom buddy status
Our 650s are running 2.0.3b. The problem still exists for us. We see the devices as members of our customer service queue stick on 'in-use' in the Queue application while the device has no active SIP channel and will accept calls. Removing 'call-limit' from the sip.conf in Asterisk for the device will fix the issue. This however will also keep the SIP channel driver in Asterisk from tracking the state of the device. Bryan M. Johns wrote: I ran into this problem with an early batch of IP650s. Polycom's firmware version 2.0.3b made this issue go away. Thanks, Bryan M. Johns Partner *Shelton | Johns Technology Group* office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 *http://www.sheltonjohns.com* http://www.sheltonjohns.com/ On Jan 23, 2007, at 10:09 AM, Chris Bullock wrote: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris ___ --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] Asterisk 1.4 Polycom buddy status
James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. ___ --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] Asterisk 1.4 Polycom buddy status
Yeah, we don't use Buddy Watch. We don't use call-limit because we want call-limit. We use it because it's the only way, that I'm aware of, to get the SIP channel driver to monitor the state of the member SIP interface. We use autopause=yes and ringinuse=no in our customer service queue configuration. Without specifying call-limit, the Queue application continues to send new calls to member interfaces that are in-use or busy. The SIP device replies to Asterisk saying it's busy and the Queue application pauses the member because of autopause. With call-limit enabled and set to any number, the Queue application knows that the member interface is busy and will not send new calls. I replied to this post describing our findings with the Queue application because it sounds like the same behavior occurs with hints and buddy watch. The state detection in the SIP channel driver appears suspect to me. Eric ManxPower Wieling wrote: James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. ___ --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
Polycom Firmware -- Was: [asterisk-users] Asterisk 1.4 Polycom buddy status
I ran into this problem with an early batch of IP650s. Polycom's firmware version 2.0.3b made this issue go away. Speaking of Polycom firmware, anyone have an up to date source for the stuff? The site I ordered from took down their FTP site that had it. :( -Kenneth ___ --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] Asterisk 1.4 Polycom buddy status
I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris ___ --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] Asterisk 1.4 Polycom buddy status
23 jan 2007 kl. 16.09 skrev Chris Bullock: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. I've seen reports on it, but haven't been able to repeat this. I need to know a way to force this to happen, repeatably. If I can get that, I can propably trace it and fix it. It can also happen if you have packet loss in the network, of course. /O ___ --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] Asterisk 1.4 Polycom buddy status
Olle E Johansson wrote: 23 jan 2007 kl. 16.09 skrev Chris Bullock: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. I've seen reports on it, but haven't been able to repeat this. I need to know a way to force this to happen, repeatably. If I can get that, I can propably trace it and fix it. It can also happen if you have packet loss in the network, of course. I've seen it happen when asterisk restarts (or possibly even just reloads SIP) without the phone being restarted - it's generally accompanied by -- Incoming call: Got SIP response 500 Internal Server Error back from 10.0.0.51 on the console. I think the status gets stuck as available most of the time, but you don't notice it because that's the default. -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --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
