[asterisk-users] 1.4.22 CALLERID(num)
We have a weird thing happening with the caller id when a call is dialed to a SIP device registered to 1.4.22. We're preparing 1.4.22 on a development machine for switch to live. The callerid number displayed on the SIP device (Polycom or soft phone) is a full SIP URL, i.e. sip:[EMAIL PROTECTED] The name is correct. It happens even when one registered SIP device dials another using the extension defined in sip.conf, i.e. 3201 calls 3213, 3213 sees sip:[EMAIL PROTECTED] for the number. Again, the name is correct. Anyone have a clue what's causing this. Our live machine is 1.4.20.1 and it handles the name and number the good old fashioned way. The same call would display only the number, 3105552121, for the callerid number. Thanks, Jay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP interface status
I've discovered that the status of a SIP device doesn't get passed as in-use when on an outbound call. Viewing the debug log the status is always passed as 'not in use' when on the outbound call. The sip_devicestate function doesn't appear to check the user object at all. The devices are configured as friends in sip.conf. Being both a peer and a user, the device is found as a peer in the sip_devicestate function but then not found in use because only the peer object is checked. If the device is configured as a user in sip.conf, then the status is returned as INVALID because in the sip_devicestate function it doesn't find a peer to check. Looking at the SVN repository, the function appears to have never checked the user object. Shouldn't the device be defined as in-use even when on an outbound call? Does this function need to be rewritten? Anyone have a solution? Thanks, James ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom headset button blinking
Does anyone know what it means when the headset button on Polycom phones is blinking? The blinking state is achieved by hitting the button twice while on-hook. First press activates the headset circuit and takes the phone off-hook. Second press deactivates the headset circuit, puts the phone on-hook, and starts the headset button blinking. Third press stops the headset button blinking. Any ideas? Thanks, 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] SIP interface status and calllimit
I've reviewed the bugs reports. I didn't see anything that applied to this. Have you? Could you point it out to me? Olle E Johansson wrote: 21 feb 2007 kl. 15.50 skrev James Fromm: Anybody seen this behavior? To determine if it's my config or a bug, could I trouble someone running Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP interface as a test? After a few hours a 'sip show inuse' should indicate the interface is on calls that it isn't. The incorrect count can be cleared up by ringing the interface for how ever many calls are incorrect. Beware, removing call-limit will require a restart to take effect. Thanks in advance for any help. A good way to check is to visit the bug tracker at bugs.digium.com If you do, you will find a few bug reports and also notice a few that has been resolved in Asterisk 1.4 svn, which is the base for the coming 1.4.1 release. Please try with latest 1.4 from subversion to test if the behaviour is fixed. Thanks, /Olle ___ --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] SIP interface status and calllimit
Nevermind, I found it. I'll put up an SVN version in my dev environment today. Thanks. James Fromm wrote: I've reviewed the bugs reports. I didn't see anything that applied to this. Have you? Could you point it out to me? Olle E Johansson wrote: 21 feb 2007 kl. 15.50 skrev James Fromm: Anybody seen this behavior? To determine if it's my config or a bug, could I trouble someone running Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP interface as a test? After a few hours a 'sip show inuse' should indicate the interface is on calls that it isn't. The incorrect count can be cleared up by ringing the interface for how ever many calls are incorrect. Beware, removing call-limit will require a restart to take effect. Thanks in advance for any help. A good way to check is to visit the bug tracker at bugs.digium.com If you do, you will find a few bug reports and also notice a few that has been resolved in Asterisk 1.4 svn, which is the base for the coming 1.4.1 release. Please try with latest 1.4 from subversion to test if the behaviour is fixed. Thanks, /Olle ___ --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] SIP interface status and calllimit
Anybody seen this behavior? To determine if it's my config or a bug, could I trouble someone running Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP interface as a test? After a few hours a 'sip show inuse' should indicate the interface is on calls that it isn't. The incorrect count can be cleared up by ringing the interface for how ever many calls are incorrect. Beware, removing call-limit will require a restart to take effect. Thanks in advance for any help. James Fromm wrote: It does. Eric ManxPower Wieling wrote: Maybe Queue doesn't consider a SIP account that returns BUSY as in use. That would be the only case where I could see needing call-limit. James Fromm wrote: We do the same thing only we use ringinuse=no and autopause=yes for the queue. With autopause, if the agent is busy their interface in the queue gets paused. Setting call-limit for the SIP interface is the only way to make ringinuse=no work. Eric ManxPower Wieling wrote: James Fromm wrote: There is an issue when using call-limit for a SIP interface in sip.conf. The call count does not properly reset when some calls end. The problem happens regardless of which side of the connection ends the call. It happens on all calls including calls from SIP interface to SIP interface (with no reinvite) within the same Asterisk server. I have not been able to determine a definite pattern. I can call from one interface to another 50 times before it happens and sometimes it happens after only 2 calls. We have to enable call-limit for our customer service queue agents so that the ringinuse option in queues.conf will work properly. Has anyone else seen this issue? Any ideas? This doesn't really help you, but might help others when deciding how to design their Asterisk system. On our phones we set call waiting off and each line appearance registers as a separate SIP user. This avoids all this silliness with call limits, group limits, etc. This also allows us total control about which call appearance a call shows up on, roll over and hunting features, etc. It does require a little more work in the dialplan, but for our needs it is well worth 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] SIP interface status and calllimit
We do the same thing only we use ringinuse=no and autopause=yes for the queue. With autopause, if the agent is busy their interface in the queue gets paused. Setting call-limit for the SIP interface is the only way to make ringinuse=no work. Eric ManxPower Wieling wrote: James Fromm wrote: There is an issue when using call-limit for a SIP interface in sip.conf. The call count does not properly reset when some calls end. The problem happens regardless of which side of the connection ends the call. It happens on all calls including calls from SIP interface to SIP interface (with no reinvite) within the same Asterisk server. I have not been able to determine a definite pattern. I can call from one interface to another 50 times before it happens and sometimes it happens after only 2 calls. We have to enable call-limit for our customer service queue agents so that the ringinuse option in queues.conf will work properly. Has anyone else seen this issue? Any ideas? This doesn't really help you, but might help others when deciding how to design their Asterisk system. On our phones we set call waiting off and each line appearance registers as a separate SIP user. This avoids all this silliness with call limits, group limits, etc. This also allows us total control about which call appearance a call shows up on, roll over and hunting features, etc. It does require a little more work in the dialplan, but for our needs it is well worth 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] SIP interface status and calllimit
It does. Eric ManxPower Wieling wrote: Maybe Queue doesn't consider a SIP account that returns BUSY as in use. That would be the only case where I could see needing call-limit. James Fromm wrote: We do the same thing only we use ringinuse=no and autopause=yes for the queue. With autopause, if the agent is busy their interface in the queue gets paused. Setting call-limit for the SIP interface is the only way to make ringinuse=no work. Eric ManxPower Wieling wrote: James Fromm wrote: There is an issue when using call-limit for a SIP interface in sip.conf. The call count does not properly reset when some calls end. The problem happens regardless of which side of the connection ends the call. It happens on all calls including calls from SIP interface to SIP interface (with no reinvite) within the same Asterisk server. I have not been able to determine a definite pattern. I can call from one interface to another 50 times before it happens and sometimes it happens after only 2 calls. We have to enable call-limit for our customer service queue agents so that the ringinuse option in queues.conf will work properly. Has anyone else seen this issue? Any ideas? This doesn't really help you, but might help others when deciding how to design their Asterisk system. On our phones we set call waiting off and each line appearance registers as a separate SIP user. This avoids all this silliness with call limits, group limits, etc. This also allows us total control about which call appearance a call shows up on, roll over and hunting features, etc. It does require a little more work in the dialplan, but for our needs it is well worth 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
[asterisk-users] SIP interface status and calllimit
There is an issue when using call-limit for a SIP interface in sip.conf. The call count does not properly reset when some calls end. The problem happens regardless of which side of the connection ends the call. It happens on all calls including calls from SIP interface to SIP interface (with no reinvite) within the same Asterisk server. I have not been able to determine a definite pattern. I can call from one interface to another 50 times before it happens and sometimes it happens after only 2 calls. We have to enable call-limit for our customer service queue agents so that the ringinuse option in queues.conf will work properly. Has anyone else seen this issue? Any ideas? Thanks, James Fromm ___ --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] End Wrap-up Time?
I patched 1.4.0 to add a command to the manager api in the queue application to implement the end wrap-up time I was asking about. All the command does is modify the 'lastcall' timestamp for the queue member by subtracting the value of the queue's defined wrapup time. Andrew Kohlsmith wrote: On 2/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: This is coming right out of left field, as I've never set up an Asterisk queue or agent system, but is it possible to pause and unpause while in the wrap-up time? What happens? Does the wrapup time go away then? Might be a counter-intuitive way around it if so... On Thursday 15 February 2007 4:34 pm, Matt wrote: I tried that. It didn't work :( What if we patched Asterisk to do just that? What could the repercussions be? They're already pausing/unpausing, so having the wrapup time auto-zero on unpause seems a non-issue... -A. ___ --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] End Wrap-up Time?
I did it really only for our use. Because we manage our queue members solely through the manager interface, the implementation only works by issuing a command while connected to the manager port. The patch also adds 'Wrapuptime' as a return value to a queuestatus on the management port and changes the manager interface to not log every command received to the debug log unless the debug option is set. The diff can be found at http://www.omnis.com/queueendwait.diff. Matt wrote: Where is this patch? On 2/16/07, *James Fromm* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I patched 1.4.0 to add a command to the manager api in the queue application to implement the end wrap-up time I was asking about. All the command does is modify the 'lastcall' timestamp for the queue member by subtracting the value of the queue's defined wrapup time. Andrew Kohlsmith wrote: On 2/15/07, Andrew Kohlsmith [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: This is coming right out of left field, as I've never set up an Asterisk queue or agent system, but is it possible to pause and unpause while in the wrap-up time? What happens? Does the wrapup time go away then? Might be a counter-intuitive way around it if so... On Thursday 15 February 2007 4:34 pm, Matt wrote: I tried that. It didn't work :( What if we patched Asterisk to do just that? What could the repercussions be? They're already pausing/unpausing, so having the wrapup time auto-zero on unpause seems a non-issue... -A. ___ --Bandwidth and Colocation provided by Easynews.com http://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 http://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] End Wrap-up Time?
Does anyone have a solution to allow an agent to selectively end his wrap-up time? We define a wrap-up time of 60 seconds to allow our agents to finish their notes from a call. In some cases, the full 60 seconds is not needed and our agents would like to end their wrap-up time. Thanks, Jay ___ --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] End Wrap-up Time?
Does anyone have a solution to allow an agent to selectively end his wrap-up time? We define a wrap-up time of 60 seconds to allow our agents to finish their notes from a call. In some cases, the full 60 seconds is not needed and our agents would like to end their wrap-up time. Thanks, Jay ___ --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] How to access environment variable?
'export MYIP' in the startup script for Asterisk.
Larry Alkoff wrote:
I was only trying to demonstrate that my special variable MYIP was
indeed in the environment of the shell. I suspect it's not in the
Asterisk process environment - why I dunno.
I'll look at that tomorrow but suspect I'll never be able to read the
MYIP variable from Asterisk.
Larry
Tzafrir Cohen wrote:
On Tue, Feb 06, 2007 at 08:04:23AM -0600, Larry Alkoff wrote:
Thanks for your reply Ioan.
Very interesting. ${ENV(PATH)} works to display the path
but ${ENV(MYIP)} does not!
There must be a list in Asterisk that only allows cerain
environmental variables to be shown. A very unnecessary bummer.
Right.
However, at the CLI prompt:
! echo $PATH and ! echo $MYIP
both work fine.
However This is incorrect: '!' only works in a remote asterisk
terminal: a connection from a different process (on the same system)
to the running
Asterisk process.
It will run a subshell of thatremote process. So it is not necessarily
related to the environment of the Asterisk process.
Also: when running something in System(), note that you run a
subprocess, and that this subprocess may have its own separate
environment.
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Re: [asterisk-users] Re: Help - Poor Voice Quality
Jim, I too am a Teliax user. Talk to their technical support. IAX2 is NOT preferred. They'll tell you to use SIP. Jim Duda wrote: Thanks for the reply Lacy. Yes, I know that I am using IAX2 and not SIP for my connection to teliax. IAX2 is the preferred protocol for connection to teliax. I have the firewall configured to prioritorize port 4569 for IAX2. I have the shorewall tcdevices file setup with 3 mbit download and 500 Kbit upload speeds. Jim Lacy Moore wrote: Jim Duda wrote: I've been on the shorewall firewall and confirmed that I have the firewall configured properly for VOIP QOS. What exactly have you done here? You do know that you are apparently using IAX2 and not SIP. Those are not the same protocols. In fact, if you configured the shorewall system for standard VoIP, that's your problem. IAX2 operates on different ports that SIP. Whereas SIP operates on a control port and then create media ports, IAX2 only uses one. As far as download speed, what have you told shorewall your download speed is? I'm not familiar with it, but just guessing that it's probably like most others. If this is the case, somewhere there is a setting to tell it what your download and upload speed is. 500kpbs up doesn't seem like enough bandwidth to support 10Mpbs down, either. ___ --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] How to access environment variable?
How do you start Asterisk? You need to make sure the environment
variable you want inside Asterisk is being exported. I use 'export
HOSTNAME' in my asterisk init script and it works like a charm.
Larry Alkoff wrote:
Thanks for your reply Ioan.
Very interesting. ${ENV(PATH)} works to display the path
but ${ENV(MYIP)} does not!
There must be a list in Asterisk that only allows cerain environmental
variables to be shown. A very unnecessary bummer.
However, at the CLI prompt:
! echo $PATH and ! echo $MYIP
both work fine.
Larry
Ioan Indreias wrote:
Hello Larry,
Probably your variable (MYIP) is not accessible to asterisk process
environment.
Test it with ${ENV(PATH)} and you will have a result there
exten = s,n,Set(test=${ENV(PATH)})
-- Executing Set(IAX2/test_iax,
test=/sbin:/usr/sbin:/bin:/usr/bin:/usr/X11R6/bin) in new stack
Larry Alkoff wrote:
How can I access an environmental variable in Asterisk 1.2.5?
It should be possible according to:
http://www.voip-info.org/wiki/view/Asterisk+variables
which says:
Environment Variables
You may access unix environment variables using the syntax:
${ENV(foo)}
${ENV(ASTERISK_PROMPT)}: the current Asterisk CLI prompt.
${ENV(RECORDED_FILE)}: the filename of the last file saved by the
Record command
I have an environmental variable MYIP which contains my current IP
address but when I execute exten _4XX the following line only says
'myip is ' and the rest is blank instead of showing
'myip is www.xxx.yyy.zzz'
exten = _4XX,n,VERBOSE(myip is ${ENV(MYIP)})
Why doesn't it work?
Larry
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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
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
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
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
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
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
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
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
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
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
[asterisk-users] Queue cmd option 'i'
Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should ignore call forward requests from queue members and do nothing when they are requested. Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding should be ignored. Using Queue(customerservice|i) causes Asterisk to crash when sending the call to the member with forwarding enabled on their SIP device. Am I misinterpreting what this option does? Thanks, 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 Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
I spent hours debugging this a few weeks ago.
The ${UNIQUEID} contains a period (.). Mine are something like
.xx. When soxmix is executed to mix the in and out files, the
file types are not specified. This causes soxmix to attempt to
determine the file type by the filename's extension. The routine in sox
that looks for the filename's extension doesn't expect multiple periods
in the filename. So it finds the file type to be xx.wav (or xx.gsm) and
that's not a format sox can handle.
You can add an AGI call to your dialplan immediately after the Queue
application to join the files.
Ex Vitorino wrote:
(1st attempt was rejected by postfix @lists.digium.com, here goes the 2nd)
-- Forwarded message --
From: Ex Vitorino [EMAIL PROTECTED]
Date: Dec 18, 2006 11:41 PM
Subject: Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Hello Asterisk Users,
I guess the subject says the most of it; here goes some more
detail:
- Running Asterisk 1.2.14
- Objective: record all calls managed by a specific queue
- Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID}
Facts:
- If the UNIQUEID chan var is used in the MONITOR_FILENAME,
before calling the Queue() application, the two legs of the call are
not mixed and I end up with the two separate -in / -out files
- If it isn't used (for example: using only ${TIMESTAMP]-${CALLERIDNUM})
then, the legs are mixed together...
Note:
- In my first attempt I never managed to get the legs mixed... Only
after some experiment, I understood (well, not 100% clear why!)
that I had to also to add to include recordagentcalls=yes and
monitor-join=yes in agents.conf !
Can anyone provide some insight into this ? Thanks in advance!
(see below for config)
--
Ex Vito
queues.conf:
[general]
persistentmembers = yes
[the_queue]
musiconhold = default
announce = the_announcement
strategy = ringall
servicelevel = 20
context = the_context
wrapuptime = 10
announce-frequency = 30
announce-holdtime = once
monitor-format = wav
monitor-join = yes
eventwhencalled = yes
eventmemberstatus = no
reportholdtime = no
member = SIP/sip0001
agents.conf:
[general]
persistentagents=yes
recordagencalls=yes
monitor-join = yes
[agents]
(no agents declared, as they are directly configured in the
queues.conf file)
extensions.conf:
...
[globals]
SUPPORT_MONITOR_PATH=/var/spool/asterisk/monitor/support
[the_context]
exten = 305,1,Answer()
exten =
305,n,Set(MONITOR_FILENAME=${SUPPORT_MONITOR_PATH}/${TIMESTAMP}-${CALLERIDNUM})
exten = 305,n,Queue(the_queue,t)
exten = 305,n,Hangup()
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Re: [asterisk-users] Polycom_acd_functions SIP trouble
Exactly. If I uncomment the secret, no SIP device or softphone will be able to register. I commented the secret so I could continue to configure using this revision of the branch. No SIP device or softphone can register as long as a secret is required. Dovid Bender wrote: I am sure you prob. know this but in your configs it shows secret commented out. Also it with a softphone if it dosent work then, then its your configs. Also did you remember to reload asterisk ? - Original Message - From: James Fromm [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, July 24, 2006 2:24 PM Subject: [asterisk-users] Polycom_acd_functions SIP trouble I'm trying to use the latest revision of Bweschke's branch from SVN for polycom_acd_functions. Asterisk builds and runs without error but all SIP devices can't register when specifying a secret in sip.conf. The Polycom 601 I'm testing with and a copy of SJphone will not register. IAX from Idefisk works without error. The error all SIP devices get is: Jul 24 10:26:48 NOTICE[31524]: chan_sip.c:14203 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.95' - Username/auth name mismatch Commenting the definition of a secret in sip.conf for the device solves this. Here's the config for one of the devices. [1003] type=friend canreinvite=no host=dynamic username=1003 ; secret=stuff context=outbound callerid=Jimmy 1003 [EMAIL PROTECTED] nat=no Why won't this revision accept the definition of a secret? Am I missing something simple (stupid)? Thanks, Jay ___ --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] Polycom_acd_functions SIP trouble
Yeah, we tried that. Tried every combination of variables in sip.conf. Only solution that works is removing the requirement for a secret. Faris Raouf wrote: Dovid Bender wrote: I am sure you prob. know this but in your configs it shows secret commented out. Also it with a softphone if it dosent work then, then its your configs. Also did you remember to reload asterisk ? - Original Message - From: James Fromm [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, July 24, 2006 2:24 PM Subject: [asterisk-users] Polycom_acd_functions SIP trouble I'm trying to use the latest revision of Bweschke's branch from SVN for polycom_acd_functions. Asterisk builds and runs without error but all SIP devices can't register when specifying a secret in sip.conf. The Polycom 601 I'm testing with and a copy of SJphone will not register. IAX from Idefisk works without error. One thing to try is setting type=peer instead of type=friend. I'm a bit dazed and confused at the moment, but if I remember correctly Polycom phones just don't work with type=friend. Of course this doesn't explain why SJPhone won't work either so maybe I'm totally off-track, but it might be worth giving it a try just the same. Faris. ___ --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] Polycom_acd_functions SIP trouble
I'm trying to use the latest revision of Bweschke's branch from SVN for polycom_acd_functions. Asterisk builds and runs without error but all SIP devices can't register when specifying a secret in sip.conf. The Polycom 601 I'm testing with and a copy of SJphone will not register. IAX from Idefisk works without error. The error all SIP devices get is: Jul 24 10:26:48 NOTICE[31524]: chan_sip.c:14203 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.95' - Username/auth name mismatch Commenting the definition of a secret in sip.conf for the device solves this. Here's the config for one of the devices. [1003] type=friend canreinvite=no host=dynamic username=1003 ; secret=stuff context=outbound callerid=Jimmy 1003 [EMAIL PROTECTED] nat=no Why won't this revision accept the definition of a secret? Am I missing something simple (stupid)? Thanks, Jay ___ --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
