Re: [asterisk-users] Queues and penalties
I'm fairly sure the patch to App Queue that was added to Asterisk 13+ should do the job... It causes agent priorities to "float up" over time so that new agents are included without excluding old agents. I can't find it right now but there can't be that many app_queue patches to ast 13 in the last 18 months Steve On Fri, 30 Nov 2018 at 09:18, Paddy Grice wrote: > Thanks Leon > > I will implement and test but I knew there would be a fix for what I > believe is a short coming in app_queue. How do I suggest this as a option > to the base code? > > Paddy > > -- > *From:* Leon Wright [mailto:lwri...@corpcloud.com.au] > *Sent:* 30 November 2018 02:17 > *To:* pa...@wizaner.com; asterisk-users@lists.digium.com > *Cc:* johnkinis...@gmail.com > *Subject:* Re: [asterisk-users] Queues and penalties > > Paddy, > > This appears to be how the queue app works. I ended up patching the queue > app: > > diff --git a/apps/app_queue.c b/apps/app_queue.c > index e3a4e22..72072d0 100644 > --- a/apps/app_queue.c > +++ b/apps/app_queue.c > @@ -4571,7 +4571,7 @@ static int ring_one(struct queue_ent *qe, struct > callattempt *outgoing, int *bus > struct callattempt *cur; > /* Ring everyone who shares this best metric (for > ringall) */ > for (cur = outgoing; cur; cur = cur->q_next) { > - if (cur->stillgoing && !cur->chan && > cur->metric <= best->metric) { > + if (cur->stillgoing && !cur->chan && > cur->metric >= qe->min_penalty * 100 && cur->metric <= qe->max_penalty > * 100) { > ast_debug(1, "(Parallel) Trying > '%s' with metric %d\n", cur->interface, cur->metric); > ret |= ring_entry(qe, cur, busies); > } > > So the penalties get calculated during the 'ringall' strategy and allowing > the queue app to exit, looping and raising the max penalty and calling the > queue app again. > > Leon > > On Thu, 29 Nov 2018 at 18:24, Paddy Grice wrote: > >> Hi John >> >> This works fine providing extensions 1001,1002 and 1003 are "Incall" or >> "Paused" - the problem appears to be that is a handset say 1002 is >> "ringing" then the 2xxx then the penalty is not honoured. >> >> This is well described in the History section of the following link >> https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue >> >> As I say this seems to be a real shortcoming in app_queue. >> >> Any ideas, suggestions, anyone want to work with me to sort this ? >> >> Paddy >> >> >> -- >> *From:* John Kiniston [mailto:johnkinis...@gmail.com] >> *Sent:* 28 November 2018 21:17 >> *To:* pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial >> Discussion >> *Subject:* Re: [asterisk-users] Queues and penalties >> >> This should work, How are you defining your timeouts in the queues.conf ? >> >> And to verify, in your extensions.conf you are calling Queue with the >> queue name and the ruleset to apply from queuerules.conf? >> >> On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote: >> >>> Hi All >>> >>> I have been looking at this problem for a few days/weeks now and after >>> some advice please. >>> >>> I currently have a customer on 11.25.3 and I am in the process of >>> upgrading versions and OS (Debian) and all things that involves mysql -> >>> PDO etc >>> >>> The problem I have is the customer want a simple call distribution like >>> this >>> >>> Extn 1001, 1002, 1003 to be called on an incoming call - if they don't >>> answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing >>> extensions and if no one answers after another 20 seconds the add in 3001, >>> 3002, 3003. >>> >>> Seems a simple queue application to me >>> >>> 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall >>> 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall >>> 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall >>> >>> and rules >>> >>> increasing the maxpenalty 1->2 after 20 seconds >>> and increasing maxpenalty 2->3 after another 20 seconds. >>> >>>
Re: [asterisk-users] Queues and penalties
Thanks Leon I will implement and test but I knew there would be a fix for what I believe is a short coming in app_queue. How do I suggest this as a option to the base code? Paddy _ From: Leon Wright [mailto:lwri...@corpcloud.com.au] Sent: 30 November 2018 02:17 To: pa...@wizaner.com; asterisk-users@lists.digium.com Cc: johnkinis...@gmail.com Subject: Re: [asterisk-users] Queues and penalties Paddy, This appears to be how the queue app works. I ended up patching the queue app: diff --git a/apps/app_queue.c b/apps/app_queue.c index e3a4e22..72072d0 100644 --- a/apps/app_queue.c +++ b/apps/app_queue.c @@ -4571,7 +4571,7 @@ static int ring_one(struct queue_ent *qe, struct callattempt *outgoing, int *bus struct callattempt *cur; /* Ring everyone who shares this best metric (for ringall) */ for (cur = outgoing; cur; cur = cur->q_next) { - if (cur->stillgoing && !cur->chan && cur->metric <= best->metric) { + if (cur->stillgoing && !cur->chan && cur->metric >= qe->min_penalty * 100 && cur->metric <= qe->max_penalty * 100) { ast_debug(1, "(Parallel) Trying '%s' with metric %d\n", cur->interface, cur->metric); ret |= ring_entry(qe, cur, busies); } So the penalties get calculated during the 'ringall' strategy and allowing the queue app to exit, looping and raising the max penalty and calling the queue app again. Leon On Thu, 29 Nov 2018 at 18:24, Paddy Grice wrote: Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to be a real shortcoming in app_queue. Any ideas, suggestions, anyone want to work with me to sort this ? Paddy _ From: John Kiniston [mailto:johnkinis...@gmail.com] Sent: 28 November 2018 21:17 To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues and penalties This should work, How are you defining your timeouts in the queues.conf ? And to verify, in your extensions.conf you are calling Queue with the queue name and the ruleset to apply from queuerules.conf? On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote: Hi All I have been looking at this problem for a few days/weeks now and after some advice please. I currently have a customer on 11.25.3 and I am in the process of upgrading versions and OS (Debian) and all things that involves mysql -> PDO etc The problem I have is the customer want a simple call distribution like this Extn 1001, 1002, 1003 to be called on an incoming call - if they don't answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing extensions and if no one answers after another 20 seconds the add in 3001, 3002, 3003. Seems a simple queue application to me 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall and rules increasing the maxpenalty 1->2 after 20 seconds and increasing maxpenalty 2->3 after another 20 seconds. But this doesn't work if users don't answer!! if user 1002 or (2001 etc) just lets his phone ring - he forgot to logoff or DND then the penalty is ignored. There seems to have been a patch for FreePBX on V13 - LazyMembers - but that is all I can find and later versions have no mention of this I guess I can use autopause and some AMI / Script but this stops phones ringing because of the timeout so the user has a ringing phone and then it stops and then it starts again whereas the penalty just adds handsets into the ringing group. This seems to be a real shortcoming in app_queue. Any ideas, suggestions, anyone want to work with me to sort this ? Paddy Grice -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being sh
Re: [asterisk-users] Queues and penalties
Paddy, This appears to be how the queue app works. I ended up patching the queue app: diff --git a/apps/app_queue.c b/apps/app_queue.c index e3a4e22..72072d0 100644 --- a/apps/app_queue.c +++ b/apps/app_queue.c @@ -4571,7 +4571,7 @@ static int ring_one(struct queue_ent *qe, struct callattempt *outgoing, int *bus struct callattempt *cur; /* Ring everyone who shares this best metric (for ringall) */ for (cur = outgoing; cur; cur = cur->q_next) { - if (cur->stillgoing && !cur->chan && cur->metric <= best->metric) { + if (cur->stillgoing && !cur->chan && cur->metric >= qe->min_penalty * 100 && cur->metric <= qe->max_penalty * 100) { ast_debug(1, "(Parallel) Trying '%s' with metric %d\n", cur->interface, cur->metric); ret |= ring_entry(qe, cur, busies); } So the penalties get calculated during the 'ringall' strategy and allowing the queue app to exit, looping and raising the max penalty and calling the queue app again. Leon On Thu, 29 Nov 2018 at 18:24, Paddy Grice wrote: > Hi John > > This works fine providing extensions 1001,1002 and 1003 are "Incall" or > "Paused" - the problem appears to be that is a handset say 1002 is > "ringing" then the 2xxx then the penalty is not honoured. > > This is well described in the History section of the following link > https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue > > As I say this seems to be a real shortcoming in app_queue. > > Any ideas, suggestions, anyone want to work with me to sort this ? > > Paddy > > > -- > *From:* John Kiniston [mailto:johnkinis...@gmail.com] > *Sent:* 28 November 2018 21:17 > *To:* pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial > Discussion > *Subject:* Re: [asterisk-users] Queues and penalties > > This should work, How are you defining your timeouts in the queues.conf ? > > And to verify, in your extensions.conf you are calling Queue with the > queue name and the ruleset to apply from queuerules.conf? > > On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote: > >> Hi All >> >> I have been looking at this problem for a few days/weeks now and after >> some advice please. >> >> I currently have a customer on 11.25.3 and I am in the process of >> upgrading versions and OS (Debian) and all things that involves mysql -> >> PDO etc >> >> The problem I have is the customer want a simple call distribution like >> this >> >> Extn 1001, 1002, 1003 to be called on an incoming call - if they don't >> answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing >> extensions and if no one answers after another 20 seconds the add in 3001, >> 3002, 3003. >> >> Seems a simple queue application to me >> >> 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall >> 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall >> 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall >> >> and rules >> >> increasing the maxpenalty 1->2 after 20 seconds >> and increasing maxpenalty 2->3 after another 20 seconds. >> >> But this doesn't work if users don't answer!! >> >> if user 1002 or (2001 etc) just lets his phone ring - he forgot to >> logoff or DND then the penalty is ignored. >> >> There seems to have been a patch for FreePBX on V13 - LazyMembers - but >> that is all I can find and later versions have no mention of this >> >> I guess I can use autopause and some AMI / Script but this stops phones >> ringing because of the timeout so the user has a ringing phone and then it >> stops and then it starts again whereas the penalty just adds handsets into >> the ringing group. >> >> This seems to be a real shortcoming in app_queue. >> >> Any ideas, suggestions, anyone want to work with me to sort this ? >> >> Paddy Grice >> >> >> >> >> >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Astricon is coming up October 9-11! Signup is available at: >> https://www.asterisk.org/community/astricon-user-conference >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.or
Re: [asterisk-users] Queues and penalties
Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to be a real shortcoming in app_queue. Any ideas, suggestions, anyone want to work with me to sort this ? Paddy _ From: John Kiniston [mailto:johnkinis...@gmail.com] Sent: 28 November 2018 21:17 To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues and penalties This should work, How are you defining your timeouts in the queues.conf ? And to verify, in your extensions.conf you are calling Queue with the queue name and the ruleset to apply from queuerules.conf? On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote: Hi All I have been looking at this problem for a few days/weeks now and after some advice please. I currently have a customer on 11.25.3 and I am in the process of upgrading versions and OS (Debian) and all things that involves mysql -> PDO etc The problem I have is the customer want a simple call distribution like this Extn 1001, 1002, 1003 to be called on an incoming call - if they don't answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing extensions and if no one answers after another 20 seconds the add in 3001, 3002, 3003. Seems a simple queue application to me 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall and rules increasing the maxpenalty 1->2 after 20 seconds and increasing maxpenalty 2->3 after another 20 seconds. But this doesn't work if users don't answer!! if user 1002 or (2001 etc) just lets his phone ring - he forgot to logoff or DND then the penalty is ignored. There seems to have been a patch for FreePBX on V13 - LazyMembers - but that is all I can find and later versions have no mention of this I guess I can use autopause and some AMI / Script but this stops phones ringing because of the timeout so the user has a ringing phone and then it stops and then it starts again whereas the penalty just adds handsets into the ringing group. This seems to be a real shortcoming in app_queue. Any ideas, suggestions, anyone want to work with me to sort this ? Paddy Grice -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and penalties
This should work, How are you defining your timeouts in the queues.conf ? And to verify, in your extensions.conf you are calling Queue with the queue name and the ruleset to apply from queuerules.conf? On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote: > Hi All > > I have been looking at this problem for a few days/weeks now and after > some advice please. > > I currently have a customer on 11.25.3 and I am in the process of > upgrading versions and OS (Debian) and all things that involves mysql -> > PDO etc > > The problem I have is the customer want a simple call distribution like > this > > Extn 1001, 1002, 1003 to be called on an incoming call - if they don't > answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing > extensions and if no one answers after another 20 seconds the add in 3001, > 3002, 3003. > > Seems a simple queue application to me > > 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall > 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall > 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall > > and rules > > increasing the maxpenalty 1->2 after 20 seconds > and increasing maxpenalty 2->3 after another 20 seconds. > > But this doesn't work if users don't answer!! > > if user 1002 or (2001 etc) just lets his phone ring - he forgot to logoff > or DND then the penalty is ignored. > > There seems to have been a patch for FreePBX on V13 - LazyMembers - but > that is all I can find and later versions have no mention of this > > I guess I can use autopause and some AMI / Script but this stops phones > ringing because of the timeout so the user has a ringing phone and then it > stops and then it starts again whereas the penalty just adds handsets into > the ringing group. > > This seems to be a real shortcoming in app_queue. > > Any ideas, suggestions, anyone want to work with me to sort this ? > > Paddy Grice > > > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues - periodic announce while ringing members
Hi, what I did, I mixed the music on hold to have the announce in at a specific time without leaving queue On 25 February 2016 at 16:53, Daniel Chavezwrote: > Ish, > I use the same version of Asterisk on CentOS 6.7. I wonder the same thing. > Hopefully we will find this out. > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues - periodic announce while ringing members
Ish, I use the same version of Asterisk on CentOS 6.7. I wonder the same thing. Hopefully we will find this out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues don't follow dialplan if no members are registered
Wow, Looks like they have really increased the options since I last looked. I just pulled down the Asterisk 13 queues.conf.sample and it's got this in it: ; paused: a member is not considered available if he is paused ; penalty: a member is not considered available if his penalty is less than QUEUE_MAX_PENALTY ; inuse: a member is not considered available if he is currently on a call ; ringing: a member is not considered available if his phone is currently ringing ; unavailable: This applies mainly to Agent channels. If the agent is a member of the queue ; but has not logged in, then do not consider the member to be available ; invalid: Do not consider a member to be available if he has an invalid device state. ; This generally is caused by an error condition in the member's channel driver. ; unknown: Do not consider a member to be available if we are unable to determine the member's ; current device state. ; wrapup: A member is not considered available if he is currently in his wrapuptime after ; taking a call. An unknown state would be a device that has a valid configuration but isn't registered. On Tue, Jul 28, 2015 at 8:51 PM, Andrew Martin amar...@xes-inc.com wrote: - Original Message - From: John Kiniston johnkinis...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 28, 2015 12:12:05 PM Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are registered In your queues.conf do you have a leavewhenempty and joinempty set? in queues.conf [myqueue] leavewhenempty = strict joinempty = strict strategy = ringall ringinuse = no John, Thanks for the fast reply! I had joinempty=yes in queues.conf, which explains why I was seeing this behavior. It looks like the strict setting is partially-deprecated, so instead I'm using the following combination: [myqueue] musiconhold=default music=default strategy=ringall joinempty=unavailable,invalid,unknown leavewhenempty=unavailable,invalid,unknown timeout=18 member = SIP/100 member = SIP/101 Is there any reason that using any of these options would be a problem, in particular unknown? It is not very well defined what an unknown state is exactly. Thanks, Andrew -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues don't follow dialplan if no members are registered
- Original Message - From: John Kiniston johnkinis...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 29, 2015 11:53:13 AM Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are registered Wow, Looks like they have really increased the options since I last looked. I just pulled down the Asterisk 13 queues.conf.sample and it's got this in it: ; paused: a member is not considered available if he is paused ; penalty: a member is not considered available if his penalty is less than QUEUE_MAX_PENALTY ; inuse: a member is not considered available if he is currently on a call ; ringing: a member is not considered available if his phone is currently ringing ; unavailable: This applies mainly to Agent channels. If the agent is a member of the queue ; but has not logged in, then do not consider the member to be available ; invalid: Do not consider a member to be available if he has an invalid device state. ; This generally is caused by an error condition in the member's channel driver. ; unknown: Do not consider a member to be available if we are unable to determine the member's ; current device state. ; wrapup: A member is not considered available if he is currently in his wrapuptime after ; taking a call. An unknown state would be a device that has a valid configuration but isn't registered. John, Thanks for the clarification and your help resolving this issue! Andrew -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues don't follow dialplan if no members are registered
- Original Message - From: John Kiniston johnkinis...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 28, 2015 12:12:05 PM Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are registered In your queues.conf do you have a leavewhenempty and joinempty set? in queues.conf [myqueue] leavewhenempty = strict joinempty = strict strategy = ringall ringinuse = no John, Thanks for the fast reply! I had joinempty=yes in queues.conf, which explains why I was seeing this behavior. It looks like the strict setting is partially-deprecated, so instead I'm using the following combination: [myqueue] musiconhold=default music=default strategy=ringall joinempty=unavailable,invalid,unknown leavewhenempty=unavailable,invalid,unknown timeout=18 member = SIP/100 member = SIP/101 Is there any reason that using any of these options would be a problem, in particular unknown? It is not very well defined what an unknown state is exactly. Thanks, Andrew -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues don't follow dialplan if no members are registered
In your queues.conf do you have a leavewhenempty and joinempty set? in queues.conf [myqueue] leavewhenempty = strict joinempty = strict strategy = ringall ringinuse = no On Tue, Jul 28, 2015 at 9:58 AM, Andrew Martin amar...@xes-inc.com wrote: Hello, I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf: exten = s,1,Queue(myqueue,rtnC,18) same = n,Background(user_unavail) same = n,WaitExten(10) exten = 1,1,Voicemail(@my-vm,s) This rings the phones in the queue for 18 seconds. If no queue members answer, the caller is then prompted to press 1 and leave a voicemail. This works well when at least 1 member is registered in the queue, however if no members are registered in the queue, the Queue() call never seems to return, and thus the remaining steps in the dialplan never execute. How can I correct this behavior so that even if the queue has no registered members, the dialplan is still followed? Thanks, Andrew -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
You need to do this when the call connects. If you can do this within a couple of seconds, this is usually good enough to be usable (that's what we do on the QueueMetrics agents pages). Thanks l. 2013/8/3 Timothy Smith timotsm...@gmail.com Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
Hi, Our queue members are Local channels, thus when dialing the agent, the dialplan will do several stuff including: Set(CALLERID(name)=${CALLERID(name)}:Sales) UserEvent(something,data: ${bunch-of-data-in-some-format}) Dial(SIP/final-agent-phone,timeout,A(Sales)) The UserEvent will be picked up by our client-register-ticket-stuff software The announcement A() will be heard by the agent upon answering the call like sales call On 4 August 2013 02:59, Mitch Claborn mitch...@claborn.net wrote: We do something very similar. Use the gosub parameter of the Queue application to call a subroutine in the dial plan when the agent answers the call. same =n,Queue(sales,tc,,sub-**QueueConnected) [sub-QueueConnected] ; this runs on the agent/member's channel exten =s,1,NoOp() ; whatever you need to do here same =n,Return() See https://wiki.asterisk.org/**wiki/display/AST/Asterisk+11+** Application_Queuehttps://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue Mitch On 08/03/2013 12:45 PM, Timothy Smith wrote: Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
Dear Mitch, Thank you so much. This partly solves my problem by a great deal, as we'll send a message to the agent immediately on picking the call. As the agents are local SIP channels, I will attempt looking up the caller's name (if it exists in our database) and set it prior to entering the queue. Is there any way of informing the agent (just) before they pick up? e.g when their phone starts ringing, so that they prepare accordingly? Regards, Wilson On Sun, Aug 4, 2013 at 4:59 AM, Mitch Claborn mitch...@claborn.net wrote: We do something very similar. Use the gosub parameter of the Queue application to call a subroutine in the dial plan when the agent answers the call. same =n,Queue(sales,tc,,sub-QueueConnected) [sub-QueueConnected] ; this runs on the agent/member's channel exten =s,1,NoOp() ; whatever you need to do here same =n,Return() See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue Mitch On 08/03/2013 12:45 PM, Timothy Smith wrote: Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
Dear Tiago, Thanks for your answer, but I have a few questions. Do you use queues? We are operating a call centre with several queues, so I don't see how we would use the Dial command. When a call comes in, we enter the caller (depending on what options he has selected) into a queue. Do you have any alternative method, which would involve dialling the agent directly as you described below? regards, T On Sun, Aug 4, 2013 at 3:47 PM, Tiago Geada tiago.ge...@gmail.com wrote: Hi, Our queue members are Local channels, thus when dialing the agent, the dialplan will do several stuff including: Set(CALLERID(name)=${CALLERID(name)}:Sales) UserEvent(something,data: ${bunch-of-data-in-some-format}) Dial(SIP/final-agent-phone,timeout,A(Sales)) The UserEvent will be picked up by our client-register-ticket-stuff software The announcement A() will be heard by the agent upon answering the call like sales call On 4 August 2013 02:59, Mitch Claborn mitch...@claborn.net wrote: We do something very similar. Use the gosub parameter of the Queue application to call a subroutine in the dial plan when the agent answers the call. same =n,Queue(sales,tc,,sub-QueueConnected) [sub-QueueConnected] ; this runs on the agent/member's channel exten =s,1,NoOp() ; whatever you need to do here same =n,Return() See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue Mitch On 08/03/2013 12:45 PM, Timothy Smith wrote: Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
Hi, You just said you use Local channels. Local channel is a dialplan that has a Dial() to a sip device? We use queues, and have a queue-macro that sends the UserEvent upon bridging the call... On 4 August 2013 16:41, Timothy Smith timotsm...@gmail.com wrote: Dear Tiago, Thanks for your answer, but I have a few questions. Do you use queues? We are operating a call centre with several queues, so I don't see how we would use the Dial command. When a call comes in, we enter the caller (depending on what options he has selected) into a queue. Do you have any alternative method, which would involve dialling the agent directly as you described below? regards, T On Sun, Aug 4, 2013 at 3:47 PM, Tiago Geada tiago.ge...@gmail.com wrote: Hi, Our queue members are Local channels, thus when dialing the agent, the dialplan will do several stuff including: Set(CALLERID(name)=${CALLERID(name)}:Sales) UserEvent(something,data: ${bunch-of-data-in-some-format}) Dial(SIP/final-agent-phone,timeout,A(Sales)) The UserEvent will be picked up by our client-register-ticket-stuff software The announcement A() will be heard by the agent upon answering the call like sales call On 4 August 2013 02:59, Mitch Claborn mitch...@claborn.net wrote: We do something very similar. Use the gosub parameter of the Queue application to call a subroutine in the dial plan when the agent answers the call. same =n,Queue(sales,tc,,sub-QueueConnected) [sub-QueueConnected] ; this runs on the agent/member's channel exten =s,1,NoOp() ; whatever you need to do here same =n,Return() See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue Mitch On 08/03/2013 12:45 PM, Timothy Smith wrote: Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
We do something very similar. Use the gosub parameter of the Queue application to call a subroutine in the dial plan when the agent answers the call. same =n,Queue(sales,tc,,sub-QueueConnected) [sub-QueueConnected] ; this runs on the agent/member's channel exten =s,1,NoOp() ; whatever you need to do here same =n,Return() See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue Mitch On 08/03/2013 12:45 PM, Timothy Smith wrote: Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and distributed device state over WAN
Oliver wrote: snip Before diving into this, I've got the following question : - let say we have two Asterisk servers A and B, - both are interconnected through PSTN (no SIP trunk) - agent Alice's phone is connected (ie registered) to server A - Alice's phone can be reached from server A phones dialing Local/6789 - Alice's phone can also be reached from server B phones dialing Local/00123456789 1. How do you configure both servers so that Alice's phone becomes a Queue Member from a server B given queue ? Simply calling AddQueueMember on server B, passing Local/00123456789 as interface value (ie AddQueueMember second argument) ? 2. Then, how should server A publish Alice's phone status ? How should server B consume Alice's phone status and associate it with the Queue member activity ? Using AddQueueMember stateinterface argument ? Before trying to get distributed device state going: Device names across all Asterisk proxy's participating in 'distributed device state' need to be unique. IE. You can't have 'SIP/cisco1' exist on server A for ALICE, and SIP/cisco1 on server B for BOB. You need to get XMPP distributed device state working. I followed https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMP P+PubSub You need a reasonbly reliable WAN links to the jabber server support the XMPP updates between servers. Asterisk segfaults if it can't contact the jabber server!!! See https://issues.asterisk.org/jira/browse/ASTERISK-18078 Then: With Alice reqistered on Server A as SIP/cisco1 With Server B hosting the queue named 'queue1'. ;(on Server B) queues.conf: [queue1] ;what makes this work with distributed states is the 'State Interface' parameter ... member = Local/00123456789,0,ALICE,SIP/cisco1 Alice will need a number to ring to login/logout of queue1 hosted on Server B; Dialplan Example: on server B: ... exten = s,n,Set(queuename=queue1) exten = s,n,Set(interface=Local/00123456789) exten = s,n,Set(penalty=0) exten = s,n,Set(stateinterface=SIP/cisco1) exten = s,n(queue-add),AddQueueMember(${queuename},${interface},${penalty},options,, ${stateinterface}) And to remove the member; ... exten = s,n(queue-remove),RemoveQueueMember(${queuename},${interface}) Alec Davis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and distributed device state over WAN
2013/1/25 Alec Davis siva...@paradise.net.nz Oliver wrote: snip Before diving into this, I've got the following question : - let say we have two Asterisk servers A and B, - both are interconnected through PSTN (no SIP trunk) - agent Alice's phone is connected (ie registered) to server A - Alice's phone can be reached from server A phones dialing Local/6789 - Alice's phone can also be reached from server B phones dialing Local/00123456789 1. How do you configure both servers so that Alice's phone becomes a Queue Member from a server B given queue ? Simply calling AddQueueMember on server B, passing Local/00123456789 as interface value (ie AddQueueMember second argument) ? 2. Then, how should server A publish Alice's phone status ? How should server B consume Alice's phone status and associate it with the Queue member activity ? Using AddQueueMember stateinterface argument ? Before trying to get distributed device state going: Device names across all Asterisk proxy's participating in 'distributed device state' need to be unique. IE. You can't have 'SIP/cisco1' exist on server A for ALICE, and SIP/cisco1 on server B for BOB. You need to get XMPP distributed device state working. I followed https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMP P+PubSub You need a reasonbly reliable WAN links to the jabber server support the XMPP updates between servers. Asterisk segfaults if it can't contact the jabber server!!! See https://issues.asterisk.org/jira/browse/ASTERISK-18078 Then: With Alice reqistered on Server A as SIP/cisco1 With Server B hosting the queue named 'queue1'. ;(on Server B) queues.conf: [queue1] ;what makes this work with distributed states is the 'State Interface' parameter ... member = Local/00123456789,0,ALICE,SIP/cisco1 Alice will need a number to ring to login/logout of queue1 hosted on Server B; Dialplan Example: on server B: ... exten = s,n,Set(queuename=queue1) exten = s,n,Set(interface=Local/00123456789) exten = s,n,Set(penalty=0) exten = s,n,Set(stateinterface=SIP/cisco1) exten = s,n(queue-add),AddQueueMember(${queuename},${interface},${penalty},options,, ${stateinterface}) And to remove the member; ... exten = s,n(queue-remove),RemoveQueueMember(${queuename},${interface}) Alec Davis I've not tried to publish device state with XMPP yet but I've discovered this issue https://issues.asterisk.org/jira/browse/ASTERISK-18078 I'm planning to install my XMPP server on the same machine as one asterisk server so hopefully, I won't be hit by the issue above but have you met this issue ? Could you get around ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and distributed device state over WAN
I've not tried to publish device state with XMPP yet but I've discovered this issue https://issues.asterisk.org/jira/browse/ASTERISK-18078 I'm planning to install my XMPP server on the same machine as one asterisk server so hopefully, I won't be hit by the issue above but have you met this issue ? Could you get around ? I installed Tigase on the asterisk server hosting the queues, our main office. Yes I have experienced https://issues.asterisk.org/jira/browse/ASTERISK-18078 But for us, we have a fibre link between 3 offices, so hardly ever see the problem, only when I reboot the server with XMPP do the other 2 asterisk's segfault. The work around for us is: Don't reboot XMPP/Asterisk server during critical periods, however they have a cron script checking to see whether asterisk is alive, and if not restart asterisk. Prior to having the luxury of private 10Mb fibre links, we had to rely on internet ADSL VPN links between our offices, no good for voice, but reliably enough for device state updates. Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and distributed device state over WAN
On 01/25/2013 01:59 PM, Alec Davis wrote: I've not tried to publish device state with XMPP yet but I've discovered this issue https://issues.asterisk.org/jira/browse/ASTERISK-18078 I'm planning to install my XMPP server on the same machine as one asterisk server so hopefully, I won't be hit by the issue above but have you met this issue ? Could you get around ? I installed Tigase on the asterisk server hosting the queues, our main office. Yes I have experienced https://issues.asterisk.org/jira/browse/ASTERISK-18078 But for us, we have a fibre link between 3 offices, so hardly ever see the problem, only when I reboot the server with XMPP do the other 2 asterisk's segfault. The work around for us is: Don't reboot XMPP/Asterisk server during critical periods, however they have a cron script checking to see whether asterisk is alive, and if not restart asterisk. Prior to having the luxury of private 10Mb fibre links, we had to rely on internet ADSL VPN links between our offices, no good for voice, but reliably enough for device state updates. Not that this is an excuse or a valid workaround for everyone, but I believe that issue won't apply if you're using Asterisk 11 and res_xmpp. res_jabber: yup, totally still a problem. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and distributed device state over WAN
Not that this is an excuse or a valid workaround for everyone, but I believe that issue won't apply if you're using Asterisk 11 and res_xmpp. res_jabber: yup, totally still a problem. Hmm. We're using Asterisk 11, but I still think res_jabber. Why havn't I changed to res_xmpp, I have no answer for that. Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and Distinctive Ring with Alert-Info
On 26/11/2012 10:14 AM, Klaverstyn, David C wrote: Hi All, I’m new to Queues and I have created one as follows which seems to work ok. [david-test] strategy = rrmemory timeout = 10 retry = 0 maxlen = 0 announce-frequency = 0 announce-holdtime = no member = SIP/121 member = SIP/122 member = SIP/123 I’m wondering how do you change the SipAddHeader/Alert-Info when a call comes from a queue so users know it is a queue that is calling? Is something like the following supposed to work? exten = 0453451564,1,SipAddHeader(Alert-Info: n=Classic-4;w=3;c=4) exten = 0453451564,2,Queue(david-test) Seems to work with Asterisk 1.8.18.0. I'm using extensions.ael and have tested the following; 400 = { SIPAddHeader(Alert-Info: n=Classic-4;w=3;c=4); Queue(400,inrt,,,30); Hangup(); }; Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues, pickup and transfers
2011/5/5 Olivier oza_4...@yahoo.fr Hi, If my memory serves me right, up to Asterisk 1.6, Queue app internals kept the application from working some other apps such as PickUp. I wonder if such things are possible (and if possible, still keep useful Queue Logs ie logs in which picked up or transfered calls are shown as such): 1- a call enters a queue, a phone rings, and a non-Queue member dials some digits and speaks with caller 2- a call enters a queue, a phone rings, basically a BLF attached to this phone blinks and someone pressing this BLS would pick the call up Which method should be used for case 1 ? Using Transfer or PickUp ? Regards Thinking over this, I told myself that in some situations, MeetMe (or ConfBridge) or Parking could be used instead of Queue : caller would hear MusicOnHold while waiting to be answered and while at the same time, a call could be originated and then transfered to appropriate channel. Anyway, my questions remain as Queue fit my other requirements. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer
The Queue() application can automatically pause members who fail to answer; this would be the solution to your problem. With that solution in place, though, the agent will still need to be able to un-pause when they return to their desk, and since that is the case, they really should be taught to go on pause when they leave their desk as well :-) Not to mention that your caller has to wait for however long your agent timeout is when this happens the first time, which is bad customer service. I am a little confused as to what the OP wants the system to do? Call the proper agent, but when they don't answer, on the next call, it shouldn't call the same agent? OK, but for how long? 5 minutes? Until they manually unpause (current option as described by Kevin), 30 minutes? Should it then up their penalty? For how long? I thought I replied back, but can`t find my own reply on the list. I just want a sequenced queue. I DO know that it means a bit longer waiting time if the first agent is unavailable, but I`m willing to live with it. I know I could write the extension to ring phones in sequence, but the queue includes other useful functionality (and logs that can be parsed and statistics created from it). I would be surprised if a queue cannot be made to ring phones always in the same order (phone A, B and C, back to A, B, C...). Is this simple use case possible? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer
I am a little confused as to what the OP wants the system to do? Call the proper agent, but when they don't answer, on the next call, it shouldn't call the same agent? OK, but for how long? 5 minutes? Until they manually unpause (current option as described by Kevin), 30 minutes? Should it then up their penalty? For how long? I should have been more precise. I don't actually expect all this to happen, but here's what I wish it did: 1) Ring agents in Round Robin fashion, but always in the same order (could simply use the already existing penalty value) 2) Always start from the top (taking into account the ringinuse value) Basically, a simple _pre-ordered_ Roundrobin. I could make this even better by (as you hinted at yourself) by using autopause and asking for an autounpause after x minutes feature. But those two things above would be wonderful, and I was actually surprised that it wasn't a possible setting. Unless I can order the agents somehow, but I seem to understand that dynamic agents are sequenced in the order in which they joined the queue, not according to some easily defined position value. How I would envision this being configured? A queue setting that would define how it handles penalty. Either in the current Ring the best agent(s) over and over again or try the good agents first, but then move on. Just a yes/no value. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer
Mike wrote: I was hoping to use this Queue not for professional agents in a call center, but for reception. When the receptionist (lowest penalty) is not at the desk, then some junior sales person can pick up those calls. We have our receptionist setup in a front-desk queue that has 2 phones in it. The incoming call rings directly to the phone for 30 seconds, if not answered, plays the, Please wait while we find someone and then drops them into a queue. At this point, it rings the operator phone again and if that fails, the 2nd phone. This will bounce back and forth between phones, until finally dropping the call into our dial-by-name directory if nobody answers. We also have both phones in a call group/pickup group, allowing to grab a call by doing a *7 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer
We have our receptionist setup in a front-desk queue that has 2 phones in it. The incoming call rings directly to the phone for 30 seconds, if not answered, plays the, Please wait while we find someone and then drops them into a queue. At this point, it rings the operator phone again and if that fails, the 2nd phone. This will bounce back and forth between phones, until finally dropping the call into our dial-by-name directory if nobody answers. We also have both phones in a call group/pickup group, allowing to grab a call by doing a *7 Thanks Doug. I realize there are many things I can do, I was just hoping to use an application command to do it all. What you described might just be what I end up doing. Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer
On Feb 3, 2011, at 11:12 AM, Kevin P. Fleming wrote: The Queue() application can automatically pause members who fail to answer; this would be the solution to your problem. With that solution in place, though, the agent will still need to be able to un-pause when they return to their desk, and since that is the case, they really should be taught to go on pause when they leave their desk as well :-) Not to mention that your caller has to wait for however long your agent timeout is when this happens the first time, which is bad customer service. I am a little confused as to what the OP wants the system to do? Call the proper agent, but when they don't answer, on the next call, it shouldn't call the same agent? OK, but for how long? 5 minutes? Until they manually unpause (current option as described by Kevin), 30 minutes? Should it then up their penalty? For how long? Maybe some more specifics would help here. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues without agent login
Barry, I'm using the Asterisk GUI. When defining a User extension (menu 'user') the only option I have is Is Agent. The SIP extension (11) is automatically created as an agent that needs to log in. In the advanced options I can manually edit queues.conf and change to member=SIP/11. This way of defining queues and members is a 2-step work. Can it be done in 1 step using the Asterisk GUI ?? Kind regards, Jonas. On Wed, 2009-11-18 at 10:42 -0500, Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 jonas kellens wrote: Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? Make the phones members of the queue. In queues.conf: [MY_QUEUE] member = SIP/1234 member = SIP/5678 etc. Barry ___ -- 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
Re: [asterisk-users] Queues without agent login
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 jonas kellens wrote: Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? Make the phones members of the queue. In queues.conf: [MY_QUEUE] member = SIP/1234 member = SIP/5678 etc. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLBBYACFu3bIiwtTARAg5oAKClAtJ98LaSXnjCDBx4xlRcLQ9l/wCgoeI+ BAi7wu2nQ6vNPZSaLCDB4DA= =egpe -END PGP SIGNATURE- ___ -- 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
Re: [asterisk-users] Queues without agent login
Simply use member=SIP/Tarek member=IAX2/JONAS member=LOCAL/whatever simple and good.. with member=SIP/extension i'm facing a CALL WAITING issue.. the agent hears a callwaiting signal whenever the queue tries to call .. so i woul dsuggest using call-limit and busy limite with all your Agents -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: jonas.kell...@telenet.be To: asterisk-users@lists.digium.com Date: Wed, 18 Nov 2009 16:21:12 +0100 Subject: [asterisk-users] Queues without agent login Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to call an extension to log in (and having to give some PIN). I just want a queue (with MoH) to collect multiple incoming calls and then one at a time transfer them to an available SIP-phone. Is this possible ? Thanks you. Jonas. _ Windows 7: I wanted simpler, now it's simpler. I'm a rock star. http://www.microsoft.com/Windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:112009___ -- 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
Re: [asterisk-users] Queues
It should be realistic, but have you considered just using followme to add the cell phones to the queue list? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an on-call queue. A call comes in and it rings the on-call extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the on-call queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- 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
Re: [asterisk-users] Queues
I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an on-call queue. A call comes in and it rings the on-call extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the on-call queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- 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 ___ -- 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
Re: [asterisk-users] Queues
Since followme is extension-based, you have at least two options. Option 1 is to have a few extensions designated for following where you punch in the cell numbers as you wish. Option 2 is to use day logic to point to the following guys based on days.If I were doing option 2, I'd try to use ASTDB to control this instead of having to code a lot of dialplan, but that's just me. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an on-call queue. A call comes in and it rings the on-call extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the on-call queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- 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 ___ -- 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
Re: [asterisk-users] Queues
Hi Travis, There's lots of different ways to attack on-call roster solutions in Asterisk - as Danny suggested, FollowMe() is definitely an option (and normally the best), but it doesn't always suit the business need. However, also as Danny suggested, in most cases using ASTDB in some way to simplify dialling plans is the way to go - then you just have to decide how you want to update the information as to the number to call, in ASTDB. For example, I had a customer a couple of years back who desperately wanted to manage his on-call roster routing using a web interface. I dollied up a simple PHP/MySQL web interface with a list of all the people (and their mobile/cell numbers) in a drop down list - they could simply select the right person, and click a First Call button to make that person the first in the roster, select another person and click a Second Call button to make that person the second in the roster, and so on. Using the Asterisk manager interface - (or even asterisk -rx command if you're not comfortable using the AMI) - you get the numbers selected into ASTDB. The dialplan just comes along then and reads the appropriate numbers from ASTDB as it steps through, and dials the people in order. As with many things in Asterisk - there is more than one way to hump the leg. Cheers Michael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, 17 November 2009 08:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queues Since followme is extension-based, you have at least two options. Option 1 is to have a few extensions designated for following where you punch in the cell numbers as you wish. Option 2 is to use day logic to point to the following guys based on days.If I were doing option 2, I'd try to use ASTDB to control this instead of having to code a lot of dialplan, but that's just me... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an on-call queue. A call comes in and it rings the on-call extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the on-call queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- 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 IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named
Re: [asterisk-users] Queues
Hi Michael, Your web interface for the on-call roster is pretty close to what we're trying to trying to achieve. I would like to have people signing into the on-call queue be the method that determined whose cell phone to call. I was hoping there was a way to pass the call exiting the queue to a variable or two that was composed of the extensions currently logged into the queue. I set up an extension number for people to call into and enter a forwarding number which writes an entry into the ASTDB. I have my dialplan check to see if there is an ASTDB entry for that extension before it tries to dial their deskphone, and if there is an entry it dials the forwarded number stored in the database instead. The closest thing so far I have found to what I am trying to achieve is to hard code a couple of spare extensions into the dialplan, and then have whoever is on-call set one of those extensions to their cell phone number. I'll definitely take another look at followme to see if I can adapt that what I'm trying to achieve. Thanks Danny Michael, Travis - Original Message - From: Michael Wyres mwy...@cdm.com.au To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 2:23:49 PM Subject: Re: [asterisk-users] Queues Hi Travis, There’s lots of different ways to attack “on-call” roster solutions in Asterisk – as Danny suggested, FollowMe() is definitely an option (and normally the best), but it doesn’t always suit the “business need”. However, also as Danny suggested, in most cases using ASTDB in some way to simplify dialling plans is the way to go - then you just have to decide how you want to update the information as to the number to call, in ASTDB. For example, I had a customer a couple of years back who desperately wanted to manage his “on-call roster” routing using a web interface. I dollied up a simple PHP/MySQL web interface with a list of all the people (and their mobile/cell numbers) in a drop down list – they could simply select the right person, and click a “First Call” button to make that person the first in the roster, select another person and click a “Second Call” button to make that person the second in the roster, and so on. Using the Asterisk manager interface – (or even “asterisk –rx command” if you’re not comfortable using the AMI) – you get the numbers selected into ASTDB. The dialplan just comes along then and reads the appropriate numbers from ASTDB as it steps through, and dials the people in order. As with many things in Asterisk – there is more than one way to “hump the leg”. Cheers Michael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, 17 November 2009 08:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queues Since followme is “extension-based”, you have at least two options. Option 1 is to have a few extensions designated for “following” where you punch in the cell numbers as you wish. Option 2 is to use “day logic” to point to the “following” guys based on days. If I were doing option 2, I’d try to use ASTDB to control this instead of having to code a lot of dialplan, but that’s just me… From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an on-call queue. A call comes in and it rings the on-call extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the on-call queue. I've got
Re: [asterisk-users] Queues
Again – lots of ways to do it – you could use a web interface to set the numbers in ASTDB for lookup – or you could create an IVR to ask for the number, and store it in ASTDB that way. Good luck! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Tuesday, 17 November 2009 10:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues Hi Michael, Your web interface for the on-call roster is pretty close to what we're trying to trying to achieve. I would like to have people signing into the on-call queue be the method that determined whose cell phone to call. I was hoping there was a way to pass the call exiting the queue to a variable or two that was composed of the extensions currently logged into the queue. I set up an extension number for people to call into and enter a forwarding number which writes an entry into the ASTDB. I have my dialplan check to see if there is an ASTDB entry for that extension before it tries to dial their deskphone, and if there is an entry it dials the forwarded number stored in the database instead. The closest thing so far I have found to what I am trying to achieve is to hard code a couple of spare extensions into the dialplan, and then have whoever is on-call set one of those extensions to their cell phone number. I'll definitely take another look at followme to see if I can adapt that what I'm trying to achieve. Thanks Danny Michael, Travis - Original Message - From: Michael Wyres mwy...@cdm.com.au To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 2:23:49 PM Subject: Re: [asterisk-users] Queues Hi Travis, There’s lots of different ways to attack “on-call” roster solutions in Asterisk – as Danny suggested, FollowMe() is definitely an option (and normally the best), but it doesn’t always suit the “business need”. However, also as Danny suggested, in most cases using ASTDB in some way to simplify dialling plans is the way to go - then you just have to decide how you want to update the information as to the number to call, in ASTDB. For example, I had a customer a couple of years back who desperately wanted to manage his “on-call roster” routing using a web interface. I dollied up a simple PHP/MySQL web interface with a list of all the people (and their mobile/cell numbers) in a drop down list – they could simply select the right person, and click a “First Call” button to make that person the first in the roster, select another person and click a “Second Call” button to make that person the second in the roster, and so on. Using the Asterisk manager interface – (or even “asterisk –rx command” if you’re not comfortable using the AMI) – you get the numbers selected into ASTDB. The dialplan just comes along then and reads the appropriate numbers from ASTDB as it steps through, and dials the people in order. As with many things in Asterisk – there is more than one way to “hump the leg”. Cheers Michael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, 17 November 2009 08:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queues Since followme is “extension-based”, you have at least two options. Option 1 is to have a few extensions designated for “following” where you punch in the cell numbers as you wish. Option 2 is to use “day logic” to point to the “following” guys based on days.If I were doing option 2, I’d try to use ASTDB to control this instead of having to code a lot of dialplan, but that’s just me… From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3
Re: [asterisk-users] queues autopause
Rilawich Ango escribió: Hi, I have 3 queue set in the table as below. name,autopause 1000,1 2000,1 3000,1 In queue 1000, the autopause works after member failed to answer call. However, other queues don't work for the autopause function. queue 1000: -- Nobody picked up in 25000 ms -- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they failed to answer. queue 2000/3000: -- Nobody picked up in 25000 ms -- SIP/1234-1544cd90 is ringing Is it the limitation of the asterisk to support one queue of autopause function? Or any setting I need to take care to make autopause function works for all queue? ango ___ -- 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 Hi, Some months ago there was a discussion about this, with a simple solution involving minimal changes to the source (1 line of code). Search the archives of this list and you will find the answer. BTW, what version of asterisk are you using? Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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
Re: [asterisk-users] queues autopause
Thanks. Finally, I find that it was caused by the use of the table wrongly. On Thu, Oct 22, 2009 at 10:23 PM, Miguel Molina mmol...@millenium.com.co wrote: Rilawich Ango escribió: Hi, I have 3 queue set in the table as below. name,autopause 1000,1 2000,1 3000,1 In queue 1000, the autopause works after member failed to answer call. However, other queues don't work for the autopause function. queue 1000: -- Nobody picked up in 25000 ms -- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they failed to answer. queue 2000/3000: -- Nobody picked up in 25000 ms -- SIP/1234-1544cd90 is ringing Is it the limitation of the asterisk to support one queue of autopause function? Or any setting I need to take care to make autopause function works for all queue? ango ___ -- 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 Hi, Some months ago there was a discussion about this, with a simple solution involving minimal changes to the source (1 line of code). Search the archives of this list and you will find the answer. BTW, what version of asterisk are you using? Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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 ___ -- 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
Re: [asterisk-users] Queues with unavailable members
C. Chad Wallace cwall...@lodgingcompany.com writes: OK, I decided to write it up in AEL. It's incomplete and untested, but it probably gets the idea across a little better. context agentcalls { _2XX = { Set(AGENT=${EXTEN}); // Assuming agent ID is extension. if (${EPOCH}${DB(AgentPaused/${AGENT})}) { // Let the call through to the cell phone Dial(...); if (cell call was rejected) { // Flag agent as paused for the next 30 seconds. Set(DB(AgentPaused/${AGENT})=$[${EPOCH}+30]); }; } else { // Agent still paused. }; }; }; I was going in the same direction at the end of my first mail, but I hadn't written any code. There is a problem though: The Queue application will keep sending calls to the Local channel, which have to be rejected, over and over. Would it perhaps work to simply Wait(30) if the call is rejected by the phone? If the Queue assumes that the phone is busy for those 30 seconds, I have accomplished my goal. It's worth a shot. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Benny Amorsen benny+use...@amorsen.dk writes: Would it perhaps work to simply Wait(30) if the call is rejected by the phone? If the Queue assumes that the phone is busy for those 30 seconds, I have accomplished my goal. It's worth a shot. This works! Actually I tried out Wait(1000), but that worked fine. After 30 seconds (the timeout in the queue) the Local channel was closed, and a short while later a new call attempt was made. Just as I was hoping. It would still be neat to have a min_dial_interval option, so that Queue never overwhelms the server with failing dial attempts. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
At 11:23 AM on 16 Oct 2009, Benny Amorsen wrote: I was going in the same direction at the end of my first mail, but I hadn't written any code. There is a problem though: The Queue application will keep sending calls to the Local channel, which have to be rejected, over and over. Would it perhaps work to simply Wait(30) if the call is rejected by the phone? If the Queue assumes that the phone is busy for those 30 seconds, I have accomplished my goal. It's worth a shot. It would only be trying one agent at a time for each waiting queue member... I don't know how expensive it is to open and close a Local channel and do a DB lookup, but I wouldn't expect it to be a real problem. You are at least avoiding multiple calls out to the cellular network. Also, if there is another agent available, the caller would be connected immediately, and it wouldn't have to make any more attempts. With the Wait() solution, that caller would be waiting for 30 seconds regardless of whether there's anyone else available. Of course, I don't know your business case, so you'll have to decide which of the two problems is worse. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
C. Chad Wallace cwall...@lodgingcompany.com writes: It would only be trying one agent at a time for each waiting queue member... Would it? Almost all our queues are on a ringall strategy. I don't know how expensive it is to open and close a Local channel and do a DB lookup, but I wouldn't expect it to be a real problem. You are at least avoiding multiple calls out to the cellular network. Not that expensive, but still a bit of a waste when done every couple of seconds. Especially if multiple agents are unavailable. Also, if there is another agent available, the caller would be connected immediately, and it wouldn't have to make any more attempts. With the Wait() solution, that caller would be waiting for 30 seconds regardless of whether there's anyone else available. This bit is solved by the ringall strategy. Of course, I don't know your business case, so you'll have to decide which of the two problems is worse. I'm fairly happy with the Wait(1000) solution for now. We'll see if testing shows any problems with it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
At 7:35 PM on 16 Oct 2009, Benny Amorsen wrote: C. Chad Wallace cwall...@lodgingcompany.com writes: Also, if there is another agent available, the caller would be connected immediately, and it wouldn't have to make any more attempts. With the Wait() solution, that caller would be waiting for 30 seconds regardless of whether there's anyone else available. This bit is solved by the ringall strategy. Of course, I don't know your business case, so you'll have to decide which of the two problems is worse. I'm fairly happy with the Wait(1000) solution for now. We'll see if testing shows any problems with it. Oh yeah, I hadn't even considered the ringall strategy! With that, your Wait() solution sounds perfect to me. Congrats! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Elliot Otchet elliot.otc...@callingcircles.com writes: Have you tried autopause=yes in your queue configuration? You can then unpause the member by either the dialplan (e.g. having the cell phone user log back in) or using an AMI based program to change the paused state. You can read more about the latter here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html That looks very interesting, thank you! First of all though I need to avoid having them autopause just because they don't answer their phone. It should only happen if the call to their phone fails completely. I guess that could be done by not doing autopause but instead pausing manually in the context that the Local call passes through. That would also solve my second problem, which is that I need to pause it in all queues, not just one queue. The last challenge is to somehow unpause them after a while. In traditional programming that would be something like keeping a list of timeout,queuemember ordered by timeout, and then when every call comes in unpause and remove the ones where timeout expired... I'm not sure that I can make an ordered list in the dialplan though. I may have to resort to AGI, but I still need somewhere to actually store the list. Tricky. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Lenz Emilitri lenz.lo...@gmail.com writes: You could configure them as agents and have them log off automatically after a while they're not responding. Agents have to log in and wait for calls though, don't they? There used to be AgentCallbackLogin, but that has been replaced by dialplan code and chan_local. Otherwise a nice idea though. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
That shouldn't be too hard to accomplish. If you've got the addons (and mysql) installed you could store them in a MySQL table (timestamp, device) and have a cron job set to run at X frequency that un-pauses the queue members via AMI. Don't want to go to MySQL? Use system() to 'touch' files named after devices. Then have your cron script go through the files by creation date. Either way gets you there. -Elliot -Original Message- From: Benny Amorsen [mailto:benny+use...@amorsen.dk] Sent: Thursday, October 15, 2009 5:06 AM To: Elliot Otchet Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Queues with unavailable members Elliot Otchet elliot.otc...@callingcircles.com writes: Have you tried autopause=yes in your queue configuration? You can then unpause the member by either the dialplan (e.g. having the cell phone user log back in) or using an AMI based program to change the paused state. You can read more about the latter here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html That looks very interesting, thank you! First of all though I need to avoid having them autopause just because they don't answer their phone. It should only happen if the call to their phone fails completely. I guess that could be done by not doing autopause but instead pausing manually in the context that the Local call passes through. That would also solve my second problem, which is that I need to pause it in all queues, not just one queue. The last challenge is to somehow unpause them after a while. In traditional programming that would be something like keeping a list of timeout,queuemember ordered by timeout, and then when every call comes in unpause and remove the ones where timeout expired... I'm not sure that I can make an ordered list in the dialplan though. I may have to resort to AGI, but I still need somewhere to actually store the list. Tricky. /Benny This message is intended only for the use of the individual (s) or entity to which it is addressed and may contain information that is privileged, confidential, and/or proprietary to Calling Circles LLC and its affiliates. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, forwarding or copying of this communication is prohibited without the express permission of the sender. If you have received this communication in error, please notify the sender immediately and delete the original message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Elliot Otchet elliot.otc...@callingcircles.com writes: That shouldn't be too hard to accomplish. If you've got the addons (and mysql) installed you could store them in a MySQL table (timestamp, device) and have a cron job set to run at X frequency that un-pauses the queue members via AMI. Don't want to go to MySQL? Use system() to 'touch' files named after devices. Then have your cron script go through the files by creation date. Either way gets you there. This seems like a very heavyweight solution. Having a cron job running every minute isn't particularly attractive, and making a daemon do the job isn't my cup of tea either. Perhaps the problem could be restated in a different way: After a queue member rejects a call (instead of just not answering), the queue should wait X amount of time before sending the next call. Queues.conf has a million settings, but I can't find one which does this. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote: Perhaps the problem could be restated in a different way: After a queue member rejects a call (instead of just not answering), the queue should wait X amount of time before sending the next call. Queues.conf has a million settings, but I can't find one which does this. To pause an agent, store the unpause time per agent in the AstDB. Then when you're deciding whether to give out a call (in the Local channel), look up ${DB(AgentPaused/agentid)} and compare it to the current time. If there is no record or the time has passed, put the call through; otherwise, skip that agent. Sorry, no example code yet... I just wanted to get the idea out there. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
At 11:32 AM on 15 Oct 2009, C. Chad Wallace wrote: At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote: Perhaps the problem could be restated in a different way: After a queue member rejects a call (instead of just not answering), the queue should wait X amount of time before sending the next call. Queues.conf has a million settings, but I can't find one which does this. To pause an agent, store the unpause time per agent in the AstDB. Then when you're deciding whether to give out a call (in the Local channel), look up ${DB(AgentPaused/agentid)} and compare it to the current time. If there is no record or the time has passed, put the call through; otherwise, skip that agent. Sorry, no example code yet... I just wanted to get the idea out there. OK, I decided to write it up in AEL. It's incomplete and untested, but it probably gets the idea across a little better. context agentcalls { _2XX = { Set(AGENT=${EXTEN}); // Assuming agent ID is extension. if (${EPOCH}${DB(AgentPaused/${AGENT})}) { // Let the call through to the cell phone Dial(...); if (cell call was rejected) { // Flag agent as paused for the next 30 seconds. Set(DB(AgentPaused/${AGENT})=$[${EPOCH}+30]); }; } else { // Agent still paused. }; }; }; -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
You could configure them as agents and have them log off automatically after a while they're not responding. l. 2009/10/14 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone=SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a member of a queue. In that case Queue() keeps sending the call to the phone, and the cell-phone=SIP gateway dutifully makes a call, which is then rejected by the cellular network. A few seconds later, Queue() tries again. This needlessly wastes resources both in Asterisk and in the cellular network. One idea is to run the call through chan_local (we do this anyway because we need to format the caller-ID differently for different phones) and then record if a call is rejected, and for the next 30 seconds just abort if we are asked to send a call to that particular phone. The downside is that we are still running a call through part of the dial plan, but at least it can be done in perhaps 3 lines of code. I would very much like to hear about smarter ways to do it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
What is the command to log off the agents ? Thx On Wed, Oct 14, 2009 at 6:45 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: You could configure them as agents and have them log off automatically after a while they're not responding. l. 2009/10/14 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone=SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a member of a queue. In that case Queue() keeps sending the call to the phone, and the cell-phone=SIP gateway dutifully makes a call, which is then rejected by the cellular network. A few seconds later, Queue() tries again. This needlessly wastes resources both in Asterisk and in the cellular network. One idea is to run the call through chan_local (we do this anyway because we need to format the caller-ID differently for different phones) and then record if a call is rejected, and for the next 30 seconds just abort if we are asked to send a call to that particular phone. The downside is that we are still running a call through part of the dial plan, but at least it can be done in perhaps 3 lines of code. I would very much like to hear about smarter ways to do it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with unavailable members
Have you tried autopause=yes in your queue configuration? You can then unpause the member by either the dialplan (e.g. having the cell phone user log back in) or using an AMI based program to change the paused state. You can read more about the latter here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html -Elliot -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen Sent: Wednesday, October 14, 2009 7:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues with unavailable members We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone=SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a member of a queue. In that case Queue() keeps sending the call to the phone, and the cell-phone=SIP gateway dutifully makes a call, which is then rejected by the cellular network. A few seconds later, Queue() tries again. This needlessly wastes resources both in Asterisk and in the cellular network. One idea is to run the call through chan_local (we do this anyway because we need to format the caller-ID differently for different phones) and then record if a call is rejected, and for the next 30 seconds just abort if we are asked to send a call to that particular phone. The downside is that we are still running a call through part of the dial plan, but at least it can be done in perhaps 3 lines of code. I would very much like to hear about smarter ways to do it. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is intended only for the use of the individual (s) or entity to which it is addressed and may contain information that is privileged, confidential, and/or proprietary to Calling Circles LLC and its affiliates. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, forwarding or copying of this communication is prohibited without the express permission of the sender. If you have received this communication in error, please notify the sender immediately and delete the original message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues load balancing
Take a look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random You should be able to do what you want with this, it obviously won't take in to account the actual amount of people still in the queue (for example if someone hangs up while on hold). I'm sure there'd be a way of integrating this in to it using some different functions, but for a quick fix random will do just fine. Cheers 2009/7/20 Joao Gomes Pereira gomespere...@startel.pt Hello I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2 How can I do that load balancing in extensions.conf? I have something like this: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer ; 2 in 3 calls go to queue_1 exten = 123,x,Queue(queue_1) ; 1 in 3 calls go to queue_2 exten = 123,x,Queue(queue_2) But how can I configure this call distribution? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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 ___ -- 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
Re: [asterisk-users] queues load balancing
Thanks for the idea. I will try it this way: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer exten = 123,4,Random(33:123,10) exten = 123,5,Queue(queue_1) exten = 123,6,Hangup exten = 123,10,Queue(queue_2) exten = 123,11,Hangup Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt Geraint Lee wrote: Take a look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random You should be able to do what you want with this, it obviously won't take in to account the actual amount of people still in the queue (for example if someone hangs up while on hold). I'm sure there'd be a way of integrating this in to it using some different functions, but for a quick fix random will do just fine. Cheers 2009/7/20 Joao Gomes Pereira gomespere...@startel.pt mailto:gomespere...@startel.pt Hello I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2 How can I do that load balancing in extensions.conf? I have something like this: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer ; 2 in 3 calls go to queue_1 exten = 123,x,Queue(queue_1) ; 1 in 3 calls go to queue_2 exten = 123,x,Queue(queue_2) But how can I configure this call distribution? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt http://www.startel.pt +351 304500650 sip: gomespere...@startel.pt mailto:gomespere...@startel.pt ___ -- 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 -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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
Re: [asterisk-users] queues load balancing
On 21/7/09 12:08 AM, Joao Gomes Pereira wrote: Thanks for the idea. I will try it this way: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer exten = 123,4,Random(33:123,10) exten = 123,5,Queue(queue_1) exten = 123,6,Hangup exten = 123,10,Queue(queue_2) exten = 123,11,Hangup Bear in mind that the Random application has been deprecated in favour of the RANDOM function: asterisk -rx 'show application random' -= Info about application 'Random' =- [Synopsis] Conditionally branches, based upon a probability [Description] Random([probability]:[[context|]extension|]priority) probability := INTEGER in the range 1 to 100 DEPRECATED: Use GotoIf($[${RAND(1,100)} number]?label) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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
Re: [asterisk-users] queues load balancing
Here is a brute force solution: [global] CALLCOUNT=0 exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer exten = 123,4,Set(CALLCOUNT)=${CALLCOUNT}+1) exten = 123,5,Gotoif($[$(CALLCOUNT} = 3]?queue2) exten = 123,6,Queue(queue_1) exten = 123,7,Hangup exten = 123,8(queue2),Set(CALLCOUNT=0) exten = 123,9,Queue(queue_2) exten = 123,10,Hangup -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Monday, July 20, 2009 7:37 AM To: gomespere...@startel.pt; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] queues load balancing On 21/7/09 12:08 AM, Joao Gomes Pereira wrote: Thanks for the idea. I will try it this way: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer exten = 123,4,Random(33:123,10) exten = 123,5,Queue(queue_1) exten = 123,6,Hangup exten = 123,10,Queue(queue_2) exten = 123,11,Hangup Bear in mind that the Random application has been deprecated in favour of the RANDOM function: asterisk -rx 'show application random' -= Info about application 'Random' =- [Synopsis] Conditionally branches, based upon a probability [Description] Random([probability]:[[context|]extension|]priority) probability := INTEGER in the range 1 to 100 DEPRECATED: Use GotoIf($[${RAND(1,100)} number]?label) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues load balancing
Danny Nicholas schrieb: Here is a brute force solution: [global] CALLCOUNT=0 exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer exten = 123,4,Set(CALLCOUNT)=${CALLCOUNT}+1) ...,Set(CALLCOUNT=$[${CALLCOUNT} + 1]) or ...,Set(CALLCOUNT=${MATH(${CALLCOUNT}+1,int)}) exten = 123,5,Gotoif($[$(CALLCOUNT} = 3]?queue2) exten = 123,6,Queue(queue_1) exten = 123,7,Hangup exten = 123,8(queue2),Set(CALLCOUNT=0) exten = 123,9,Queue(queue_2) exten = 123,10,Hangup Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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
Re: [asterisk-users] Queues recording CDR
Hi, My apologies Nicolas, a mistake from my part. And I appreciate for correcting me. Asternic is a good piece of work. Regards, Kurian Thayil. On Mon, 2009-07-06 at 09:41 -0300, Nicolás Gudiño wrote: Hello, Just a correction, Asternic Call Center Stats is not from asteriskguru. Asteriskguru has its own statistic program that is not open source, but free to use. Asternic was written by me (not asteriskguru) and has an open source version and a commercial one. Best regards, -- Nicolás Gudiño Buenos Aires - Argentina On Mon, Jul 6, 2009 at 12:08 AM, Kurian Thayilkurianmtha...@gmail.com wrote: Hi Sriram, 1. Set the channel variable MonitorFilename before Queue() in dialplan and you can give some meaningful filename for record. 2. I guess you can use an AGI to capture events and then integrate this with a DB in the Backend. This should help you to track the activity. 3. asternic from asteriskguru is kind of OK. Gives you a live and detailed report. Parses the queue_log to the MySQL DB and works. This parse program could be used in your AGI which I mentioned in point 2. Hope this helps. Regards, Kurian Thayil. On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote: Hi 1. I want to record all calls that land to an agent via a queue using a meaningful name - as of now i name the recorded file on the fly using {CALLERID} variable so that the file gets stored using the caller id iunder /var/spool/asterisk/monitor , now if i want to store it as CALLERIDEXTEN where call landed from queue how can i do this ? 2. I have a CDR issue - when A calls he is put in Queue and say he is answered by Agent B ..Agent B transfers the Call to agent C as it is to Agent C whom A wants to talk..when the call gets d/c the CDR for that call shows the destination field as B whereas it shd be C...how do i take care of this ...in my call center agents are paid on the basis of talk time on inbound calls - this way an agent who just transfers calls is at merry !! 3. Are their any GPL based queue reporting software - hows the asterisk queue statistics program from asteriskguru.com has anyone tried it ? Thanks Sriram ___ -- 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 -- Kurian Mathew Thayil. (GPG KeyID: E232394F) ___ -- 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 ___ -- 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 -- Kurian Mathew Thayil. (GPG KeyID: E232394F) signature.asc Description: This is a digitally signed message part ___ -- 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
Re: [asterisk-users] Queues recording CDR
Hello, Just a correction, Asternic Call Center Stats is not from asteriskguru. Asteriskguru has its own statistic program that is not open source, but free to use. Asternic was written by me (not asteriskguru) and has an open source version and a commercial one. Best regards, -- Nicolás Gudiño Buenos Aires - Argentina On Mon, Jul 6, 2009 at 12:08 AM, Kurian Thayilkurianmtha...@gmail.com wrote: Hi Sriram, 1. Set the channel variable MonitorFilename before Queue() in dialplan and you can give some meaningful filename for record. 2. I guess you can use an AGI to capture events and then integrate this with a DB in the Backend. This should help you to track the activity. 3. asternic from asteriskguru is kind of OK. Gives you a live and detailed report. Parses the queue_log to the MySQL DB and works. This parse program could be used in your AGI which I mentioned in point 2. Hope this helps. Regards, Kurian Thayil. On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote: Hi 1. I want to record all calls that land to an agent via a queue using a meaningful name - as of now i name the recorded file on the fly using {CALLERID} variable so that the file gets stored using the caller id iunder /var/spool/asterisk/monitor , now if i want to store it as CALLERIDEXTEN where call landed from queue how can i do this ? 2. I have a CDR issue - when A calls he is put in Queue and say he is answered by Agent B ..Agent B transfers the Call to agent C as it is to Agent C whom A wants to talk..when the call gets d/c the CDR for that call shows the destination field as B whereas it shd be C...how do i take care of this ...in my call center agents are paid on the basis of talk time on inbound calls - this way an agent who just transfers calls is at merry !! 3. Are their any GPL based queue reporting software - hows the asterisk queue statistics program from asteriskguru.com has anyone tried it ? Thanks Sriram ___ -- 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 -- Kurian Mathew Thayil. (GPG KeyID: E232394F) ___ -- 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 ___ -- 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
Re: [asterisk-users] Queues recording CDR
Hi Sriram, 1. Set the channel variable MonitorFilename before Queue() in dialplan and you can give some meaningful filename for record. 2. I guess you can use an AGI to capture events and then integrate this with a DB in the Backend. This should help you to track the activity. 3. asternic from asteriskguru is kind of OK. Gives you a live and detailed report. Parses the queue_log to the MySQL DB and works. This parse program could be used in your AGI which I mentioned in point 2. Hope this helps. Regards, Kurian Thayil. On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote: Hi 1. I want to record all calls that land to an agent via a queue using a meaningful name - as of now i name the recorded file on the fly using {CALLERID} variable so that the file gets stored using the caller id iunder /var/spool/asterisk/monitor , now if i want to store it as CALLERIDEXTEN where call landed from queue how can i do this ? 2. I have a CDR issue - when A calls he is put in Queue and say he is answered by Agent B ..Agent B transfers the Call to agent C as it is to Agent C whom A wants to talk..when the call gets d/c the CDR for that call shows the destination field as B whereas it shd be C...how do i take care of this ...in my call center agents are paid on the basis of talk time on inbound calls - this way an agent who just transfers calls is at merry !! 3. Are their any GPL based queue reporting software - hows the asterisk queue statistics program from asteriskguru.com has anyone tried it ? Thanks Sriram ___ -- 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 -- Kurian Mathew Thayil. (GPG KeyID: E232394F) signature.asc Description: This is a digitally signed message part ___ -- 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
Re: [asterisk-users] Queues in memory after startup
Gabriel Ortiz Lour wrote: Hi all, After * starts the command queue show would not show any of the realtime queues, but just the ones that are in the queues.conf file. In this state de AMI would not send any QueueMemberStatus for that queues until a call is received by that realtime queue. Anyone knows any whay to load this information in *'s memory without the need of the queue receiving a call? Thanks, Gabriel Ortiz There is a bit of a hack you can use. If you instead use the command queue show some_specific_queue_name then Asterisk will load from realtime. Then type queue show again and you'll see all the queues. I'm not sure why it was written this way. If you use any 1.6 version of Asterisk, you will find that it does not behave this way. queue show will always show all queues. Mark Michelson ___ -- 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
Re: [asterisk-users] Queues in memory after startup
Mark Michelson escribió: Gabriel Ortiz Lour wrote: Hi all, After * starts the command queue show would not show any of the realtime queues, but just the ones that are in the queues.conf file. In this state de AMI would not send any QueueMemberStatus for that queues until a call is received by that realtime queue. Anyone knows any whay to load this information in *'s memory without the need of the queue receiving a call? Thanks, Gabriel Ortiz There is a bit of a hack you can use. If you instead use the command queue show some_specific_queue_name then Asterisk will load from realtime. Then type queue show again and you'll see all the queues. I'm not sure why it was written this way. If you use any 1.6 version of Asterisk, you will find that it does not behave this way. queue show will always show all queues. Mark Michelson I was noticing the same behavior with 1.4 realtime queues. I thought this was the standard, but it's good to know that 1.6 behaves well, no matter if the queue is static from the queues.conf file or defined in realtime way. Some of this little improvements makes 1.6 a bit more attractive to try for us (1.4 users). Moreover, some of the changes made to trunk to improve IAX2 performance, and some other big changes are just awesome. Thanks. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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
Re: [asterisk-users] Queues Announce help request.
The most popular answer I've seen here is to replace the regular music with a streamed audio feed which can be anything you have access to. I'd try and give you details, but they wouldn't be correct. This information is pretty easy to locate in the digium site, viop-info.org or google. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Friday, March 20, 2009 12:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Queues Announce help request. I am trying to get a queue to do more than just play music and hold calls. Specifically, making some comforting voice announcements would be nice. Below is the queues.conf file relevant portions. Member phone number is munged to protect the guilty. We shouldn't need the announcement source info, but I have been trying everything. The problem is with the member busy, we get no voice announcements. (For test purposes is being on hold busy? We have also just laid the phone on the desk.) We will settle for expected hold time, Thank you announcements, Position in queue, or Dow Jones 30 Industrials news. :-) Anyone have a tip? Cary Fitch Here are some relevant errors from console. [Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12406 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up. -- SIP/3617001000-009a3930 is circuit-busy -- Nobody picked up in 6 ms -- Exiting on time-out cycle - [Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12949 handle_response: Remote host can't match request CANCEL to call '495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up. -- Stopped music on hold on SIP/3617001401-fc0359f0 [usa-queue] queue-youarenext = queue-youarenext queue-thereare = there-thereare queue-callswaiting = queue-callswaiting queue-holdtime = queue-holdtime queue-thankyou = queue-thankyou music = default maxlen = 0 strategy = ringall context = leave-message periodic-announce = thank-you-message periodic-announce-frequency = 30 announce-frequency = 30 announce-holdtime = yes announce-round-seconds = 10 joinempty = no leavewhenempty = yes retry = 30 timeout = 300 member = SIP/3617001402 ___ -- 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 ___ -- 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
Re: [asterisk-users] Queues Announce help request.
- Cary Fitch ca...@usawide.net wrote: I am trying to get a queue to do more than just play music and hold calls. Specifically, making some comforting voice announcements would be nice. You may want to take the quotes off of the filenames in your queues.conf config file... they're not needed, and could very well be causing your problems. -- Jared Smith Training Manager Digium, Inc. ___ -- 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
Re: [asterisk-users] Queues Announce help request.
On 3/20/09, Cary Fitch ca...@usawide.net wrote: I am trying to get a queue to do more than just play music and hold calls. Specifically, making some comforting voice announcements would be nice. Below is the queues.conf file relevant portions. Member phone number is munged to protect the guilty. We shouldn't need the announcement source info, but I have been trying everything. The problem is with the member busy, we get no voice announcements. (For test purposes is being on hold busy? We have also just laid the phone on the desk.) We will settle for expected hold time, Thank you announcements, Position in queue, or Dow Jones 30 Industrials news. :-) Anyone have a tip? Cary Fitch I just thought I'd mention that ViciDial has the ability to play a periodic announcement on inbound queue calls, as well as music on hold, place in line, estimated hold time and lots of other inbound-only features. ViciDial does not use Asterisk Queues so the way we got it working probably wouldn't help you much, but I just wanted to mention it's functionality and that it is open source if you wanted to give it a try. Thanks, MATT--- ___ -- 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
Re: [asterisk-users] * Queues with legacy pbx extensions ?
hi we ALWAYS use sip phone IP*-E1-pstn or sip phone-IP-*-E1-legasypbx-?- pstn we use the digiums cards whit echo canceller and we havent any echo problem. more than 20 or 30 installations. whit almost every provider in the country. dont be SO scared. David 2009/1/26 Sriram d_r_sri...@hotmail.com Hello Everybody I am using Trixbox 2.4 (with TE420P PRI lines) .. my setup is like Calls --Asterisk--legacy pbx---analog extensions(agents). Whenver a call comes in , asterisk dials the ACD number of the legacy pbx which in turn decides to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span and 5432 is the ACD number of legacy pbx under which agents like 102,103,104 are present. Now my question is (might be very silly) : 1. I dont have a queue stats software for my legacy pbx, the agents stil logon and logout on their legacy pbx..I was wondering if its possible that all these 102,103,104 etc login to a queue which resides on asterisk ? Can they login and logout of the asterisk queue..I am asking this since there are very nice asterisk queue reporting utilities out there for free..and i wud like to use them ...this will also help me using screen pop-ups for my agents...i know SIP extensions can simplify my setup but echo problems scare me also i dont want to throw my legacy pbx on which i invested heavily.. Can anyone throw some pointers ? Thanks in advance Sriram ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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
Re: [asterisk-users] Queues, SIP channel and In Use
Benoit wrote: Hi, I'm a little surprised, up until 1.4.22 my agents where using an IAX channel to ZoIPer Softphone, however since after the upgrade to .22 we experienced a problem with hangup failure between zoiper and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i made them switch to SIP instead. Weird thing is that the 'Not In Use' warning message keep showing (WARNING[1863]: app_queue.c:3136 try_calling: The device state of this queue member, Agent/136, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings.) However, when i look at the queue a few sec after the Agent is marked 'in use' which wasn't the case with IAX iirc Agent are defined using a Local channel, but we used the '/n' flag to passthru the status: Agent/136 (Local/1...@queues/n) .. As for the SIP peer definition i have the limitonpeer=yes in the general section and all peers are templated based on this: [poste](!) type=friend host=dynamic qualify=yes call-limit=6 Is their something more in can do to avoid the warning ? I believe the use of the Local channel is what is causing the warning to appear. The problem is that the device state is not updated until after app_queue checks to see if it should display that warning. This has been brought up before, but since it doesn't actually adversely affect the operation of the queue, not much has been done. The worst you have to worry about is that warning. If you find the warning to be annoying, the best I can offer you is to either not log warnings (a bad idea, imho) or just remove that line of code from the source. Mark Michelson ___ -- 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
Re: [asterisk-users] Queues, SIP channel and In Use
Mark Michelson a écrit : Benoit wrote: Hi, I'm a little surprised, up until 1.4.22 my agents where using an IAX channel to ZoIPer Softphone, however since after the upgrade to .22 we experienced a problem with hangup failure between zoiper and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i made them switch to SIP instead. Weird thing is that the 'Not In Use' warning message keep showing (WARNING[1863]: app_queue.c:3136 try_calling: The device state of this queue member, Agent/136, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings.) However, when i look at the queue a few sec after the Agent is marked 'in use' which wasn't the case with IAX iirc Agent are defined using a Local channel, but we used the '/n' flag to passthru the status: Agent/136 (Local/1...@queues/n) .. As for the SIP peer definition i have the limitonpeer=yes in the general section and all peers are templated based on this: [poste](!) type=friend host=dynamic qualify=yes call-limit=6 Is their something more in can do to avoid the warning ? I believe the use of the Local channel is what is causing the warning to appear. The problem is that the device state is not updated until after app_queue checks to see if it should display that warning. This has been brought up before, but since it doesn't actually adversely affect the operation of the queue, not much has been done. The worst you have to worry about is that warning. If you find the warning to be annoying, the best I can offer you is to either not log warnings (a bad idea, imho) or just remove that line of code from the source. Mark Michelson No it's fine for me, i was just wondering if i missed something thanks ___ -- 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
Re: [asterisk-users] queues and MEMBERINTERFACE for AGI script
Thomas Winter wrote: Hi, iam using and queue and starting an AGI script after caller connected to agent. How to find out in the script the connected agent, MEMBERINTERFACE seemed to be not work, either as variable in the queue command and also not in the AGI script. How to found out which agent is connected to calling channel? I try to avoid to using LOCAL channels, because I like the function ringinuse. regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thomas, In queues.conf you need to set the variable setinterfacevar=yes You'd then pass the AGI to the Queue application with something like this: exten = some_extension,n,Queue(somequeue,some.agi) Then within the AGI you'd retrieve the variable like this: my $memberinterface = $AGI-get_variable('MEMBERINTERFACE'); # Perl example. Hope this helps. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues, monitor-join=yes, and volume
Thanks. If I find out some settings for soxmix, do you maybe know where can I change Asterisk settings for soxmix (parameters)? Regards, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg Sent: Tuesday, May 13, 2008 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues, monitor-join=yes, and volume On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote: Is there any way to modify the volume (either lower the volume of the clients, or increase the volume of the agents) while doing the join of the -in and -out files into one recording? Uh-huh. Read the documentation for soxmix. ___ -- 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 ___ -- 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
Re: [asterisk-users] Queues, monitor-join=yes, and volume
You can call the sox binary directly from your dialplan, or any other binary that fits your needs. If you post your dialplan where you're doing the recording, we can give input about where to put the calls to sox. On Wed, May 14, 2008 at 4:16 AM, Asterisk [EMAIL PROTECTED] wrote: Thanks. If I find out some settings for soxmix, do you maybe know where can I change Asterisk settings for soxmix (parameters)? Regards, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg Sent: Tuesday, May 13, 2008 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues, monitor-join=yes, and volume On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote: Is there any way to modify the volume (either lower the volume of the clients, or increase the volume of the agents) while doing the join of the -in and -out files into one recording? Uh-huh. Read the documentation for soxmix. ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] Queues, monitor-join=yes, and volume
On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote: Is there any way to modify the volume (either lower the volume of the clients, or increase the volume of the agents) while doing the join of the -in and -out files into one recording? Uh-huh. Read the documentation for soxmix. ___ -- 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
Re: [asterisk-users] Queues +Exiting
-Ursprüngliche Nachricht- Von: Rob Schall [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 9. April 2008 15:50 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Queues +Exiting I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message thanks for holding. press # to leave a message or stay on the line to continue holding. I set up the context in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my confs: queues.conf [custserv] music=default strategy=ringall ;timeout=10 retry=20 wrapuptime=0 maxlen=0 context = queue-out periodic-announce=cont_holding periodic-announce-frequency=15 ;announce-frequency=15 ;announce-holdtime=yes member = SIP/2001 member = SIP/2002 member = SIP/1004 extensions.conf [queue-out] exten = s,1,Voicemail(u${vmbox}) exten = s,2,Hangup Perhaps it's the s extension, did you try with exten = 1,1,Voicemail(u${vmbox}) exten = 1,2,Hangup Regards, Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues +Exiting
That fixed it. I always thought the s would be the fall back from all extensions that didn't match. I guess that doesn't work in this case. Thanks! Rob Guido Hecken wrote: -Ursprüngliche Nachricht- Von: Rob Schall [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 9. April 2008 15:50 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Queues +Exiting I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message thanks for holding. press # to leave a message or stay on the line to continue holding. I set up the context in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my confs: queues.conf [custserv] music=default strategy=ringall ;timeout=10 retry=20 wrapuptime=0 maxlen=0 context = queue-out periodic-announce=cont_holding periodic-announce-frequency=15 ;announce-frequency=15 ;announce-holdtime=yes member = SIP/2001 member = SIP/2002 member = SIP/1004 extensions.conf [queue-out] exten = s,1,Voicemail(u${vmbox}) exten = s,2,Hangup Perhaps it's the s extension, did you try with exten = 1,1,Voicemail(u${vmbox}) exten = 1,2,Hangup Regards, Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ -- 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 ___ -- 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
Re: [asterisk-users] queues without 302 redirects?
On Wed, Oct 31, 2007 at 06:11:47PM +0100, Louis-David Mitterrand wrote: Hi, Using 1.4.13 is it possible to ignore 302 redirects from sip devices belonging to a queue? For a queue that rings the whole office it doesn't seem very useful to obey a redirect programmed on a phone. It seems this was the default behaviour in 1.2. For the record and google the answer is the 'i' option in Queue(). Thanks again to Strom_M on #asterisk! god I love IRC... ___ --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
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
Tim Groeneveld wrote: On Tuesday 21 August 2007 12:32:12 am Mark Michelson wrote: When users call 510 then, it actually does ring everyone who has called 511. The problem is when the operator comes to pick up the call. The operator hears nothing, and the user still hears the Music on Hold. Not only that, but after about 5 seconds, the operators call gets dropped. Is there anything that I am doing wrong? Remove the Answer() before the call to Queue(). See if that corrects the problem. No, that did not help at all. Maybe I should use AgentLoginCallback? Thanks a million, Tim Groeneveld AgentCallbackLogin is NOT recommended as it has always been very buggy. In the interest of sanity, I just tried the same setup you have using 1.4 and then using trunk. It worked in 1.4, but I had the same problem as you when trying it with trunk. I'd recommend opening a bug report so that this can be further analyzed. ___ --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 ___ --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
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote: I am running r79979 of Asterisk Trunk, and I am having problems trying to use app_queue.so. I want to use the extension 510 to be a line where users can call technical support. Extensions 511 and 512 are used by the operators to dynamically make themselves a Queue Member or not. So, operators call 511, and they should get added to the Queue as a Queue member. When users call 510 then, it actually does ring everyone who has called 511. The problem is when the operator comes to pick up the call. The operator hears nothing, and the user still hears the Music on Hold. Not only that, but after about 5 seconds, the operators call gets dropped. Is there anything that I am doing wrong? Thanks, Tim here are snipits of my config files: == extensions.conf == [default] exten = 510,1,Answer exten = 510,2,Queue(techsupport,t) exten = 511,2,Set(CALLBACKNUM=${CALLERID(number)}) exten = 511,3,AddQueueMember(techsupport) exten = 511,4,Playback(queue-techsupport-in) exten = 511,5,Hangup == queues.conf == [techsupport] music=default strategy = ringall timeout = 10 retry = 2 maxlen = 0 announce-frequency = 10 announce-holdtime = yes == agents.conf == [general] ackcall=no Can you also provide output of show queues and show channels? Plus the logfile of dial to 511. I'm using QueueAdd after AgentCallbackLogin (trough manager API). Maybe you need to use AgentCallbackLogin first? Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --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
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
On Monday 20 August 2007 8:16:32 pm Atis wrote: On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote: Can you also provide output of show queues and show channels? Plus the logfile of dial to 511. I'm using QueueAdd after AgentCallbackLogin (trough manager API). Maybe you need to use AgentCallbackLogin first? Regards, Atis == Call to 511 == Saved useragent Linksys/SPA3102-3.3.6(GW) for peer 101 == Using TOS bits 0 == Using CoS mark 5 -- Executing [EMAIL PROTECTED]:2] Set(SIP/101-0821ca48, CALLBACKNUM=101) in new stack -- Executing [EMAIL PROTECTED]:3] AddQueueMember(SIP/101-0821ca48, techsupport) in new stack [Aug 20 22:28:26] NOTICE[6238]: app_queue.c:3441 aqm_exec: Added interface 'SIP/101' to queue 'techsupport' -- Executing [EMAIL PROTECTED]:4] Playback(SIP/101-0821ca48, queue-techsupport-in) in new stack -- Executing [EMAIL PROTECTED]:5] Hangup(SIP/101-0821ca48, ) in new stack == Spawn extension (mor, 511, 5) exited non-zero on 'SIP/101-0821ca48' == Outputs == *CLI show channels No such command 'show channels' (type 'help' for help) *CLI show queues No such command 'show queues' (type 'help' for help) *CLI queue show techsupport has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers signature.asc Description: This is a digitally signed message part. ___ --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
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote: On Monday 20 August 2007 8:16:32 pm Atis wrote: On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote: Can you also provide output of show queues and show channels? Plus the logfile of dial to 511. I'm using QueueAdd after AgentCallbackLogin (trough manager API). Maybe you need to use AgentCallbackLogin first? Regards, Atis == Call to 511 == Saved useragent Linksys/SPA3102-3.3.6(GW) for peer 101 == Using TOS bits 0 == Using CoS mark 5 -- Executing [EMAIL PROTECTED]:2] Set(SIP/101-0821ca48, CALLBACKNUM=101) in new stack -- Executing [EMAIL PROTECTED]:3] AddQueueMember(SIP/101-0821ca48, techsupport) in new stack [Aug 20 22:28:26] NOTICE[6238]: app_queue.c:3441 aqm_exec: Added interface 'SIP/101' to queue 'techsupport' -- Executing [EMAIL PROTECTED]:4] Playback(SIP/101-0821ca48, queue-techsupport-in) in new stack -- Executing [EMAIL PROTECTED]:5] Hangup(SIP/101-0821ca48, ) in new stack == Spawn extension (mor, 511, 5) exited non-zero on 'SIP/101-0821ca48' == Outputs == *CLI show channels No such command 'show channels' (type 'help' for help) *CLI show queues No such command 'show queues' (type 'help' for help) *CLI queue show techsupport has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers Ok, i just noticed that you are running trunk. Probably you should write to asterisk-dev then. Seems that agent get's added correctly. So you could try to view (and post us) log of call to queue, maybe it says something. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --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
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
Tim Groeneveld wrote: I am running r79979 of Asterisk Trunk, and I am having problems trying to use app_queue.so. I want to use the extension 510 to be a line where users can call technical support. Extensions 511 and 512 are used by the operators to dynamically make themselves a Queue Member or not. So, operators call 511, and they should get added to the Queue as a Queue member. When users call 510 then, it actually does ring everyone who has called 511. The problem is when the operator comes to pick up the call. The operator hears nothing, and the user still hears the Music on Hold. Not only that, but after about 5 seconds, the operators call gets dropped. Is there anything that I am doing wrong? Thanks, Tim here are snipits of my config files: == extensions.conf == [default] exten = 510,1,Answer exten = 510,2,Queue(techsupport,t) exten = 511,2,Set(CALLBACKNUM=${CALLERID(number)}) exten = 511,3,AddQueueMember(techsupport) exten = 511,4,Playback(queue-techsupport-in) exten = 511,5,Hangup == queues.conf == [techsupport] music=default strategy = ringall timeout = 10 retry = 2 maxlen = 0 announce-frequency = 10 announce-holdtime = yes == agents.conf == [general] ackcall=no ___ --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 Remove the Answer() before the call to Queue(). See if that corrects the problem. ___ --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
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
On Tuesday 21 August 2007 12:32:12 am Mark Michelson wrote: When users call 510 then, it actually does ring everyone who has called 511. The problem is when the operator comes to pick up the call. The operator hears nothing, and the user still hears the Music on Hold. Not only that, but after about 5 seconds, the operators call gets dropped. Is there anything that I am doing wrong? Remove the Answer() before the call to Queue(). See if that corrects the problem. No, that did not help at all. Maybe I should use AgentLoginCallback? Thanks a million, Tim Groeneveld signature.asc Description: This is a digitally signed message part. ___ --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
Re: [asterisk-users] Queues with logged in agents that are not reachable
On 7/30/07, voiplist [EMAIL PROTECTED] wrote: I noticed that if I have an agent logged in using AgentCallBackLogin and that agent is unreachable for some reason (SIP phone unplugged) calls to him/her will completely yack. For example: 1-Agent 500 is the only one logged into queue number 1. 2-A call comes into queue number 1 3-The call is pushed to agent 500 at extension 21 which is unreachable because the ethernet cable is unplugged to extension 21's handset. 4-The caller gets hungup on entirely instead of the call going to another agent or leaving the caller in the queue I don't expect this to happen but I want to be sure all bases are covered on light days during shift changes etc. This is either a problem with your dialplan or your queue configuration. If you always want your callers to enqueue regardless of agent status, make sure that joinempty=yes and leavewhenempty=no in queues.conf for that queue. You may also want to add a exten = whatever,n,NoOp(${QUEUESTATUS}) right after your call to Queue() to see why the calls are leaving the dialplan. I suspect that you've got one or the other of those settings not set properly, so when there are no available agents, your calls exit the Queue() application with $QUEUESTATUS set to JOINEMPTY or LEAVEEMPTY, but you don't have anything in your dialplan following Queue(), so they run off the end of the extension and * hangs up on them. Note that there is a problem with 1.4.9 that breaks joinempty=yes/leavewhenempty=no - there's a patch offered to my bug report ( http://bugs.digium.com/view.php?id=10320), but due to other strange instability observed in 1.4.8 and 1.4.9, I'm back on 1.47.1, so I haven't tested it yet. -- j. ___ --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
Re: [asterisk-users] Queues strategy leastrecent
Yes it seems... but not just with that strategy. Now I'm working on a queue witch I need to go to agent 1004, then 1018 and then 1010 so I wrote this in queue.conf: [general] persistentmembers = yes autofill = yes monitor-type = MixMonitor [FAC] musicclass = default strategy = roundrobin servicelevel = 60 timeout = 10 retry = 10 weight=0 wrapuptime = 0 maxlen = 0 periodic-announce-frequency=60 periodic-announce = Track1 joinempty = no leavewhenempty = yes ringinuse = no timeoutrestart = yes member = Agent/1004 member = Agent/1018 member = Agent/1010 ;member = Agent/1011 But what happens is it behaves has rrmemory since it remembers the last agent it rang, and instead if using the order I specified, it goes sorted (1004, then _1010_, and only then 1018). Asterisk is a great piece of software, a lot more stable then what it was 2y ago, but there are some basic bugs that I can't understand why they still happen... I'm one of the very few persons that use it for queues? Thank you for your comments and help. Cheers, Marco. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jakub Glazik Sent: sexta-feira, 27 de Julho de 2007 12:55 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Queues strategy leastrecent Dnia 2007-07-27, o godz. 11:09:37 Marco Campos [EMAIL PROTECTED] napisał(a): I think this strategy should work like this: a) Make a list of available agents and their idle time (time since last call) and update it each time a call gets answered or ends b) When a call comes in, try to pass it to the first agent on that list c) If the timeout (timeout = 10) is reached it should try the next (second in this case) agent on that list d) .and so on. But what I see is that it is always trying to pass the call to the same agent, since it thinks that that agent is idle longer than any other. Are any of your seen a similar problem, and eventually know how to fix this? I think the same and I have the same problem with leastrecent (1.4.7.1). I believe that current behaviour is buggy and not useful. -- .: Jakub Głazik, .: email jabber: zytekatnuxi.pl ___ --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 ___ --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
Re: [asterisk-users] Queues strategy leastrecent
Dnia 2007-07-27, o godz. 11:09:37 Marco Campos [EMAIL PROTECTED] napisał(a): I think this strategy should work like this: a) Make a list of available agents and their idle time (time since last call) and update it each time a call gets answered or ends b) When a call comes in, try to pass it to the first agent on that list c) If the timeout (timeout = 10) is reached it should try the next (second in this case) agent on that list d) .and so on. But what I see is that it is always trying to pass the call to the same agent, since it thinks that that agent is idle longer than any other. Are any of your seen a similar problem, and eventually know how to fix this? I think the same and I have the same problem with leastrecent (1.4.7.1). I believe that current behaviour is buggy and not useful. -- .: Jakub Głazik, .: email jabber: zytekatnuxi.pl ___ --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
Re: [asterisk-users] Queues monitoring software
You might want to have a look at QueueMetrics: http://queuemetrics.loway.it/ I am not sure if it supports all features you are looking for but it should be a good start. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ --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
Re: [asterisk-users] Queues monitoring software
On 7/12/07, Stefan Reuter [EMAIL PROTECTED] wrote: You might want to have a look at QueueMetrics: http://queuemetrics.loway.it/ I am not sure if it supports all features you are looking for but it should be a good start. QueueMetrics is working well for me in a 75 seat call center, but it won't do everything you ask for. - It doesn't have an interface to let you manage your Asterisk Configuration (though it can pull info from your Asterisk configuration apparantly - I opted to set things up manually) - it has a realtime view that uses the Manager API, but this is to show queue status, not other aspects of your * operation (like Zap status) Other than that, it's relatively easy to set up. A tad annoying if your * and QM live on different boxes, and I regularly lament the inability to modify the output much (all of the calculations and most of the output are tied up in compiled Java classes), but it took care of 80% or so of my reporting needs in a few days when the alternative was to roll my own reporting system. A combination of QM plus some homegrown stuff, or maybe QM plus one of the other Asterisk Management web portals might do the trick for you. -- j. ___ --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
Re: [asterisk-users] Queues monitoring software - OrderlyStats now FREE
Hello Voipcrazy, It's funny you should mention that - we've just released (as in today) a FREE version of our OrderlyStats service for call centre and queue monitoring and management. OrderlyStats features realtime (synchronous/message-based) display of all queue, agent and caller events so you can see what's happening in your call centre as it happens. The control panel feature allows you to reassign agents and penalties on the fly. Our Agent Bar tool also helps agents log in and out of their queues, and enter Pause for wrap-up at the touch of a button, in addition to displaying realtime wallboard-style queue information. Furthermore, we automatically produce a wealth of historical statistics, allowing you to track caller trends and analyse staffing requirements across your queues. You can sign up now at http://www.orderlyq.com/freestats.html . It would certainly be worth your while to take a look, especially if you're considering spending money on QueueMetrics. Thanks for reading, Matt. [EMAIL PROTECTED] wrote Hello all, A client of us, needs a queue monitoring system. In realtime he needs to now the PRI status, the agents logged in and logged out, the number of received calls by agent, ,etc. I am not a call center specialist and i want to find a call center software to offer to my client that fits his needs. I need a monitoring solution for incomming and outgoing calls and a queue management interface to create and/or modify queues or agents. Any one of you could has instalesd this kind of software? Which one? Which one could you recomend me? Thanks in advance. Voipcrazy. ___ --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
Re: [asterisk-users] Queues with announce
Ciao Andrea, Hello *, do queues allow me to set an announce like the A() option of the Dial() cmd? The announce that I've found is a message that is heard by the caller. I'd like to send a message to the member of the queue that picks up the call. In order to help people that find this message through a search engine: the announce = XX option in queues.conf allows me to solve my problem. I can set it dynamically through the fourth parameter of the Queue() application. -- Dott. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. ___ --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] Queues: Play a list of sound file n round-robin at a specific interval
Why don't you make up the MOH in order to play your sound files, as you need? l. On Mon, 07 May 2007 16:29:28 +0200, Andre Courchesne - Consultant [EMAIL PROTECTED] wrote: Hi, Anyone knows if there is a way to play a list of sound file in a round robin mode (at specific interval) while someone in waiting in moh in a queue? Ok, you enter a queue and wait listening to moh, every X minutes a sound file is played from a list of sound files to be played. If that possible and if so how? Thanks for any pointers. Andre -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] Queues: Play a list of sound file n round-robin at a specific interval
It is indeed a way, but not very flexible for my customer that want to change the sound files weekly or even daily. I found this patch that might do the trick: http://bugs.digium.com/view.php?id=6681 Need to test it. So far I could not find a way to do so without patching asterisk. Lenz wrote: Why don't you make up the MOH in order to play your sound files, as you need? l. On Mon, 07 May 2007 16:29:28 +0200, Andre Courchesne - Consultant [EMAIL PROTECTED] wrote: Hi, Anyone knows if there is a way to play a list of sound file in a round robin mode (at specific interval) while someone in waiting in moh in a queue? Ok, you enter a queue and wait listening to moh, every X minutes a sound file is played from a list of sound files to be played. If that possible and if so how? Thanks for any pointers. Andre --Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] queues
You can have the agent login once and newer log out. You can certainly set up your asterisk box to persit the login over the reload and the restart. persistentagents=yes Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group (FIG) may transmit information that is confidential and privileged information of Featherstone Informatics Group (FIG). Unless you are the intended addressee, you may not use, copy or disclose to anyone this communication or any information transmitted by this communication. If you have received such communication in error, please advise the sender by e-mail and/or telephone and destroy this communication immediately. This communication and any information transmitted by this communication may also be considered protected health information as defined under the Health Insurance Portability and Accountability Act and its related regulations (a.k.a., HIPAA) or any other similar state law. Please exercise due care and ensure that you comply with its contractual and legal obligations. - Original Message - From: Voip Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 18, 2007 5:23:18 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] queues ___ --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] queues
Is there anyway to setup a queue with only one agent (device) which is always logged in. So when a call hits that queue the device will ring (if not already on a call) or will be put in the queue if the call is already in place? Sure, in queues.conf you can add many type of members (not just agents) like SIP or Local channels. So you don't need to use AgentLogin/CallBackLogin ej. [recepcion] musicclass = default monitor-format = wav49 strategy = ringall timeout = 15 retry = 2 autopause = no maxlen = 3 context = voicemail setinterfacevar = yes announce-frequency = 15 periodic-announce-frequency = 0 announce-holdtime = yes announce-round-seconds = 10 joinempty = strict leavewhenempty = strict eventwhencalled = yes eventmemberstatus = yes ringinuse = yes timeoutrestart = no member = SIP/9001,1 member = SIP/9005,2 -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --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] Queues
Yes. On Sat, 17 Mar 2007, Steve Kennedy wrote: A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Queues
On Sat, Mar 17, 2007 at 11:44:52AM -0700, Steve Edwards wrote: Yes. to which bit? auto-agent (as per resource) or voicemail to an agent? Steve On Sat, 17 Mar 2007, Steve Kennedy wrote: A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Queues
If you make a SIP device a queue member, that member will be rung so long as the device state of the SIP interface shows as not in use. With regard to voicemail, are you trying to get a queue call answered by voicemail or is that not your intent? On 3/17/07, Steve Kennedy [EMAIL PROTECTED] wrote: On Sat, Mar 17, 2007 at 11:44:52AM -0700, Steve Edwards wrote: Yes. to which bit? auto-agent (as per resource) or voicemail to an agent? Steve On Sat, 17 Mar 2007, Steve Kennedy wrote: A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Queues do not accept calls if all agent are busy?
On 2/15/07, Angel Heart [EMAIL PROTECTED] wrote: cud any one help me figuring out the problem... When the agent in a queue is engaged, it cannot accept anymore calls, below is the script; Angel, Check your queues.conf, specifically the joinempty parameter. See below the relevant part in the queues.conf sample file: ... ; This setting controls whether callers can join a queue with no members. There ; are three choices: ; ; yes- callers can join a queue with no members or only unavailable members ; no - callers cannot join a queue with no members ; strict - callers cannot join a queue with no members or only unavailable ; members ; ; joinempty = yes ... Cheers, -- Ex Vito ___ --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] Queues do not accept calls if all agent are busy?
Hi Ex vito, Thank you for your response below is my current config. I defined it into my queues_addintional.conf where the definitions of queues defined. Do I need to defined it in the general portion of queues.conf? But anyway, there's no harm for trying. I am using Asterisk 1.2.13 svn rev 47264 with AudioCodes FXO/FXS Gateway. queues.conf [general] ; ; Global settings for call queues ; (none exist currently) ; ; Note that a timeout to fail out of a queue may be passed as part of application call ; from extensions.conf: ; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout]) ; example: Queue(dave|t|||45) [default] ; ; Default settings for queues (currently unused) ; #include queues_custom.conf #include queues_additional.conf queues_additional.conf [7001] wrapuptime=5 timeout=20 strategy=leastrecent retry=5 queue-youarenext= queue-thereare= queue-thankyou=custom/client-in-queue queue-callswaiting= music=default monitor-join=yes monitor-format= maxlen=0 leavewhenempty=no joinempty=Yes context=ivr-6 announce-holdtime=yes announce-frequency=30 Kindest regards. Angel Ex Vitorino [EMAIL PROTECTED] wrote: On 2/15/07, Angel Heart wrote: cud any one help me figuring out the problem... When the agent in a queue is engaged, it cannot accept anymore calls, below is the script; Angel, Check your queues.conf, specifically the joinempty parameter. See below the relevant part in the queues.conf sample file: ... ; This setting controls whether callers can join a queue with no members. There ; are three choices: ; ; yes- callers can join a queue with no members or only unavailable members ; no - callers cannot join a queue with no members ; strict - callers cannot join a queue with no members or only unavailable ; members ; ; joinempty = yes ... Cheers, -- Ex Vito ___ --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 - Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games.___ --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] queues and LOCAL for members
Am Friday 02 February 2007 23:48 schrieb BJ Weschke: On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote: Hi, I have an queue stored in relatime and defined members called through LOCAL/ I found out that if I call the members through the LOCAL think the queue statistics is not updated. Any idea, or isnt possible to call members with LOCAL channel. There's been some efforts to have Local channels as viable queue members. I'm not quite sure that I understand your issue. Can you post some more details possibly in a bug on bugs.digium.com ? Thanks, I found out LOCAL is working. I have been confused because an change of queue-members and an reload has reset the queue statistic. ___ --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