Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
I´ll be upgrading my box this weekend and let you know the consequences. I´m new at the community and it would be good for me to know what was the problem with 1.4.17 Thanks for taking some time for me. Daniel On Wed, Jul 23, 2008 at 10:59 AM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote: > On Wed, Jul 23, 2008 at 10:44:12AM -0500, Tilghman Lesher wrote: > > On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote: > > > On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote: > > > > On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: > > > > > My * version: 1.4.17 > > > > > > > > Please upgrade to 1.4.21.2. > > > > > > Just a suggestion, Tilghman: it might have been nice to add "because it > > > fixes your specific problem", so that we wouldn't assume "because we > > > don't want to talk to you if you rev is too old". :-) > > > > It probably fixes his specific problem, AND because I don't like > > diagnosing an issue that we've already solved and that he would have > > figured out, if he had bothered to try the latest release. 1.4.21.1 > > should have been fixed, as well, but at that point, I had just spent 3 > > hours working frantically to get two security advisories out the door, > > so that the community wouldn't be vulnerable to two critical issues, > > and suggesting that he try a version that was vulnerable would have > > been bad. > > Oh, sure. > > I'm just sayin... > > It's pretty clear to me that while you're playing in the NFL, he may > not be. > > Cheers, > -- jra > -- > Jay R. Ashworth Baylink > [EMAIL PROTECTED] > Designer The Things I Think RFC > 2100 > Ashworth & Associates http://baylink.pitas.com '87 > e24 > St Petersburg FL USA http://photo.imageinc.us +1 727 647 > 1274 > > Those who cast the vote decide nothing. > Those who count the vote decide everything. > -- (Josef Stalin) > > ___ > -- 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 > ___ -- 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] 3-way calling for IAX channels
We are developing an softphone based on IAX client version 1.2 (my current SIP softphone has many eoors), but it doesn´t have a specific function for Conferencing (3-way calling) or to place the other party on HOLD. I´m trying to do it through the PBX because our softphone´s lack of functions. I´ll be gratefull for further comments. Thanks again, Daniel On Tue, Jul 22, 2008 at 11:49 PM, Noah Miller <[EMAIL PROTECTED]> wrote: > Hi Daniel - > > > There is no way to enable it at the softphone itself? As is the case for > > hardphones like my Polycom. > > A phone can definitely do conference mixing. As you asked about IAX > channels on the asterisk-users list, I assumed you were asking about > how to do this in asterisk. > > My experience with IAX softphones is somewhat limited, but maybe if > you indicate which phone you're using, somebody could provide you with > assistance. > > > - Noah > > > > > Daniel > > On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller <[EMAIL PROTECTED]> > > wrote: > >> > >> Hi Daniel - > >> > >> > How can I made a 3-way conference betwwen IAX channels? > >> > My current version is: 1.4.21.1 > >> > >> Anytime you need a call with more than 2 parties, you need to use some > >> kind of conferencing application. The "default" conference > >> application for asterisk is meetme. You can use meetme with any kind > >> of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() > >> application in extensions.conf, and create your conference rooms in > >> meetme.conf > >> > >> > >> - Noah > >> > >> ___ > >> -- 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 > > > > > > ___ > > -- 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 > > > > ___ > -- 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 > ___ -- 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] 3-way calling for IAX channels
Thanks for answering Noah, There is no way to enable it at the softphone itself? As is the case for hardphones like my Polycom. Daniel On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller <[EMAIL PROTECTED]> wrote: > Hi Daniel - > > > How can I made a 3-way conference betwwen IAX channels? > > My current version is: 1.4.21.1 > > Anytime you need a call with more than 2 parties, you need to use some > kind of conferencing application. The "default" conference > application for asterisk is meetme. You can use meetme with any kind > of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() > application in extensions.conf, and create your conference rooms in > meetme.conf > > > - Noah > > ___ > -- 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 > ___ -- 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
[asterisk-users] Suddenly my Asterisk Box Hanged up all calls
Hi everybody, I hope someone could help me. Today one of my PBXs seemed as it were frozen. When I typed *asterisk -rx "core show channels"; "sip show peers" *or *"queue show" *it showed the last activity before the crash. Finally I realise that something was wrong because there were no activity at CLI and incoming calls to IVR didn´t get through, then I decided to restart asterisk and things were good again. What logs say is: /var/log/boot.log: Jul 22 13:51:49 pbx asterisk: Interruption of asterisk succeeded Jul 22 13:51:49 pbx asterisk: Start of safe_asterisk succeeded /var/log/asterisk/messages.1983: [Jul 22 13:40:36] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies [Jul 22 13:40:36] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies [Jul 22 13:40:36] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies [Jul 22 13:40:36] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies [Jul 22 13:40:37] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies *[Jul 22 13:51:49] NOTICE[3328] loader.c: 1 modules will be loaded.* *[Jul 22 13:51:49] NOTICE[3328] cdr.c: CDR simple logging enabled.* *[Jul 22 13:51:49] NOTICE[3328] loader.c: 157 modules will be loaded.* *[Jul 22 13:51:50] NOTICE[3328] config.c: Registered Config Engine odbc* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Connecting asterisk* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Registered ODBC class 'asterisk' dsn->[asterisk]* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Connecting mysql1* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Registered ODBC class 'mysql1' dsn->[MySQL-asterisk]* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Connecting mysql2* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Registered ODBC class 'mysql2' dsn->[MySQL-asterisk]* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: res_odbc loaded.* *[Jul 22 13:51:50] NOTICE[3328] config.c: Registered Config Engine mysql* *[Jul 22 13:51:50] NOTICE[3328] app_queue.c: Queue members successfully reloaded from database.* *[Jul 22 13:51:50] NOTICE[3328] pbx_ael.c: Starting AEL load process.* *[Jul 22 13:51:50] NOTICE[3328] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'.* *[Jul 22 13:51:50] NOTICE[3328] pbx_ael.c: File /etc/asterisk/extensions.ael not found; AEL declining load* [Jul 22 13:51:50] NOTICE[3437] chan_sip.c: Peer '806' is now Reachable. (61ms / 2000ms) [Jul 22 13:51:50] NOTICE[3437] chan_sip.c: Peer '831' is now Reachable. (62ms / 2000ms) [Jul 22 13:51:50] NOTICE[3437] chan_sip.c: Peer '819' is now Reachable. (30ms / 2000ms) [Jul 22 13:51:50] NOTICE[3437] chan_sip.c: Peer '813' is now Reachable. (30ms / 2000ms) [Jul 22 13:51:51] NOTICE[3437] chan_sip.c: Peer '856' is now Reachable. (26ms / 856ms) First *channel.c *errors **happens when someone use the HOLD and MUTE function in a SIP softphone based on PortSIP Development Kit (we are building an IAX client based softphone to fix these ERRORS and hidden -at logger.conf- WARNINGS). At *Jul 22 13:51:49 *I decided to reboot asterisk *[service asterisk restart] * 4 days ago the system had reboot itself but PBX functions went back automatically, that were no the case today :(. Agents told me they were receiving congestion messages (the most for invalid numbers) in outbound calls just before the crash. My * version: 1.4.17 We have round about 60 users with no more than 30 simultaneous calls at the peak hour (inbound and outbound) My inbound campaing has 10 agents in a simple queue (Local Channels and SIP extensions) My hardware is an IBM iSeries 386 *[Linux pbx_daniel.com 2.6.9-67.0.1.ELsmp #1 SMP i686 i686 i386 GNU/Linux]* I`ll really appreciate every help you can give me. Thanks in advance, Daniel Arohuanca +51 1 3594122 ___ -- 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] features on dial pad
I`m looking for the same for an IAX cliente based softphone I´m trying to build. Did you get some info about it? Thanks in advance, Daniel Arohuanca +51 1 3594122 On Wed, Apr 9, 2008 at 10:29 PM, nhadie ramos <[EMAIL PROTECTED]> wrote: > Hi All, > > If i were to develop a softphone, how can i add call transfer, call on > hold and 3-way conference on it? linksys Ip phone has those built-in button > to xfer, conf, on hold. > and x-lite also has those, how can i have those if i develop my own? > > Thank You > > Regards, > Nhadie > ___ > -- 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 -- 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
[asterisk-users] 3-way calling for IAX channels
How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 ___ -- 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
[asterisk-users] Help for an IAX_Client-based softphone
Hi everyone, I`ve been having several problems with my current softphone and I`m trying to develop an IAX Client based one. Does anyone know how can I get help or useful resources about it? Specially with Conference function and management of incoming call events to launch an AGI at that time. I´ll be very gratefull for any help you have. Daniel Arohuanca Lagos +51 1 3594122 ___ -- 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] CLI> show queues NOT WORKING WELL
I forgot it!, I'm using Asterisk 1.4.19.1 version. On Mon, Jun 30, 2008 at 1:47 PM, Chento Arohuanca <[EMAIL PROTECTED]> wrote: > Hi Atis and friends, > > I think it has direct relation with calls not being delivered to available > agents, as you can see at following lines, I´ve confirmed i*n situ* there > were real available agents: > > [EMAIL PROTECTED] apps]# asterisk -rx "queue show" > QUEUE1 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), > W:1, C:722, A:13, SL:83.7% within 0s >Members: > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has > taken 1 calls (last was 2040 secs ago) > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has > taken 3 calls (last was 1045 secs ago) > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has > taken 3 calls (last was 1207 secs ago) > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has > taken 8 calls (last was 351 secs ago) > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has > taken 3 calls (last was 450 secs ago) >No Callers > > QUEUE2 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4, > C:249, A:1, SL:87.6% within 0s >Members: > *[agents' status are similar to QUEUE5]* >No Callers > > QUEUE3 has 0 calls (max 100) in 'rrmemory' strategy (3s holdtime), W:3, > C:582, A:6, SL:84.7% within 0s >Members: > *[agents' status are similar to QUEUE5]* >No Callers > > QUEUE4 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:3, > C:254, A:9, SL:78.3% within 0s >Members: > *[agents' status are similar to QUEUE5]* >No Callers > > > QUEUE5 has 2 calls (max 100) in 'rrmemory' strategy (7s holdtime), W:1, > C:531, A:6, SL:79.3% within 0s >Members: > * Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has > taken no calls yet* > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has > taken 1 calls (last was 1253 secs ago) > * Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has > taken 9 calls (last was 795 secs ago* > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has > taken 4 calls (last was 251 secs ago) > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has > taken 2 calls (last was 283 secs ago) > * Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has > taken 5 calls (last was 264 secs ago* > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (In use) has > taken 1 calls (last was 15 secs ago) > * Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has > taken no calls yet > * > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has > taken 1 calls (last was 3376 secs ago) > * Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has > taken no calls yet > * > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has > taken no calls yet > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has > taken 3 calls (last was 1162 secs ago) > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (In use) has > taken 2 calls (last was 2046 secs ago) >Callers: > * 1. Zap/76-1 (wait: 0:28, prio: 0) > 2. Zap/63-1 (wait: 0:04, prio: 0)* > > QUEUE6 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4, > C:360, A:0, SL:84.2% within 0s >Members: > *[agents'
Re: [asterisk-users] CLI> show queues NOT WORKING WELL
uot;queue show" |grep wait 1. Zap/76-1 (wait: 0:42, prio: 0) 2. Zap/63-1 (wait: 0:18, prio: 0) 1. Zap/80-1 (wait: 0:07, prio: 0) Looking at the related logs I see: 1214848558|1214848504.226198|QUEUE5|NONE|ENTERQUEUE||1XYZABCD 1214848573|1214848504.226198|QUEUE5|Local/[EMAIL PROTECTED]/n|RINGNOANSWER|0 . . 1214848712|1214848504.226198|QUEUE5|Local/[EMAIL PROTECTED]/n|RINGNOANSWER|0 1214848712|1214848504.226198|QUEUE5|Local/[EMAIL PROTECTED]/n|RINGNOANSWER|0 1214848713|1214848504.226198|QUEUE5|NONE|ABANDON|1|2|155 The call has ringed: 604 - 92 times, 605 - 88; 606 - 88, 609 - 88, 610 - 88, 611 - 9, 614 - 88, 622 - 88, 623 - 9; 624 - 88; 627 - 27; 628 - 88; 630 - 64 times. I will very appreciate any comment you have, it would be possible that agent (or device) status are not being updated correctly? Thanks in advance, Elder Arohuanca, Cell Phone: +51 1989541408 On Thu, Jun 19, 2008 at 5:15 PM, Atis Lezdins <[EMAIL PROTECTED]> wrote: > On Thu, Jun 19, 2008 at 10:06 PM, Chento Arohuanca <[EMAIL PROTECTED]> > wrote: > > Just about 30 minutes that I can´t get real information from my Asterisk > > box. All agents seem to be available but is not true: > > > > > > QUEUE_01 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4, > > C:0, A:0, SL:0.0% within 0s > >Members: > > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) > has > > taken no calls yet > > Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) > has > > taken no calls yet > > > > > > [EMAIL PROTECTED] asterisk]# asterisk -rx "core show channels" > > Channel Location State Application(Data) > > SIP/641-08cef808 (None) Up Bridged > > Call(Local/[EMAIL PROTECTED] > > Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/641|120|rtT) > > Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged > > Call(Zap/65-1) > > Zap/65-1 [EMAIL PROTECTED]:1 Up > Queue(QUEUE_01|tT|||1800) > > Zap/64-1 [EMAIL PROTECTED]: Up (None) > > SIP/625-09766788 (None) Up Bridged > > Call(Local/[EMAIL PROTECTED] > > Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/625|120|rtT) > > Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged > > Call(Zap/66-1) > > Zap/66-1 [EMAIL PROTECTED]:1 Up > Queue(QUEUE_02|tT|||1800) > > SIP/620-09358088 (None) Up Bridged > > Call(Local/[EMAIL PROTECTED] > > Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/620|120|rtT) > > Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged > > Call(Zap/63-1) > > Zap/63-1 [EMAIL PROTECTED]:1 Up > Queue(QUEUE_01|tT|||1800) > > Zap/94-1 (None) Up Bridged > > Call(SIP/623-b2b1d070) > > SIP/623-b2b1d070 [EMAIL PROTECTED]:3 Up > > Dial(Zap/g3/2714269||tTrRS) > > SIP/615-08a892c0 (None) Up Bridged > > Call(Local/[EMAIL PROTECTED] > > > > Please help me with this issue! > > > > Local channels don't support state information in Asterisk 1.4. For > that you either need to use 1.6 or backport of state_interface for > 1.4. Then you have to set call-lim
[asterisk-users] CLI> show queues NOT WORKING WELL
Just about 30 minutes that I can´t get real information from my Asterisk box. All agents seem to be available but is not true: QUEUE_01 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet [EMAIL PROTECTED] asterisk]# asterisk -rx "core show channels" Channel Location State Application(Data) SIP/641-08cef808 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/641|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/65-1) Zap/65-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/64-1 [EMAIL PROTECTED]: Up (None) SIP/625-09766788 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/625|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/66-1) Zap/66-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_02|tT|||1800) SIP/620-09358088 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/620|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/63-1) Zap/63-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/94-1 (None) Up Bridged Call(SIP/623-b2b1d070) SIP/623-b2b1d070 [EMAIL PROTECTED]:3 Up Dial(Zap/g3/2714269||tTrRS) SIP/615-08a892c0 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Please help me with this issue! Thanks, ___ -- 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
