[Asterisk-Users] PickUpChan and Intercept
Hello everyone,
I have been asked for "directed pickup" and saw
that both "PickupChan" from bristuff and "Intercept" applications
do the dirty work.
I have tried both on asterisk-1.0.9 (
BRIstuffed-0.2.0-RC8o ) but I always got an error when trying to pick the
ringing call.
the debug says:
SIP/marco-73a0 is
ringing -- SIP/marco-73a0 is ringing
-- SIP/marco-73a0 is ringing -- SIP/marco-73a0 is
ringing -- Executing PickupChan("SIP/edevena-a940",
"SIP/marco") in new stackOct 13 12:35:48 WARNING[1675]: channel.c:513
ast_channel_walk_locked: Avoided initial deadlock for 'SIP/edevena-a940',
10 retries! -- No channel found
SIP/marco. -- SIP/marco-73a0 is ringing
the problems seems in ast_channel_walk_locked.
Will someone help on this matter?
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[Asterisk-Users] qozap(!) problem
As I said before, I can not get help from junghanns, so I ask the list. I installed * version 1.0.7 bristuffed latest version and this solves the music on hold problem. But this introduces a new problem that I did not have before. Every 1 second pops up the message: May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 8 z1 64 z2 40 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 26 z2 0 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 1 z1 32 z2 15 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 63 z2 42 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 2 z1 34 z2 16 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 33 z2 7 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 11 z1 104 z2 77 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 4 z1 77 z2 57 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 3 z1 61 z2 42 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 31 z2 5 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 10 z2 112 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 100 z2 74 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 11 z1 62 z2 35 there are no IRQ conflicts ( checked with lspci -v) and everything works. What does this message mean? Thanks for any help Eugenio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] qozap(!) problem
Which version of * and bristuff did you install, I had bristuff-0.2.0-RC8a and now I am trying bristuff-0.2.0-RC8c - Original Message - From: Altus Snyman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 09, 2005 3:15 PM Subject: Re: [Asterisk-Users] qozap(!) problem Ya well let me know when u solved this We have the same thing Do you have any other cards in with it We have a diguim fxs/fxo card in so maybe its a error with working together Anyway Let me know when you get a fix for it because no one seems to know(or check their /var/log/messages) This lets my asterisk hang at lest one daily and I needed to schedule regular reboots On Mon, 2005-05-09 at 14:45, Eugenio De Vena wrote: As I said before, I can not get help from junghanns, so I ask the list. I installed * version 1.0.7 bristuffed latest version and this solves the music on hold problem. But this introduces a new problem that I did not have before. Every 1 second pops up the message: May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 8 z1 64 z2 40 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 26 z2 0 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 1 z1 32 z2 15 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 63 z2 42 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 2 z1 34 z2 16 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 33 z2 7 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 11 z1 104 z2 77 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 4 z1 77 z2 57 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 3 z1 61 z2 42 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 31 z2 5 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 10 z2 112 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 100 z2 74 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 11 z1 62 z2 35 there are no IRQ conflicts ( checked with lspci -v) and everything works. What does this message mean? Thanks for any help Eugenio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] music on hold on R key not working.
Oh boy I am getting crazy... I installed an asterisk with J4BRI ( 3 BRI point to point ) , Snom phones and everything works fine. Where's the problem? Well I can not get music on hold.. Well really MusicOnHold works, works on Queue, works on # transfer , but hell it does not work when I press the Hold Key (R) on phones. It is not my first installation but it is the first time I get this problem. It is not a problem of mpg123 as the thing works in other parts of Asterisk, it is just a matter of the damned hold key. When I press the hold key in a working installation asterisk says Started music on hold, class 'default', on but in this installation it gets the INVITE messages and the call is really put on hold, but music does not play. Another strange thing is that when I call I could not get the ringtone and I had to state progressinband in sip.conf, but in other installation this was not necessary. The release is Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k Will someone give me a hint? Eugenio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 802.1p , precedence and TOS
well I thought that with diffserv it could be done, I will double check and let you know, thanks for you hint. Eugenio - Original Message - From: SCollins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 21, 2005 1:49 PM Subject: Re: [Asterisk-Users] 802.1p , precedence and TOS I Don't know if this is a solution that is not better suited for your Linux Distro, using vconfig and a 802.1q/1p aware NIC, on the * Server, to create and prioitize the Ingress and Egress Voice VLAN traffic. Would you not agree? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 802.1p , precedence and TOS
well tos is already set and working ( I debugged the packet in and out ) and they work but to and precedence work at layer 3 i.e. in the ip header so they are considered by routers and layer3 switches. I want the 802.1p switch to pass voice traffic priorized on the lan not the wan. The way to do this is at level 2 ( mac level ) . The only language switch understand, because they do not go at IP level ( at least 3com 3300 ... ) Eugenio - Original Message - From: Steve Blair [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 21, 2005 6:56 PM Subject: Re: [Asterisk-Users] 802.1p , precedence and TOS Why wouldn't you just use tos=some value like 0xb8 or tos=lowdelay depending upon the config file in question? Remember there is an overlap between IP precedence bits and some DSCPs for backward compatability. Honor that overlap and you can use DiffServ processing logic even if your device can only set an IP prec. value. Eugenio De Vena wrote: well I thought that with diffserv it could be done, I will double check and let you know, thanks for you hint. Eugenio - Original Message - From: SCollins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 21, 2005 1:49 PM Subject: Re: [Asterisk-Users] 802.1p , precedence and TOS I Don't know if this is a solution that is not better suited for your Linux Distro, using vconfig and a 802.1q/1p aware NIC, on the * Server, to create and prioitize the Ingress and Egress Voice VLAN traffic. Would you not agree? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 802.1p , precedence and TOS
Thanks for your kind help, I understand ip precedence and that's ok. I also found on Snom phones how to mark 802.1p ( which is what I need now ). On the 3Com 3300 802.1p is enabled and correctly priorized . The only thing I miss is how to tell asterisk to originate rtp packet marked with 802.1p tag ( at layer 2 ) Any idea? - Original Message - From: SCollins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 19, 2005 8:45 PM Subject: Re: [Asterisk-Users] 802.1p , precedence and TOS Questions: 1) How can I set the 802.1p precedence field from Snom and asterisk? 2) Which is the differences between the IP Precedence field and TOS field? IP Precednce and ToS are synonymous. There is a 3-bit ToS field in the IP header that allows for 8 levels of classification of IP traffic - these are known as the IP precedence bits. 802.1p or CoS (Class of Service) is a Layer 2 Prioitization protocol. 802.1p uses a 3-bit prioitization field, in an 802.1q TAG,to define 8 priority levels. To obtain the correct Layer 2 CoS on your 3300 I believe you need to set the prioritization on the IP phone, using 802.1q and then setting the prioritization, and then configure the clasifiers and profiles on the 3Com 3300 so that it will remark the packets on ingress. The CoS traffic classification will need to be configured on ALL Switches in your network for the 802.1p prioritzation of the Voice traffic. Most layer 3 Switches support both ToS and CoS and will re-mark the 802.1p Prioritization into the ToS field in the IP header, so that if the traffic is being routed, between VLANs for example, the traffic will maintain it's priority. Hope this helps. Sean On Tue, 19 Apr 2005 15:28:43 +0200, Eugenio De Vena [EMAIL PROTECTED] wrote: Hello everyone, I have some doubt on the QoS matter and I hope that someone could bring me some light. I see my 3Com 3300 switch supports 802.1p priorization , I see through Microsoft network monitor that the packet coming from my SIP phones have in IP header a field which is marked Precedence and when the packet is sent from phone to asterisk the value is CRITICAL/EXP , when the packet is sent back from asterisk to phone the value is Normal. To correct this I set the value Tos in sip.conf to 160 and now the packet sent back from asterisk to phone is marked CRITICAL/EXP too. Fine. Under the field Precedence there is a value called Type of Service ( TOS ) and the values is Normal both ways. Googling around I see that 802.1p is a Layer2 and not Layer3 specification so it must work at MAC level, so I think that these field which are contained into the IP header are not considered by my switch and are just ignored. The Ip precedence field I think would be considered when routed by a QoS router ( which is not my case ). Questions: 1) How can I set the 802.1p precedence field from Snom and asterisk? 2) Which is the differences between the IP Precedence field and TOS field? Thanks Everyone Eugenio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.1and1.com/?k_id=8358073 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghans QuadBRI and fax detection
Hello, I have QuadBRI and asterisk 1.0.6 bristuffed but fax reception works ugly. My faxes are missing many rasters and even sending does not work well. Can you tell me with version of asterisk , spandsp, app_sndfax etc you use to have a good result? Thanks for you kind help - Original Message - From: Kristof Hardy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Tuesday, April 19, 2005 9:07 AM Subject: Re: [Asterisk-Users] Junghans QuadBRI and fax detection Marc Storck wrote: does the Junghans QuadBRI Card and qozap module support Fax detection? You could have a look at the fax detection code in AMP, maybe that helps? But I think it should work. We're not using it, because we're using a fixed number for fax, so if that number gets a call, the (software-) fax always picks up. We didn't like the small delay you need for fax detection, and this way it always works perfect. cheers Kristof ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 802.1p , precedence and TOS
Hello everyone, I have some doubt on the QoS matter and I hope that someone could bring me some light. I see my 3Com 3300 switch supports 802.1p priorization , I see through Microsoft network monitor that the packet coming from my SIP phones have in IP header a field which is marked Precedence and when the packet is sent from phone to asterisk the value is CRITICAL/EXP , when the packet is sent back from asterisk to phone the value is Normal. To correct this I set the value Tos in sip.conf to 160 and now the packet sent back from asterisk to phone is marked CRITICAL/EXP too. Fine. Under the field Precedence there is a value called Type of Service ( TOS ) and the values is Normal both ways. Googling around I see that 802.1p is a Layer2 and not Layer3 specification so it must work at MAC level, so I think that these field which are contained into the IP header are not considered by my switch and are just ignored. The Ip precedence field I think would be considered when routed by a QoS router ( which is not my case ). Questions: 1) How can I set the 802.1p precedence field from Snom and asterisk? 2) Which is the differences between the IP Precedence field and TOS field? Thanks Everyone Eugenio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom and hint priority
Hello, I had the same problem. I solved it by putting the context of the phone in sip.phone as the same context where the hint statement is: i.e.: sip.conf [1713] context=phones extensions.conf [phones] ;1713 exten = 1713,hint,sip/1713 exten = 1713,1,Playback(transfer,skip) ; Please hold while... exten = 1713,2,Macro(stdexten,1713,sip/1713) hope it will work for you too - Original Message - From: Lance Grover [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 17, 2005 6:23 AM Subject: Re: [Asterisk-Users] snom and hint priority I have set up Hint on all my extensions, according to all I have found out, the correct way, however I do not get anything on the phone. Is there something I am missing? I have one of these snom 220's with the side car, and another 220. I am running an RPM version of asterisk and have also tried this on a compiled version of asterisk from the CVS tree. Neither way did it work, is there some thing else I am missing? I set it up as Destination with the sip URL of the extension and my dial plan looks like this: ;1713 exten = 1713,hint,sip/1713 exten = 1713,1,Playback(transfer,skip) ; Please hold while... exten = 1713,2,Macro(stdexten,1713,sip/1713) as you can see I use a Macro but I do not try to put the hint in the Macro, also I have tried this without the Macro. I have rebooted the phone and restarted asterisk after each change. Can someone please help me out? Thanks a ton, -Lance On 4/13/05, Josh Dady [EMAIL PROTECTED] wrote: (boy mail in this list piles up fast when I can't check it) On Apr 8, 2005, at 10:03 AM, Michael George wrote: - It appears that the extension used with the hint must be the same as the extension used to dial that channel. So if extension 22 will ring Zap/2, then exten = 22,hint,Zap/2 will work, but exten = 222,hint,Zap/2 will not. Why is that? The extension is how asterisk maps SIP URLs to chunks of your dialplan -- if you program a button on a snom to dest sip:[EMAIL PROTECTED], the phone will use that same URL for both dialing and subscribing to extension state. Unless you have a phone that lets you specify different URLs for dialing and subscribing to state, they have to match in asterisk. - If I am correct in the above, then there is no way for me to monitor a channel that is not an extension. As an example, I have a TDM400 with 3 FXS (Zap/1-3 on extensions 21-23) and 1 FXO (Zap/4) as well as a VoIP channel for dialing out. I can monitor the states of the extensions with extension entries like exten = 21,hint,Zap/1 but I cannot monitor the state of the FXO with exten = 0,hint,Zap/4 because 0 is not the extension of Zap/4. Indeed, Zap/4 has no extension. Is it not possible to monitor that line, then? There has to be a SIP URL for the phone to subscribe to -- if you put: exten = zap4,hint,Zap/4 in your extensions.conf (with no zap4,1,... entry) it wouldn't be dialable (although the phone would still try if you pushed it) but would have a valid SIP URL. -- Joshua P. Dady ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Lance Grover ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI and signalling
The problem of Q931 SETUP I have also depends on Junghanns and they do not support in any way the products the sell... - Original Message - From: Stefan Gofferje [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 15, 2005 8:30 AM Subject: Re: [Asterisk-Users] ISDN BRI and signalling Bob van der Moezel schrieb: I want to signal BUSY condition to a bristuffed HFC-S ISDN line. However: exten = s,1,Busy has no effect, exten = s,1,Playtones(Busy) is not audable over unanswered line (I live in the Netherlands...) So I currently do: + exten = s,1,Answer + exten = s,2,Playtones(Busy) + exten = s,3,Busy Which obviously is not an ideal solution. Is there a way to get real signalling out a bristuffed HFC-S ? I have the same problem with a Fritz!PCI and chan_capi as external line. I have a HFC-S card in NT mode as internal ISDN. On this card, Busy and Congestion do work. As both drivers came from Junghanns, I suppose, this is a problem with the drivers... Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VIC2BRI and J4BRI
Hello, the problem has been solved. J4BRI is not compatible with VIC-2BRI unless the VIC-BRI are programmed in point-to-multipoint mode. Nobody knows why ( nor Cisco or Junghanns ) but at least we have a workaround. Eugenio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Q931 Setup message
Hello Everyone, I am working with a J4BRI and it works ok withour national provider ( TelecomItalia ) and when connected to ISDN pabx ( Ericsson and Tenovis ). We now have switch to a new telco provider ( Fastweb ) which brings you S0 via Cisco 1760 VIC-2BRI interface. They say( and Cisco says ) that VIC-2BRI emulate perfectly the national isdn standard, but I found that it is not true. First of all I have to switch 3-4 and 5-6 pin of the isdn cable, otherways is not going to work, not too bad but anyway... The critical thing is that I can not receive calls! The normal ( telco and pabx ) Q931 SETUP message arrives like this: 08 01 2c 05 a1 04 03 80 90 a3 where: 08 means Q931 message 01 length of call reference field 2c call reference field a1 sending complete 04 next is bearer cap 03 80 90 is bearer cap and so on when I connect the J4BRI to the VIC-2BRI I receive 08 01 2f 0504 03 80 90 a3 ... a1 so the a1 ( sending complete ) arrives at the end of the SETUP message. After the bearer cap , the calling number and the called number. Asterisk does not recognize this byte stream as a SETUP message and so discards it. Is there anything I can do or is a developer problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 190 and lamp field
Hello, I tried for almost 2 day to make the Function Keys of Snom phone work. Now that they work, maybe I can help someone. The first two steps are simple and documented. 1) Go to Function Keys and make them "destination", type in the extension to be monitored. 2) add a "hint" statement to the extension to be monitored this is what I discovered and that make the whole thing work on not. The extension to be monitored must be defined in the same context defined in sip.conf and that is really important. I had in sip.conf context=sipphones and then in my extensions.conf I defined some variables and include the [day] and [offduty] context. The phones were defined in [day] context and the "lamp field" of the Snom phone did not work. The lamp were on and did not went on and off with busy/clear line. If I set in sip.conf the context to [day] the lamp field start to work , this seem decisive in how the busy/clear NOTIFY process work ( do not know why ). the subscribecontext= keywork does not work in release 1.0.2, maybe in cvs.. I hope this can be useful Eugenio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
