[asterisk-users] Function CHANNELS
I appreciate it if someone can post an an example for function CHANNELS showing the usage of the regular expression filter. Basically I would like to get a count of active channels having a certain criteria. Is it possible to search for a channel having a custom variable set to specific value (directly in dialplan or thru agi)? Our Asterisk version is 1.8. -- _ -- 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] ringing in queues
> Reduce the timeout in the queue configuration (but not in the Queue application in the dialplan), when the timeout > (and the retry) value has elapsed, all available members will be rung again. > Thanks, that should do it. Date: Fri, 13 Mar 2015 14:16:34 + From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ringing in queues On 13 March 2015 at 14:04, Matt Hamilton wrote: We use the ringall strategy for a small queue with 4 members. When a call comes in, if one of the members is busy, all the phones except the busy phone rings (as intended). While the other phones are ringing, if this busy phone becomes available again, we would like to have it start ringing. Right now it just sits idle. Is this possible? I played with ringinuse (queues.conf) and callcounter (sip.conf) values, but wasn't able to get it going. Thanks, Matt -- _ -- 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 Reduce the timeout in the queue configuration (but not in the Queue application in the dialplan), when the timeout (and the retry) value has elapsed, all available members will be rung again. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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
[asterisk-users] ringing in queues
We use the ringall strategy for a small queue with 4 members. When a call comes in, if one of the members is busy, all the phones except the busy phone rings (as intended). While the other phones are ringing, if this busy phone becomes available again, we would like to have it start ringing. Right now it just sits idle. Is this possible? I played with ringinuse (queues.conf) and callcounter (sip.conf) values, but wasn't able to get it going. Thanks, Matt -- _ -- 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] No of parked calls limit
Thanks. I will try to test how many parked calls Asterisk can handle using SIPp. > Date: Wed, 30 Oct 2013 21:41:10 -0500 > From: rnew...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] No of parked calls limit > > I doubt there is a hard-coded limit for parked calls specifically, but > I don't know the answer. If you ask in #asterisk-dev when devs are > around, a developer familiar with the parking code could probably tell > you. > > As you probably already know, you can limit calls in a variety of > ways, one of the simplest and probably most general is the maxcalls > option in asterisk.conf. > > On Tue, Oct 29, 2013 at 6:17 PM, Matt Hamilton > wrote: > > Is there a limit to the number of parked calls Asterisk can handle? > > > > Thanks, > > Matt > > > > -- > > _ > > -- 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 > > > > -- > Rusty Newton > Digium, Inc. | Community Support Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct: +1 256 428 6200 > > 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 -- _ -- 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
[asterisk-users] No of parked calls limit
Is there a limit to the number of parked calls Asterisk can handle? Thanks, Matt -- _ -- 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
[asterisk-users] multiple parking lot best practice
We are planning to have about 100+ parking lots defined in features.conf , each with about 4 unique park positions. Asterisk will be handling all the parking and unparking (we don't exclusively use Park/ParkedCall in the dialplan): [parkinglot_a] parkpos => 1-4 context=parked [parkinglot_b] parkpos => 5-8 context=parked As far as I can tell, Asterisk adds/removes extensions to the parking context(s) dynamically as the calls are parked/unparked. I'm curious which one is preferred (as far as performance and/or stability) - to use one context for all parking lots or to use a separate context for each one? Thanks, Matt -- _ -- 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
[asterisk-users] Number of parked calls in a parking lot
Is there a way to find out the no of parked calls in a parking lot by name (in a multiple parking lot environment) from within the dialplan (not CLI) other than writing a custom function (like VMCOUNT)? Thanks, Matt -- _ -- 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] Call parking issue with Cisco SPA phone
I'll answer my own question: Setting "Keep Referee When REFER Failed" to Yes on the Cisco phone seems to do the trick. From: mistral9...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 20 Oct 2013 11:56:17 -0400 Subject: [asterisk-users] Call parking issue with Cisco SPA phone I'm trying to implement call parking with asterisk and Cisco SPA504G phones: features.conf parkext => 700 parkpos => 701-702 context => parkedcalls I defined one of the unused keys to park the calls: Key2: fnc=sd;ext=700@10.0.1.103;vid=1;nme=Park I also defined two other keys to pickup/unpark the calls: Key3: fnc=blf+sd+cp;sub=701@10.0.1.103 Key4: fnc=blf+sd+cp;sub=702@10.0.1.103 Parking using these works smoothly. I answer the incoming call, press Key2 to park the call. Call is parked, Key3 turns red showing there is a parked call. If I want to unpark the call, I hit Key3 and the call is unparked. My problem happens when Key3 and Key4 are idle (no parked calls): I answer the incoming call and without first parking the car, I hit one of the idle keys (Key3 or Key4), the phone sends a REFER message, and the incoming call hangs up. I'm trying to find out why the call hangs up and how to prevent that? Thanks, Matt -- _ -- 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
[asterisk-users] Call parking issue with Cisco SPA phone
I'm trying to implement call parking with asterisk and Cisco SPA504G phones: features.conf parkext => 700 parkpos => 701-702 context => parkedcalls I defined one of the unused keys to park the calls: Key2: fnc=sd;ext=700@10.0.1.103;vid=1;nme=Park I also defined two other keys to pickup/unpark the calls: Key3: fnc=blf+sd+cp;sub=701@10.0.1.103 Key4: fnc=blf+sd+cp;sub=702@10.0.1.103 Parking using these works smoothly. I answer the incoming call, press Key2 to park the call. Call is parked, Key3 turns red showing there is a parked call. If I want to unpark the call, I hit Key3 and the call is unparked. My problem happens when Key3 and Key4 are idle (no parked calls): I answer the incoming call and without first parking the car, I hit one of the idle keys (Key3 or Key4), the phone sends a REFER message, and the incoming call hangs up. I'm trying to find out why the call hangs up and how to prevent that? Thanks, Matt -- _ -- 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] parking - why doesn't this work?
Thanks Noah.
I thought my dialplan did that: Check to see if the parking space is occupied,
if so, pickup the call in the parking space, if not, park the call to that
space.
This works fine for the first time around, but the second time the
phone/asterisk behaves differently. Still not clear why it's failing. Device
states are updated correctly btw. I assumed when you unpark a call, that space
becomes available/free, and you can park the same call to the same parking
space more than once; e.g. park it to 701, unpark/pick it up from 701, park it
to 701 again, unpark it, etc.
I'm only using one phone (in the lab) - the race condition is not an issue for
now, but thanks for reminding that because my approach might not work the way I
thought it would. I will look into your solution. In any case, it might be
better to let Asterisk pick a random spot (4 lines, 4 parking spots). The good
thing is, we don't need to know where it is (anyone can pick it up) as long as
BLF works properly and we know there is a call parked.
Matt
From: nengelbe...@team-meta.net
To: asterisk-users@lists.digium.com
Date: Mon, 14 Oct 2013 20:31:06 +
Subject: Re: [asterisk-users] parking - why doesn't this work?
You can hack together a way with custom device states and manual use of the
Park() and ParkPickup() functions, but it won’t be particularly pretty. A rough
dialplan might look like the following (adjust to match your requirements,
especially if a park fails or something similar):
exten => _70[1234],1,Verbose(5,Park pickup or park call for slot ${EXTEN})
same => n,GotoIf($[${DEVICE_STATE(park:${EXTEN}@parkinglot)} =
NOT_INUSE]?park,pickup)
; Currently no call parked – park call
same => n(park),Set(__PARKINGEXTEN=${EXTEN})
same => n,Set(__RETURNTO=${CALLERID(num)})
same => n,Dial(Local/s@park,)
; Park failed, clear the device state and return
same => n,Goto(parking-return,${RETURNTO},1)
; Currently a call parked – pick up
same => n(pickup),ParkedCall(${EXTEN}@parkinglot)
same => n,Hangup()
[park]
exten => s,1,Verbose(5,Park call)
same => n,Park(,parking-return,${RETURNTO},1,s,parkinglot)
[parking-return]
exten => _X.,1,Verbose(5,Return parked call to internal phone)
same => n,Set(CALLERID(name)=PK:${CALLERID(name))
same => n,Dial(Local/${EXTEN}@users,)
same => ; some fallback for if the return user doesn’t answer
Basically, the idea is, check to see if the parking space is occupied. If it
is occupied, someone is trying to pickup the parked call, so connect them with
the ParkedCall() application. If it is not occupied, someone is trying to
park a call, so set up the PARKINGEXTEN variable with “where to park it” (e.g.
701), and set up the RETURNTO variable with “where to return if the park fails
or times out” (in my example,
based on the caller ID number of the parking channel – make sure it’s set to
something that will return either via a local channel like I have in my example
or a direct dial to a SIP/ or other channel).
By putting the double underscore (__) in front of the variable name when we set
it, we tell Asterisk to automatically set that variable on any channel spawned
as a descendant of this channel (necessary for parking via a Local channel).
I’m suggesting parking via local channel so that the RETURNTO variable survives
on an attended transfer. Also, the specific example I have above
will not work properly with unattended (blind) transfers to the parking
extension. If you want to support a blind transfer to the parking space, you
need to find a way to use the BLINDTRANSFER and BLIND_XFER_PEER channel
variables to set RETURNTO
correctly.
There’re plenty of other ways to do it, but the core of what you’ll need to
investigate for “SLA parking” is to use ${PARKINGEXTEN} to tell Asterisk where
to
park the call, and use the features.conf settings for the parking lot to
prevent Asterisk from automatically hunting into additional spaces (if you
allow Asterisk to hunt into a new slot, and two people try to park on 701 at
the same time, one of the two calls
will wind up on 702 and the ,s, option in the Park() application means
Asterisk won’t be reading back parking slots to the Parker, so you won’t know
which call lost the race. Practically speaking, it’s not a huge problem, but
the best practice would be to
prevent the auto-hunting and avoid the race condition altogether).
Thank you,
Noah Engelberth
MetaLINK Technologies
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Matt Hamilton
Sent: Monday, October 14, 2013 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] parking - why doesn't this work?
Parking/unparking will be done from multiple phones so that someone else can
pickup/unpark the call from their phone that I parked on mine. I
Re: [asterisk-users] parking - why doesn't this work?
Parking/unparking will be done from multiple phones so that someone else can pickup/unpark the call from their phone that I parked on mine. I'm just testing it on one phone now. I'm trying to simulate the SLA functionality (which Asterisk has, but it's not very scalable and they haven't really been doing much development/improvement on that lately). We have been using SLA for a while, but we are also looking at other options. Unfortunately, conventional parking (pressing #700 and announcing the parking space) is not suitable for our very fast paced environment. > Date: Mon, 14 Oct 2013 16:15:22 +0200 > From: webaccou...@jgoettgens.de > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] parking - why doesn't this work? > > Hmmm, do I understand you correctly that you park and unpark a call using the > same phone? > > If yes, why does simply "holding" the call does not work? The SPA504 has an > extra large button > on the right for this and you don't need any support in the dialplan. > > jg > > -- > _ > -- 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
[asterisk-users] parking - why doesn't this work?
I'm trying to implement parking with only one button to park and unpark a call.
Scenario:
Call is answered, I press the button (on a Cisco SPA504) to park the call, it
comes to [from-office] context where the call is parked successfully (there is
no parking lot number announcement though).
To unpark, I press the same button, it comes to [from-office] context, and the
call is picked up/unparked successfully. (I know there is no need to check the
device state, since Park will go to the next line if it fails, but this is only
for testing.)
The problem happens when I try to park the same call to the same parking lot
space AGAIN by pressing the same button. This time I hear the announcement, but
something else happens which I'm not sure how to describe - the incoming call
seems to be put on hold and parked at the same time (to end the call, you need
to press "end call" twice).
This is my first attempt at parking, and I know this is not the common way to
do this, but would like to know why this is failing.
Here is my config:
[from-office]
exten => _70X,1,SET(devstate=${DEVICE_STATE(park:${EXTEN}@parkedcalls)})
same => n,GotoIf($["${devstate}"="INUSE"]?unpark)
same => n,Set(PARKINGEXTEN=${EXTEN})
same => n,Park() ;
same => n,hangup()
same => n(unpark),ParkedCall(${EXTEN})
same => n,hangup()
[parked_stations]
exten => 701,hint,park:701@parkedcalls
exten => 702,hint,park:702@parkedcalls
;--
features.conf
parkext => 700
parkpos => 701-702
context => parkedcalls
parkingtime => 300
;--
sip.conf
..
context=from-office
subscribecontext=parked_stations
Thanks,
Matt
--
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Re: [asterisk-users] Phones flashing but not ringing
> Have you tried restarting the phone instead of Asterisk? I don't think that > Asterisk sends > separate commands to the bell and to the call LED. Since the LED is flashing, > it is likely that > the "SIP INVITE" signal from Asterisk is ok. Also the ring tone normally does > not come from > Asterisk itself. > We tried it and it doesn't help. It's not one phone, multiple phones do it at the same time. I think it's related to Asterisk SLA - maybe device states get messed up. Matt -- _ -- 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
[asterisk-users] Phones flashing but not ringing
We have been using Asterisk SLA for a while with Cisco SPA series phones. Once in a while the phones flash, but not ring when a call comes in. We can pick it up and talk to the caller even if that's the case. This is pretty random (might not happen for couple of weeks). The quick solution is to restart Asterisk which we are trying to avoid. What might cause this? Thanks, Matt -- _ -- 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
[asterisk-users] strange NAT issue?
We have a couple of cisco SPA phones and 3CX softphones behind a NAT firewall in a remote location. Firewall is connected to a bridged router which connects them to the public internet. Router 5.6.7.8 Firewall 5.6.7.9(gateway 5.6.7.8) Cisco SPA phone 192.168.1.4 Softphone 192.168.1.5 When these phones try to register, this is what we see on the Asterisk side: The source IP of the softphones appear to be 5.6.7.9 (ip of the firewall) whereas the source IP of the hard phones are 5.6.7.8 (ip of the router) for some reason. What might cause the discrepancy? We would like to have the source to show up as 5.6.7.9 (IP of the remote firewall). Thanks, Matt -- _ -- 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] multiple provider for incoming
Greg, > At some point you need to consider how much is too much... I agree. Up until a few days ago, we thought (and were told by our "major" provider) that their network is extremely reliable. Over the weekend all the T1s to our stores were down. Our bonded T1s were supposed to be redundant - they went down together. That was fixed after many hours, and the next day they had an unrelated major outage which took out all our DIDs (T1s were up, but no calls coming thru from the provider). We are in the process of getting fiber from another provider like you for at least some critical locations - that will alleviate the connectivity issues, but our biggest concern is the DIDs now. Matt > Date: Tue, 30 Apr 2013 23:33:22 -0500 > From: gmals...@coastalacq.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] multiple provider for incoming > > Matt, > > At some point you need to consider how much is too much... > > I run a call center with more then 125 commissioned phone sales reps and more > than 60 customer service reps. We run dual servers, fiber from one provider > and 6 bonded T1's from another provider. We purchase our so trunks from a > wholesale company who is a major provider to resellers. Being so, their > network is extremely reliable. However, late last week an upstream/downstream > provider had am outage which affected some of our DIDs but not others. > > Your assumption that porting a number from one provider to another is > correct. If I remember correctly it's an FCC mandate that a number cannot be > ported within 30 days of a previous port. > > Greg > > Matt Hamilton wrote: > > >Don, > > > >Inbound reliability is very important. We don't use toll-free numbers, but > >we will look into that. I thought porting numbers - not sure about toll-free > >though - from one provider to the other took days (not technically, but > >paperwork, etc.) > > > >Thanks, > >Matt > > > >From: d...@donkelly.biz > >To: asterisk-users@lists.digium.com > >Date: Tue, 30 Apr 2013 22:38:44 -0500 > >Subject: Re: [asterisk-users] multiple provider for incoming > > > >If inbound reliability is important, you may be able to accomplish what you > >want with redundant servers, multiple sip providers and toll-free numbers > >that can be readily switched between the sip providers.--Don From: > >asterisk-users-boun...@lists.digium.com > >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton > >Sent: Tuesday, April 30, 2013 10:25 PM > >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [asterisk-users] multiple provider for incoming >The process > >will depend on your provider, of course, but I know some have an option that > >if they are unable to reach > >>your box, then they can auto-forward to another DID, or to a voicemail box, > >>or to a user-defined function, etc. > > > >Forwarding to another DID will/should work for us assuming they are going to > >be able to do that during a failure on their side. During a recent outage (I > >think they had some major issues at one of their switches), they were not > >able to send the calls to our box which was online. > > > >Thanks, > >Matt > > > >Date: Tue, 30 Apr 2013 20:38:19 -0500 > >From: wcse...@selbytech.com > >To: asterisk-users@lists.digium.com > >Subject: Re: [asterisk-users] multiple provider for incomingOn Tue, Apr 30, > >2013 at 7:50 PM, David Wessell wrote:Hi Matt, You can't > >have multiple providers for inbound traffic. You can have multiple providers > >for outbound traffic though. ThanksDavid David, I'm not sure where you got > >this information, but it's not accurate. I've had multiple inbound and > >outbound SIP providers for years going to a single box. You just get a > >separate DID from each provider. Matt, > > > >The process will depend on your provider, of course, but I know some have an > >option that if they are unable to reach your box, then they can auto-forward > >to another DID, or to a voicemail box, or to a user-defined function, etc. > >-- > >Thanks, > >--Warren Selby, dCAP > >http://www.SelbyTech.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
Re: [asterisk-users] multiple provider for incoming
Don, Inbound reliability is very important. We don't use toll-free numbers, but we will look into that. I thought porting numbers - not sure about toll-free though - from one provider to the other took days (not technically, but paperwork, etc.) Thanks, Matt From: d...@donkelly.biz To: asterisk-users@lists.digium.com Date: Tue, 30 Apr 2013 22:38:44 -0500 Subject: Re: [asterisk-users] multiple provider for incoming If inbound reliability is important, you may be able to accomplish what you want with redundant servers, multiple sip providers and toll-free numbers that can be readily switched between the sip providers.--Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton Sent: Tuesday, April 30, 2013 10:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] multiple provider for incoming >The process will depend on your provider, of course, but I know some have an option that if they are unable to reach >your box, then they can auto-forward to another DID, or to a voicemail box, or >to a user-defined function, etc. Forwarding to another DID will/should work for us assuming they are going to be able to do that during a failure on their side. During a recent outage (I think they had some major issues at one of their switches), they were not able to send the calls to our box which was online. Thanks, Matt Date: Tue, 30 Apr 2013 20:38:19 -0500 From: wcse...@selbytech.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] multiple provider for incomingOn Tue, Apr 30, 2013 at 7:50 PM, David Wessell wrote:Hi Matt, You can't have multiple providers for inbound traffic. You can have multiple providers for outbound traffic though. ThanksDavid David, I'm not sure where you got this information, but it's not accurate. I've had multiple inbound and outbound SIP providers for years going to a single box. You just get a separate DID from each provider. Matt, The process will depend on your provider, of course, but I know some have an option that if they are unable to reach your box, then they can auto-forward to another DID, or to a voicemail box, or to a user-defined function, etc. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.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 -- _ -- 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] multiple provider for incoming
>The process will depend on your provider, of course, but I know some have an option that if they are unable to reach >your box, then they can auto-forward to another DID, or to a voicemail box, or to a user-defined function, etc. Forwarding to another DID will/should work for us assuming they are going to be able to do that during a failure on their side. During a recent outage (I think they had some major issues at one of their switches), they were not able to send the calls to our box which was online. Thanks, Matt Date: Tue, 30 Apr 2013 20:38:19 -0500 From: wcse...@selbytech.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] multiple provider for incoming On Tue, Apr 30, 2013 at 7:50 PM, David Wessell wrote: Hi Matt, You can't have multiple providers for inbound traffic. You can have multiple providers for outbound traffic though. Thanks David David, I'm not sure where you got this information, but it's not accurate. I've had multiple inbound and outbound SIP providers for years going to a single box. You just get a separate DID from each provider. Matt, The process will depend on your provider, of course, but I know some have an option that if they are unable to reach your box, then they can auto-forward to another DID, or to a voicemail box, or to a user-defined function, etc. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.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 -- _ -- 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
[asterisk-users] multiple provider for incoming
We have couple of stores with published phone numbers. We are currently using a telecom company that hosts those numbers and provides us with SIP trunking. Recently we experienced couple of outages with this company, so we are looking into getting a backup provider just in case. As far as I know it's straight forward to setup multiple providers in Asterisk, but I'm not quite sure how the incoming calls will work as far as the phone nos. Do we need to have the first provider temporarily forward the calls to the second provider's DIDs? What if first provider goes down completely? This is not really an Asterisk question, but any help is appreciated. Thanks, Matt -- _ -- 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] MySQL InnoDB or MyISAM for CDR
Date: Thu, 27 Sep 2012 10:23:35 +0200 From: lenz.lo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR I'd go for MyISAM and would set up a remote replica if data integrity is important. If you have like 1000 calls of (say) 30 seconds avg length, and you create 10 events per call, you would expect an event every three seconds. This is about 300 inserts per second. Say 600 at peaks. This should be feasible with server-grade hardware without much difficulty. Also as you always INSERT it behaves as a log file (no seeking, no locking) if the table is optimized. l. We decided to go with MyISAM since it supports concurrent inserts (as you suggested). Data integrity (a slight loss of call records) is something we can live by. Right now we use DRBD for replication, but I guess with MyISAM it doesn't make much sense if the db crashes. We are looking into other options as well. Thanks. -- _ -- 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] MySQL InnoDB or MyISAM for CDR
Our top priority is the raw Write (INSERT) performance, Read (SELECT) performance is not important. Strict ACID compliance is not necessary either. MySQL (on a separate database server) should be able to handle inserting CDR records (approximately up to 10 records for each call) for about 1000 concurrent calls coming from an Asterisk cluster. Matt Date: Tue, 25 Sep 2012 18:19:50 -0500 From: lo...@keobi.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR Very good point. For revenue critical data like CDRs, being ACID compliant is important. MyISAM is compliant. And like InnoDB, can have the features making it compliant turned off. On Sep 25, 2012 6:12 PM, "Patrick Lists" wrote: On 09/25/2012 11:18 PM, Logan Bibby wrote: MyISAM would be best, in my opinion. The features that cause the little bit of performance overhead in InnoDB wouldn't be necessary for CDR storage. Iirc InnoDB is ACID compliant so might be preferable if MyISAM is not. More information here: http://en.wikipedia.org/wiki/ACID https://blogs.oracle.com/MySQL/entry/comparing_innodb_to_myisam_performance Regards, Patrick -- _ -- 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
[asterisk-users] MySQL InnoDB or MyISAM for CDR
Which one (InnoDB or MyISAM) is preferred for CDR as far as write performance is concerned? Thanks, Matt -- _ -- 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
[asterisk-users] dynamic hints and SLA
I would like to have the SLA hints created on the fly as the phones register
dynamically. Is this possible?
The static version that works:
[sla_stations]
exten => 10041_ln10041,hint,SLA:10041_ln010041
exten => 10041_ln10042,hint,SLA:10041_ln010042
exten => 10042_ln10041,hint,SLA:10042_ln010041
exten => 10042_ln10042,hint,SLA:10042_ln010042
The following didn't work.
exten => _1004.,hint,SLA:${EXTEN}
I came across couple of posts suggesting to use EXTENSION_STATE somehow. Any
hints? :)
Thanks,
Matt
--
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Re: [asterisk-users] asterisk and meetme
> ConfBridge is the preferred conference application in Asterisk 10+. While > MeetMe is currently deprecated, you can still enable it and run it in > Asterisk 10+. What's going to happen to SLA (which is heavily integrated with MeetMe)? Will the functionality be ported to ConfBridge? Thanks, Matt > Date: Fri, 10 Aug 2012 08:34:32 -0500 > From: mjor...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] asterisk and meetme > > > > - Original Message - > > From: "Jerry Geis" > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Sent: Friday, August 10, 2012 8:25:54 AM > > Subject: Re: [asterisk-users] asterisk and meetme > > > > On 08/10/2012 09:00 AM, Jerry Geis wrote: > > > My bad - "make menuconfig" was not coming up as my window was too > > small, > > was confused as the help page I was on for asterisk 10 and meetme did > > not say > > anything about it being deprecated (as menuconfig does). > > Which help page was it? If its the application description on the Asterisk > wiki, you're right - those don't currently display the 'support' status of > the module that the application resides in. > > You can find out all of the support statuses of the various modules here: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States > > > So meetme is deprecated in asterisk 10. Looks like I need to move to > > app_confbridge. > > ConfBridge is the preferred conference application in Asterisk 10+. While > MeetMe is currently deprecated, you can still enable it and run it in > Asterisk 10+. > > -- > 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 -- _ -- 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] Automate SLA testing
Thanks Paul, we are looking into the testsuite. > Date: Sun, 3 Jun 2012 18:32:57 -0400 > From: paul.belan...@polybeacon.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Automate SLA testing > > On 12-06-03 03:56 PM, Matt Hamilton wrote: > > > > We would like to automate Shared Line Appearance testing (e.g. phoneA > > answers a call, puts in on hold, phoneB picks up the call on hold) in our > > lab. Are there any tools/SIP call generators/clients that may help us > > create such a scenario? > > > Check out the asterisk testsuite for some examples[1]. You could use a > combination of StarPy and pjsua (python bindings) to do this. > > [1] http://svnview.digium.com/svn/testsuite/asterisk/trunk/ > > -- > Paul Belanger | PolyBeacon, Inc. > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > Github: https://github.com/pabelanger | Twitter: > https://twitter.com/pabelanger > > -- > _ > -- 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
[asterisk-users] Automate SLA testing
We would like to automate Shared Line Appearance testing (e.g. phoneA answers a call, puts in on hold, phoneB picks up the call on hold) in our lab. Are there any tools/SIP call generators/clients that may help us create such a scenario? Thanks, Matt -- _ -- 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] device state of a realtime queue member
> You can use system() to do this from the dialplan I'll give that a try. Seems like there is no dialplan function for that yet. I guess querying the database via func_odbc is another option. Thanks. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 18 Apr 2012 12:10:55 -0500 Subject: Re: [asterisk-users] device state of a realtime queue member You can use system() to do this from the dialplan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton Sent: Wednesday, April 18, 2012 11:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] device state of a realtime queue member Thanks Ishfaq, I need something from within the dialplan though. > From: i...@pack-net.co.uk > To: asterisk-users@lists.digium.com > Date: Wed, 18 Apr 2012 10:06:38 +0100 > Subject: Re: [asterisk-users] device state of a realtime queue member > > On Tue, 2012-04-17 at 11:53 -0400, Matt Hamilton wrote: > > I'm trying to find if a realtime queue member is paused or not from > > the dialplan. > > > > For a "paused", "not in use" phone, DEVICE_STATE returns "not in use" > > only. Is there a function that will tell if the phone is paused or not > > (other than querying the database directly)? > > > > Thanks, > > Matt > > > Hi > > You could use > queue show > or > queue show > asterisk console commands > > Ish > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 > > > -- > _ > -- 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] device state of a realtime queue member
Thanks Ishfaq, I need something from within the dialplan though. > From: i...@pack-net.co.uk > To: asterisk-users@lists.digium.com > Date: Wed, 18 Apr 2012 10:06:38 +0100 > Subject: Re: [asterisk-users] device state of a realtime queue member > > On Tue, 2012-04-17 at 11:53 -0400, Matt Hamilton wrote: > > I'm trying to find if a realtime queue member is paused or not from > > the dialplan. > > > > For a "paused", "not in use" phone, DEVICE_STATE returns "not in use" > > only. Is there a function that will tell if the phone is paused or not > > (other than querying the database directly)? > > > > Thanks, > > Matt > > > Hi > > You could use > queue show > or > queue show > asterisk console commands > > Ish > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 > > > -- > _ > -- 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
[asterisk-users] device state of a realtime queue member
I'm trying to find if a realtime queue member is paused or not from the dialplan. For a "paused", "not in use" phone, DEVICE_STATE returns "not in use" only. Is there a function that will tell if the phone is paused or not (other than querying the database directly)? Thanks, Matt -- _ -- 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
[asterisk-users] meetme timeout if only one participant
Is it possible to have a meetme conference timeout (and go to the next line in the dialplan) if there is only one participant left? Thanks, Matt -- _ -- 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] SendText causes Retransmission errors
> Date: Mon, 19 Mar 2012 10:31:52 -0500 > From: kpflem...@digium.com > > 502 10.0.1.103 10.0.1.57 Request: CANCEL sip:104@10.0.1.57:5060 > > Why did Asterisk CANCEL the call here? I assume it's part of the SLA implementation. As I mentioned in my original email, I'm using SendText to send a text message when the user picks up a line in a SLA setup. In this case, ext 124 is calling 104, and one of the lines on 104 is picking it up. Asterisk is connecting to that line and cancelling the first request?? (just guessing) same => n,SendText(hi) same => n,SLAStation(4*104_line104) > > > *503 (for 493) 10.0.1.57 10.0.1.103 Status: 200 OK* > > 524 (503) 10.0.1.57 10.0.1.103 Request: ACK > > sip:8*104_line104@10.0.1.103:5060 > > This appears to be broken. The listing here claims this ACK is in > response to the '200 OK' in packet 503, which itself was a final > response to the MESSAGE request in packet 493. However, MESSAGE requests > do not use ACK for a three-way handshake like INVITE requests do. In > addition, this packet is going the wrong direction to be an ACK for > packet 503, since it's going the same direction as packet 503 did. I use Wireshark to capture the packets, and Wireshark is reporting it that way; i.e. saying that Request Frame for the ACK is the OK (for MESSAGE). I guess it's incorrect. The order and direction of messages I posted in my previous email are taken directly from Wireshark. Frame 15 is MESSAGE Frame 19 is OK (for MESSAGE) Frame 20 is ACK (Wireshark is saying the Request Frame is 20 ??) I tried to post the full SIP capture here, but it got rejected because of the size of the post (about 280k). -- _ -- 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] SendText causes Retransmission errors
Kevin, thanks for your response. Here is the more detailed Wireshark capture of the SIP traffic between phone (10.0.1.57) and Asterisk (10.0.1.103). The numbers between parentheses are Request Frames. I don't see an ACK for the 200 OK to the INVITE (491) for the dialplan that gives us Retransmission errors (without WAIT), but there is also no ACK for the same INVITE for the dialplan that works (with WAIT). If you still want to take a look at the full packet capture, I'll post it. Matt - Without WAIT(1) - we get Retransmisson errors 48610.0.1.5710.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 48710.0.1.103 10.0.1.57Status: 401 Unauthorized 490 (486) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103 49110.0.1.5710.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 49210.0.1.103 10.0.1.57Status: 100 Trying 49310.0.1.103 10.0.1.57Request: MESSAGE sip:104@10.0.1.57:5060 (text/plain) 500 (for 491) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP 50110.0.1.103 10.0.1.57Request: NOTIFY sip:104@10.0.1.57:5060 50210.0.1.103 10.0.1.57Request: CANCEL sip:104@10.0.1.57:5060 503 (for 493) 10.0.1.5710.0.1.103 Status: 200 OK 524 (503) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 525 (501) 10.0.1.5710.0.1.103 Status: 200 OK 52610.0.1.5710.0.1.103 Status: 487 Request Terminated 527 (for 502) 10.0.1.5710.0.1.103 Status: 200 OK 528 (502) 10.0.1.103 10.0.1.57Request: ACK sip:104@10.0.1.57:5060 585 (524) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP (resend of 500) 588 (524) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 803 (588) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP (resend of 500) 806 (588) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 1223 (806) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP (resend of 500) 1229 (806) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 2042 (1229) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP (resend of 500) 204410.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 288610.0.1.103 10.0.1.57Status: 200 OK, with SDP 288810.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 375210.0.1.103 10.0.1.57Status: 200 OK, with SDP 375510.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 - with WAIT(1). There is no more messages beyond 672 until the call is over. Everything is normal. There is no ACK for the OK for INVITE in 430 here either. 42510.0.1.5710.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 42610.0.1.103 10.0.1.57Status: 401 Unauthorized 429 (425) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103 43010.0.1.5710.0.1.103 Request: INVITE sip:8*104_line104@10.0.1.103, with SDP 43110.0.1.103 10.0.1.57Status: 100 Trying 43210.0.1.103 10.0.1.57Request: MESSAGE sip:104@10.0.1.57:5060 (text/plain) 443 (for 432) 10.0.1.5710.0.1.103 Status: 200 OK 645 (for 430) 10.0.1.103 10.0.1.57Status: 200 OK, with SDP 64610.0.1.103 10.0.1.57Request: NOTIFY sip:104@10.0.1.57:5060 64710.0.1.103 10.0.1.57Request: CANCEL sip:104@10.0.1.57:5060 667 (443) 10.0.1.5710.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 668 (646) 10.0.1.5710.0.1.103 Status: 200 OK 67010.0.1.5710.0.1.103 Status: 487 Request Terminated 671 (647) 10.0.1.5710.0.1.103 Status: 200 OK 672 (for 647) 10.0.1.103 10.0.1.57Request: ACK sip:104@10.0.1.57:5060 > Date: Fri, 16 Mar 2012 10:22:49 -0500 > From: kpflem...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] SendText causes Retransmission errors > > On 03/16/2012 0
[asterisk-users] SendText causes Retransmission errors
Hi, I'm using SendText to send a text message when the user picks up a line in a SLA setup (even though I'm not sure the problem is related to SLA). I'm on Asterisk 10.2.1 (same in 1.8.9) [from-office] .. same => n,SendText(hi) same => n,SLAStation(line1234) .. Here is a simplified version of the SIP messages: 1 phone => Asterisk INVITE 2 Asterisk => phone Trying 3 Asterisk => phone MESSAGE 4 Asterisk => phone OK (for the INVITE at 1) 5 phone => Asterisk OK (for the MESSAGE at 3) 6 Asterisk => phone OK (for the INVITE at 1)*** RESEND of 4 7 Asterisk => phone OK (for the INVITE at 1)*** RESEND of 4 .. The text message is sent and the call is connected, but Asterisk keeps resending OK for the INVITE, and eventually drops the call after Transmission timeout. If I insert a WAIT after SendText, the order of the OKs changes, and everything works: same => n,SendText(hi) same => n,Wait(1) same => n,SLAStation(line1234) Here is the SIP message flow with WAIT (4 and 5 above are swapped): 1 phone => Asterisk INVITE 2 Asterisk => phone Trying 3 Asterisk => phone MESSAGE 4 phone => Asterisk OK (for the MESSAGE at 3) 5 Asterisk => phone OK (for the INVITE at 1) Is there anything else I can do other than using WAIT (which might not be a consistent solution anyway)? Thanks, Matt -- _ -- 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
[asterisk-users] SLA and call transfer
Hi, Is it possible to transfer a call in Asterisk SLA (shared line appearance) e.g., call comes in via SLATrunk, it's answered, and transferred to an outside number? Thanks, Matt -- _ -- 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] skip authentication for REGISTER
Thanks Kevin. Seems like remotesecret takes over if secret is not defined - I'll do further tests.. The authentication for REGISTERs and SUBSCRIBEs are done at a sip proxy (opensips) - I'll try to take care of the UAC authorization request for NOTIFY there (if possible). Regards, Matt > Date: Tue, 14 Feb 2012 09:44:38 -0600 > From: kpflem...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] skip authentication for REGISTER > > On 02/14/2012 08:43 AM, Matt Hamilton wrote: > > Hi, > > > > For REGISTER and SUBSCRIBE requests coming from UACs, is it possible to > > make Asterisk skip authentication even if a "secret" is defined in > > sip.conf for the peer; i.e. similar to insecure=invite for INVITE requests? > > > > If I leave "secret" blank, Asterisk doesn't require any authentication - > > this works as I want. However, I also use "SIP NOTIFY" to contact UACs > > (UACs are set to require authorization for NOTIFY), but without the > > "secret" defined, Asterisk can't send the correct authorization. > > You can use 'remotesecret' to set the secret string for Asterisk to use > to respond to authentication challenges. There isn't any way to make > REGISTER/SUBSCRIBE/etc. insecure like there is for INVITEs... I can't > imagine that many people would want unauthenticated REGISTERs to be > allowed :-) > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.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 -- _ -- 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
[asterisk-users] skip authentication for REGISTER
Hi, For REGISTER and SUBSCRIBE requests coming from UACs, is it possible to make Asterisk skip authentication even if a "secret" is defined in sip.conf for the peer; i.e. similar to insecure=invite for INVITE requests? If I leave "secret" blank, Asterisk doesn't require any authentication - this works as I want. However, I also use "SIP NOTIFY" to contact UACs (UACs are set to require authorization for NOTIFY), but without the "secret" defined, Asterisk can't send the correct authorization. Thanks, Matt -- _ -- 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
[asterisk-users] criteria for setting registration expiration
Hi, Are there any guidelines/recommended values for setting the registration expiration and subscription expiration for SIP phones? The default values for those settings on our phones are 60 secs. Any disadvantages for making them longer; e.g. 300 secs or more? Thanks, Matt -- _ -- 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] Dialing multiple numbers
Thanks Carlos, using local channels works. Asterisk inserts a separate CDR record for each with only one of them having disposition=ANSWERED (the rest NO ANSWER). Matt Date: Sun, 5 Feb 2012 12:48:57 -0500 From: crt.ro...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dialing multiple numbers Hello, Maybe exten => 100,1,Dial(Local/14153456789@provider&Local/15606305670@provider) Regards On Sun, Feb 5, 2012 at 12:16 PM, Matt Hamilton wrote: Hi, When dialing multiple numbers, we get connected to the first channel that answers and others are hung up. exten => 100,1,Dial(SIP/provider/14153456789&SIP/provider/15606305670) How can I find out the channel (actually which destination no) the connection is made to (so that I can write it to CDR)? Thanks, Matt -- _ -- 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
[asterisk-users] Dialing multiple numbers
Hi, When dialing multiple numbers, we get connected to the first channel that answers and others are hung up. exten => 100,1,Dial(SIP/provider/14153456789&SIP/provider/15606305670) How can I find out the channel (actually which destination no) the connection is made to (so that I can write it to CDR)? Thanks, Matt -- _ -- 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] meetme - Unable to write frame to channel
I'm not using meetme directly - I'm using SLA which internally uses meetme and creates conferences for SLA trunks. There are no sound problems for me, either, but when the caller hangs up and this error occurs, the trunk statuses are not updated properly and the phones still show them as in use or hold. It's really hard to duplicate it - it seems to happen more under heavier load though. Matt Date: Sun, 22 Jan 2012 13:36:07 +0100 From: li...@jttech.se To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] meetme - Unable to write frame to channel 2012-01-20 20:09, Matt Hamilton skrev: Hi, Once in a while when a SIP channel connected to meetme conference is hung up, I start getting the following error multiple times: WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to channel Local/100203@h The status of the channel is not updated, and the only way to get back to normal is to restart Asterisk. Any thoughts? Is this a timing issue? As you write I have seen this also with SIP in Meetme conferences sometimes when sip-channels is hung up. I havn't found any real problem or bad sound related to this, so I usually ignore this error. -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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] meetme - Unable to write frame to channel
Other posts on the internet suggests that this is a timing issue. According to Asterisk wiki, the preferred timing module is res_timing_timerfd.so (then res_timing_dahdi.so and the least preferred res_timing_pthread.so). res_timing_timerfd.so is the one I'm using, but still getting these errors. Is timing hardware such as TDM410P, Sangoma UT51, the only reliable option? Thanks. From: mistral9...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 20 Jan 2012 14:09:21 -0500 Subject: [asterisk-users] meetme - Unable to write frame to channel Hi, Once in a while when a SIP channel connected to meetme conference is hung up, I start getting the following error multiple times: WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to channel Local/100203@h The status of the channel is not updated, and the only way to get back to normal is to restart Asterisk. Any thoughts? Is this a timing issue? Thanks a lot, Matt -- _ -- 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
[asterisk-users] meetme - Unable to write frame to channel
Hi, Once in a while when a SIP channel connected to meetme conference is hung up, I start getting the following error multiple times: WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to channel Local/100203@h The status of the channel is not updated, and the only way to get back to normal is to restart Asterisk. Any thoughts? Is this a timing issue? Thanks a lot, Matt -- _ -- 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
[asterisk-users] public ip issue with asterisk cluster
Hi, I have an Opensips server dispatching to 3 Asterisk servers. I would like to assign public IPs to all of these servers and avoid NAT altogether - phones will also have public IPs. The way I set this in the lab, all the SIP traffic goes thru the SIP proxy (Opensips) and RTP goes directly between the Asterisk servers and the UAs. The issue is that our provider (they will be both sip trunk and internet access provider for us) wants to assign us only 1 public IP on their voice network - they are saying that the above design is unusual. I'm new to this, is it? If we end up getting only 1 public IP, I assume putting all behind NAT (or assigning the public IP to opensips and putting the asterisk servers behind NAT) will do it. rtpproxy is also setup on the Opensips server just in case - I can use it to force the RTP traffic thru the sip proxy. Any other way? All I want to do is load balance the RTP traffic, avoid any unnecessay processing and bottlenecks (rtpproxy, etc.). Any thoughts? Thanks, Matt -- _ -- 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
[asterisk-users] UPDATE RE: registration not authorized - stale nonce
Asterisk destroys SIP dialogs in 32 secs, so increased the UAC registry expiration to 300 secs just in case, but that didn't help either. From: mistral9...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 1 Jan 2012 18:13:07 -0500 Subject: [asterisk-users] registration not authorized - stale nonce I have a very basic setup where a UAC registers with Asterisk 1.8.7.2 - both on the same subnet, no nat. The following is the flow of messages: 1. UAC sends the registration request 2. Asterisk responds with 401 Unauthorized with a new nonce 3. UAC sends a new digest with the nonce received from Asterisk 4. Asterisk authorizes UAC and sends OK This works as expected when there is no load on the Asterisk server. For testing asterisk under load, I use Sipp (runnig at a server on the same subnet) with a very basic scenario (call is placed, moh played, sipp hangs up after 20 secs). Max calls on the system at one time is 50. My problem happens when the above UAC tries to register while the Sipp test is running. Steps 1 and 2 above happen as expected, Asterisk sends a new nonce to the UAC, but at step 3, UAC sends the old digest (old nonce) back to Asterisk. Asterisk doesn't authorize the UAC with "Correct auth, but based on stale nonce" warning. UAC registration expiry is 60 seconds. As I mentioned, if the test is not running, everything seems to be OK. If I adjust the max calls to 10 in the test and reduce the load, the registrations go thru. Any ideas? Thanks, Matt -- _ -- 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
[asterisk-users] registration not authorized - stale nonce
I have a very basic setup where a UAC registers with Asterisk 1.8.7.2 - both on the same subnet, no nat. The following is the flow of messages: 1. UAC sends the registration request 2. Asterisk responds with 401 Unauthorized with a new nonce 3. UAC sends a new digest with the nonce received from Asterisk 4. Asterisk authorizes UAC and sends OK This works as expected when there is no load on the Asterisk server. For testing asterisk under load, I use Sipp (runnig at a server on the same subnet) with a very basic scenario (call is placed, moh played, sipp hangs up after 20 secs). Max calls on the system at one time is 50. My problem happens when the above UAC tries to register while the Sipp test is running. Steps 1 and 2 above happen as expected, Asterisk sends a new nonce to the UAC, but at step 3, UAC sends the old digest (old nonce) back to Asterisk. Asterisk doesn't authorize the UAC with "Correct auth, but based on stale nonce" warning. UAC registration expiry is 60 seconds. As I mentioned, if the test is not running, everything seems to be OK. If I adjust the max calls to 10 in the test and reduce the load, the registrations go thru. Any ideas? Thanks, Matt -- _ -- 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
[asterisk-users] performance/memory
I have a couple of performance/memory related questions: Is there any downside to using long URIs as far as memory or database (mysql) performance is concerned, e.g. sip:1234567890_1234567...@abc.com? Or is this negligible? Also is there a performance hit if no pattern matching is used? e.g. exten => _XXX,Noop(... vs exten => 100,Noop(.. exten => 101,Noop(... exten => 102,Noop(... ... exten => 999,Noop(... If a call comes to 999, does Asterisk go through each extension sequentially from 100 to 999 until it finds the matching one? Thanks, Matt -- _ -- 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] queue not skipping ringing phone
Thanks Sebastian. It was a phone related issue. Factory resetting the phones and reconfiguring them fixed it. It probably was a CW issue as you suggested. > I think it is up to your phones to allow only one concurrent session, > you could check call-waiting is deactivated on your phones?! > > If your phones allow more than one active dialog you probably wont have that > much fun with queues... > > And make sure you have read the "Queue Empty Options" section of the > queues.conf example as some parameters changed to be more flexible > (joinempty = ringing etc... ). That could be interesting too... > > hth, > Sebastian Denz -- _ -- 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
[asterisk-users] queue not skipping ringing phone
I have a queue that distributes calls among 3 phones. When a phone is in use (including on hold), queue skips that device and sends the call to the next available one as expected. On the other hand, if a call comes in while one of the phones is ringing, the queue doesn't seem to recognize that phone as "in use" and sends the second call to the ringing phone. If the first call is answered, the second call is sent to the next available phone right away. I'm new to asterisk and wondering if this is normal; I thought the ringing phone would be skipped "as in use" as well. Is there a setting on the asterisk side that I can use to force the queue to skip the "ringing" phone, or should this somehow be done on the phone itself? Thanks, Matt Below is the queues.conf: [qtemplate] announce-frequency=0 announce-holdtime=no announce-position=no autofill=yes eventmemberstatus=no eventwhencalled=no joinempty=strict leavewhenempty=strict maxlen=0 memberdelay=0 penaltymemberslimit=0 periodic-announce-frequency=0 queue-callswaiting=silence/1 Sendqueue-thereare=silence/1 queue-youarenext=silence/1 reportholdtime=no ringinuse=no servicelevel=60 strategy=rrmemory timeout=0 timeoutpriority=app timeoutrestart=no retry=0 weight=0 wrapuptime=0 musicclass=default monitor-type=MixMonitor monitor-format=wav [q1000](qtemplate) member=Local/1001@handle-queue,,,SIP/1001 member=Local/1002@handle-queue,,,SIP/1002 member=Local/1003@handle-queue,,,SIP/1003 -- _ -- 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
[asterisk-users] including conf using static realtime
Is it possible to load some parts of the extensions.conf file via static realtime? For example, extensions.conf [some_context] #include abc.conf extconfig.conf abc.conf => mysql,asterisk,ast_config So far, I wasn't able to get it going - Asterisk crashes at startup. Maybe abc.conf doesn't exist yet (hasn't been read from DB yet) when asterisk loads the extensions.conf file? Just wanted to check if this is possible before playing it with it more. That context in the extensions.conf will have lots of repetitions and will be updated once in a while, so loading it from db for a bunch of asterisk servers seems logical. Thanks, Matt -- _ -- 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
[asterisk-users] asterisk registrations by SER proxy
I integrated Opensips with Asterisk Realtime (Asterisk sipusers/peers point to Opensips subscribe table via a view). Opensips handles the registrations. However, when a call comes in (INVITE is routed to Asterisk), it seems like Asterisk doesn't know about the user (or sees the users as not authorized), so can't create the SIP channel. (I use queues and conferencing also.) If I route the REGISTER to Asterisk after authorizing in Opensips, Asterisk does the authorization/registration again from scratch. In that case call goes through, but I end up duplicating the authorization process. I was hoping to take the load of handling registrations from Asterisk. I know this is a very common scenario, but I'm not very clear about the process. Is it possible to make Asterisk be aware of those registrations made by the proxy server? Thanks, Matt -- _ -- 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] check if devices reachable in queue
Thanks Dale, you pointed me in the right direction. > You are calling the Dial() application here. If you are using queues, you should use the Queue() application. I'm using Local channels with the queue: - queues.conf [support] member => Local/1001@handle-queue ---extensions.conf- [incoming] Queue(support) [handle-queue] ; some preprocessing here same => n,Dial() I found out that since I'm using the Local channel as the queue member, the Queue() doesn't know the state the call is in. It monitors the state of the Local channel, and not the device. However, it seems to be possible to give the queue the actual device to monitor and associate that with the Local channel by modifiying the member in the queues.conf: ; queues.conf [support] member => Local/1001@[handle-queue],,,SIP/1001 I won't be able to test this until this evening, but it seems like it's going to work... -- _ -- 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] check if devices reachable in queue
>Have you tried, instead of pre-processing the caller before calling
Queue(), checking the ${QUEUESTATUS} variable.
Even when the phones are UNREACHABLE, QUEUE is still trying until it times out
- ${QUEUESTATUS} = TIMEOUT
I get the following for all the members of the queue, in a loop, until it times
out.
Executing [1001@handle-queue:3] Dial("Local/1001@handle-queue-6d01;2",
"SIP/1001") in new stack
[Nov 21 18:57:42] WARNING[4780]: app_dial.c:2196 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'Local/1001@handle-queue-6d01;2' status is
'CHANUNAVAIL'
-- Local/1001@handle-queue-6d01;1 is circuit-busy
-- Nobody picked up in 0 ms
[Nov 21 18:57:42] WARNING[4780]: channel.c:4622 ast_prod: Prodding channel
'Local/1001@handle-queue-6d01;2' failed
queue.conf-
joinempty=no
joinunavailable=no
leavewhenempty=yes
timeout=0(for testing purposes, I set the timeout in the
application to 10 secs)
timeoutpriority=app
timeoutrestart=no
retry=0
Is it possible to make the queue not wait for the timeout and return with
JOINUNAVAIL after 1 round of testing the peers?
Thanks.
Matt
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Re: [asterisk-users] Deleting AstDB family at start
Thanks Paul. The following works.. --cli.conf --- [startup_commands] ; ; Any commands listed in this section will get automatically executed ; when Asterisk starts as a daemon or foreground process (-c). ; ;sip set debug on = yes ;core set verbose 3 = yes ;core set debug 1 = yes database deltree example = yes --- Matt > Date: Mon, 21 Nov 2011 16:33:47 -0500 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Deleting AstDB family at start > > On 11-11-21 03:46 PM, Steve Edwards wrote: > > On Sun, 20 Nov 2011, Matt Hamilton wrote: > > > >> Is it possible to delete the keys belonging to a family in AstDB at > >> Asterisk startup? I would like to repopulate it from another source > >> each time Asterisk is restarted. > > > > How about: > > > > [sudo] /usr/sbin/asterisk -r -x 'database deltree example' > > > > in /etc/init.d/asterisk or safe_asterisk? > > > Easier to use cli.conf > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > 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] CDR uniqueid - across multiple servers?
Mike, Just enter a unique "systemname" into asterisk.conf for each box. This system identifier is appended to the front of the unique id field in cdr. /etc/asterisk/asterisk.conf [options] systemname=asterisk1 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 21 Nov 2011 13:50:26 -0600 Subject: Re: [asterisk-users] CDR uniqueid - across multiple servers? Since the MYSQL CDR is not the standard /var/log/asterisk/cdr-csv/Master.csv file, but an add_on where uniqueid is just a table field varchar(32), you could create an AGI to touch the field during the hangup extension and append the servername or a number to the front, so instead of 123456.111 you could have server1.123456.111 or you could make a daemon running outside of Asterisk to do the same thing. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, November 21, 2011 11:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] CDR uniqueid - across multiple servers? Hi, Is there a way to add a uniqueid prefix to each server to make sure that the CDRs uniqueids are indeed unique across multiple servers? I am using MYSQL tables to keep these records. 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 -- _ -- 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] Deleting AstDB family at start
Thanks Danny. > [clearkeys]> Exten => start,1,answer()> Exten => start,n,dbdeltree(foo)> > Exten => start,n,hangup > Set and retrieve Global variables for small > searches. I will try the "local call" option to [clearkeys]. I guess I can also use a global flag to call dbdeltree only once in the existing context before entering anything into AstDB. Matt -- _ -- 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
[asterisk-users] check if devices reachable in queue
I would like to perform 2 checks on a queue:
1. if the caller stays in the queue for a certain time, I would like to forward
him to phone A.
2. if the devices/members in the queue are not reachable, I would like to
forward him to a phone B.
The first one is straight-forward via the timeout.
I'm looking for a fast/practical way of accomplishing the second one. In other
words, before sending a call to a queue, I would like to see if the
members/devices in that queue are available/reachable.
I define the members statically in queue.conf and QUEUE_MEMBER_COUNT gives the
count of those - doesn't care if they are available/reachable or not (even if
phone is unhooked, still counted).
I should be able to loop through each member and use ${DEVICE_STATE()}. for every incoming call, isn't this overkill? Any other way?
Thanks a lot,
Matt
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[asterisk-users] Deleting AstDB family at start
Is it possible to delete the keys belonging to a family in AstDB at Asterisk startup? I would like to repopulate it from another source each time Asterisk is restarted. I know there is a DBdeltree() function. Is there a context that only runs once (automatically) at Asterisk startup (so that I can call this function)? Also is AstDB lookup faster than a func_odbc lookup? Is there a faster way to perform a lookup in Asterisk; e.g. create a lookup table in memory perhaps? I'm new to Asterisk... Thanks, Matt -- _ -- 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
