Re: [asterisk-users] Siemens Gigaset + Asterisk Query?
Hi! > I was wondering if you can use the base station as a outbound pots > connection for asterisk. > > I currently have a tdm410 to do fxs/fxo ports and would like to get rid of > it, I used to use a spa3102, but it only had 1 fxo (telephone connector). > I like the idea of the siemans but I would like to control the pots > fallover from asterisk. > > if not the siemans are there any other bases that would fit the bill ? The AVM Fritz!Box 7270 could do the job, but I am not sure if you can get that with an English language web interface. DECT and SIP registrar (for LAN only) are available with a recent firmware. This box might be a bit oversized for what you are trying to do, though. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ivvr with asterisk
On 01/22/2010 09:02 PM, Pham Quy wrote: > Hi all, > > First I'm very new. I want to build an Interactive Video-voice Response > system. There is number of choice I have found so far: FreePBX, TriBox, > Asterisk. > > Which is the best in my case? and what do i need to build such IVVR > system? All use Asterisk underneath; FreePBX and Trixbox are simply administrative GUI layers that provide you with a different way to manage the configuration files. Using Asterisk straight would require that you edit them by hand. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ivvr with asterisk
Quyps, It looks like you mis-read the picture. Asterisk is the core, it need to be there regardless you use FreePBX or Tribox. FreePBX is a GUI web interface to manage asterisk. Itself is not an IP-PBX. Trixobx, still based on the Asterisk + freePBX, adds some more additional applications based on the community feed back and requirement. Trixbox is an easy go, but there may be some unwanted stuff with it. elastix.org is also a nice package, give it a try. Regards MT Kondela kevesystems.com On Sat, Jan 23, 2010 at 7:32 AM, Pham Quy wrote: > Hi all, > > First I'm very new. I want to build an Interactive Video-voice Response > system. There is number of choice I have found so far: FreePBX, TriBox, > Asterisk. > > Which is the best in my case? and what do i need to build such IVVR > system? > > Thanks. > Quyps > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set CDR userfield for Queues
I just added a line with 'h'extension.
My dialplan is like this
[mycontext]
exten => s,1,Queue(6000)
exten => h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
On Sat, Jan 23, 2010 at 12:14 AM, William Stillwell (Lists)
wrote:
> "setinterfacevar=yes"
>
> Needs to be under each queue
>
> What does your dialplan end up looking like?
>
> I would like to add to mine, and stop running a cron job..
>
> exten => 5000,1,Answer
> exten => 5000,n,Queue(5000|rn)
> exten => 5000,n,VoiceMail(5000,u)
> exten => 5000,n,Hangup
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
> Sent: Friday, January 22, 2010 1:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Set CDR userfield for Queues
>
> The 'h' extension worked. Thanks.
>
> The other option of 'memebermacro' did not work. On the asterisk
> console I could see that the macro is executed and cdr userfield is
> set when agent answers the call, but the userfield doesn't show up in
> the generated cdr.
>
> Also I had one more question. Doesn't "setinterfacevar=yes" work when
> it is declared in the general section? I had to declare it for each
> queues.
>
>
>
> On Fri, Jan 22, 2010 at 10:37 PM, Carlos Chavez
> wrote:
>> On Fri, 2010-01-22 at 20:25 +0530, Deep D wrote:
>>> I want to do something like this
>>> exten = 1234,n,Queue(6000,c)
>>> exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
>>> agent who answered the call
>>> exten = 1234,n,Hangup
>>>
>> Actually because the user will hangup within the Queue application
> you
>> cannot do that. You will have to use the h extension to make the change
>> to the userfield. Something like this:
>>
>> h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
>>
>> Make sure you have setinterfacevar=yes in your queue.conf so that
>> variable is created when the user is connected to the agent. Another
>> possibility is to run a macro by using "membermacro=somemacro" and set
>> the userfield within that macro. I think that option is only available
>> on Asterisk 1.6.X and not for older ones though. You can also run an
>> AGI script (you can set it as an option in the Queue commando) that will
>> set the userfield as this AGI is run just before the call is bridged to
>> the agent but the ${MEMBERINTERFACE} is already set.
>>
>>
>> --
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Carlos Chávez Prats
>> Director de Tecnología
>> +52-55-91169161 ext 2001
>>
>> --
>> _
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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> asterisk-users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Re: [asterisk-users] Siemens Gigaset + Asterisk Query?
Hi I was wondering if you can use the base station as a outbound pots connection for asterisk. I currently have a tdm410 to do fxs/fxo ports and would like to get rid of it, I used to use a spa3102, but it only had 1 fxo (telephone connector). I like the idea of the siemans but I would like to control the pots fallover from asterisk. if not the siemans are there any other bases that would fit the bill ? Alex On Fri, Jan 22, 2010 at 07:14:56PM -0500, John Hurley wrote: > From my experience, unless you have another base station for sets you would > want to configure separately it is not possible. > > I may be wrong, hopefully. > > Sent from my Android phone > > -Original Message- > From: Alan Lord (News) [alansli...@gmail.com] > Received: 1/22/10 7:01 PM > To: asterisk-users@lists.digium.com [asterisk-us...@lists.digium.com] > Subject: [asterisk-users] Siemens Gigaset + Asterisk Query? > > When you configure the Siemens gigaset handsets (I have S685IP), there > is a single option for all handsets to use either the POTS interface or > VOIP as the default outbound destination - you then need to add a dial > suffix if you want to use an alternate outbound route. > > Does anyone have any suggestions as to how to make just *one* of the > DECT handsets only use the POTS but others default to their Asterisk SIP > subscriptions? > > The POTS is on the Gigaset base station & not on the Asterisk server. > > TIA. > > Al > -- "If you found somebody that had information about an attack on America, you'd want to know as best as we can to find out what the facts are." - George W. Bush 12/12/2005 Philadelphia, PA signature.asc Description: Digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ivvr with asterisk
Hi all, First I'm very new. I want to build an Interactive Video-voice Response system. There is number of choice I have found so far: FreePBX, TriBox, Asterisk. Which is the best in my case? and what do i need to build such IVVR system? Thanks. Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset + Asterisk Query?
> > Does anyone have any suggestions as to how to make just *one* of the > DECT handsets only use the POTS but others default to their Asterisk SIP > subscriptions? > Hi Al, I've played with the Siemens Gigaset in the past and I don't recall being able to do this. Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX ans SS7
On 01/22/2010 07:12 PM, mickael ropars wrote: > Hi all, > > what is the signalling of IAX? > > Currently I want to connect two switch through IP using asterik > signaling, and I want to transfer SS7 over IP (between the 2 asterisk), > will IAX can transfer SS7 signalling through IP (like TDMoIP does) If no > which solution can I use? > > see below the architecture > > switch1-Asterisk--Asterisk > --Switch >E1 IP > E1 > To some extent, yes, but only very basically. IAX2 doesn't have the ability to pass through ISUP attributes. For that, you would need SIP-T. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset + Asterisk Query?
>From my experience, unless you have another base station for sets you would >want to configure separately it is not possible. I may be wrong, hopefully. Sent from my Android phone -Original Message- From: Alan Lord (News) [alansli...@gmail.com] Received: 1/22/10 7:01 PM To: asterisk-users@lists.digium.com [asterisk-us...@lists.digium.com] Subject: [asterisk-users] Siemens Gigaset + Asterisk Query? When you configure the Siemens gigaset handsets (I have S685IP), there is a single option for all handsets to use either the POTS interface or VOIP as the default outbound destination - you then need to add a dial suffix if you want to use an alternate outbound route. Does anyone have any suggestions as to how to make just *one* of the DECT handsets only use the POTS but others default to their Asterisk SIP subscriptions? The POTS is on the Gigaset base station & not on the Asterisk server. TIA. Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX ans SS7
Hi all, what is the signalling of IAX? Currently I want to connect two switch through IP using asterik signaling, and I want to transfer SS7 over IP (between the 2 asterisk), will IAX can transfer SS7 signalling through IP (like TDMoIP does) If no which solution can I use? see below the architecture switch1-Asterisk--Asterisk --Switch E1 IP E1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siemens Gigaset + Asterisk Query?
When you configure the Siemens gigaset handsets (I have S685IP), there is a single option for all handsets to use either the POTS interface or VOIP as the default outbound destination - you then need to add a dial suffix if you want to use an alternate outbound route. Does anyone have any suggestions as to how to make just *one* of the DECT handsets only use the POTS but others default to their Asterisk SIP subscriptions? The POTS is on the Gigaset base station & not on the Asterisk server. TIA. Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Handling SIP error codes/ISDN codes
Hi,
I was trying to use 2 of asterisk servers and interconnected, one of them as
a peer to other sever (configured in sip.conf), so all the calls to server 1
will just be passed to server 2 (has PRI Card, TE 412P, only one PRI
connected), i was sending calls to server 1 and that would send to server 2
and then dial out using Dahdi, but the problem that i got was the hangup
cause codes, i was not able to pass the appropriate ones to the server 0
(Test server) that sent a call to server 1. For example, when the user at
server 0 (test server) made a call to server 1, that sends it to server 2
and connects to the appropriate destination, but in the mean while if we
just cancel the call, we need to see the SIP error code as 487 - Request
terminated, but I was only able to see the ISDN core in PRI debug on server
2, but was not able to see '487' in sip debug, even though if i am handling
the error code conditionsIs there any way of handling the error
codes properly?
Asterisk version: 1.4.22.1
Libpri: 1.4.10.1
dahdi: 2.2.0.2 are the versions that I am using.
The way I was handling the codes for the server 2:
[macro-result]
exten => s,1,Wait(1)
exten => s,2,ResetCDR(w)
exten => s,3,NoCDR()
exten => s,4,GotoIf($[${ISNULL(${ARG1})}]?7:5)
exten => s,5,Set(RC=${ARG1})
exten => s,6,Goto(s|9)
exten => s,7,GotoIf($[${ISNULL(${DIALSTATUS})}]?8:rc-${DIALSTATUS}|1)
exten => s,8,Set(RC=${IF($[${ISNULL(${HANGUPCAUSE})}]?0:${HANGUPCAUSE})})
exten => s,9,Goto(rc-${RC}|1)
exten => s,10,Hangup(${RC})
exten => i,1,Set(RC=0)
exten => i,2,Goto(s|9)
exten => rc-ANSWER,1,Set(RC=16)
exten => rc-ANSWER,2,Goto(s|9)
exten => rc-BUSY,1,Set(RC=17)
exten => rc-BUSY,2,Goto(s|9)
exten => rc-CANCEL,1,Set(RC=16)
exten => rc-CANCEL,2,Goto(s|9)
exten => rc-CHANUNAVAIL,1,Set(RC=44)
exten => rc-CHANUNAVAIL,2,Goto(s|9)
exten => rc-CONGESTION ,1,Set(RC=19)
exten => rc-CONGESTION ,2,Goto(s|9)
;exten => rc-NOANSWER,1,Set(RC=19)
;exten => rc-NOANSWER,2,Goto(s|9)
exten => rc-0,1,NoOp(NOTDEFINED)
exten => rc-0,n,Goto(s|10)
exten => rc-1,1,NoOp(UNALLOCATED)
exten => rc-1,n,Goto(s|10)
exten => rc-2,1,NoOp(NO_ROUTE_TRANSIT_NET)
exten => rc-2,n,Goto(s|10)
exten => rc-3,1,NoOp(NO_ROUTE_DESTINATION)
exten => rc-3,n,Goto(s|10)
exten => rc-6,1,NoOp(CHANNEL_UNACCEPTABLE)
exten => rc-6,n,Goto(s|10)
exten => rc-7,1,NoOp(CALL_AWARDED_DELIVERED)
exten => rc-7,n,Goto(s|10)
exten => rc-16,1,NoOp(NORMAL_CLEARING)
exten => rc-16,n,Goto(s|10)
exten => rc-17,1,NoOp(USER_BUSY)
;exten => rc-17,n,Busy()
exten => rc-17,n,Goto(s|10)
exten => rc-18,1,NoOp(NO_USER_RESPONSE)
exten => rc-18,n,Goto(s|10)
exten => rc-19,1,NoOp(NO_ANSWER)
exten => rc-19,n,Goto(s|10)
exten => rc-21,1,NoOp(CALL_REJECTED)
exten => rc-21,n,Goto(s|10)
exten => rc-28,1,NoOp(INVALID_NUMBER_FORMAT)
exten => rc-28,n,Goto(s|10)
exten => rc-29,1,NoOp(FACILITY_REJECTED)
exten => rc-29,n,Goto(s|10)
exten => rc-30,1,NoOp(RESPONSE_TO_STATUS_ENQUIRY)
exten => rc-30,n,Goto(s|10)
exten => rc-31,1,NoOp(NORMAL_UNSPECIFIED)
exten => rc-31,n,Goto(s|10)
exten => rc-34,1,NoOp(NORMAL_CIRCUIT_CONGESTION)
exten => rc-34,n,Congestion()
exten => rc-34,n,Goto(s|10)
Thank you for your help.
Regards
Sandesh
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Re: [asterisk-users] queue groups in asterisk 1.4
We run with ringall strategy, and have had shared_lastcall on for six months. afaik, this only shares the last call data for things like wrapup time; it definitely does not share how the queues with ringall hand off their queued calls. As for the patch below, I have since tried it on 1.4.29, and it does not apply cleanly due to the interface_state patch from digium added from 1.6, as well as some other minor code changes. I have added a patch for 1.4.29 to the same directory listed below. It compiles and passes very basic testing, but has not been run in production. -Steve On 01/22/2010 07:53 AM, Lenz Emilitri wrote: Maybe I'm saying something stupid, but I thought this was what shared_lastcall would do with a leastrecent strategy. ; shared_lastcall will make the lastcall and calls received be the same in ; members logged in more than one queue. ; This is useful to make the queue respect the wrapuptime of another queue ; for a shared member ; shared_lastcall=no I did not really try it, just read the docs. l. 2010/1/20 Steven Alligood mailto:st...@bluehost.com>> This email is not a question, but a potential solution to any who have had the same issue I have faced. If you have agents logged in to multiple queues at the same time, Asterisk does not handle the answering of those queues in any set order or sequence. It has no way of prioritizing calls in between the shared queues, to guarantee that the queue with the call that has the longest hold time will be answered next. If Asterisk is configured with several queues and all queues have calls waiting, when an Agent becomes available, Asterisk randomly picks the queue that will be answered next. This can be a problem adversely affecting service levels since the queue with the call having the longest hold time may be essentially ignored for several cycles of available agents. To fix this issue, we had a custom patch written for the 1.4 branch. Digium has decided to not include it in the 1.6 branch, so I am making the 1.4 patch available for any who want or need it. It works well on 1.4.19.1, but I have not tried it on anything after that (I am currently running 1.6 and have the issue back again). Questions are always welcome, though I may or may not have the answers, as I am not the coder, just the asterisk admin. As always, if you need it/ want it in your production systems, it would be good to send an email to Digium letting them know that. -Steve Patch at: http://mirrors.bluehost.com/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom vs Polycom
On Fri, Jan 22, 2010 at 05:06:17PM -0300, Andrew Latham wrote: > I have worked on many snom phones over the years I have never had > a snom phone go bad... I have had about 10 in the last 12-18 months, I had 1 with a fault hand set plug - the reseller replaced it. Other wise they have been great. > > I have repaired stuck screens and overheated sticky bits but all in > all snom are great phones. I recently showed my personal phone to > some people including a VoIP engineer that fell in love with my snom > 360... It has scratches and dents from use and abuse but the real > shocker was when I turned the phone over and showed them the date of > manufacture ... 2004... > > I am working with some Polycom phones right now. They look ok. I > don't hear any "better speaker" that people talk about. They are > troublesome to administrate or provision as a single phone. provision the snom is a "pain" to setup - well you have to do a lot of work, but its work while - lots of documentation on the snom site on mass distribution - makes life simple I would recommend it for anything more than 3 phones (all you need is a deamon provided by snom written in perl and a http server) [snip] > signature.asc Description: Digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom vs Polycom
- "Andrew Latham" wrote: > Having demo phones is priceless. Sometimes I show off the phones I > like with some phones I don't like to show the end users why it > matters... Bad screens, cost, features, color (snom snow rocks!), > etc... Anybody else remember the white Grandstream GXP101's with the shiny 'pearl' buttons? I haven't seen one of those since 2006, 2005 maybe. Now *THAT* was a good looking phone. :-D --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom vs Polycom
I have worked on many snom phones over the years I have never had a snom phone go bad... I have repaired stuck screens and overheated sticky bits but all in all snom are great phones. I recently showed my personal phone to some people including a VoIP engineer that fell in love with my snom 360... It has scratches and dents from use and abuse but the real shocker was when I turned the phone over and showed them the date of manufacture ... 2004... I am working with some Polycom phones right now. They look ok. I don't hear any "better speaker" that people talk about. They are troublesome to administrate or provision as a single phone. With Polycom the LDAP requires an additional license. With snom the firmware is mostly OSS and you can download and play. It does really cool things with LDAP and HTTP... I have worked with Aastra and Grandstream also in the past amongst others and I have found that some models are better than others and each vendor must make something good once in a while just to stay in business... Having demo phones is priceless. Sometimes I show off the phones I like with some phones I don't like to show the end users why it matters... Bad screens, cost, features, color (snom snow rocks!), etc... ~ Andrew "lathama" Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Fri, Jan 22, 2010 at 4:30 PM, Doug Lytle wrote: > Randy R wrote: >> http://twitpic.com/z8n36 >> >> > > > *snicker* > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom vs Polycom
Randy R wrote: > http://twitpic.com/z8n36 > > *snicker* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom vs Polycom
http://twitpic.com/z8n36 On Fri, Jan 22, 2010 at 8:11 PM, Philipp von Klitzing wrote: > Hey hey! > >> Anyone got any subjective (!) views on the merits of these two ranges >> , using asterisk 1.4 ? I need to supply approx 30 handsets to a new >> client, with the senior managers (6) having some slightly more >> "managerial" phones -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom vs Polycom
Hey hey! > Anyone got any subjective (!) views on the merits of these two ranges > , using asterisk 1.4 ? I need to supply approx 30 handsets to a new > client, with the senior managers (6) having some slightly more > "managerial" phones * Let the customer test and decide himself * Polycom: great speaker phone, but no built-in BLF * Snom: poor speaker phone, many BLF buttons & more * Configuration and flexibility is great with the SNOMs, if you like you can even stream video to a snom 370 or 820 * The snom 8xx firmware still has its quirks (not fully mature yet), so I'd say better stick with the 3xx series for the moment Yours, Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set CDR userfield for Queues
"setinterfacevar=yes"
Needs to be under each queue
What does your dialplan end up looking like?
I would like to add to mine, and stop running a cron job..
exten => 5000,1,Answer
exten => 5000,n,Queue(5000|rn)
exten => 5000,n,VoiceMail(5000,u)
exten => 5000,n,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
Sent: Friday, January 22, 2010 1:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set CDR userfield for Queues
The 'h' extension worked. Thanks.
The other option of 'memebermacro' did not work. On the asterisk
console I could see that the macro is executed and cdr userfield is
set when agent answers the call, but the userfield doesn't show up in
the generated cdr.
Also I had one more question. Doesn't "setinterfacevar=yes" work when
it is declared in the general section? I had to declare it for each
queues.
On Fri, Jan 22, 2010 at 10:37 PM, Carlos Chavez
wrote:
> On Fri, 2010-01-22 at 20:25 +0530, Deep D wrote:
>> I want to do something like this
>> exten = 1234,n,Queue(6000,c)
>> exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
>> agent who answered the call
>> exten = 1234,n,Hangup
>>
> Actually because the user will hangup within the Queue application
you
> cannot do that. You will have to use the h extension to make the change
> to the userfield. Something like this:
>
> h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
>
> Make sure you have setinterfacevar=yes in your queue.conf so that
> variable is created when the user is connected to the agent. Another
> possibility is to run a macro by using "membermacro=somemacro" and set
> the userfield within that macro. I think that option is only available
> on Asterisk 1.6.X and not for older ones though. You can also run an
> AGI script (you can set it as an option in the Queue commando) that will
> set the userfield as this AGI is run just before the call is bridged to
> the agent but the ${MEMBERINTERFACE} is already set.
>
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
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Re: [asterisk-users] Set CDR userfield for Queues
The 'h' extension worked. Thanks.
The other option of 'memebermacro' did not work. On the asterisk
console I could see that the macro is executed and cdr userfield is
set when agent answers the call, but the userfield doesn't show up in
the generated cdr.
Also I had one more question. Doesn't "setinterfacevar=yes" work when
it is declared in the general section? I had to declare it for each
queues.
On Fri, Jan 22, 2010 at 10:37 PM, Carlos Chavez wrote:
> On Fri, 2010-01-22 at 20:25 +0530, Deep D wrote:
>> I want to do something like this
>> exten = 1234,n,Queue(6000,c)
>> exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
>> agent who answered the call
>> exten = 1234,n,Hangup
>>
> Actually because the user will hangup within the Queue application you
> cannot do that. You will have to use the h extension to make the change
> to the userfield. Something like this:
>
> h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
>
> Make sure you have setinterfacevar=yes in your queue.conf so that
> variable is created when the user is connected to the agent. Another
> possibility is to run a macro by using "membermacro=somemacro" and set
> the userfield within that macro. I think that option is only available
> on Asterisk 1.6.X and not for older ones though. You can also run an
> AGI script (you can set it as an option in the Queue commando) that will
> set the userfield as this AGI is run just before the call is bridged to
> the agent but the ${MEMBERINTERFACE} is already set.
>
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Hardware issue, was Using HASH() and REALTIME_HASH()
Le 13/01/2010 09:57, Benoit a écrit : > Le 12/01/2010 16:35, Tilghman Lesher a écrit : > >> On Tuesday 12 January 2010 04:44:36 Benoit wrote: >> >> >>> I just experienced another problem however i have two rnis cards, one >>> b410p and one te220, >>> while the later works prefectly i can't really make the first one work, >>> using DAHDI or mISDN >>> under asterisk 1.6. >>> >>> >> If you're having trouble with any Digium hardware, the best thing to do is to >> call Digium support and get your free installation support provided with our >> hardware. >> >> >> > Hi, > > I didn't think of this, since it looked like more of an asterisk problem > (asterisk 1.4/misdn => ok asterisk 1.6/misdn => fail, asterisk 1.6/dahdi > => fail). > > Audio (both way) is working (voicemail/playback), but it fail when > Dial'ing a device. > Looks like a problem with signalling ... > > But anyway i just opened a support case, thanks > Well, in fact it wasn't an hardware issue: when calling thru the B410p the callerid string is prepended with an Id, looks like the length of the resulting string is a problem to initial a SIP call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
On Fri, 22 Jan 2010, Zhang Shukun wrote: > as you know, we can use MYSQL command to visit mysql database > > but if i use other database like Oracke,sybase,etc, Could i use MYSQL > command ? ODBC will do what you want. Personally, I'd vote for an AGI using whatever C API your DB provides -- like Pro*C to access Oracle. You will have access to all of the features of your DB and your dialplan will be a lot cleaner and easier to maintain. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc question
On Friday 22 January 2010 09:53:33 Giedrius Augys wrote:
> 2010/1/22 Tilghman Lesher
>
> > On Friday 22 January 2010 01:49:13 Giedrius Augys wrote:
> > > 2010/1/21 Tilghman Lesher wrote:
> > > > On Thursday 21 January 2010 09:51:13 Giedrius Augys wrote:
> > > > > Is it possible to free idle connections? When limit was 40, I had
> >
> > lost
> >
> > > > > part of data. My asterisk version is 1.6.0.20 .
> > > >
> > > > We intentionally do not, since the maximum number of connections is
> > > > always the maximum concurrent number of queries, and there's a
> > > > significant penalty in establishing a connection.
> > >
> > > Yes, I'm using FreeTDS. In Asterisk Console I can see 200 SET command
> >
> > (for
> >
> > > inserting to MSSQL: Set("Local/123456...@from-trunk-2f45;2",
> > > "ODBC_INSERT_RECORD()=11,123456789")), but in MS SQL I can see only
> >
> > 60
> >
> > > or 43 records. Maybe it's possible wait some time , if all connections
> >
> > are
> >
> > > busy and retry again insert records?
> >
> > Are you getting an error or a warning? We cannot begin to guess at what
> > might
> > be wrong without seeing more information.
>
> No errors and warnings! If you need, I can make a test again.
My guess is that there's actually nothing wrong, then, and your SQL query is
not doing what you think it should be doing.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Set CDR userfield for Queues
On Fri, 2010-01-22 at 20:25 +0530, Deep D wrote:
> I want to do something like this
> exten = 1234,n,Queue(6000,c)
> exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
> agent who answered the call
> exten = 1234,n,Hangup
>
Actually because the user will hangup within the Queue application you
cannot do that. You will have to use the h extension to make the change
to the userfield. Something like this:
h,1,Set(CDR(userfield)=${MEMBERINTERFACE})
Make sure you have setinterfacevar=yes in your queue.conf so that
variable is created when the user is connected to the agent. Another
possibility is to run a macro by using "membermacro=somemacro" and set
the userfield within that macro. I think that option is only available
on Asterisk 1.6.X and not for older ones though. You can also run an
AGI script (you can set it as an option in the Queue commando) that will
set the userfield as this AGI is run just before the call is bridged to
the agent but the ${MEMBERINTERFACE} is already set.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Asterisk 403 Forbidden message with port translation
Hello, I managed to get it working. Seems like i was overwriting fields used in computation of the digest response. Once i turn off authentication the call flow works perfectly. I will need to make necessary modifications to work with digest authentication. As a next step i will be implementing encryption/decryption on the F.W server. Thanks and Regards, Vikram. Vikram Ragukumar wrote: > Hello, > > - --- > |Sip Softphone|---|Internet||F.W|-|Asterisk| > - --- >IP addresses: a.b.c.dq.w.e.r > > The SIP softphone(x-lite) is configured to register with the asterisk > server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as > the outbound proxy for the softphone(Outbound proxy a.b.c.d:9090). > Authentication credentials for the softphone match the user registered > in asterisk's sip.conf. F.W runs Kamailio and rtpproxy, with Kamailio > listening on port 5060. > > The asterisk server is setup to listen on port 5060. > > The Firewall(F.W), uses a libnetfilter_queue based program to : > > (a) Rewrite the destination port 9090 as 5060, and rewrite all other > occurrences of 9090 as 5060 in the SIP message, for packets from the > softphone to the asterisk server. > > (b) Rewrite the source port 5060 as 9090, and rewrite all other > occurrences of 5060 as 9090 in the SIP message, for packets from the > asterisk server to the softphone. > > The following exchange of SIP messages take place > -Sip softphone sends a REGISTER message to asterisk > -Asterisk responds with a 401 UNAUTHORIZED > -Sip softphone replies with a REGISTER message containing auth. info. > -Asterisk responds with a 403 FORBIDDEN : BAD AUTHORIZATION > > The above setup works when the softphone uses port 5060, so there > problem here does not have anything to do with Authorization credentials. > > Is it possible i might be modifying parts of the packet that shouldn't > be modified or i might not be modifying some relevant parts of the packet ? > > Thanks in advance, > Vikram. > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom phone DND state
Hi, I know having Asterisk aware of Polycom "Do No Disturb" state wasn't working before (1.4), but is this working in any recent version? Is there any "custom" way of doing this? Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc question
2010/1/22 Tilghman Lesher
> On Friday 22 January 2010 01:49:13 Giedrius Augys wrote:
> > 2010/1/21 Tilghman Lesher wrote:
> > > On Thursday 21 January 2010 09:51:13 Giedrius Augys wrote:
> > > > Is it possible to free idle connections? When limit was 40, I had
> lost
> > > > part of data. My asterisk version is 1.6.0.20 .
> > >
> > > We intentionally do not, since the maximum number of connections is
> > > always the maximum concurrent number of queries, and there's a
> > > significant penalty in establishing a connection.
> >
> > Yes, I'm using FreeTDS. In Asterisk Console I can see 200 SET command
> (for
> > inserting to MSSQL: Set("Local/123456...@from-trunk-2f45;2",
> > "ODBC_INSERT_RECORD()=11,123456789")), but in MS SQL I can see only
> 60
> > or 43 records. Maybe it's possible wait some time , if all connections
> are
> > busy and retry again insert records?
>
> Are you getting an error or a warning? We cannot begin to guess at what
> might
> be wrong without seeing more information.
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
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Hi,
No errors and warnings! If you need, I can make a test again.
--
Pagarbiai / Best Regards,
Giedrius
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Re: [asterisk-users] Snom vs Polycom
Julian Lyndon-Smith wrote: > Anyone got any subjective (!) views on the merits of these two ranges > , using asterisk 1.4 ? > > I need to supply approx 30 handsets to a new client, with the senior > managers (6) having some slightly more "managerial" phones than the > base phones which will be used for one line only. > > TIA > > Julian Surely you mean objective rather than subjective? I work for a hosted VoIP provider and we use and resell snom pretty exclusively and I think they are a very good, feature rich and easy to use handset. I've played with Polycoms a couple of times and thought they looked a bit cheap (although costing much more!) and from an admin point of view were not nearly as good as the snom. But that's just my subjective opinion! Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odbc question
On Friday 22 January 2010 01:49:13 Giedrius Augys wrote:
> 2010/1/21 Tilghman Lesher wrote:
> > On Thursday 21 January 2010 09:51:13 Giedrius Augys wrote:
> > > Is it possible to free idle connections? When limit was 40, I had lost
> > > part of data. My asterisk version is 1.6.0.20 .
> >
> > We intentionally do not, since the maximum number of connections is
> > always the maximum concurrent number of queries, and there's a
> > significant penalty in establishing a connection.
>
> Yes, I'm using FreeTDS. In Asterisk Console I can see 200 SET command (for
> inserting to MSSQL: Set("Local/123456...@from-trunk-2f45;2",
> "ODBC_INSERT_RECORD()=11,123456789")), but in MS SQL I can see only 60
> or 43 records. Maybe it's possible wait some time , if all connections are
> busy and retry again insert records?
Are you getting an error or a warning? We cannot begin to guess at what might
be wrong without seeing more information.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Setting MixMonitor options from Queue
Unless you're using MixMonitor to actively monitor calls, the optimum solution would indeed (IMO) be postprocess adjusting with SOX or something similar. That being said, >From this link; http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf I got this notion monitor-format = wav49 wav49 presents much louder than regular wav and gsm in my experience -- _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Friday, January 22, 2010 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Setting MixMonitor options from Queue I know this is not what you need, but you might postprocess recordings to raise the volume level. I know this is not optimal but it's a start. l. 2010/1/21 Scott Gifford Hello, We are recording our calls to queues by putting the appropriate options in our "queue.conf". This is all working properly. We would now like to set the MixMonitor option to adjust the caller volume (which is very quiet). With the regular MixMonitor application, we would just add the "v4" option to make it much louder. I don't see a way to set this option when MixMonitor is started from the queue. Does anybody know of a way to set MixMonitor options from the queue.conf? Thanks, Scott. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set CDR userfield for Queues
I want to do something like this
exten = 1234,n,Queue(6000,c)
exten = 1234,n,Set(CDR(userfield)=${Agent}) ;; where Agent is the
agent who answered the call
exten = 1234,n,Hangup
On Fri, Jan 22, 2010 at 8:01 PM, Danny Nicholas wrote:
> Since userfield is a variable, that would be a qualified yes.
> Here's how I use it in my dialplan
>
> - exten => NXX,1,noop(Answer())
> - exten => NXX,n,Verbose (Dial DAHDI g1 ${EXTEN})
> - exten => NXX,n,Dial(DAHDI/g1/${EXTEN},60,miKkTtg)
> - exten => NXX,n,Set(CDR(userfield)=${CDR(userfield)}
> -Hangupcause:${HANGUPCAUSE})
> - exten => NXX,n,Hangup(${HANGUP_CAUSE})
>
> How I envision what you're asking:
> - exten => s,n,Set(CDR(userfield)=${CDR(userfield)} -Agent:${AGENT})
>
> --
> Danny Nicholas
> --
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
> Sent: Friday, January 22, 2010 8:23 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Set CDR userfield for Queues
>
> Hello,
>
> I am using Queue application with multiple agents in each queue. I
> want to set the CDR(userfield) for each cdr based on the agent
> answering the call. Is it possible to do this?
>
> Thanks
>
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>
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Re: [asterisk-users] queue groups in asterisk 1.4
Maybe I'm saying something stupid, but I thought this was what shared_lastcall would do with a leastrecent strategy. ; shared_lastcall will make the lastcall and calls received be the same in ; members logged in more than one queue. ; This is useful to make the queue respect the wrapuptime of another queue ; for a shared member ; shared_lastcall=no I did not really try it, just read the docs. l. 2010/1/20 Steven Alligood > This email is not a question, but a potential solution to any who have had > the same issue I have faced. > > If you have agents logged in to multiple queues at the same time, Asterisk > does not handle the answering of those queues in any set order or sequence. > It has no way of prioritizing calls in between the shared queues, to > guarantee that the queue with the call that has the longest hold time will > be answered next. > > If Asterisk is configured with several queues and all queues have calls > waiting, when an Agent becomes available, Asterisk randomly picks the queue > that will be answered next. This can be a problem adversely affecting > service levels since the queue with the call having the longest hold time > may be essentially ignored for several cycles of available agents. > > To fix this issue, we had a custom patch written for the 1.4 branch. > Digium has decided to not include it in the 1.6 branch, so I am making the > 1.4 patch available for any who want or need it. It works well on 1.4.19.1, > but I have not tried it on anything after that (I am currently running 1.6 > and have the issue back again). > > Questions are always welcome, though I may or may not have the answers, as > I am not the coder, just the asterisk admin. > > As always, if you need it/ want it in your production systems, it would be > good to send an email to Digium letting them know that. > > -Steve > > Patch at: http://mirrors.bluehost.com/asterisk > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
On Friday 22 January 2010 02:06:08 Zhang Shukun wrote: > as you know, we can use MYSQL command to visit mysql database > > but if i use other database like Oracke,sybase,etc, Could i use MYSQL > command ? > > if not, is there any other alternative could do the same > function(visit database in dailplan)? func_odbc will do what you want. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GoToIfTime issue
On Friday 22 January 2010 04:06:29 Zhang Shukun wrote: > 2010/1/22 Randy R : > > On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun wrote: > >> exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1) > >> > >> but what should i do. if i want to set seperate weekdays,like mon,wed. > >> not continuous weekday like mon-fri. > > > > I couldn't find any reference to multiple, non-contiguous days on a > > quick Google, but this would work at the cost of an extra line: > > > > exten => 222,1,GoToIfTime(11:00-14:00|mon|*|*?1:3,1) > > exten => 222,2,GoToIfTime(11:00-14:00|wed|*|*?1:3,1) > > Thank you, but why don't it to be comma seperate to represent seperate > weekdays? as | mon,wed,fri | > > it's also very understandable. Starting in 1.6.2, you can use the ampersand to join days. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble getting feature codes to work
hugolivude wrote: > I have this in features.conf: > > [applicationmap] > testfeature1 => #9,caller,Playback,tt-monkeys > testfeature2 => #8,callee,Playback,tt-monkeys > > and this in the context where the dial takes place: > > include => featuremap > include => applicationmap You need to re-read the sample features.conf; the categories of features defined there are *not* dialplan contexts, so using "include =>" for them in the dialplan is not going to accomplish anything except to generate a warning message when the dialplan is parsed. Immediately below the [applicationmap] category heading in the sample features.conf it describes exactly how to enable the features you have defined in that category. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting MixMonitor options from Queue
I know this is not what you need, but you might postprocess recordings to raise the volume level. I know this is not optimal but it's a start. l. 2010/1/21 Scott Gifford > Hello, > > We are recording our calls to queues by putting the appropriate options in > our "queue.conf". This is all working properly. > > We would now like to set the MixMonitor option to adjust the caller volume > (which is very quiet). With the regular MixMonitor application, we would > just add the "v4" option to make it much louder. I don't see a way to set > this option when MixMonitor is started from the queue. > > Does anybody know of a way to set MixMonitor options from the queue.conf? > > Thanks, > > Scott. > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set CDR userfield for Queues
Let me know if you figure it out, I am interested in this as well. Right now I have a cron job that executes this every 5 minutes.. UPDATE cdr SET userfield = MID( dstchannel, 1 , LOCATE( '-', dstchannel )-1) WHERE disposition = 'ANSWERED' AND LOCATE( '-', dstchannel ) > 0 and lastapp = 'Queue' and userfield <> '' -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D Sent: Friday, January 22, 2010 9:23 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set CDR userfield for Queues Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set CDR userfield for Queues
Since userfield is a variable, that would be a qualified yes.
Here's how I use it in my dialplan
- exten => NXX,1,noop(Answer())
- exten => NXX,n,Verbose (Dial DAHDI g1 ${EXTEN})
- exten => NXX,n,Dial(DAHDI/g1/${EXTEN},60,miKkTtg)
- exten => NXX,n,Set(CDR(userfield)=${CDR(userfield)}
-Hangupcause:${HANGUPCAUSE})
- exten => NXX,n,Hangup(${HANGUP_CAUSE})
How I envision what you're asking:
- exten => s,n,Set(CDR(userfield)=${CDR(userfield)} -Agent:${AGENT})
--
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deep D
Sent: Friday, January 22, 2010 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set CDR userfield for Queues
Hello,
I am using Queue application with multiple agents in each queue. I
want to set the CDR(userfield) for each cdr based on the agent
answering the call. Is it possible to do this?
Thanks
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Re: [asterisk-users] Snom vs Polycom
On Fri, Jan 22, 2010 at 1:26 PM, Julian Lyndon-Smith wrote: > Anyone got any subjective (!) views on the merits of these two ranges , > using asterisk 1.4 ? The choice of phones is crucial. Setting aside my tastes, you really need to get a couple of typical users to try them before committing to buying a bunch IMO. I'll bet someone like e4strategies.com would work something out if you called and talked to them. Even the one-liners need to be ergonomic and so many SIP phones are horrible at that. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set CDR userfield for Queues
Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble getting feature codes to work
Thanks for the GREAT tip. Changing to a single feature digit of * for blindxfer worked which led me to changing featuredigittimeout = 2000. Now I can do blindxfer w/ ##. Why I didn't try changing featuredigittimeout long ago is beyond me! *blush* Thanks again. One thing that still doesn't work tho is applicationmap. I have this in features.conf: [applicationmap] testfeature1 => #9,caller,Playback,tt-monkeys testfeature2 => #8,callee,Playback,tt-monkeys and this in the context where the dial takes place: include => featuremap include => applicationmap Any ideas? I'd love to hear tt-monkeys from either side of the call! In the end tho, I'm trying to provide alternate method for hanging up since I don't want to base it on adding the h option to the Dial command. As you know that would hangup on a single *, which is not so good when calling an IVR. Cheers! H On Fri, Jan 22, 2010 at 7:02 AM, Karsten Wemheuer wrote: > Hi, > > Am Donnerstag, den 21.01.2010, 21:08 -0500 schrieb hugolivude: > > Hi, > > > > I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. > > Features.conf contians this: > > > > blindxfer=## > > atxfer=*2 > > automon=*1 > > disconnect=** > > > > I'm really most interested in getting disconnect to work so that I > > hear "Goodbye" when I press ** during a call connected this way in my > > dial plan: > > > > exten => 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT) > > exten => 1,n,Playback(vm-goodbye) > > > > The call works fine and the CLI tells me that ** is an active feature: > > > > Builtin Feature Default Current > > --- --- --- > > Pickup*8 *8 > > Blind Transfer# ## > > Attended Transfer *2 > > One Touch Monitor *1 > > Disconnect Call * ** > > Park Call > > > > When I press ** during a call though, nothing appears in the CLI > > (verbosity = 4). I do it very quickly so I don't believe timeout is > > an issue. > > As DTMF recognition is not the problem (as You told in the other post), > You can check two other things: > 1) Exclude the timing issue: Are the other 2-character feature codes > working? What about testing with a 1-character code setting or with a > featuretimeout in the conf-file (I believe the default is very short) > 2) If this is a sip-to-sip call, check if asterisk stays in the audio > path (you can check it with a network sniffer like tcpdump or > wireshark). > > HTH, have a nice weekend, > > Karsten > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OfficeSIP Softphone
Hello, I would like to see this as well, from an Asterisk CLI log perspective with "sip debug" turned on. The .log file for login and invite is attached, I have use asterisk -vr command. Is it correct? Yes, here is two INVITEs (I have missed first invite before), but the server respond 401 on first invite and softphone send ACK. Here is softphone log. If Asterisk receives the ACK *after* the second INVITE I understand it. The softphone uses single tcp connection, so messages must arrive in same order as them was sent. Best regards, Vitali Fomine login-invite.log Description: Binary data -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom vs Polycom
Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? I need to supply approx 30 handsets to a new client, with the senior managers (6) having some slightly more "managerial" phones than the base phones which will be used for one line only. TIA Julian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Popular Gigabit Phones
http://www.snom.com/en/products/ip-phones/snom-870-touchscreen-voip-phone/ http://www.aastraintecom.com/cps/rde/xchg/SID-3D8CCB6A-935A2A1B/30/hs.xsl/38707.htm ~ Andrew "lathama" Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Fri, Jan 22, 2010 at 12:00 AM, Matt Darnell wrote: > On Thu, Jan 21, 2010 at 3:30 PM, Jonathan Thurman > wrote: >> On Thu, Jan 21, 2010 at 4:56 PM, Matt Darnell wrote: >>> Most manufacturers charge in excess of $80 to upgrade from a 10/100 >>> switch to a 10/100/1000 switch built into the phone. >>> The cost might have been in the chipset 5 years ago but I can get a 5 >>> port gigabit switch for $30. >>> >>> What are most folks using for people that need gigabit to the desktop >>> and don't want to run another cable? >> >> For our engineering staff we use Polycom SoundPoint IP 560's. Cubes >> with two drops for heavy users who have to be dual homed were build >> without VoIP in mind (or an tech department at all for that matter)... >> I haven't run iperf through them, so I don't have any performance >> statistics. No one has complained except for our fiscal department, >> the phones do come at a premium above the standard phones =). >> >> -Jonathan > > Thanks for the feedback on the 560's. Polycom's are very well built phones. > > I am surprised they don't offer a phone in the 3 series form factor > that has gigabit. People must not be asking for it because the only > folks that seem to offer it are Polycom and Cisco (not the Linksys > rebrand). > > -Matt > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 mysql 'NO ANSWER' disposition problem
Hi all!
I have installed a quite old Asterisk 1.6.2.0-rc2 with latest DAHDI on
Ubuntu 9.10 from repository. It is working now but mysql logging is very
strange. All calls have logged in mysql cdr table, which is fine, but
disposition is 'NO ANSWER' even if I had talked on phone. Duration is
correct but billsec is zero. Any idea why? Unfortunately I cannot upgrade to
newer version because repository only have this one.
Console log with verbose 9:
[Jan 22 12:52:19] -- Attempting call on dahdi/g1/xxx for y...@default:1
(Retry 1)
[Jan 22 12:52:19] -- Requested transfer capability: 0x00 - SPEECH
[Jan 22 12:52:23] WARNING[3285]: chan_dahdi.c:2682 dahdi_enable_ec: Unable
to enable echo cancellation on channel 1 (No such device)
[Jan 22 12:52:24] WARNING[3285]: chan_dahdi.c:2682 dahdi_enable_ec: Unable
to enable echo cancellation on channel 1 (No such device)
[Jan 22 12:52:24] WARNING[3285]: chan_dahdi.c:2682 dahdi_enable_ec: Unable
to enable echo cancellation on channel 1 (No such device)
[Jan 22 12:52:24]> Channel DAHDI/1-1 was answered.
[Jan 22 12:52:24] == Starting DAHDI/1-1 at default,yyy,1 failed so falling
back to exten 's'
[Jan 22 12:52:24] -- Executing [...@default:1] Wait("DAHDI/1-1", "1") in
new stack
[Jan 22 12:52:25] -- Executing [...@default:2] Answer("DAHDI/1-1", "") in
new stack
[Jan 22 12:52:25] -- Executing [...@default:3] Set("DAHDI/1-1",
"TIMEOUT(digit)=5") in new stack
[Jan 22 12:52:25] -- Digit timeout set to 5.000
[Jan 22 12:52:25] -- Executing [...@default:4] Set("DAHDI/1-1",
"TIMEOUT(response)=10") in new stack
[Jan 22 12:52:25] -- Response timeout set to 10.000
[Jan 22 12:52:25] -- Executing [...@default:5] BackGround("DAHDI/1-1",
"demo-congrats") in new stack
[Jan 22 12:52:25] -- Playing 'demo-congrats.gsm' (language
'hu')
[Jan 22 12:52:33] WARNING[3285]: chan_dahdi.c:2682 dahdi_enable_ec: Unable
to enable echo cancellation on channel 1 (No such device)
[Jan 22 12:52:35] -- Channel 0/1, span 1 got hangup request, cause 31
[Jan 22 12:52:35] -- Hungup 'DAHDI/1-1'
[Jan 22 12:52:35] NOTICE[3311]: pbx_spool.c:349 attempt_thread: Call
completed to dahdi/g1/xxx
I saw that new mysql addon have the possibility to aliasing columns. What
kind of fields, internal variables Asterisk have for that columns? For
example is there any variable that tells what *original* extension I had
called? Not the last but the initial. Some variables are at the end of the
cdr_mysql.conf but is there any other?
Bye,
a
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Re: [asterisk-users] Trouble getting feature codes to work
Hi, Am Donnerstag, den 21.01.2010, 21:08 -0500 schrieb hugolivude: > Hi, > > I'm having trouble getting feature codes to work in Asterisk 1.4.21.2. > Features.conf contians this: > > blindxfer=## > atxfer=*2 > automon=*1 > disconnect=** > > I'm really most interested in getting disconnect to work so that I > hear "Goodbye" when I press ** during a call connected this way in my > dial plan: > > exten => 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT) > exten => 1,n,Playback(vm-goodbye) > > The call works fine and the CLI tells me that ** is an active feature: > > Builtin Feature Default Current > --- --- --- > Pickup*8 *8 > Blind Transfer# ## > Attended Transfer *2 > One Touch Monitor *1 > Disconnect Call * ** > Park Call > > When I press ** during a call though, nothing appears in the CLI > (verbosity = 4). I do it very quickly so I don't believe timeout is > an issue. As DTMF recognition is not the problem (as You told in the other post), You can check two other things: 1) Exclude the timing issue: Are the other 2-character feature codes working? What about testing with a 1-character code setting or with a featuretimeout in the conf-file (I believe the default is very short) 2) If this is a sip-to-sip call, check if asterisk stays in the audio path (you can check it with a network sniffer like tcpdump or wireshark). HTH, have a nice weekend, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme conferencing - large deployment SIP or ZAP?
I've been asked by my company to setup a conferencing system to support up to 400 people on a conference calls, where all users will be dialling in frpm the PSTN. I am exploring using Asterisk meetme to do this. I have two questions in relation to this:- For Meetme conferences is it better to have all participants to dial in via SIP provider terminating to Asterisk via SIP/IAX, or use something like the Digium TE420P cards connected to 16xE1's (I'm in Australia). Are there pro's and con's to each method. For this number of users, should I be looking at multiple asterisk servers to distribute the load, or a single server (I have some Dual core Dell 2950 servers available). If its multiple, can you simply daisy chain conference participants across servers? Thanks Steve Sydney, Australia -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - SPA3102 not detecting CID - Which settings to tune ?
Hi, I'm connecting a Linksys SPA3102 to 3 different PSTN analog lines. With only one of those, CID is shown. Beside that, everything is working OK. Lines have different providers and/or locations. All are located in France and CID Detection Method is ETSI FSK / Bell 202. If I'm connecting a TDM400-enabled Asterisk system, to one of those 2 non-working lines : it does work. The only non-default meaningful setting for this TDM card is loadzone=fr. Which 3102 setting would you suggest change to get CID working ? Here and there, I've read that increasing PSTN Answer Delay might help. Setting this to different values (up to 10 secs) didn't change end result. What about increasing "PSTN to SPA Gain" ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OfficeSIP Softphone
22 jan 2010 kl. 11.51 skrev Vitali Fomine: >>> report 491 Request Pending on invite message. Why server report the >>> error? >> The server reports this when we already have an INVITE to handle. >> Please check that you did not transmit two invites without waiting for >> a response and sending an ACK from your softphone. > > Yes, here is two INVITEs (I have missed first invite before), but the server > respond 401 on first invite and softphone send ACK. Here is softphone log. > Unfortunately, I do not know how to enable (where to find) log of SIP > messages at server? I would like to see that log with Asterisk messages in between, so I understand when Asterisk receives the ACK. If Asterisk receives the ACK *after* the second INVITE I understand it. > > I have find one more issue, the server sends two replies on register > message, first with 200 and second one 403 with same CSeq. I am not sure is > that relayted to INVITE issue. But the asterisk show user as > connected-unmonitored in control panel (trixbox). I would like to see this as well, from an Asterisk CLI log perspective with "sip debug" turned on. /O > > Best regards, > Vitali Fomine > > 01/22/2010|13:37:36.097 > INVITE sip:5...@trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:58238 > Max-Forwards: 70 > From: ;tag=60b512cec9;epid=08fd7dc31f > To: > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 1 INVITE > Contact: > ;proxy=replace;+sip.instance="" > User-Agent: UCCAPI/2.0.6362.67 > Supported: timer > Supported: ms-sender > Supported: ms-early-media > Supported: Replaces > ms-keep-alive: UAC;hop-hop=yes > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:5...@trixbox1.local", nonce="1746fc14", > response="c57700c482c83cbeb411398d92f94113" > Content-Type: application/sdp > Content-Length: 2147 > > v=0 > o=- 0 0 IN IP4 192.168.1.15 > s=session > [...session description removed...] > > > 01/22/2010|13:37:36.102 > SIP/2.0 401 Unauthorized > Via: SIP/2.0/TCP 192.168.1.15:58238;received=192.168.1.15 > From: ;tag=60b512cec9;epid=08fd7dc31f > To: ;tag=as63c5f412 > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 1 INVITE > User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39ce8842" > Content-Length: 0 > > > > 01/22/2010|13:37:36.102 > ACK sip:5...@trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:58238 > Max-Forwards: 70 > From: ;tag=60b512cec9;epid=08fd7dc31f > To: ;tag=as63c5f412 > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 1 ACK > User-Agent: UCCAPI/2.0.6362.67 > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:5...@trixbox1.local", nonce="1746fc14", > response="a3e2da49ec1a432115871eb965f4aad3" > Content-Length: 0 > > > > 01/22/2010|13:37:36.103 > INVITE sip:5...@trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:58238 > Max-Forwards: 70 > From: ;tag=60b512cec9;epid=08fd7dc31f > To: > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 2 INVITE > Contact: > ;proxy=replace;+sip.instance="" > User-Agent: UCCAPI/2.0.6362.67 > Supported: timer > Supported: ms-sender > Supported: ms-early-media > Supported: Replaces > ms-keep-alive: UAC;hop-hop=yes > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:5...@trixbox1.local", nonce="39ce8842", > response="5925e73eeaf067412c6b1c73cf520d0e" > Content-Type: application/sdp > Content-Length: 2147 > > v=0 > o=- 0 0 IN IP4 192.168.1.15 > s=session > c=IN IP4 192.168.1.15 > [...removed..] > > > 01/22/2010|13:37:36.106 > SIP/2.0 491 Request Pending > Via: SIP/2.0/TCP 192.168.1.15:58238;received=192.168.1.15 > From: ;tag=60b512cec9;epid=08fd7dc31f > To: ;tag=as63c5f412 > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > Content-Length: 0 > > > 01/22/2010|13:37:36.106 > ACK sip:5...@trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:58238 > Max-Forwards: 70 > From: ;tag=60b512cec9;epid=08fd7dc31f > To: ;tag=as63c5f412 > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 2 ACK > User-Agent: UCCAPI/2.0.6362.67 > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:5...@trixbox1.local", nonce="39ce8842", > response="2ebb3595b1af94e67f7e880478c82171" > Content-Length: 0 > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- ___
Re: [asterisk-users] OfficeSIP Softphone
>> report 491 Request Pending on invite message. Why server report the >> error? > The server reports this when we already have an INVITE to handle. > Please check that you did not transmit two invites without waiting for > a response and sending an ACK from your softphone. Yes, here is two INVITEs (I have missed first invite before), but the server respond 401 on first invite and softphone send ACK. Here is softphone log. Unfortunately, I do not know how to enable (where to find) log of SIP messages at server? I have find one more issue, the server sends two replies on register message, first with 200 and second one 403 with same CSeq. I am not sure is that relayted to INVITE issue. But the asterisk show user as connected-unmonitored in control panel (trixbox). Best regards, Vitali Fomine 01/22/2010|13:37:36.097 INVITE sip:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: ;tag=60b512cec9;epid=08fd7dc31f To: Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 1 INVITE Contact: ;proxy=replace;+sip.instance="" User-Agent: UCCAPI/2.0.6362.67 Supported: timer Supported: ms-sender Supported: ms-early-media Supported: Replaces ms-keep-alive: UAC;hop-hop=yes Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:5...@trixbox1.local", nonce="1746fc14", response="c57700c482c83cbeb411398d92f94113" Content-Type: application/sdp Content-Length: 2147 v=0 o=- 0 0 IN IP4 192.168.1.15 s=session [...session description removed...] 01/22/2010|13:37:36.102 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.1.15:58238;received=192.168.1.15 From: ;tag=60b512cec9;epid=08fd7dc31f To: ;tag=as63c5f412 Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39ce8842" Content-Length: 0 01/22/2010|13:37:36.102 ACK sip:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: ;tag=60b512cec9;epid=08fd7dc31f To: ;tag=as63c5f412 Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 1 ACK User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:5...@trixbox1.local", nonce="1746fc14", response="a3e2da49ec1a432115871eb965f4aad3" Content-Length: 0 01/22/2010|13:37:36.103 INVITE sip:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: ;tag=60b512cec9;epid=08fd7dc31f To: Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 2 INVITE Contact: ;proxy=replace;+sip.instance="" User-Agent: UCCAPI/2.0.6362.67 Supported: timer Supported: ms-sender Supported: ms-early-media Supported: Replaces ms-keep-alive: UAC;hop-hop=yes Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:5...@trixbox1.local", nonce="39ce8842", response="5925e73eeaf067412c6b1c73cf520d0e" Content-Type: application/sdp Content-Length: 2147 v=0 o=- 0 0 IN IP4 192.168.1.15 s=session c=IN IP4 192.168.1.15 [...removed..] 01/22/2010|13:37:36.106 SIP/2.0 491 Request Pending Via: SIP/2.0/TCP 192.168.1.15:58238;received=192.168.1.15 From: ;tag=60b512cec9;epid=08fd7dc31f To: ;tag=as63c5f412 Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 01/22/2010|13:37:36.106 ACK sip:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: ;tag=60b512cec9;epid=08fd7dc31f To: ;tag=as63c5f412 Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 2 ACK User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:5...@trixbox1.local", nonce="39ce8842", response="2ebb3595b1af94e67f7e880478c82171" Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GoToIfTime issue
2010/1/22 Randy R : > On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun wrote: >> exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1) > >> but what should i do. if i want to set seperate weekdays,like mon,wed. >> not continuous weekday like mon-fri. > > I couldn't find any reference to multiple, non-contiguous days on a > quick Google, but this would work at the cost of an extra line: > > exten => 222,1,GoToIfTime(11:00-14:00|mon|*|*?1:3,1) > exten => 222,2,GoToIfTime(11:00-14:00|wed|*|*?1:3,1) Thank you, but why don't it to be comma seperate to represent seperate weekdays? as | mon,wed,fri | it's also very understandable. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Call Xfer issue between DataCenter and User Site
Sorry to bump this one... Anyone have any other ideas on it? Regards Steven Davison Net Technial Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Davison Sent: 21 January 2010 08:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Xfer issue between DataCenter and User Site Thanks for the responses on this one David Gibbons: reinvite=no is set, as we need the asterisk box to maintain the audio for recording... (I believe even if we didn't have this option, MixMonitor would have the same effect anyway.) Peder: the firewall is integrated into the router, and is a Zyxel 660H-D1... which hasn't caused NAT issues in the past, but it is something that we can switch out and see if a different make/model has the same problem. In answer to your questions, the Data Center IP is the external address that has been 1 to 1 Nat'd to the internal address. The phone site has no static Nat in place for Sip or RTP, so we are reliant on the routers ability to sort that out. There is a firewall on that router, which allows ALL traffic out, and also allows SIP and RTP in. Hope that clears up a few things! :) Steven Davison - Network Engineer t: 0845 0034567 f: 0845 0034543 w: www.ntsols.com Net Technical Solutions | Suite 1 Wesley Chambers | Queens Road | Aldershot | Hampshire | GU11 3JD -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 20 January 2010 18:24 To: asterisk-users@lists.digium.com Cc: Alistair Mackenzie Subject: Re: [asterisk-users] Call Xfer issue between DataCenter and User Site Admittedly I didn't read your SIP debug (on the mobile), but do you have reinvite=no set for the extensions and SIP trunks (providers)? This sounds on the surface like a classic case of the Mondays. Erm reinvites I mean. 1. Incoming call from pstn/viop provider 2. Call is answered by a user 3. Call needs to be transferred 4. Xfer button is pushed, other user is called, answered, and they speak about the call 4b. The incoming call is held, listening to MoH 5. Xfer is pushed again, 6. 7. MoH stops, 8. Office user gets no audio 9. Incoming call is silent, and then call is dropped 10. Office user gets fed up of saying ‘hello??!?’ and hangs up. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OfficeSIP Softphone
22 jan 2010 kl. 10.13 skrev Vitali Fomine: > Hello, > > I am developing the free SIP softphone (audio+video) for Windows. And I have > some issues with asterisk 1.6 compatibility. I am new in asterisk, so I > guess, I have no enough skills to config asterisk properly. I have enable > tcp transport mode and register client, but can not make a call. The server > report 491 Request Pending on invite message. Why server report the error? The server reports this when we already have an INVITE to handle. Please check that you did not transmit two invites without waiting for a response and sending an ACK from your softphone. /O > > Here is link to the softphone: > http://www.officesip.com/download/officesip-softphone-1.0.msi > > Best regards, > Vitali Fomine > > > INVITE sip:5...@trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:52774 > Max-Forwards: 70 > From: ;tag=39be813029;epid=f918608aea > To: > Call-ID: 738a7dd4d06d4c439c29fb703e491533 > CSeq: 2 INVITE > Contact: > ;proxy=replace;+sip.instance="" > User-Agent: UCCAPI/2.0.6362.67 > Supported: timer > Supported: ms-sender > Supported: ms-early-media > Supported: Replaces > ms-keep-alive: UAC;hop-hop=yes > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:5...@trixbox1.local", nonce="601d7934", > response="36c795437dc4088ac5947f923e8dbb0f" > Content-Type: application/sdp > Content-Length: 2146 > > v=0 > o=- 0 0 IN IP4 192.168.1.15 > s=session > c=IN IP4 192.168.1.15 > b=CT:99980 > t=0 0 > m=audio 46080 RTP/AVP 114 111 112 115 116 4 8 0 97 101 > k=base64:D3YQHD+33y6crQYg5HKB5+xk+uzWWjx1Nqu92I0yqiNXO3u4Neq5AsqMPOA0 > a=candidate:y0T/m81+hR/z4InuWh6XB7ix0Az1iTjwcaqYENbLia4 1 > JzFksprV3LGT4h5qLzK+gA UDP 0.830 192.168.1.15 46080 > a=candidate:y0T/m81+hR/z4InuWh6XB7ix0Az1iTjwcaqYENbLia4 2 > JzFksprV3LGT4h5qLzK+gA UDP 0.830 192.168.1.15 8960 > a=candidate:JNZVPl8eF6E19v/HMq1oWE/6ScDVqz05i2XXLGhNhjw 1 > qIgZJbPCW/xDOWMMN5ZWJQ UDP 0.840 192.168.56.1 17792 > a=candidate:JNZVPl8eF6E19v/HMq1oWE/6ScDVqz05i2XXLGhNhjw 2 > qIgZJbPCW/xDOWMMN5ZWJQ UDP 0.840 192.168.56.1 30080 > a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 > inline:5zuBoVK+RVi6Yw/Po02VsVrbZVQLPVy4VxColZpZ|2^31|1:1 > a=crypto:2 AES_CM_128_HMAC_SHA1_80 > inline:bB2j++RDWPo/sLbSBVijJy8lKyy/dd2bEKyxdC+k|2^31|1:1 > a=maxptime:200 > a=rtcp:8960 > a=rtpmap:114 x-msrta/16000 > a=fmtp:114 bitrate=29000 > a=rtpmap:111 SIREN/16000 > a=fmtp:111 bitrate=16000 > a=rtpmap:112 G7221/16000 > a=fmtp:112 bitrate=24000 > a=rtpmap:115 x-msrta/8000 > a=fmtp:115 bitrate=11800 > a=rtpmap:116 AAL2-G726-32/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:97 RED/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=encryption:optional > m=video 34432 RTP/AVP 121 34 > k=base64:ve6wgVJQaeIkcDokUVyKXuQM2JzIBIoyiJUDPcH27R89T80GhLRVF+JPZPtI > a=candidate:VNyfYW0mHvCAJsHrBO+jt+l0bZHFIuiSfHfcjYORL8U 1 > Xy99337KGr+R6TEcalNuaQ UDP 0.850 192.168.1.15 34432 > a=candidate:VNyfYW0mHvCAJsHrBO+jt+l0bZHFIuiSfHfcjYORL8U 2 > Xy99337KGr+R6TEcalNuaQ UDP 0.850 192.168.1.15 12032 > a=candidate:h4gtcS9V5HP54MByMbgC3wJpFfg1WciRe/VqLZ6qiFc 1 > gjQT1OVzQRIyzR2eq1i7kw UDP 0.860 192.168.56.1 52608 > a=candidate:h4gtcS9V5HP54MByMbgC3wJpFfg1WciRe/VqLZ6qiFc 2 > gjQT1OVzQRIyzR2eq1i7kw UDP 0.860 192.168.56.1 37120 > a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 > inline:fcLhKq245/mep3k6sYBdnnusNq8mfwAN6aXBpbot|2^31|1:1 > a=crypto:2 AES_CM_128_HMAC_SHA1_80 > inline:q2Atyjw2+REUuFXxkQYrE3nuzsVT5xFkt+2xcaDD|2^31|1:1 > a=maxptime:200 > a=rtcp:12032 > a=rtpmap:121 x-rtvc1/9 > a=rtpmap:34 H263/9 > a=encryption:optional > > > SIP/2.0 491 Request Pending > Via: SIP/2.0/TCP 192.168.1.15:52774;received=192.168.1.15 > From: ;tag=39be813029;epid=f918608aea > To: ;tag=as5c7a7ed8 > Call-ID: 738a7dd4d06d4c439c29fb703e491533 > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > Content-Length: 0 > > > ACK sip:5...@trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:52774 > Max-Forwards: 70 > From: ;tag=39be813029;epid=f918608aea > To: ;tag=as5c7a7ed8 > Call-ID: 738a7dd4d06d4c439c29fb703e491533 > CSeq: 2 ACK > User-Agent: UCCAPI/2.0.6362.67 > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:5...@trixbox1.local", nonce="601d7934", > response="35dca1911b5bb614b1cadfda53e7d8f4" > Content-Length: 0 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-d
Re: [asterisk-users] GoToIfTime issue
On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun wrote: > exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1) > but what should i do. if i want to set seperate weekdays,like mon,wed. > not continuous weekday like mon-fri. I couldn't find any reference to multiple, non-contiguous days on a quick Google, but this would work at the cost of an extra line: exten => 222,1,GoToIfTime(11:00-14:00|mon|*|*?1:3,1) exten => 222,2,GoToIfTime(11:00-14:00|wed|*|*?1:3,1) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OfficeSIP Softphone
Hello, I am developing the free SIP softphone (audio+video) for Windows. And I have some issues with asterisk 1.6 compatibility. I am new in asterisk, so I guess, I have no enough skills to config asterisk properly. I have enable tcp transport mode and register client, but can not make a call. The server report 491 Request Pending on invite message. Why server report the error? Here is link to the softphone: http://www.officesip.com/download/officesip-softphone-1.0.msi Best regards, Vitali Fomine INVITE sip:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:52774 Max-Forwards: 70 From: ;tag=39be813029;epid=f918608aea To: Call-ID: 738a7dd4d06d4c439c29fb703e491533 CSeq: 2 INVITE Contact: ;proxy=replace;+sip.instance="" User-Agent: UCCAPI/2.0.6362.67 Supported: timer Supported: ms-sender Supported: ms-early-media Supported: Replaces ms-keep-alive: UAC;hop-hop=yes Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:5...@trixbox1.local", nonce="601d7934", response="36c795437dc4088ac5947f923e8dbb0f" Content-Type: application/sdp Content-Length: 2146 v=0 o=- 0 0 IN IP4 192.168.1.15 s=session c=IN IP4 192.168.1.15 b=CT:99980 t=0 0 m=audio 46080 RTP/AVP 114 111 112 115 116 4 8 0 97 101 k=base64:D3YQHD+33y6crQYg5HKB5+xk+uzWWjx1Nqu92I0yqiNXO3u4Neq5AsqMPOA0 a=candidate:y0T/m81+hR/z4InuWh6XB7ix0Az1iTjwcaqYENbLia4 1 JzFksprV3LGT4h5qLzK+gA UDP 0.830 192.168.1.15 46080 a=candidate:y0T/m81+hR/z4InuWh6XB7ix0Az1iTjwcaqYENbLia4 2 JzFksprV3LGT4h5qLzK+gA UDP 0.830 192.168.1.15 8960 a=candidate:JNZVPl8eF6E19v/HMq1oWE/6ScDVqz05i2XXLGhNhjw 1 qIgZJbPCW/xDOWMMN5ZWJQ UDP 0.840 192.168.56.1 17792 a=candidate:JNZVPl8eF6E19v/HMq1oWE/6ScDVqz05i2XXLGhNhjw 2 qIgZJbPCW/xDOWMMN5ZWJQ UDP 0.840 192.168.56.1 30080 a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:5zuBoVK+RVi6Yw/Po02VsVrbZVQLPVy4VxColZpZ|2^31|1:1 a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:bB2j++RDWPo/sLbSBVijJy8lKyy/dd2bEKyxdC+k|2^31|1:1 a=maxptime:200 a=rtcp:8960 a=rtpmap:114 x-msrta/16000 a=fmtp:114 bitrate=29000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:112 G7221/16000 a=fmtp:112 bitrate=24000 a=rtpmap:115 x-msrta/8000 a=fmtp:115 bitrate=11800 a=rtpmap:116 AAL2-G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=encryption:optional m=video 34432 RTP/AVP 121 34 k=base64:ve6wgVJQaeIkcDokUVyKXuQM2JzIBIoyiJUDPcH27R89T80GhLRVF+JPZPtI a=candidate:VNyfYW0mHvCAJsHrBO+jt+l0bZHFIuiSfHfcjYORL8U 1 Xy99337KGr+R6TEcalNuaQ UDP 0.850 192.168.1.15 34432 a=candidate:VNyfYW0mHvCAJsHrBO+jt+l0bZHFIuiSfHfcjYORL8U 2 Xy99337KGr+R6TEcalNuaQ UDP 0.850 192.168.1.15 12032 a=candidate:h4gtcS9V5HP54MByMbgC3wJpFfg1WciRe/VqLZ6qiFc 1 gjQT1OVzQRIyzR2eq1i7kw UDP 0.860 192.168.56.1 52608 a=candidate:h4gtcS9V5HP54MByMbgC3wJpFfg1WciRe/VqLZ6qiFc 2 gjQT1OVzQRIyzR2eq1i7kw UDP 0.860 192.168.56.1 37120 a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:fcLhKq245/mep3k6sYBdnnusNq8mfwAN6aXBpbot|2^31|1:1 a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:q2Atyjw2+REUuFXxkQYrE3nuzsVT5xFkt+2xcaDD|2^31|1:1 a=maxptime:200 a=rtcp:12032 a=rtpmap:121 x-rtvc1/9 a=rtpmap:34 H263/9 a=encryption:optional SIP/2.0 491 Request Pending Via: SIP/2.0/TCP 192.168.1.15:52774;received=192.168.1.15 From: ;tag=39be813029;epid=f918608aea To: ;tag=as5c7a7ed8 Call-ID: 738a7dd4d06d4c439c29fb703e491533 CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 ACK sip:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:52774 Max-Forwards: 70 From: ;tag=39be813029;epid=f918608aea To: ;tag=as5c7a7ed8 Call-ID: 738a7dd4d06d4c439c29fb703e491533 CSeq: 2 ACK User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:5...@trixbox1.local", nonce="601d7934", response="35dca1911b5bb614b1cadfda53e7d8f4" Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GoToIfTime issue
hi , all
what's wrong with this command?
exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1)
as i got the error:
-- Executing [...@95040:1] GotoIfTime("SIP/1001-0099",
"11:00-14:00|mon|wed|*|*?1:3|1") in new stack
[Jan 20 11:21:11] WARNING[16804]: pbx.c:4118 get_range: Invalid day
'wed', assuming none
but what should i do. if i want to set seperate weekdays,like mon,wed.
not continuous weekday like mon-fri.
--
Best regards,
Sucan
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[asterisk-users] MYSQL problem
hi,all as you know, we can use MYSQL command to visit mysql database but if i use other database like Oracke,sybase,etc, Could i use MYSQL command ? if not, is there any other alternative could do the same function(visit database in dailplan)? Thanks! -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
