Re: [asterisk-users] file and on SayNumber() app
No, it will not have "and" if You just put and.ulaw. You should correct file say.conf - there are rules how to read numbers, and You should add "and" there if You want to hear it. 2012/7/23 נפתלי מאיר : > It`s not will to be: ; "one - thousand - two - hundred - and - thirty - four > ?? > > I put and.ulaw file on en dir and on en/digits dir. -- _ -- 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] Use of rtptimeout
If you properly link users.conf to sip.conf you can use it it there too. 2012/6/12 Rabary : > Hi list, > > I want to use rtptimeout function on asterisk 1.4 but any docs I read, it is > said that I need to configure it in sip.conf file, > But can I use rtptimeout in users.conf file or do I need to configure all > the SIP accounts on sip.conf file before I can use rtptimeout ? > > thanks. > > -- > Rabary Teddy > > Inutile d'imprimer ce mail > > > -- > _ > -- 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] Asterisk 1.8.10
Hi
As far as I know "Alert-Info" is as far as vendor specific extension
to SIP used by CISCO VOIP-gateways only. Didn't noticed any other
vendors to support that. Software clients neither. So such trick is
only usable in conjunction with CISCO.
Anyway, wait another answer, probably somebody knows more.
2012/6/12 motty.cruz :
> Hello,
> How to change ring tone on interncal call? I'm using Centos 5.8 Asterisk 1.8
>
> exten =>
> 666,1,SIPAddHeader("Alert-Info:")
> exten => 666,n,Dial(SIP/10)
>
> The above would not how to defirenciate from internal call or external call?
>
>
> Thanks,
> motty
>
>
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Re: [asterisk-users] Asterisk AMI SIP channel detect phone ringing
2012/5/3 JIMMY GATHAGE : > I am using a SIP trunk to make outgoing calls. Outgoing calls are > going through okay. I am using the AMI to Originate a call. The > channel is not returning any event when the phone on the PSTN is > ringing. How can i detect the phone ringing on the SIP channel? It is possible you made synchronous origination. Did you specified Async: true header in origination action? If I remember correctly, synchronous origination blocks AMI session until it done origination routine. -- _ -- 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] Incoming SIP call is rejected always.
2012/4/18 Matthew Jordan : > I imagine that this is the case, as ASTERISK-19601 noted that > when this situation occurs, the NOTICE message indicates that > there is a failure to match the extension, as opposed to a failure > to match an allowed domain. Yes, it was hell to detect real error cause(I was forced to learn how to debug in KDevelop in less than four hours). Yes, it looks like ASTERISK-19601. But still I cannot understand why asterisk extracts wrong domain from request. > However, in your SIP configuration you have set allowexternaldomains to no. Yes, it is intended. > Without knowing the URI the INVITE request was addressed to, its > difficult to say what might be the actual cause of this. I first letter I have provided CLI log which contains full request packets(Authless and authed INVITE included). Probably I do not understand how to configure Asterisk: I have one asterisk. It serves SIP domain example.com. This asterisk must be able to establish session with registered client of this account and also must be able to accept incoming sessions. No sessions with 3rd-party accounts on 3rd-party domains allowed to established. How I should setup this asterisk? -- _ -- 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] Incoming SIP call is rejected always.
2012/4/17 Danny Nicholas : > Maybe it needs to be _4001020? > Not, it doesn't. Actually I have traced this incoming call step by step. Real reason it refuses - wrong domain. But why it wrong - have not any idea. -- _ -- 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] Incoming SIP call is rejected always.
Hi Have an asterisk. Setup a couple of friends. Sip.conf - http://pastebin.com/zUgiYbBi Trying to make incoming call, and have such error(cli output) http://pastebin.com/zFfgYcNR NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from 'RMT20' (192.168.8.1:5062) to extension '4001020' rejected because extension not found in context 'rmt-context'. But, as you see, there is such extension. What I'm doing wrong? -- _ -- 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] Multiprocess Asterisk
2012/2/20 M Takahashi : > Is anybody running multiprocess of Asterisk on a server ? Does it work well? > My configuration is too complicated. I know Asterisk on a virtual machine > works well. but OS overhead is considerable. that is why I want to divide a > process. Running 3 instances of Asterisk on the same Server. My configuration not very complicated and Server not very overloaded but it works. Main task was to separate for each instance own network port range, and to make separate File-System sandbox(and so get rid of some absolute path in dialplans). But you should properly calculate hardware requirements in order to achieve required performance. Also, if You use Linux, you can user native kernel virtualisation technologies. Personally I recommend to use LXC. It has almost zero overhead, because works on native, logically separated environments. -- _ -- 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] How to play audio file in background in dialplan?
2012/2/4 Steve Edwards :
> does DO-SOME-ACTION-BASED-ON-PREVIOUS-CALL-RETURN-CODE involve some
> interaction with Asterisk?
Yes, DO-SOME-ACTION-BASED-ON-PREVIOUS-CALL-RETURN-CODE interacts with Asterisk.
> Does the entire ${PROFILE_MUSIC} file need to be played or
> does it need to be interrupted when
> DO-SOME-ACTION-BASED-ON-PREVIOUS-CALL-RETURN-CODE completes?
It should be stopped only after
DO-SOME-ACTION-BASED-ON-PREVIOUS-CALL-RETURN-CODE completes. It is
actually my longtime operation. taskproxy works very fast - fetches
from external storage data for DO-SOME-ACTION...
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Re: [asterisk-users] How to play audio file in background in dialplan?
2012/2/3 Danny Nicholas : > In my PERL AGI I use > - print STDOUT "SET MUSIC ON HOLD DEFAULT\n"; > - print STDOUT "SET MUSIC ON HOLD OFF\n"; > > Ignore the "-" - stupid outlook needs them. There is one problem: I have not any MOH class, and cannot pre-create it. File I will play does not exists until incoming call arrives, I even don't know its exact name. -- _ -- 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] How to play audio file in background in dialplan?
2012/2/3 Danny Nicholas : > > Fork or shell task 1 > Fork or shell task 2 > What exactly commands I should invoke in AGI instead of and ? STREAM FILE returns only after file ends, this is not what I want. -- _ -- 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] [asterisk-dev] How to play audio file in background in dialplan?
2012/2/3 Steve Edwards : > Do your processing in an AGI. Before you get into your 'longtime task' > create another thread to play your file. > > When you finish your 'longtime task' join the background thread. Yes, part of task is executed in AGI. But, I still do not understand how I can do something with Asterisk while it locks pbx execution for channel on particular application. I prefer simple solution like: same=>n,start-play-in-background-app(file-to-play) same=>n,do-one-stuff() a-lot-of-`same=>n,do-something`s-here same=>n,do-other-stuff() same=>n,stop-play-in-background-app() In generally I need something like PlayTones/StopPlayTones pair, but instead of giving list of tones/name-of-list I want give name of freshly generated file. -- _ -- 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-dev] How to play audio file in background in dialplan?
Hi I have a task. While serving incoming call I should do some longtime task(consumes more than few tens of seconds). So I decided to turn on background music in order to entertain caller. `core show applications` showed me 3 potential candidates: Background, Playback and StartMusicOnHold. Unfortunately, all of them does not meet my requirements: 1. Background and Playback block execution of dialplan. 2. StartMusicOnHold I cannot say which file it should play. Only class, which is inpossible for me because of file to play is of dynamically generated content with runtime obtained name, and I cannot predict and configure all possible pairs of file-name/class-name.(And don't want to use realtime to link class-name and file-name on fly). Any ideas? regards, Yaroslav -- _ -- 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] Is there any way to terminate async origination initialized by AMY?
Hi I have an application. It connects to Asterisk via AMI, and when user decides it begins asynchronous origination to some device. But very often user decides to break origination and make another call. How can I achieve that? As much as I see, Asterisk doesn't returns any ID of dial process and has not nay StopOriginate actions so AMI client can use them. What to do? This function is essentially important. Regards, Yaroslav -- _ -- 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] how to listen on different sip port for a device?
2011/12/26 sean darcy : > So how do I get * to listen to two different ports? sip.conf section [general] bindport=whatever-port-you-want -- _ -- 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] rasterisk not knowing config path?
2011/12/6 Danny Nicholas : > You don't state your Asterisk version, but this sounds like a task for > chan_skinny perhaps? Or it might just be as simple as hitting an RTP range. Asterisk >=1.8 No, I want manually connect to asterisk via ssh console. I.e. like: ssh user@ast-host /sbin/rasterisk - I know but did not found any in manuals. -- _ -- 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] rasterisk not knowing config path?
Hello I have machine running a couple of instances of asterisk. Each instance create own control pipe (asterisk.ctl). How I can remotely connect into asterisk which own pipe I know? I know I can do it if path to pipe specified in asterisk.conf, but I have not any asterisk.conf accessible, only control pipe. -- _ -- 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] AMI: anything to glue originate to events?
After origination successfully complete and channel will be created
you probably should link ActionID and channel name.
Origination action will be next:
Action: Originate
Channel: Local/1@internal
Exten: 384087
Context: SIP-UA-00128
Priority: 1
CallerID: 601
ActionID: FFA02C6A03
Variable: ActionID=FFA02C6A03
[SIP-UA-00128]
exten =>
384087,1,UserEvent(LinkOriginate,CHANNEL:${CHANNEL(name),ACTIONID:${ActionID}}
UserEvent application will generate event into AMI in form
Event: LinkOriginate
CHANNEL: channle-name (channel id created by asterisk)
ACTIONID: FFA02C6A03 (action id you set in originate)
They you(AMI client side) should associate received CHANNLE value with
ActionID, so later when you receive any event which contains channel
name you can easily find ActionID and do your work.
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Re: [asterisk-users] Licensing question.
I shall contact when(and if) decision will be made. But such decision cannot be made basing only on this paragraph, because it does not describes anything. There are no description of licensing procedure, nor pricing, nor liability, rights or freedoms(at least in general approximation) of sides. So I'm here asking and asking again. In any case, even usage of GPL-ed copy of Asterisk(or any other software) is illegal in my country. regards, Yaroslav. -- _ -- 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] Licensing question.
Greetings I have found next paragraph in Licence file(source root) "Digium, Inc. (formerly Linux Support Services) holds copyright and/or sufficient licenses to all components of the Asterisk package, and therefore can grant, at its sole discretion, the ability for companies, individuals, or organizations to create proprietary or Open Source (even if not GPL) modules which may be dynamically linked at runtime with the portions of Asterisk which fall under our copyright/license umbrella, or are distributed under more flexible licenses than GPL." What does it mean? Does it mean I can write non-GPL modules(BSD, MIT, etc)? Can I build my modules in common asterisk source tree(i.e. using LOCAL_MOD_SUBDIRS="my_mod_subdirs_list" make ) or must use separate tree? If so, then since Asterisk core does not accepts anything except AST_MODULE_INFO(ASTERISK_GPL_KEY, ) what I should do here? regards, Yaroslav. -- _ -- 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] custom automated meeting
You need simple dialplan of four steps:
same =>n,Set(conf_name=conf-${RAND(1,1000)})
same =>n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =>n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same =>n,MeetMe(${conf_name},dFI1xAC)
same =>n,Noop(do post conference stuff)
2011/10/31 Thanasis :
> I need your help in implementing the following scenario:
>
> A certain extension will ring two sip phones simultaneously and when one
> of them answers, the other keeps ringing until it answers too, and then
> all three (the caller and the other two) are immediately placed in a
> conference room (same room for all three).
>
> Can we do it?
>
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Re: [asterisk-users] Asterisk does not accepts SIP registration
2011/10/25 Tarek Sawah : > Hello, > Is L6 a remote device? is there any firewall residing between the server and > UA? > > > Tarek Sawah > > Information Technology Adviser > > Integrated Digital Systems > > CCNP, MCSE, RHCE, TELECOM > > USA: +1 386 492 9993 > L6 is account of DLINK DVG7022S VoIP router in LAN(via 100MBits switch). No firewall on asterisk's host. All packets are intact. -- _ -- 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 does not accepts SIP registration
Hello Always returns 401 Unauthorized, because of [Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on stale nonce received from '"L6" ;tag=31b9dc9e-684902' L6 is realtime device of type FRIEND (DLINK DVG7022S) Reviewed SIP conversation - no results. SIP debug <--- SIP read from UDP:172.30.8.18:5060 ---> REGISTER sip:172.30.8.13:5060 SIP/2.0 v:SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bKe3a0c285855ae0e5 f:"L6" ;tag=31b9dc9e-684902 t:"L6" i:BD2F-1923-466848179B9BEAA6258E-001@SipHost CSeq:23 REGISTER m: Expires:0 Max-Forwards:70 User-Agent:dlink 12-36-9924913 l:0 <-> <--- Transmitting (no NAT) to 172.30.8.18:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bKe3a0c285855ae0e5;received=172.30.8.18 From: "L6" ;tag=31b9dc9e-684902 To: "L6" ;tag=as1a9dabcb Call-ID: BD2F-1923-466848179B9BEAA6258E-001@SipHost CSeq: 23 REGISTER Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1555e540" Content-Length: 0 <> REGISTER sip:172.30.8.13:5060 SIP/2.0 v:SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bK629009e8cee7f7e3 f:"L6" ;tag=31b9dc9e-684902 t:"L6" i:BD2F-1923-466848179B9BEAA6258E-001@SipHost CSeq:24 REGISTER m: Expires:0 Max-Forwards:70 Authorization:Digest username="L6",realm="asterisk",nonce="1555e540",uri="sip:172.30.8.13:5060",response="f47b23120619eb9e6d184bafa48b92c9",algorithm=MD5 User-Agent:dlink 12-36-9924913 l:0 <-> [Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on stale nonce received from '"L6" ;tag=31b9dc9e-684902' [Oct 25 11:59:48] VERBOSE[2501] chan_sip.c: <--- Transmitting (no NAT) to 172.30.8.18:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bK629009e8cee7f7e3;received=172.30.8.18 From: "L6" ;tag=31b9dc9e-684902 To: "L6" ;tag=as014cd348 Call-ID: BD2F-1923-466848179B9BEAA6258E-001@SipHost CSeq: 24 REGISTER Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11195a41", stale=true Content-Length: 0 <> sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 100 t38pt_udptl = no ;tos_audio = none ;tos_sip = none ;tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = alaw type = friend host=dynamic context = noop-context dtmfmode=rfc2833 ;language = ru ;sipdebug=yes nat=no rtcachefriends=yes qualify=1 deny=0.0.0.0/0.0.0.0 permit=172.30.8.0/255.255.255.0 sip show settings Global Settings: UDP Bindaddress:0.0.0.0:5060 TCP SIP Bindaddress:Disabled TLS SIP Bindaddress:Disabled Videosupport: No Textsupport:No Ignore SDP sess. ver.: No AutoCreate Peer:No Match Auth Username:No Allow unknown access: No Allow subscriptions:Yes Allow overlap dialing: Yes Allow promisc. redir: No Enable call counters: No SIP domain support: Yes Realm. auth:No Our auth realm asterisk Use domains as realms: No Call to non-local dom.: No URI user is phone no: No Always auth rejects:No Direct RTP setup: No User Agent: Asterisk PBX SDP Session Name: Asterisk PBX 1.8.5.0 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify:No Legacy userfield parse: No Caller ID: asterisk From: Domain: Record SIP history: Off Call Events:Off Auth. Failure Events: Off T.38 support: No T.38 EC mode: Unknown T.38 MaxDtgrm: -1 SIP realtime: Enabled Qualify Freq : 6 ms Q.850 Reason header:No Network QoS Settings: --- IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text:CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text:5 Jitterbuffer enabled: No Network Settings: --- SIP address remapping: Disabled, no localnet list Externhost: externaddr: (null) Externrefresh: 10 Global Signalling Settings: --- Codecs: 0x8 (alaw) Codec Order:alaw:20 Relax DTMF:
[asterisk-users] device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable
Hello Have a setup of asterisk with realtime SIP devices. Trying to organise monitoring of my SIP devices. Once device registered, its state becomes NOT_INUSE (result of DEVICE_STATE(SIP/device) function). Simulating of device breakage - powerdown it. Waiting for a while (minute or two), retrieving DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI: sip qualify peer device load no result. What I did not configured? My sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 100 t38pt_udptl = no ;tos_audio = none ;tos_sip = none ;tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = alaw type = friend host=dynamic context = noop-context dtmfmode=rfc2833 nat=no rtcachefriends=yes qualify=1 deny=0.0.0.0/0.0.0.0 permit=172.30.8.0/255.255.255.0 regards, Yaroslav -- _ -- 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] Question on meetme and t option
2011/10/12 Kevin P. Fleming : > then Asterisk *could* stop sending audio towards the device connected to that > channel. > then Asterisk *could* send it a message telling it to not bother sending any > audio. I think in any case Asterisk must not halt any audio data stream. It is task of application(MeetMe) to do that(drop inbound frames or not to send outbound frames). The reason is simple: something may listen that particular channel(for example chanspy, or audio hook, or (Mix)Monitor application or something else), and it should hear audio in any case. I.e. audio/video data streams of active channel are not exclusively accessed by currently running application at any time. regards, Yaroslav -- _ -- 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] Passive wait in dialplan?
Unfortunately I don't know behaviour of Progress() function, so cannot make any conclusions. As far as I traced it back to tech- implementation, this call does not changes any state of channel. But I analysed only sip and dahdi drivers. Neither it plays any indication tones. 2011/10/5 Sammy Govind : > So here's what I think about your scenario: > CALL-FLOW > 1- Call come in to asterisk (channel not answered) > 2- Event is triggered and User decides what to do with call > 3- On basis of what user decided a variable is set. > 4- Asterisk on the base of that variable route the call further. > If this is the intended behaviour I'd make the dialplan which would be > something like. > DIAL-PLAN ALGO > 1- Progress() ; Won't Answer the channel and put the call in trying... mode. > 2- Generate User Evnt > 3- While(USERDECISION == "") > 4- Endwhile > 5- Execute anything on base of USERDECISION > This has some limitation due to progress. GUI user needs to decide fast as > progress will time-up and the caller will get NO_ANSWER from the system. > Queue can be used to put call on wait until something is decided by GUI user > but for that you'll have to use system resources and also answer the channel > first. > I hope some real expert here guide you in a better direction. > -- _ -- 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] Passive wait in dialplan?
Yes, something like that, but hold"-state should not answer channel. answer command will be given explicitly. or call can be transfered to Dial command, etc. 2011/10/5 Sammy Govind : > Can you please explain what you are trying to do? What I've perceived from > this thread is that you want to put call on hold (passively as in no > resources usage) and then on the base of some User's input from UI proceed > with the call accordingly !! > -- _ -- 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] Passive wait in dialplan?
I don't know much about queues, but if channel enter into queue it
should not change its state. I.e. not answer, no moh, no interacting
with user input(DTMF). Less I use unknown helpers, better my
configuration is.
Second issue which can appear using queues - its async state. User can
issue 2 serial commands, and I should have synchronisation tools. In
dialplan I using UserEvent application - which issues event in AMI,
with given data headers. Queue - is there any possibility to customise
queue join(or like) AMI event? Without patches(I already have made
some patches to core and wrote additional module to make * work as I
require).
2011/10/5 Nasir Iqbal :
> What about waiting in "queues"?
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.com/
>
>
>
> On Wed, Oct 5, 2011 at 1:35 PM, Yaroslav Panych wrote:
>>
>> Hello, everyone
>>
>> Here part of my dialplan context
>> [globals]
>> CMD_NOOP=0
>> CMD_DOSTUFF1=1
>> CMD_DOSTUFF2=2
>> CMD_DOSTUFF3=2
>>
>> [blah-context]
>> same => n,Set(COMMAND=${CMD_NOOP})
>> same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
>> same =>
>> n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1)
>> same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF2}"]?LBL_DO_STUFF2)
>> same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF3}"]?LBL_DO_STUFF3)
>> same => n,Wait(0.2)
>> same => n,GoTo(COMMAND_SWITCH)
>> same => n,NoOp(--- NOT REACHED ---)
>>
>> UserEvent sends blah-event via AMI to high-level UI, user makes
>> decision and issues some command via Action:SetVar, then dialplan
>> continues to work.
>>
>> The problem is, in dialplan there is an active wait loop, i.e. waiting
>> mechanism which rapidly checks some var(consuming processor resources
>> and flooding logs). Is it possible to create passive waiting loop
>> within current abilities of Asterisk 1.8?
>>
>> regards, Yaroslav
>>
>> --
>> _
>> -- 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 --
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[asterisk-users] Passive wait in dialplan
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same => n,Set(COMMAND=${CMD_NOOP})
same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =>
n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1)
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF2}"]?LBL_DO_STUFF2)
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF3}"]?LBL_DO_STUFF3)
same => n,Wait(0.2)
same => n,GoTo(
regards, Yaroslav
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[asterisk-users] Passive wait in dialplan?
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same => n,Set(COMMAND=${CMD_NOOP})
same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =>
n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1)
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF2}"]?LBL_DO_STUFF2)
same => n,GoToIf($["${COMMAND}"="${CMD_DOSTUFF3}"]?LBL_DO_STUFF3)
same => n,Wait(0.2)
same => n,GoTo(COMMAND_SWITCH)
same => n,NoOp(--- NOT REACHED ---)
UserEvent sends blah-event via AMI to high-level UI, user makes
decision and issues some command via Action:SetVar, then dialplan
continues to work.
The problem is, in dialplan there is an active wait loop, i.e. waiting
mechanism which rapidly checks some var(consuming processor resources
and flooding logs). Is it possible to create passive waiting loop
within current abilities of Asterisk 1.8?
regards, Yaroslav
--
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