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  • asterisk/freeswitch in nat/no-nat setup

    - by pQd
    hi, my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party voip providers [ since it's in different countries - those are different providers, but that's irrelevant ]. i was thinking about terminating sip calls on something like asterisk/freeswitch server and having all sip-devices log on just once to such server[s] - mostly to provide things like voicemail, groupcalls, redirections etc. it seems perfectly doable but there is one problem - i cannot find examples how to prepare for nat/no nat. for calls routed to from/to 3rd party voip operator - i'll need handling for nat/stun etc, but for handling of internal calls - i do not want any nat, all traffic should go via vpns to different branches. can you provide me some hints how to configure it? any tutorials? thanks!

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  • FreePBX: Asterisk in the Cloud (EC2) Audio Problems

    - by neezer
    Please pardon the newbie question, but I can't seem to figure this out. I followed the Voxilla's tut to the tee: http://voxilla.com/2009/10/15/voxill...p-by-step-1457 But in making calls, my softphones connect, yet no audio (in either direction). I know from poking around the forums that this is generally caused by two factors: NAT and audio codecs. I (being new to the arena), however, don't know which. I believe I have Asterisk and the clients restricted to just ulaw, and I also believe I have the correct ports open, and my externip set correctly (I think the Voxilla AMI does this automatically, since it's in the cloud). I'm a bit lost. I'd be happy to post whatever configuration files that might help, provided you tell me where they are on the filesystem. But like I said before, this is effectively a vanilla install of Voxilla's own FreePBX AMI. I'd appreciate any help or guidance here. Thanks!

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  • FreePBX: Asterisk in the Cloud (EC2) Audio Problems

    - by neezer
    Please pardon the newbie question, but I can't seem to figure this out. I followed the Voxilla's tut to the tee: http://voxilla.com/2009/10/15/voxill...p-by-step-1457 But in making calls, my softphones connect, yet no audio (in either direction). I know from poking around the forums that this is generally caused by two factors: NAT and audio codecs. I (being new to the arena), however, don't know which. I believe I have Asterisk and the clients restricted to just ulaw, and I also believe I have the correct ports open, and my externip set correctly (I think the Voxilla AMI does this automatically, since it's in the cloud). I'm a bit lost. I'd be happy to post whatever configuration files that might help, provided you tell me where they are on the filesystem. But like I said before, this is effectively a vanilla install of Voxilla's own FreePBX AMI. I'd appreciate any help or guidance here. Thanks!

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  • Search for specific call in asterisk log files

    - by chiborg
    In my Asterisk log file, I have a line like this (truncated): Executing [123@mycontext:1] Set("SIP/myhost-b7111840", "__INCOMINGCLI=4711") Now I want to do the following filtering while looking at the log file with tail -f: Match lines with a specific value for "INCOMINGCLI", storing the call ID (the "SIP/myhost-b7111840" part) Output all subsequent lines that contain the call ID. As a bonus, having a grep-like option like -A would be nice. I could do that easily in various programming languages, but how would I do it with standard UNIX commands like sed or awk? Can it be done with these commands?

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  • Asterisk: forward if peer unreachable

    - by Cedric H.
    I would like to respond to incoming calls by checking is a specific peer is reachable, and dial the appropriate number accordingly. Presently I did this: exten => 1200,1,Answer() same => n,Set(reachable=${SHELL(asterisk -rx "sip show peers" | grep ^cedrich-phone.*OK)}) same => n,GotoIf($["${LEN(${reachable})}" = "0"]?extoffline) same => n,Dial(SIP/cedrich-phone,20) same => n(extoffline),Dial(SIP/another-phone,20,tr) same => n,Hangup() Could you tell me if this is acceptable and if it can be improved ?

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  • handle actions diferent to Command with Asterisk::AMI

    - by rkmax
    I'm learning Asterisk :: AMI, but all examples deal with the action Command. i've tryed to run the following action (no success) my $action = $astman->action({ Action => "Agents" }); i have the following sub for print response work fine for Action => 'Command' if i try other thing diferent i dont get response in CMD, how i can get response from others Actions? sub print_response { my $action = shift; print "\nResponse: ", $action->{'Response'}; print "\nMessage: ", $action->{'Message'}; print "\nActionID: ", $action->{'ActionID'}; if(defined $action->{'CMD'}) { print "\nCMD: ", scalar(@{$action->{'CMD'}}); print "\n-------------------------------------------\n"; foreach (@{$action->{'CMD'}}) { print $_, "\n"; } print "\n-------------------------------------------\n"; } print "\nCompleted: ", $action->{'COMPLETED'}; print "\nGood: ", $action->{'GOOD'}; }

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  • asterisk outbound caller id

    - by MCS
    I'm trying to set the caller id number for an outbound call. My asterisk .call file looks like this: Channel: SIP/flowroute/1234567890 Context: test Extension: 1234567890 Priority: 1 Here's my extensions.conf: [test] exten => _1NXXXXXXXXX,1,Set(CALLERID(num)=8005552222) exten => _1NXXXXXXXXX,n,Dial(SIP/${EXTEN}@flowroute) exten => _1NXXXXXXXXX,n,Playback(hello-world) When I receive the call, the caller id number is 1-206-445-6979, even though the CDR log has both src and clid set to 8005552222. I'm using flowroute as my carrier. Is there something wrong on their side?

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  • Asterisk Register username with special character like "@"

    - by Najibul Huq
    I am using a SIP provider that has provided me with a username like: [email protected] (Note this is only the username part) And has a numerical password. My Register string looks something like this: [email protected]:[email protected] But this is not working, as asterisk is only sending the first part +112223344 before the first @. My provider is adamant about having the full form of it. This is the first time I am facing this issue that is quite unusual for me. Please help.

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  • asterisk Dial-plan?

    - by Rev
    Hi I want to make a dial plan for asterisk to do this: for incoming-call check the caller-id and if caller id is equal with specific number (for example 666) then hangup that call.(this dial-plan also known as anti ex-girlfriend ) also I wrote this dial-plan for doing this but it doesn't work well.(don't hangup then incoming call from 666 and go to queue macro) [macro-queue] exten => s, 2, Queue(${ARG1}) [default] exten => s, 1, Answer exten => s/666, 2 ,Hangup exten => s, 2, BackGround(welcome) exten => s, 3, Macro(queue,operator)

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  • A Deep Dive into Transport Queues - Part 1

    Submission queues? Poison message queues? Johan Veldhuis unlocks the mysteries of MS Exchange's Transport queues that used to temporarily store messages waiting until they are passed through to the next stage, and explains how to manage these queues.

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  • asterisk extensions.conf & sip.conf

    - by Josh
    I'm trying to get my Dialplan to work. When I call, the only thing I get is a dial tone to enter extension "no Background(thanks-calling) is played". When extension 123 is dialed, busy signal is triggered and asterisk CLI get frozen. Any help will be appreciate it. Conf files below. ; PSTN on sip.conf [pstn] type=friend host=dynamic context=pstn username=pstn secret=password nat=yes canreinvite=no dtmfmode=rfc2833 qualify=yes insecure=port,invite disallow=all allow=ulaw ; PSTN on extensions.conf [pstn] exten => s,1,Answer exten => s,2,Wait,2 exten => s,4,DigitTimeout,5 exten => s,5,ResponseTimeout,10 exten => s,6,Background(thanks-calling) exten => 0,1,Goto(incoming,123,1) ; (Member Services) [incoming] exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console exten => 123,n,Ringing() exten => 123,n,Answer() exten => 123,n,Playback(silence/1) exten => 123,n,Playback(connecting1) exten => 123,n,Wait(3) exten => 123,n,Dial(SIP/line1,60) exten => 123,n,Congestion

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  • Transfering call asterisk to different context

    - by Necronet
    I have a Small and basic PBX, and with two contexts wich basicly are sales and supervisor both have different roles and privileges. I notice that it is possible to transfer call from the same context but it have been imposible to transfer anything to another context. Any insight, i am kinda a rookie on asterisk but currently there is no one else in charge... Thanks Edit This is the extension.conf [supervisor] include => from-internal exten => _40XX,1,Answer exten => _40XX,n,Set(calltime=${STRFTIME(${EPOCH},,%C%y%m%d.%H.%M.%S)}) exten => _40XX,n,Set(CALLEDNUMBER=${EXTEN}) exten => _40XX,n,MixMonitor(/tmp/Para_${CALLEDNUMBER}-${calltime}-De_${CALLERID(num)}.wav) exten => _40XX,n,Dial(SIP/${EXTEN},40,TtRr) exten => _40XX,n,Hangup [sales] include => out-trunksip exten => _41XX,1,Answer exten => _41XX,n,Set(calltime=${STRFTIME(${EPOCH},,%C%y%m%d.%H.%M.%S)}) exten => _41XX,n,Set(CALLEDNUMBER=${EXTEN}) exten => _41XX,n,MixMonitor(/tmp/Para_${CALLEDNUMBER}-${calltime}-De_${CALLERID(num)}.wav) exten => _41XX,n,Dial(SIP/${EXTEN},40,TtRr) exten => _41XX,n,Hangup and the sip.conf looks like this: [supervisor] username=sales secret=ASUPERSECRETPASSWORD type=peer ..... context=supervisor mailbox=supervisor [sales] username=sales secret=ASUPERSECRETPASSWORD type=peer ..... context=sales mailbox=sales What do you suggest in order to get the supervisor with the same privileges that he already has and the sales been able to transfer calls to him

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  • asterisk extensions.conf & sip.conf

    - by Josh
    I'm trying to get my Dialplan to work. When I call, the only thing I get is a dial tone to enter extension "no Background(thanks-calling) is played". When extension 123 is dialed, busy signal is triggered and asterisk CLI get frozen. Any help will be appreciate it. Conf files below. ; PSTN on sip.conf [pstn] type=friend host=dynamic context=pstn username=pstn secret=password nat=yes canreinvite=no dtmfmode=rfc2833 qualify=yes insecure=port,invite disallow=all allow=ulaw ; PSTN on extensions.conf [pstn] exten => s,1,Answer exten => s,2,Wait,2 exten => s,4,DigitTimeout,5 exten => s,5,ResponseTimeout,10 exten => s,6,Background(thanks-calling) exten => 0,1,Goto(incoming,123,1) ; (Member Services) [incoming] exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console exten => 123,n,Ringing() exten => 123,n,Answer() exten => 123,n,Playback(silence/1) exten => 123,n,Playback(connecting1) exten => 123,n,Wait(3) exten => 123,n,Dial(SIP/line1,60) exten => 123,n,Congestion

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  • Asterisk dialplan context and label clarifications

    - by liv2hak
    I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the web.Can someone confirm if the IVR below is correct? Correct in the sense that if the below is used will it work.I know the IVR does not do much yet.But I am just trying to clarify my understanding. [incoming] exten => 123,1,Answer() same => n(menuprompt),Background(main-menu) exten => 1,1,Playback(digits/1) same => n,Goto(incoming,menuprompt,123) exten => 2,1,Playback(digits/2) same => n,Goto(incoming,menuprompt,123) exten => 9,1,Hangup() [main-menu] exten => n(menuprompt),Background(main-menu) exten => 3,1,Playback(digits/3) same => n,Goto(main-menu,menuprompt,n) exten => 4,1,Playback(digits/4) same => n,Goto(main-menu,menuprompt,n) exten => 9,1,Hangup()

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  • Ubuntu: package installed, but files missing?

    - by jeckyll2hide
    I have been playing around with the /etc/asterisk directory, installing the related pacakge (asterisk-config), removing it, removing the directory manually (just playing around to get the configuration synced to my configuration repo). Now I just want to reinstall the official package, so I do: root@tethys:/etc# apt-get install asterisk-config root@tethys:/etc# tree asterisk/ asterisk/ +-- manager.d What?! Empty?!? Have I installed it? root@tethys:/etc# dpkg --get-selections | grep asterisk asterisk install asterisk-config install asterisk-core-sounds-en install asterisk-core-sounds-en-gsm install asterisk-modules install asterisk-moh-opsound-gsm install asterisk-voicemail install Indeed! Let me check the contents of the package: root@tethys:/etc# dpkg -L asterisk-config ... /etc /etc/asterisk /etc/asterisk/res_snmp.conf /etc/asterisk/dbsep.conf /etc/asterisk/cel_custom.conf /etc/asterisk/cel.conf /etc/asterisk/meetme.conf /etc/asterisk/jingle.conf /etc/asterisk/queuerules.conf ... So, what have I done that the package will get installed, but the contents are nowhere to be seen? And, more importantly, how can I force the contents to be installed, no matter what I have done before?

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  • Type Declaration - Pointer Asterisk Position

    - by sahs
    Hello, in C++, the following means "allocate memory for an int pointer": int* number; So, the asterisk is part of the variable type; without it, that would mean something else (that's why I usually don't separate the asterisk from the variable type). Then what is the reason the asterisk is considered something else, instead of being part of the type? For example, it seems better, if the following meant "allocate memory for two int pointers": int* number1, number2; Am I wrong?

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  • asterisk public ip and clients with public / private ip

    - by user1165435
    I am using asterisk with a public ip. I have 4 clients which could be behind a nat or with public ip. I did set for all the clients nat=yes and canreinvite=no and qualify=yes. I did notice that if the clients were behind the nat everything went ok, but if the clients had public the call did not establish (no ringing on the asterisk server). WHere is the problem? Is there a bug in asterisk? As I've no there should be no problem for public ip and server with public ip.

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  • Asterisk auto Call recording

    - by Manjoor
    We are running asterisk with 8 port FXO. FXO connects to our old PBX (Samsung Office Serv 100). Now we want to record all calls routed through FXO (if it was dialed to outside or comming from outside). Here is the diagram |------|--------------------------------- | |--------------24 Lines ---------- Other clasic Phones PRI------ | PBX |--------------------------------- | | | | | |-----------|---------| | |--8 lines--| |--------- | |-----------|Asterisk |---------- 50 SIP phone |------| | |---------- |---------|---------- Is there a simple way to do this?

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  • asterisk : add application

    - by karthi-27
    Hi all, I want to know the way to add new asterisk applications and modules.For example I don't have the SetGlobalVar application in my asterisk machine.I want to add that.Is there any way. Thanks in advance .

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  • cant make outbound calls - asterisk

    - by deanvz
    I have a basic Atcom IP01 with the following config Registered Voip (SIP) Trunk Registered Voip Phone - ext Dial Plan Outbound Call rule I made use of this manual that the manufacturer supplies: http://www.atcom.cn/cn/download/pbx/ip01/ATCOM%20IP01-User%20Manual-V1.0-EN.pdf Whenever I try and make a call, it seems that the outbound call rule that i defined does not get regarded as the default rule even though the dial plan lists this as the only outbound call rule. When dialling I see in the log file the following [Jan 1 09:10:07] NOTICE[176]: chan_sip.c:14377 handle_request_invite: Call from '6001' to extension '00765243679' rejected because extension not found. The 00765243679 is a cellular number. Am I missing a configuration in order to make outbound calls? Land line, other Voip numbers and cellular calls have been tried

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  • Play music by Operator in asterisk?

    - by Rev
    Hi I want in call duration between operator and caller,play sound for operator(something like hold music). But in order to play this sound, operator must dial unique code and then sound will be play for caller, and caller only hear that sound file! After that (sound fully played), caller back to operator's queue or something like this. So is this possible to do or not? (if possible, post dial-plan for this too)

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  • Play music by Operator in asterisk?

    - by Rev
    Hi I want in call duration between operator and caller,play sound for operator(something like hold music). But in order to play this sound, operator must dial unique code and then sound will be play for caller, and caller only hear that sound file! After that (sound fully played), caller back to operator's queue or something like this. So is this possible to do or not? (if possible, post dial-plan for this too)

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  • Determine which user initiated call in Asterisk

    - by adaptive
    I had the following code in my extensions.conf file: [local] exten => _NXXNXXXXXX,1,Set(CALLERID(name)=${OUTGOING_NAME}) exten => _NXXNXXXXXX,n,Set(CALLERID(num)=${OUTGOING_NUMBER}) Now I want to change this code to set the CallerID and number based on the user/extension that is making the call. In fact I have four(4) users/extensions in my sip.conf and only one of them (the one I use for business) is supposed to send a different caller id/number. Everything is in the same context (for simplicity) since all lines need to be able to pick up an incoming call. The only difference is when line1 needs to make a call, it has to send a different caller id/number and use a different provider. This is what I have so far: [local] exten => _NXXNXXXXXX,1,Set(line=${SIP_HEADER(From)}) exten => _NXXNXXXXXX,n,Verbose(line variable is <${line}>) exten => _NXXNXXXXXX,n,Set(CALLERID(name)=${IF($[ ${line} = line1 ]?${COMPANY_NAME}:${FAMILY_NAME})}) exten => _NXXNXXXXXX,n,Set(CALLERID(num)=${IF($[ ${line} = line1 ]?${COMPANY_NUMBER}:${FAMILY_NUMBER})}) exten => _NXXNXXXXXX,n,Dial(${IF($[ ${line} = line1]?SIP/${EXTEN}@${COMPANY_PROVIDER}:SIP/${EXTEN}@${FAMILY_PROVIDER})}) I really don't know if this is correct and I'm afraid to commit these changes to my extensions.conf before validating. Any help will be greatly appreciated.

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