Re: [asterisk-users] Nobody picked up in 20000 ms
On 06/20/09 21:49, Joseph wrote:
>When I call my internal extension the phone rings only once and goes to
>voicemail.
>It suppose to ring for 30sec. before calls transfer to voicemail;
>
> -- Executing [...@internal:1] Dial("SIP/11-00780380", "SIP/218|20|rw") in
> new stack
> -- Called 218
> -- SIP/218-00786910 is ringing
> -- Nobody picked up in 2 ms
> -- Executing [...@internal:2] VoiceMail("SIP/11-00780380", "11") in new
> stack
> -- Playing 'vm-intro' (language 'en')
> == Spawn extension (internal, 315, 2) exited non-zero on 'SIP/11-00780380'
>
>Does anybody knows why is it doing it?
>I'm using Asterisk 1.4.22.1
I forgot to mention that this is happening only when I try to dial locally
between some extensions when I come from PSTN line and dial the same extension
the
phones are ringing OK for 30sec.
--
Joseph
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[asterisk-users] Nobody picked up in 20000 ms
When I call my internal extension the phone rings only once and goes to
voicemail.
It suppose to ring for 30sec. before calls transfer to voicemail;
-- Executing [...@internal:1] Dial("SIP/11-00780380", "SIP/218|20|rw") in
new stack
-- Called 218
-- SIP/218-00786910 is ringing
-- Nobody picked up in 2 ms
-- Executing [...@internal:2] VoiceMail("SIP/11-00780380", "11") in new
stack
-- Playing 'vm-intro' (language 'en')
== Spawn extension (internal, 315, 2) exited non-zero on 'SIP/11-00780380'
Does anybody knows why is it doing it?
I'm using Asterisk 1.4.22.1
--
Joseph
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[asterisk-users] Limit transfers
I have asterisk installed in a callcenter: 60 DAHDI external lines and 72 SIP extensions, and have a BIG PROBLEM. Image a friend of one of the agents wants to call abroad paying the local fee. He dials to the callcenter, uses DID to get to his friend, asks him to place the call on hold and dial abroad and then hangup to bridge the call!!! I discovered this by chance when one of the agents, after answering an external call, tryed to hangup and place a new call. He really pressed the hook for a few milliseconds and was interpreted as a flash. After finishing the call he hanged, bridging the first call with the second. I cannot disable call forwarding, as the agents need it under certain circumstances. What I need to do is disable bridging two external lines. Has anybody faced this problem? Thanks, Daniel ___ -- 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] 400 calls at g711 how much cpu power
> > I am fairly certain he was simply reporting the results (for posterity) of > the event having already happened. Good to know (I guess?) that such > small hardware can acheive the performance that was squeezed out of it. > Impressive. > > All THAT said, I am unconvinced that there was no sales effort involved in > sending out millions of unsolicited calls. Claim if you like that this > was some public information event (which you fail to expand much upon) and > convict me of mistrust, but who would have paid for such a thing. TV ads, > radio spots, billboards, etc., are much more effective for public > information. Unsolicited calls on that order mean only one thing to me - > SPAM. So what wonderful product were you "informing" the public about > with regard to the looming threat of illness? Jeff, indeed i was posting for posterity. Maybe someone will benefit in an outbound-only scenario that he/she will not need a supercomputer to pump a 20sec audio clip. Again, this was a public service. And indeed TV and radio was used. Unless you live in a bubble, you may have heard about AH1N1 virus. Which unfortunately hit us (Panama, Republic of Panama, Central America) very hard. I foud very repetitive to tell in my posts that i am from panama, central america, blah,blah blah. Anyways, a quick google search of this forum will also revealed that i am kind of a regular poster and even my cellphone is listed here (Jon Pounder, my cellphone is +507 6675 5083 in case YOU want to sell me a car loan, i dont mind getting a call. Im a IT consultant and i have a chargeback line. Please call me as many times as you want...please do so between 10pm and 6am where my chargeback is the most expensive). Guys, Grow up! Next time someone needs to learn mouth-to-mouth and CPR lessons, please DONT teach him. Because, following your inmature way of thinking, the person who wants to learn CPR may as well be looking for information to learn how to suffocate people. Next time your son wants to know how gasoline works or how is being produced. Please keep your familiy in ignorance. You may be training the next crazy person who will burn things all around the world. But, you wont do that, do you? Again, I always tell my familiy that keeping others in ignorance is bad. but sometimes it must be done for the sake of a greater good, and my comment is always followed with good and sound examples (atomic technology, viruses, etc). But I forgot that Asterisk, the phone lines and a calling system is the way the world is going to be dominated by the martians. So the secret about phone system calculations must be keept in Area 51. Now I understand Kevin Mitnick. Cheers to all. Bye. > > Erick Perez > Cel +(507) 6675-5083 > > ___ -- 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 Talker Optimization
Hello, all. I've been playing with MeetMe and talker optimization seemed like a great idea. I activated it as follows: exten => 201,1,MeetMe(100201,cTo) However, although I can see who is the talker on the CLI pbx01*CLI> meetme list 100201 User #: 01 1001 Denise Dion-Sullivan Channel: SIP/1001-1e1db7c8 (not talking) 00:00:33 User #: 02 1000 John A. Sullivan III Channel: SIP/1000-1e1e11a8 (talking) 00:00:24 No one can hear the talker. It doesn't matter how loudly the talker shouts! Is there something else I'm supposed to do to enable this? I am using Asterisk 1.6.1.1 with CentOS 5.3 and Snom 320 and 360 phones. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Configuring Deltacom pri in Florida
Hi guys, Trying to do a pri pass-through with a duel port t1 card from Digium. I have port 1 setup to come in and out from Deltacom and port 2 to feed the pbx. Incoming calls work from Deltacom just fine, but outgoing do not. Anyone know what switch type I may need to get the dial-outs to work or anything else I might need? Thanks, Tom ___ -- 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] PRI cause codes
I am trying to retrieve the cause code of a outgoing call over a PRI
where the number called is out of service. When an out service number is
called I get a recording that the number dialed is not a working
number. I see cause code 1 in in the CLI as soon as the call is dialed
the Telco recording goes on for 30 sec. then hangs up. Any idea on how
retrieve info that the called number is is out of service. My
understanding is cause code 1 is an unallocated number.
thanks
Steve Casto
Asterisk 1.4.21.1
-- Executing [17609199...@admin3:1] Dial("SIP/Bob-00aaf150",
"Zap/G1/17609199147||g") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/17609199147
-- Zap/23-1 is proceeding passing it to SIP/Bob-00aaf150
-- PROGRESS with cause code 1 received
-- Zap/23-1 is making progress passing it to SIP/Bob-00aaf150
-- Channel 0/23, span 1 got hangup, cause 102
-- Hungup 'Zap/23-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [17609199...@admin3:2] NoOp("SIP/Bob-00aaf150", "102")
in new stack
== Auto fallthrough, channel 'SIP/Bob-00aaf150' status is 'CHANUNAVAIL'
from extensions.conf
exten => _1NX,1,Dial(Zap/G1/${EXTEN},,g)
exten => _1NX,n,NoOp(${HANGUPCAUSE})
from zapata.conf
> context=pri
> group=1
> switchtype=national
> signalling = pri_cpe
> priindication = outofband
> channel =>1-23
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Re: [asterisk-users] Dail in modem
If i understand correctly you need users to be able to dial in using a modem to your servers then you are going to share your internet connection with those who dial your server. So, no, it has nothing to do with asterisk... you want to be looking at wvdial for the clients (assuming they are linux) and whatever the equivalent server would be (don't know as i've never done it). Good luck 2009/6/19 ABBAS SHAKEEL > Geraint lee > > > I also dont know .what kind of requirements are these :P > > i am just looking if it can happen > > > On Fri, Jun 19, 2009 at 9:33 PM, Geraint Lee wrote: > > is it just me or am i right in thinking this has nothing to do with > > asterisk? > > > > 2009/6/19 ABBAS SHAKEEL > >> > >> Hello > >> > >> Actually i am required to make two application > >> > >> 1) that user use > >> 2) that is deployed on server > >> > >> > >> Application for user will be just like the windows standard connection > >> using dail up modem but user will dail my PSTN number instead of the > >> number we inter provided by ISP. > >> > >> on deployed server side we will get he usename and pass and other > >> parameters of application and then use them in java code > >> > >> > >> is it possible ? (nothing is impossible but for a Asterisk and java > >> developer with limited time frame) > >> > >> Thanks > >> > >> > >> On Fri, Jun 19, 2009 at 7:24 PM, Bob Pierce > wrote: > >> > > >> > On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: > >> >> I am required to do some thing like Dail in modem . > >> >> User will have to call a modem just like we do in dail up connection > >> >> now we need to handle that request and retrieve some parameters > >> >> from that send a HTTp request to a web server and then after getting > >> >> http response send user a feed back .. > >> >> > >> > > >> > Why do you need a modem? What will be dialing into the Asterisk > system, > >> > a human or a machine? > >> > > >> > ___ > >> > -- 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 > > > > ___ > -- 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] [SPAM] RE: newbie questions
C. Savinovich wrote: > Let me see if I get you: you inserted the installation CD, then you > restarted the computer, and now you want to know what to do next? > > CS > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Poe > Sent: Friday, June 19, 2009 11:25 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] newbie questions > > I have an Asterisknow.org CD. When I boot up, it seems ready for me to > choose update, console, etc. I'm assuming I need to do something at the > CLI prompt. Is there a tutorial that would take me from loading CD to > making first test call? > > Computer is Dell Optiplex GX260 > 50GB free disk space > 1.5GB RAM > P4 processor > external mic > speakers > Skype is on board, and would be good to use it, if possible. > > If I want to use Skype, do I need anything additional? Would it be > better to install CD on my hard drive? Any help appreciated. > Tom > > ___ > -- 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 > > Hello: Cancel these questions. It's looking more and more like I need to have a dedicated computer for proper Asterisk implementation, even for a home single line user. I found the link for ordering O'Reilly Asterisk book. Will spend time with that, first. Thanks for those who have responded. By the way, my AsteriskNow CD had no install guide with it. Tom ___ -- 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] Join Asterisk Global Meeting Sunday June 21 using VOIP - BerkeleyTIP
Come help BTIP get our new Asterisk box configured for VOIP conferencing. Especially - install a monitor that shows the people in the conference, & perhaps info like volume level, etc. :) == Join the friendly Asterisk Global [& ALL FREE SW HW & CULTURE] community Meeting: this Sunday, June 21, using VOIP, 10A - 6P Pacific USA time [GMT - 8? 7? hours] = 1P - 9P Eastern USA = 6P - 2A??? GMT - Daylight savings correction? +7 hours? at the BerkeleyTIP Global Free SW HW & Culture meeting http://sites.google.com/site/berkeleytip/ CONNECT VIA VOIP (& IRC): Join IRC channel #berkeleytip on freenode.net, & we'll help get you connected on VOIP. Have a VOIP headset. http://sites.google.com/site/berkeleytip/remote-attendance LOCAL MEETING NEW LOCATION: Free speech cafe closed Sundays in summer. Watch the BTIP local list for final in person UCB meeting location details: http://groups.google.com/group/BerkTIP GENERIC HOURLY SCHEDULE, & MARK YOUR CALENDAR - NEXT 3 MEETINGS: Sun June 21, Sat July 4, Sun July 19 http://sites.google.com/site/berkeleytip/schedule Join the mailing list, introduce yourself, tell us what projects you are interested in, invite others to join your project: BTIP-Global http://groups.google.com/group/BerkTIPGlobal = HOT TOPICs: Oracle owns OpenOffice & MySQL - What help is needed? KDE 4 aps need work - especially Calendar?! Open Hardware - Robotics? POLL: ** How about 2x per month Weekday Evening BTIP-Global Meetings? ** 1) The Wednesday & Thursday _after_ the BTIP weekend meetings? 2) The Monday & Tuesday _before_ the BTIP weekend meetings? 3) Other? Your suggestions? - Join the mailing list & send in your opinions/thoughts/suggestions. GROUP PROJECT - Asterisk VOIP conference server: We've now got our own Asterisk VOIP conference server. [Thanks, Windsor & Jack! :) ] Help: - get a user channel members status page working - get SIP & Skype ability? http://sites.google.com/site/berkeleytip/remote-attendance YOUR PROJECT - LET US KNOW, GET SOME VOLUNTEER HELP: http://groups.google.com/group/BerkTIPGlobal VIDEOS - OPPORTUNITY - FINDER VOLUNTEER NEEDED No videos this month, cause David & I are too busy. Do you want to find some for us all to watch? Check out this link, email the list & let us know you'd like to volunteer. :) http://sites.google.com/site/berkeleytip/talk-videos See the mailing lists for the latest info/changes: http://sites.google.com/site/berkeleytip/mailing-lists JOIN THE ANNOUNCEMENT LIST - 1 or 2 announcements per month: http://groups.google.com/group/BerkTIPAnc FOR FORWARDING: You are invited to forward this message to anywhere appropriate. ___ -- 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] newbie questions
On Sat, 20 Jun 2009, C. Savinovich wrote: > Let me see if I get you: you inserted the installation CD, then you > restarted the computer, and now you want to know what to do next? How about: 1) Turn off the computer. 2) Read the installation guide for the CD. 3) Install the software. 4) Read ATOF to get a clue to the scope of what Asterisk can do. 5) Get frustrated trying to do really cool things within the GUI. 6) Format the drive. 7) Install CentOS. 8) Install Asterisk from source. 9) Learn to configure the configuration files "by hand." But then, I gladly admit to being a command line weenie. 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] Fw: RE:Nagios under *[solved]
On Sat, 20 Jun 2009, Sriram wrote: > Thanks for all your help, i followed your answers and found on that > nagios was being run as user nagiosand if i executed the last > command it asked for a password [i tried nagios password,root password > etc] but it did not work..it the end i opened nagios.cfg and changed the > NAGIOS_USER to root and changed the ownership permissons on the script > also to root..I now get the correct status on the Nagios interface.. It would be better to run Nagios as nagios and add nagios to sudo so it can execute "asterisk -nrx" as the user executing Asterisk. The following line in /etc/sudoers should do the trick. nagios ALL=(ALL) NOPASSWD: ALL This, unfortunately, will allow the nagios user (or anybody who hacks into nagios) to execute any command as root, but it is safer than running nagios as root. Sudo has the facility to allow a user to execute a single command as another user, I just don't know the syntax off the top of my head. Google is your friend... Something like: nagios ALL=(user-running-asterisk) NOPASSWD: /usr/sbin/asterisk should get you close. 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
