seems a possible culprit, now that I think about it more.) > > Anyway, here's the bits of: > ./chan_dahdi.conf > ./dahdi-channels.conf > ./extensions.conf > > [trunkgroups] > [channels] If you Are you able to dial in a different fashion? i see you have already tryed. Any way to make sure that Number1 picks up before calling number2? http://popupjammer.com/asterisk-unable/asterisk-unable-to-load-module-chan-dahdi-so.html
now that's solve.. we encounter this kind of problem Apr 16 10:39:29 WARNING: chan_zap.c:11062 setup_zap: Ignoring switchtype Apr 16 10:39:29 WARNING: chan_zap.c:11062 setup_zap: Ignoring signalling Apr 16 10:39:29 WARNING: chan_zap.c:11062 setup_zap: Ignoring rxwink -- as i've said on my first post at first dialer 1 and dialer 2 passed through the gateway server and suddenly we decide not to use gateway. cancelling call to 880193444344. http://forums.asterisk.org/viewtopic.php?t=72700
sorry im a newbie in asterisk and vicidial.. Rus-----------SfinxSofthttp://sfinxsoft.com Sfinx Сообщений: 672Зарегистрирован: 21 июн 2011, 23:40Откуда: Odessa Сайт Вернуться наверх Re: Received response: "Forbidden" from '"Anonymous" [email protected] » 11 ноя 2013, 13:45 Извиняюсь - неправильно использовал тэг. and lastly if i logged in a sip phone registered in dialer 1, dialer 2 will be the one who's dialing...
High Jump Champion "Weird" topology errors in QGIS In what spot would the new Star Wars movie "Rogue One" go in the Machete Order? But when I try to call an outside number, apparently there is a problem with the authentication, because although I define the dial account in the campaign, I can see in share|improve this answer edited Oct 13 '14 at 14:41 answered Oct 9 '14 at 18:37 dkwiebe 61135 You're right! Recommend This Page Join our mailing list * indicates required Email Address * First Name Last Name Search My Blog My Blog TagsApache Apple Asterisk Blog book Capture CentOS Cisco Cloud
Reply ab permalink Instead of "SIP/14075551234" can be whatever?? dial [email protected] dial = the command 14075551234 = the digits to send, so this could be anything you want it just has to match something in the context you specify @internal Reply Bhuvnesh permalink Hi I looking for AMI command to originate call to a specific number and once called party pickup the call and play ivr and then further according to why not try these out so how will i know of how many available lines my trunk has?
Thx Reply motekpc permalink wow, thanks for this. Reply Isaías permalink Thanks man, very helpful! Categories Business Cars Cisco Deals Internet Linux Random Tech Udemy Uncategorized VOIP Search Copyright © 2016 jonathanmanning.com. and leads are stock on hopper..
my manager.conf file: [general] enabled = yes ;webenabled = yes port = 5038 bindaddr = 0.0.0.0 [salman] secret=salman permit=0.0.0.0/0.0.0.0 read=all write=all my extension.conf file : [general] static=yes writeprotect=yes clearglobalvars=no autofallthrough=yes priorityjumping=no check over here Can three +1/+1 counters be considered one +3/+3 counter? Then i added 2 Phone Mapping entries:Username | Device | Extension | Caller ID user1 | SIP/user1 | 2000 | user1 user2 | SIP/user2 | 2001 | user2When i try Related posts: Monitoring Agents in Asterisk with ChanSpy How To: Install Asterisk Voicemail GUI Centos vmail.cgi How To: Asterisk Queue Configuration Example How To: Capture DTMF Digits(rfc 2833) from Asterisk with
more stack exchange communities company blog Stack Exchange Inbox Reputation and Badges sign up log in tour help Tour Start here for a quick overview of the site Help Center Detailed wooltear2n Posts: 32Joined: Fri Feb 05, 2010 5:55 am Top Reply with quote by okli » Fri Apr 16, 2010 11:43 pm Set it to the number of available lines The 1.1.1 release should occur in the near future but I cannot give you an exact date, it mostly depends on when I can get some bandwidth go through the issues. http://popupjammer.com/asterisk-unable/asterisk-unable-to-create-find-sip-channel-for-this-invite.html I have configured my manager.conf, I have a dialplan in my extensions.conf and I have created a user on iax.conf.
Do read manager's manual, look at the few pinned topics how to get started if you need to. and another is channel.c:2514 __ast_request_and_dial: Unable to request Local/[email protected] Does "Excuse him." make sense?
still the calls by the past campaign is there and so the new leads for campaign cant get through... [email protected] Сообщений: 40Зарегистрирован: 07 сен 2012, 10:06 Вернуться наверх Re: Received response: "Forbidden" from '"Anonymous" [email protected] » 11 ноя 2013, 13:47 Sfinx писал(а):Как защититься от этого?не юзать дефективные телефонные дистры, originate SIP/[email protected] extension [email protected] == Using SIP RTP CoS mark 5 [Jul 21 08:21:31] ERROR[C-00000001]: netsock2.c:271 ast_sockaddr_resolve: getaddrinfo(“sip-outbound”, “(null)”, …): Name or service not known [Jul 21 08:21:31] WARNING[C-00000001]: chan_sip.c:6058 create_addr: sort of.
wooltear2n Posts: 32Joined: Fri Feb 05, 2010 5:55 am Top Reply with quote by boybawang » Fri Apr 16, 2010 1:47 am you should try upgrading your version first Vicidial Actually i decided to use vicidialnow since i'm using centos.. Should I find punctures by immersing inner tube in water or hearing brezze or feeling breeze or how else? weblink If there's a 0 being added it's coming from someplace.
Replace elements in list larger than x times the magnitude of the previous value with the mean of its neighbours A published paper stole my unpublished results from a science fair acl.c:244 ast_get_ip_or_srv: Unable to lookup 'dyanmic' whenever this kind of problem exist the dialer stops dialing.. Blog of Asterisk Tools navaismo Salt of the Asterisk Posts: 1610Joined: Mon Dec 07, 2009 1:30 pmLocation: Mexico City, Mexico E-mail navaismo Top Re: Unable to request channel DAHDI by Right now i've changed my server and install vicidialnow.
or hang and then we need to reboot the dialers to dial again and after how many minutes, it will pause or hang again.. Leave a Reply Name: (required): Email: (required): Website: Comment: Note: XHTML is allowed. Reply Click here to cancel reply. Bought agency bond (FANNIE MAE 0% 04/08/2027), now what?
The other numbers got the REJECTED status in the Wombat logs.If I check the Asterisk logs, I have these lines:(...)[Jun 12 18:10:04] VERBOSE func_timeout.c: Channel will hangup at 2013-06-12 18:12:04.980 ART.[Jun Where should a galactic capital be? Do you have a CLI trace of what happens (make sure you send only one call - FreePBX logs get very hard to read quickly). My AGI is basically a java code which uses Asterisk-java.jar library for communication with asterisk server.
ded Сообщений: 11701Зарегистрирован: 26 авг 2010, 19:00 Вернуться наверх Re: Received response: "Forbidden" from '"Anonymous" Sfinx » 11 ноя 2013, 13:41 Как защититься от этого?не юзать дефективные телефонные дистры, безмозглые i think the default is 96 right? It seems that something is adding a 0 to all outbound number on the ISDN trunk ONLY. Why did Vader dislike Krennic?
will it be ok?