similar to: Outgoing PRI CID?

Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and 1.4.7.1 on FreeBSD 6.2) [general] priorityjumping=yes With n+101: exten => 1337,1,Dial(SIP/zytek,5,Ttj) exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS}) exten => 1337,n,Hangup -- Executing [1337 at firma:1] Dial("SIP/113-087a3000", "SIP/zytek|5|Ttj") in new stack -- Called zytek

I'm calling 'deliver' from Postfix and in some cases from Procmail. I set this system up more than six months ago and it's been working flawlessly until yesterday (16:52:19 local time) when it, without any apparent reason, just stopped delivering mails! Lots of checking and googling (I've forgot how exacly I setup the system :), I made 'deliver' SUID and it worked

I can't seem to get the [s]tart to work in my extensions... ----- s n i p ----- [default] exten => s,n,Goto(s-${DIALSTATUS},1) exten => s-BUSY,1,Voicemail(${EXTEN}, b) exten => 2403,1,Dial(sip/${EXTEN},20,t) exten => _X.,2,Playback(pbx-invalid) ----- s n i p ----- If I dial '2403' with is off-hook, I don't get to the voice mail, I get the playback... Setting

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

Hi all, I've been beating my head against this for some time now. For incoming calls, I'd like to send my users a "localized" caller id number. By "localized," I mean one with out the 1+areacode for local calls and only 10 digits (minus the leading 1) for long distance calls. For example: I get a call from 15055551234. Since I live in the 505, I should see:

I'm trying to get my home directory (automaticly) mounted from the AD server via SMB - I can't (don't want to) use NFS, because that fucks up the file modes (even with SFU it's not compatible enough). Using LibNSS/LDAP and LibPAM/{LDAP,Krb5} will authenticate me, get me a ticket and overall work exactly as planed. However, my home directory isn't availible (I had to create

I'm using the kickstart file we developed for v5 on 6.2 in VM (kvm/qemu running on a CentOS 5.6), but it fails: Error partitioning. Could not allocate requested partitions: not enough free space on disks. There's nothing wrong with the disk. Both fdisk and parted works fine, I can view, partition and write a partition table and the disk is 10.5GB. However, looking through

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

I'm using Swedish on version 1.4.13. The full part of the log is: [Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any format [Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett (format 0x8 (alaw)): No such file or directory The word 'ett' means 'one'. We have two words for one: 'en' and 'ett'. Any idea how to fix/solve

Hi All, Asterisk 1.4.12 CentOS 5 My ISP account includes nearly 500 minutes of VOIP calls per month but the service is expensive for unbundled minutes. So I'm trying to find a way to keep an accumulated total of calls made through that trunk so that I can automatically switch to a lower-cost provider when my bundled minutes are used. The plan is to store the accumulated time in AstDB and

Sorry for this. This is most likely a HOWTO or FAQ question, but it's so much information and documentation to wade through so I hope someone could take a minute to answer anyway. If not, no worries. I'll get to it sooner or later :) I'm trying to understand what Asterisk actually is and the basic workings... I think I've understand what I need to get going, except one thing.

I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>

Is it at all possible to set a Global Variable freely whenever a context gets used without having to enter an extension priority to use SetGlobalVar? This is really limiting the dialplan for me. Heres an example of what I would like to be able to do. [globals] AREACODE= [local] exten=_NXXXXXX,1,Dial(SIP/${AREACODE}${EXTEN}/blah) [anyoldcontext1] AREACODE=313 include=local [anyoldcontext2]

Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty

hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission

Thanks for your reply, [globals] AREACODE=381 [outbound] exten => _NXXXXXX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN},80) did not work for me, any ideas? Thanks, On 04/27/2015 01:59 PM, Phil Reynolds wrote: > > > On 27 April 2015 21:32:42 BST, Motty Cruz <motty.cruz at gmail.com> wrote: > >Hello, > > > >I would like to add area code if clients dial 7

Hi All, Asterisk 1.4.12 on CentOS 5 Sorry for a question that I'm guessing is obvious to most of you. I'm trying to revamp my dialplan. When I first created it, I had something like: exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => s,2,Dial(${rgMain},${RINGTIME},t) exten => s,3,VoiceMail(main at default) exten => s,103,VoiceMail(main at default)

Will this work to match any number from the 770,404, or 678 area codes? _[404|770|678]NXXXXXX If this won't work, is there a pattern that will do this? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> .

Took some examples from voip-info.org to deal with call forwarding etc: exten => _*21*X.,1,NoOp(Unconditional Call Forward on extension ${CALLERID(num)} to ${EXTEN:4}) exten => _*21*X.,n,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) exten => _*21*X.,n,Hangup() Problem is that * don't hangup cleanly: Spawn extension (default, *21*2403, 3) exited non-zero on 'SIP/2401-081e7048'

Hi list! I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this; We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ... The customer wants that incoming call should go to one group with some phones in it, if the group is