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...the block that is connected to the TDM card. Does anyone have suggestions on where I should start trying to troubleshoot the root cause of the FXS problem? Obviously having to manually restart Asterisk/DAHDI every time the server reboots isn't a practical long term solution. Thank you, Noah Engelberth Direct Link Computer Systems -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100128/aab4bb3b/attachment.htm

...'t use a Goto/Gosub, or "Extension gotoextension, priority 0 returned normally even though call was hungup" if I do. I'm running Asterisk SVN-branch-11-r371592M on CentOS 6.3 64-bit. Asterisk is able to send using JabberSend via other processing in my dialplan. Thank you, Noah Engelberth MetaLINK Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120831/bee99fd3/attachment.htm>

...eries ONTs. IAX peers are ATCOM AG198 ATA gateways, either behind the ONTs (but on the same voice VLAN the ONTs use to talk to Asterisk) or on my Asterisk server's local network. The voice VLAN is a different subnet than Asterisk is on, but no NAT exists between the subnets. Thank you, Noah Engelberth System Administration MetaLINK Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120802/6e17f148/attachment.htm>

...seems to be returning a lot of things that are more or less abandoned projects. Does anyone have any recommended solutions for a CentOS 6 / Asterisk 10 "vanilla" server? Not opposed to something commercial, provided it actually works and isn't a disaster to set up. Thank you, Noah Engelberth MetaLINK Technologies System Administration nengelberth at team-meta.net<mailto:nengelberth at team-meta.net> 419-636-0999 ext 100 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120227/edf4aaa...

...ds PIN via XMPP 4) Asterisk receives PIN, validates, completes processing 5) Asterisk sends process response to client (if applicable), and then lets the call fall through 6) Asterisk fires the sendtodialplan incoming logic for the message that was sent in step 3 Thank you, Noah Engelberth MetaLINK Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120920/63f5d81a/attachment.htm>

...ses. I have verified the SPA phones BLFs do still show the correct Ringing and InUse&Ringing statuses if they subscribe directly to a SIP device's state with the hint - the issue only seems to be effecting Custom devices. Can anyone think of anything else I should check? Thank you, Noah Engelberth MetaLINK Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120820/94414517/attachment.htm>

...hanging the GotoIf - as long as the DUNDi switch is not active in [users]. With the DUNDi switch active, I get the warning message above and the call hangs up. Is there something I'm doing wrong in my config? Is this basically expected behavior that I need to adjust around? Thank you, Noah Engelberth MetaLINK Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120823/f0d2426b/attachment.htm>

...urned up very little for documentation on dependencies, gotchas, etc for either item, so I'm hoping someone here can help me get IMAP set up for my Asterisk box. The server's specifications are as follows: CentOS 5.3, 32-bit Asterisk 1.6.1.6 with addons 1.6.1.1 Thank you, Noah Engelberth Direct Link Computer Systems -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091008/0d9cdced/attachment.htm

Hello, our gateway receive incoming calls from an outside gateway for multiple DIDs. For some of them we want fax detection, for other no. To do so, faxdetect is set to yes, but how to disable the fax detection for a specific incoming DID? For those DIDs, we just want to forward the call to a real fax machine DID which will do the job. Thanks for any hint Regards -- Daniel

Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short

...sion if the version number changes". Given that I don't have ignoresdpversion=yes either globally or for this peer, does this mean that Asterisk will only honor new SDP packets if the version is higher, or will it honor any change? Or should I be looking somewhere else? Thank you, Noah Engelberth MetaLINK Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130826/16bcf3fa/attachment.htm>

...te to RINGING when the queue call gets created. Obviously, I can use membergosub to set the agent to "INUSE" after they pick up the call (like Dial() U), but is there anything that I can use to manipulate the channel that is calling the agent while/before it is ringing? Thank you, Noah Engelberth MetaLINK Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120820/cf8c4bb7/attachment.htm>

The easiest way is to just run the Dial() command to forward the call to the hard fax without ever Answer()-ing the call. Without an Answer() on the call, Asterisk can't listen for fax detection (because the call hasn't been set up and there is no audio leg yet). Thank you, Noah Engelberth -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Friday, January 16, 2015 5:59 AM To: Asterisk Users Subject: [asterisk-users] Disable fax detect on specific incoming DID Hello,...

...eing received "XMPP received from 'test account'", but no XMPP debug output occurs when I try to place my test call to send the XMPP message. Kinda going cross-eyed from looking at this - is there anything else I should try or anything wrong in my configuration? Thank you, Noah Engelberth MetaLINK Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120821/82865192/attachment.htm>

I'm trying to to to "dial1" if caller id match: but dial plan execute 220,n(dial1) regardless exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1) exten => 220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w) exten => 220,n, I was under impression that if condition is met it will jump to

...based on both servers' nonces, but Asterisk still rejects it with a 401. Is there something I should be looking at on the Asterisk side that would fix this problem? Does anyone know if there have been changes in authorization handling since 11.5.1 that would fix the issue? Thank you, Noah Engelberth MetaLINK Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140724/51b017be/attachment.html>

Hi ?? I have been trying for several days get 3 Cisco spa508g phones (firmware 7.5.5) to work with asterisk 11.6 cert1 and sla. I can get the phones to all ring when an incoming call arrives, and I see the slatrunk working. However the blf function does not work. If one extension picks up the call the others do not show the trunk in use.? And as you might expect the hold and outbound dialing does

Hi all, I've got a provisioning file that I use to configure Cisco SPA112's. I'm wanting to get this file to do 3 things for me. I want to turn T.38 on, Call forwarding off, and Call waiting, off for both lines. but it's not working. This is what I'm using to enable T.38 for line 1. <FAX_Enable_T38_1_>Yes</FAX_Enable_T38_1_>

...ustom Device state") same => n,Verbose(Updating: ${ARG1} Calls: ${GROUP_COUNT(${ARG1}@activecalls)}) same => n,ExecIf($[${GROUP_COUNT(${ARG1}@activecalls)} > 1]?Set(DEVICE_STATE(Custom:${ARG1})=INUSE):Set(DEVICE_STATE(Custom:${ARG1})=NOT_INUSE)) same => n,Return() Thank you, Noah Engelberth -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130312/9897babc/attachment.htm>

Hopefully someone can point me in the correct direction. I had a 1.4x system die on me yesterday, while I was prepping a new machine to replace it. Took the machine on site yesterday and spent the day and part of the evening getting things working. This morning, I finished up converting my dial plan, knowing there'd be calls of things that I missed. While testing, I've noted that all