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Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd...

...ustomizations allowed to setup the calls. Does anyone know how I could place automated outgoing calls that would have the proper sip headers added to it that would allow the call to be auto-answered? I've also posted this question to the forums here: http://forums.asterisk.org/viewtopic.php?f=1&t=89190 Many thanks, Steve -------------------- http://www.stevemccann.net -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140128/c2c72ccc/attachment.html>

Getting the some errors making dahdi 2.11.0. Seems same as listed here http://forums.asterisk.org/viewtopic.php?f=1&t=96455 In that link they say to use 2.10.2 but that's from December. Is there a fix yet for this? Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street Lafayette, IN 47905...

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1&t=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate it all here, I've posted my logs and conf files on that thread, too. Problem is that while there are quite a few sip examples...

...y tries to dial that extension. Features.conf is extremely barebones so I'm not sure where any problems would come up: [general] transferdigittimeout => 3 xfersound = beep [featuremap] blindxfer => # atxfer => *2 Tried change transferdigittimeout to something insanely long like 100, but still the same thing. I even fully restarted asterisk (just in case it was some weird fluke that crept in). This is 1.6.2.17.2. Someone else posted something extremely similar, though it's in version 1.8.2.3: http://forums.digium.com/viewtopic.php?f=1&t=77154 hose

Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1&t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was Music On Hold. Eventually I tracked this down to using br...

...e these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David Klaverstyn -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130219/7c9d1d62/attachment.htm>

Hi all, I am trying to enable SIP SIMPLE communication in my test environment. I have the following env : - one server (192.168.50.126) with Asterisk 11 - 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143 I successfully had a phone call between clients. I used the following link to enable SIMPLE messaging between my clients : http://highsecurity.blogspot.ca/2012/03/asterisk-10-110-sms-messaging-o...

Hi list, I've been googling this issue and found some good resources however I am still running into problems with the following combo ... Here's my story; - Asterisk 13.4 with FreePBX 12. - Migrating from Asterisk 11 / FreePBX 2.11 - Mix of Cisco 79xx handsets, mostly 7940G's. My problems started with (the very common) issue of the 7940 not replying to 401 UNAUTHORIZED with a second REGISTER containing the auth digest details. A quick Google fou...

Very rarely, but it has repeated, we see openssh on the client side hanging. On the server side there is no indication of connection in the logs. These are always scripted remote commands that do not have user interaction when we find it. This seems to be happening only in vm environments but I could be wrong. It seems surprising to me that there would not be timeouts and retries on the protocol,

Hi, I'm just trying to install the DPMA on my Asterisk. I already made the upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: *mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185 * *compiling Asterisk-Cert2 1.8.11* *./configure make make install make config * Afther that i register the DPMA license, and finally copied the * res_digium_phone.so* to */usr/lib/asteris...

...is wrong. If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly. Unfortunately, I?m using FreePBX, so it owns this file and my changes won?t persist a FreePBX reload. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>) www.OntheNet.com.au<http://www.onthenet.com.au/> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nilesh Govindrajan Sent: Wednesday, 22 July 2015 1...

Hi, I've already post this to the forum three days ago, sorry if it's sounds like a crosspost, but I've got no replies, so I'm trying other channels :) This is the link to the forum post if someone prefer to reply here: http://forums.asterisk.org/viewtopic.php?f=1&t=86985 I'm using Asterisk 1.8.20.0 (the freepbx build) with CEL logging activated. I'm using CEL because in our pbx we have different queues and trunks serving different customers (we are an inbound call center) and we need to detect when and how we have to bill our customers. I'm...

Hello, I made a post to the forums (http://forums.digium.com/viewtopic.php?f=1&t=72901&sid=3d5c2717ca5ab7ad676957ae436d4b51) but haven't received any replies, so thought I'd try here. On my debian machine running asterisk 1:1.4.21.2~dfsg-3, I've been noticing that there's a problem with conferences (using both meetme and app_conference) and the aud...

Hi, With latest asterisk-stat (2.0.1) : 1. In call-log.php file, there are lines with src="images/clear.gif" but there is no such images/clear.gif file. This produces : [Sun Apr 18 14:03:14 2010] [error] [client 192.168.102.102] File does not exist: /var/www/asterisk-stat/images/clear.gif, referer: http://192.168.102.240/a...

After upgrading to 1.8.3.2 today, I notice that my Aastra 480i SIP phones no longer initiate hold music when a call is placed on hold. I seem to be having the same issue as the person here: http://forums.digium.com/viewtopic.php?f=1&t=77553 Has anyone else run into this issue? -- Anthony - http://messinet.com -...

Hi, I would like to install K-Pacs (www.k-pacs.net) under WINE. If I open the K-pacs installer ("Installer V1.6.0 English.exe"), I receive the following error: wine: cannot find L"C:\\windows\\system32\\Installer.exe" I am using the latest version of Ubuntu & Wine. Any idea what the problem might be? I'm new to Ubuntu & have not used WINE before (so there is a big chance of me m...

hello, I'm trying to connect Asterisk to Google Calendar. The connection work fine but Asterisk don't retrieve any programmed event present on the calendar. Asterisk version 1.8.20.1 Any hint? Thank you - Bakko

Hello, I've got a test setup with 2 asterisk boxes: Asterisk1 with: asterisk 11.5.1 dahdi 2.7.0.1 Digium TDM400 with 2 FXO ports Asterisk2 with: asterisk 11.5.1 dahdi 2.7.0 Digium TDM400 with 2 FXS ports Asterisk1 has the following AEL Dialplan: context remote { s => { Answer(); Dial(DAHDI/g1/7005); }; }; When a call from Ast...

Hey all, I've got a question about including the SIP method "MESSAGE" in the SIP Allow header sent by Asterisk (version 1.8). I found a post (http://forums.asterisk.org/viewtopic.php?f=1&t=83638) on this question, but there was no response. I've also done a bunch of searching for "asterisk sip allow method message", but I'm not finding much. The problem is that the client, the softphone Jitsi, i...