similar to: Dial Out IVR

Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both the software but when i start Asterisk, it exits with the following dump. Error Text Start========================= [res_crypto.so] => (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key

I installed Asterisk CVS-NHEAD-05/13/05-01:59:30 and placed few call in and through successfully. I was trying to set up the Realtime - picking the sip.conf and extensions.conf from mysql. I was going through some wiki pages, but what i don't understand is - which configuration change makes asterisk stop looking at extensions.conf and sip.conf for sip peers and pick the same from database.

Hi all, Can i make Asterisk stay out of the media path for call park feature? In the 'sip.conf' i made canreinvite=yes in the general section but it does not seem to take effect. I don't see any reason for Asterisk to withhold sending re-invite. I am testing the call park in the single LAN,both on caller side and reciever side i am using X-Lite phones. Any suggestions?? Thanks,

Hi, I have installed libvirt 4.9. libvirtd 4.9 takes a long time to come up. I enabled debug prints and also put my own prints Logs are below 2019-02-06 05:55:49.082+0000: 377: info : libvirt version: 4.9.0 2019-02-06 05:55:49.082+0000: 377: info : hostname: draco 2019-02-06 05:55:49.082+0000: 377: info : virObjectNew:248 : OBJECT_NEW: obj=0x558e782d8bb0 classname=virAccessManager

Hi, I have installed libvirt 4.9. libvirtd 4.9 takes a long time to come up. I enabled debug prints and also put my own prints Logs are below 2019-02-06 05:55:49.082+0000: 377: info : libvirt version: 4.9.0 2019-02-06 05:55:49.082+0000: 377: info : hostname: draco 2019-02-06 05:55:49.082+0000: 377: info : virObjectNew:248 : OBJECT_NEW: obj=0x558e782d8bb0 classname=virAccessManager

2015-02-27 10:25 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > O.K. So what does your existing Dial() statement in extensions.conf look > like? > apology, put the gateway was sangoma but is a openvox , all my outgoing calls out for this context: [my-mobile-out] exten => _NXXXXXXX,n,Dial(SIP/1003/${EXTEN},55,rT) exten =>

Hi! Is there any experience of converting vmware''s vmdk file to a domU image? Maybe via extracting vmdk (how?) -> build tar -> untar in domU? That would be very nice despite replacing the original kernel with a self-built one and worthy to write a script... :) Cheers, Sven. _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com

Hello all. I'm having a problem debugging an IVR I'm building. I can't see any reason this shouldn't be working. Firstly the asterisk version is: Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a i686 running Linux on 2006-02-17 22:44:48 UTC Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily

Hi? 1. When a mobile user dial an IVR short code , mobile network able to divert that call to Asterisk platform.? 2. There would be web servers which are holds Voice XML . 3. Asterisk would be able to redirect the mobile request to certain Voice XML server accordingly. Just for like this setup , how do we install asterisk with minimum of asterisk modules ? Do we need to install complete

hi, i facing a problem here. in my sip.conf, i specify a account like this, [1234] type=friend context=from-sip username=1234 secret=1234 nat=no canreinvite=yes dtmfmode=info mailbox=1234@default disallow=all allow=ulaw so i am able to login with username 1234 and password 1234 but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus

I was just reading the release notes for the latest SPA-841 firmware, and noticed that Sipura added support for "SIP-B" to this release. This apparently adds support for bridged line appearances, parking softkeys, called party ID, external missed call summary support, and a handful of other useful features. The release notes are available at

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind

Hi group. I have huge problem. My pickup exten #8 isn't working. This is what I have configured. pbx*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 #8 In sip.conf I have callgroup=2 pickupgroup=2 For called party and same for person that is trying to pick up the call. The person that is trying

Hi, I have several queues configured at my call center for different support levels. Today, something weird happened: - A client called queue 1 and was answered by an agent - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf - The user transferred the client to another Queue, by using the second channel and the XFer key of her

have you set up the db schema? and have you entered any sip data into the db? Sean -----Original Message----- From: Chandra [mailto:chandra@digital.com.np] Sent: Tue 1/6/2004 10:57 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] no results. i have been working with the retrieve_sip_conf_from_mysql.pl file and i have set everything as required. but when i

This evening I posted a new set of Digium G.729 codec modules to our FTP server and web site, for Linux x86 and x86-64 processors. They were built using GCC 4.0.1, and they now report the processor they were optimized for when they are loaded. The previous x86-64 module required a non-standard Asterisk binary configuration, so this was corrected. In addition, there was only a generic version

Hi, This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the "New York" time? What value I have to give to GMT offset value? Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

Greetings everyone, I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy allows you to communicate with multiple Asterisk boxes from a single point of contact using a variety of I/O formats, now including support for XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format. Astmanproxy is