similar to: delay to dial

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/53055c0c/attachment.htm

Hi! I am searching for the description of the available dialstrin options for the DAHDI channel (and also other channel types). I am not looking for outdated voip-info links, but for the authoritative source, e.g. something like "core show application Dial" Does such thing exists? thanks Klaus

Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source & destination? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/48076c6b/attachment.htm

When Asterisk establish a call through an outbound trunk, Is there any way I can know who hang up the call first? The caller or the party called? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091022/c8beaabb/attachment.htm

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf

How can I let asterisk immediately dials a trunk when off hook?

Hi all, I am using Asterisk CVS, and I am getting a huge delay in dialing SIP. This Asterisk box is taking calls from a PABX over ZAP, then dialing SIP users. So, a user '0251' dials from his phone, the PABX sends it the my Asterisk box, no delay, then I get a 15 sec delay, before it actually dials the end SIP user. 1 -- Accepting call from '0251' to '0834541083' on

Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk

I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next

AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: --

Hi, When I setup a user, I give them an extension like 570xxxxxxx. This is fine and dandy while in one area code, but we've since gone to other area codes. I'd like the user's to retain the ability to dial 7 digits no matter what number they have. Any thoughts on how to do that? EXAMPLE: User has number 7175551212. I want that when they dial 3235555 it dials 717-323-5555.

A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the

Hi recently i have this error on our server.... how i can fix it? # systemctl status libvirtd ● libvirtd.service - Virtualization daemon Loaded: loaded (/usr/lib/systemd/system/libvirtd.service; enabled; vendor preset: enabled) Active: active (running) since Sat 2017-07-01 11:08:50 EDT; 2min 54s ago Docs: man:libvirtd(8) http://libvirt.org Main PID: 5233 (libvirtd)

I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message " all our circuits are busy now" then few second later, the phone rings, going to the second route! And the call can be established, how can I get rid of this message?? thanks --------------

Are there any cards that work with * that do the VoIP-to-TDM processing on the cards, with multiple interfaces? The QuickNet Internet LineJack meets the description I believe, but it only has a single FXS or FXO. Are there any cards that have more than one FXS? Thanks. __ Dana Olson Disclaimer: The information transmitted in this message is intended only for the person or entity to which it

I'm currently running asterisk@home v 2.7. However I believe asterisk has inbuilt a system wide speed dial system. Preserved number range starting at 300. Just wondering if it's possible to view/backup/restore/modify this data without having to enter it in manually. e.g. 300 301 12345678 (to save phone number 12345678 in speed dial 301?) I'm looking at creating a new installation

I'm debugging a TxFax problem whereby the fax transmission fails. I suspect interrupt latency--some interrupt routine is holding its interrupt too long. I have all unnecessary services switched off and X is not running when I perform these tests. Some transmission are successful while others fail at random points. I've noticed that after I boot Linux, load zaptel, wcfxo, and wcfxs,