similar to: Changing audio message to text message

Hi, I'd like to know if it's possible to configure Asterisk to automaticaly close calls when the BYE request hasn't been sent by any clients and the call still exists for Asterisk ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour 93120 LA COURNEUVE FRANCE E-mail : anthony at mbdsys.com Tel : +33 (0) 143 11 09 14 ou +33 (0) 148 35 20 46

Hi, Does Asterisk manager allow multiple clients to connect to an Asterisk instance using the same user account ? Thanks,

Hello, I have a queue with only one element and one agent member. I want that my call leave the queue after 30s. My problem is that my call stays 60s in the queue and my agent is called 2 times. Can you say me how can i do it please?? -------------------------------- [queue] music=default strategy=ringall timeout=30 maxlen=1 context=mbdsys announce-frequency=0 announce-holdtime=no

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

Hi, For a few weeks now, our asterisk server has been experiencing something very odd. From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707

I am having difficulty with the calculation of the power law exponent for set of nodes within a graph. Specifically, I am interested in the distribution of in-degree and out-degree among communities of web pages where the web pages are the nodes of the graph and the hyperlinks the edges. According to the literature, the distribution of incoming and outgoing links obeys a power law distribution

I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are

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Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get "May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite:

Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the "soon-to-become-available-in-the-uk" S685IP. Both systems have great feature sets and, on-paper at least, look to be the bee's knees. Anyone got any skeletons on them? Thanks Alan -- The way out is open!

Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo

Hello all! Does anyone have a good howto to setup Asterisk and Skype. Thanks Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080526/831b3824/attachment.htm

Hi, Anyone know what the settings in SIPDefault.cnf should be for Cisco 7940 phones this year? Came in today to find they'd all moved one hour ahead (NTP server is correct and ok). Found the "day" was set to "26", but on trying to change the settings to the below, my test phone isn't changing back: dst_start_month: March ; Month in which DST starts dst_start_day:

Sorry, I'm doing a mail test since I was not able to send any mails to the mailing list for about a week... Thanks,

All, I'm trying to configure queue agents w/ a DUNDi setup so that an agent can login to whatever server they please w/o any custom setup. In general this seems to work, agents login w/ AgentCallbackLogin into the incoming context (not a special queue context) and can receive queue calls. The problem is that since the incoming context is the same context as the normal incoming call context,

I am trying to build a simple queue for the receptionist phone. In other words, there is only 1 agent and that is the receptionist phone. I just defined a few lines in queues.conf [console] strategy = ringall member => SIP/4000 ;4000 is the console extension In extensions.conf, it is: exten => 4000,1,Answer() exten => 4000,n,Queue(console) exten => 4000,n,HangUp() I pressed

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EXTEN:2}) exten => _**XXX,n,Hangup() [BLF] ; Defines a BLF Hint for phones exten =>

I've a big problem with SIP forwarding back into 'ringing groups' creating what can only be described as call storms :-( I have a 'ringing groups' of SIP phones with an effective dialplan (much simplified) like so: ; Purchase ledger [ptsn_inbound] exten => _846061,1,Dial(Local/6061 at groups) .... [groups] exten =>

Hi, in the earlier version there was a sounds.txt with the transcript of the soundfiles. Does this still exist somewhere? Is there a plan to make speech synthesis available the same way as soundfiles, ie. instead of playing language/soundfile.wav, send the text to the speechengine and play the output...? Jay... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone know how to customize the order of the soft keys on a 7960 running SIP? All the documentation I could find is CallManager related. Specifically, I want to move the transfer function to the first set of buttons during a call.