search for: versaevel

...When I call in (with an agent logged in) I get to hear the MOH on the client side, hover no matter how high the hold time is, I NEVER get an announcement over my queue position or my estimated wait time? Both the incoming call and the agent are on SIP channels. What is wrong ? Kind regards, E. Versaevel

...call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads asterisk@host instead of X-Liteusername@host. Any idea how this is possible? Kind reagards, E. Versaevel

...t) Exten => s, 3, playtones(busy) But I can't hear a busy tone on my sip phone, the call is answered, I hear the test file playback, but no busy tone. I tried to enter the values directly into playtones, but that didn't work either. Am I missing something? Kind regards, E. Versaevel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041125/293c68fe/attachment.htm

...pickup the extension 712 is dialed again. I'm thinking something like putting a value in the Asterisk DB if the extension is busy, after that extension hangs up it should trigger the dial to the originating extension and on pickup it should dial the other extension :-) Kind regards, E. Versaevel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041221/021fdbeb/attachment.htm

...es (sync the conf file with rsync for example), however if the phones use a host=dynamic they wont be able to be called until they have reregistered themselves at the backup asterisk. Is there a SER like t_relay kinda thingy to let the backup know the locations of the Sip Phones? Kind regards, E. Versaevel

...softphone calls out to the sip provider and the sip provider returns an error (404 Not Found for example) the sip message is not relayed back to my sip phone, it just sits and waits for a timeout. Is it possible to relay the returned SIP (error) message to the softphone? Kind reagards, E. Versaevel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/3699ce44/attachment.htm

hi Username : 1000012 Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (gsm|g729|g726|alaw|ulaw) the above is from SIP SHOW PEER 1000012, and as it clearly shows, g.729 is preferred before alaw. If I dial this SIP - * - SIP from a phone with G.729 enabled, it uses G.729. However, if I dial from my cell phone - GSM - PSTN - * - SIP, the call uses ALAW, which I thought it

...rk, i can allso see the invite back to the phone when getting the call out of hold. Because of this problem attented transfers won't work correct either (since the other side of the call gets dropped before the call is transferred). All calls are SIP<-->SIP. Any ideas? Kind regards, E. Versaevel

...talyst Epress 500 to power the phones (poe), however if i power hem via the adapter and hook them up to a hub instead of the switch the phone does retrieve it's config but we would like to use the poe to power them. Anyone else using a Catalyst Express 500 with snom's ? Kind regards, Erik Versaevel

...eue_1 or queue_2 it allways rings everyone directly without checking if Agent1 is available or not. It should distribute the calls from queue_1 to the other agents only when agent/1 is unavailable and agent/1 should only get calls from queue_2 when all other agents of queue_2 are unavailable Erik Versaevel

Hi All, I have been reading through the list quite a bit, and I am going to post this more as a poll than anything else. I am working on setting up a very small business with something that resembles a professional voice system. My idea is to use Asterisk with a SIP provider and SIP clients. I currently have a Vonage account already. So adding the 9.99 a month Soft Phone would be easy.

Asterisk Monitor seems to be working fine. Though the problem I am having is the two files (in & out) muxing. I added ,m to the string, yet the call records two files still, and I get the resulting error, at the bottom. monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4 8:23-in.gsm

hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck.I know very well this is not kind a problem discussed in this group but i try my best and all in vein so finally i am here hoping you ppl helping me out.I discussed this problem in asterisk's-users group and adding feedback from asterisk-users group my configs are sip.conf [general] port=5060

I just recompiled my version from this morning's CVS Head. My systems voice files (voicemail, time etc) were playing nicely. Until that is I added an extension and now the files won't play. Worse than that, * thinks the files have played and goes to the next step in the dial plan. What gives? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem