Re: [asterisk-users] Microsoft will enter VoIP market in earnest next year, says Ballmer
Everyone here is saying how it would be so great to have native desktop/outlook/exchange/etc support, but seriously, do you really think M$ is going to develop these products to use with the open source market? They're going to want to try monopolizing it and creating an environment where you need to use M$ VoIP products to take advantage of to try forcing users to buy their products like they do with everything else. On Tuesday 07 November 2006 05:28 pm, Dean Collins wrote: http://www.siliconvalley.com/mld/siliconvalley/business/international/as ia/15944981.htm There's not much in the article so only click through if super interested but I'm curious and looking for people's opinions. What application integration would you like to see between MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook and number pop I'm kind of curious what other functionality there is to be developed (I'd also like to see drop and drag from outlook into conference calls. What would you like to see in asterisk, if we get some solid responses we'll see about organizing some bounties to get it developed. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip answer one side , ring other side
Try: progressinband=no in your sip.conf. -Brodie On Friday 29 September 2006 08:07 am, antonio wrote: Hi, the scheme is this : xlite --- Asterisk --- SIP gateway --- PSTN When i make a call with xlite (sip) to asterisk on the display of xlite i see that the call is connected but the phone is still ringing .. What is the problem ?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install HUDLite Server
Yeah there are some problems with the docs, and the product itself isn't very impressive -- still bugs that existed for months that basically make it worthless for me to use. Anyway, since they didn't include ircd and the perl mods in the new package, just download and install ircd-hybrid from ircd-hybrid.com, and the perl modules it references using CPAN. If you use queues in your setup, don't even bother..it still won't track calls that come in on a queue. -Brodie On Thursday 14 September 2006 12:48 am, Zeeshan Zakaria wrote: The Linux documentation on installing HUDLite doesn't really say how to install it. I can download the hudlite RPM, but where are the rest of the RPMs indicated in the documentation. And then how and where is the fonality folder is created? Somebody needs to re-write the documentaiton page. Please guide me on how to install HUD Server, if anybody has installed it successfully. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell hardware ...
I'm using a Dell SC1430 that includes the Intel NIC and don't have any problems at all. Also using a TE210P and TDM400P w/ 4 FXS in the box. I've never had to reboot the box or restart Asterisk (except for kernel upgrades and * upgrades of course). -Brodie On Monday 11 September 2006 05:12 pm, Alan Bunch wrote: I was going to use a Dell 1425 for Asterisk build but I see on Digium's website that hardware may be problematic. Can anyone shed a litle more light on the problem. I see the Intel ethernet cards seem to cause problems. If I need to disable the onboard Intel on the Dell hardware I can I just need to know what to expect. How about the 850, any word there ? TIA Alan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText Queue Notification
I know this isn't answering your question, but what I did for queue notification was use softkeys on the phones that call a PHP script on the * box that'll output XML for the phone to parse and display the queue stats on demand. Of course your phone would need to have an XML parser or some other type of minibrowser. For sending SIP messages to my Snom phones I use Sipsak to display agent login info and their associated queue(s) so that it's easy for agents to know what their status is. -Brodie On Thursday 24 August 2006 10:33 am, John D. Coleman wrote: I was wondering if anyone was able to execute custom commands on a channel once a caller connects to an agent after being in a queue. The reason I ask, is because I would like to use SendText to send a message to the agent receiving the call to let the agent know how many calls are waiting in the queue. I tried using ChanSpy, but then SendText will send messages only to and from the caller who initiated the ChanSpy. One way I could get around this is if I found out how to use SendText from the commandline, like smsq. I'm pretty sure that's not possible because of the nature of SIP MESSAGE but I figured I'd ask. Thanks, John Coleman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E61
Using STUN isn't a solution to NAT either, as it won't work with symmetrical NAT, which is very common (or for at least to partially use symmetrical). I'll be interested to see how the Paragon Wifi phone fares out when it starts making an appearance in the US. -Brodie On Thursday 24 August 2006 08:23 am, Andreas Sikkema wrote: Anyone here use the Nokia E61 ? I am looking to invest in a wifi phone and I want to get the best. Is it good as far as reception ? That is of most importance to me. Thanks. I've tried it in the last couple of days. The biggest issue for me ist that it HAS to be on the same side of a NAT as the server it talks to (asterisk, ser, etc). If it is on the private side of a NAT and the server is on the public side, it doesn't work. I've read something on the Nokia forums that Nokia is aware of the problem and it will be solved. My problem is that they want to solve this using STUN etc, while I would prefer they also wouldn't have the software care if it is on the inside of a NAT like most other CPE's so our platform can take care of things. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [asterisk-users] Snom360 with 6.2.2 firmware
Well, you could just press the transfer button when the line starts to ring instead of waiting for someone to answer. -Brodie On Wednesday 23 August 2006 02:07 am, Giordano Grandis wrote: Thanks, but my problem is that I need to transfer a call, while the called party is ringing. I cannot wait that the called to call. Thanks again Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Brodie Macleod Inviato: martedì 22 agosto 2006 16.44 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] Snom360 with 6.2.2 firmware Although I'm not using this firmware, attended transfers on these phones are done like this (while talking to the person you want to transfer): 1. Press one of the other line keys and dial the destination number of the person you are transferring to (your caller on line 1 will be put on hold). 2. If the person answers and is ready to accept the call, press the Transfer button, and line 1 line 2 will be bridged along with you, after which time you can hangup the phone, leaving the caller and callee connected. -Brodie On Tuesday 22 August 2006 02:44 am, Giordano Grandis wrote: Hi all, I'm using a Snom360 with bristuffed asterisk and i want to known if is possibile realize somthing of this: I receive an incoming call and then answered I want to transfer it to a cell phone (or another pubblic number), so press transfer on the phone, call the number and only if the called party is avaible i want to transfer the call. Infact with the transfer key, when i send the number, i lost the state of call, and i do not known if the called party was avaible or no. Is there a way to realize this ? Thanks very much in advance Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom360 with 6.2.2 firmware
Although I'm not using this firmware, attended transfers on these phones are done like this (while talking to the person you want to transfer): 1. Press one of the other line keys and dial the destination number of the person you are transferring to (your caller on line 1 will be put on hold). 2. If the person answers and is ready to accept the call, press the Transfer button, and line 1 line 2 will be bridged along with you, after which time you can hangup the phone, leaving the caller and callee connected. -Brodie On Tuesday 22 August 2006 02:44 am, Giordano Grandis wrote: Hi all, I'm using a Snom360 with bristuffed asterisk and i want to known if is possibile realize somthing of this: I receive an incoming call and then answered I want to transfer it to a cell phone (or another pubblic number), so press transfer on the phone, call the number and only if the called party is avaible i want to transfer the call. Infact with the transfer key, when i send the number, i lost the state of call, and i do not known if the called party was avaible or no. Is there a way to realize this ? Thanks very much in advance Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing after answered on zaptel
Try setting: progressinband=no in your sip.conf -Brodie On Monday 14 August 2006 10:20 pm, Don Fanning wrote: Greetings List, I'm having a strange problem with my X100p card still ringing after the call is connected. Any idea on how to solve this? Using latest asterisk (not svn) along with latest zaptel driver. Thanks, Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
