http://readlist.com Asterisk-users Mailing List - New Threads feed en http://readlist.com/readlist-logo-tiny.gif http://readlist.com/ 156 30 http://readlist.com/lists/lists.digium.com/asterisk-users/15/78117.html Fri, 16 May 2008 09:41:21 GMT Hi, hope not to be OT :) after more than 3 years of PBX installations we can adfirm Asterisk is stable enough to be considered a good product but still we encounter a lot of problems when deploying a new PBX. It seems that the biggest problems are all networking related: one way voice (also inside a LAN), calls drops, etc... How do you face this kind of problems? Which diagnose tools/methods do you use? Thank you. ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78107.html Fri, 16 May 2008 07:54:46 GMT Hi Asterisk Users, I'm interested in how many concurrent calls Asterisk can process without troubles. I mean 1 Asterisk server (software) like either proxy or media server (any numbers will be appropriate). 1. Is there any limitations by the software? What is this number? 2. What is the maximum count of concurrent calls you've ever seen/tested? -- Thanks in advance Alexander Olekhnovich ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78106.html Fri, 16 May 2008 07:45:36 GMT Good day, I'm using a dial string as follows: Dial(SIP/${phonenumber},30,grM(screen^${pin})L(${300000}[:60000])); When I set a variable in the macro screen, it doesn't get passed back to the extension from where the dial was called. I can always put the result in the MySQL database, but that feels a bit overkill... the macro looks as follows: macro screen (arg1) { Wait(0.2); ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78101.html Fri, 16 May 2008 03:42:22 GMT Hi Everyone, I'm pretty new to asterisk but coming from a call center background; needless to say I am amazed. Here is my current dilemma; but first some info on my setup. I have 3 public IP's from my provider...my LAN sits under one behind a Sonicwall TZ-180, while my trixbox sits on another behind a Linksys home router. The trixbox is running off a Visionman Server and I have port forwarded all ports (-0-65555?) to the trixbox. When I am on my ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78100.html Fri, 16 May 2008 03:42:02 GMT Hi all, There is a setting called autopause in queue.conf to pause a queue member if they fail to answer a call. The autopause setting will pause the agent even when they are on the line. I want to know if it is possible to pause the queue member only when they don't answer after timeout? ango _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78098.html Fri, 16 May 2008 02:50:29 GMT Dear All Please provide me the details how to configure hard phone with Asterisk? If any one used Nortel phones Or Cisco IP phone 7940 Please let me know how to configure at the asterisk side as well as in the Device. Thanks in advance Regards Ovia Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78087.html Thu, 15 May 2008 23:36:09 GMT I have a lot of recordings from asterisk in a .gsm format. I would like to play these files from a web browser (IE, firefox and opera) What do I need to do in order to achieve this goal ? Thanks Julian _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78085.html Thu, 15 May 2008 21:47:01 GMT When I connect to various asterisk services such as VoicemailMain(), MeetMe() as examples, I do not get to hear the first greeting messages. I've tried adding a Wait(1) before or after the application but this seems to have no effect. Is there another setting/parameter I can play with to delay the start of the playback of these messages? TIA Al -- The way out is open! ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78084.html Thu, 15 May 2008 21:46:18 GMT I don't see why you couldn't use asterisk in a setup that large. It would require a number of servers, and SER to handle the registrations, and call routing and use asterisk for what its good at, ivr/vm. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78083.html Thu, 15 May 2008 21:43:51 GMT I will have a small shop with ~4 phones using an HP server with Asterisk on it, it has two NICS and so I planned on plugging one into the cable modem, and the other into the switch. I was going to let this box perform NAT for the company but I am concerned about QOS for the VOIP portion. Anyone got a similar setup and care to share what they successfully implemented? Thanks! jlc _______________________________________________ -- ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78078.html Thu, 15 May 2008 21:07:25 GMT I am receiving a NOTICE on CLI Received SIP subscribe for peer without mailbox I have installed 1.4.19.1 first and then upgraded to 1.4.19.2, In both the version I am getting the same notice. I have been using 1.4.15 earlier, where I never got this. Regards, Sanjay Rajdev _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78070.html Thu, 15 May 2008 19:16:29 GMT Just a quick question, wanted to see if anyone knew where the menuselect app stored your choices. I think it's menuselect.makeopts but I'm not sure...just thought someone might know. Sherwood McGowan P.S. I'll post here if I figure it out before there's a response :) _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78068.html Thu, 15 May 2008 19:02:57 GMT Everyone- We're looking at using some Citel gateways to serve one of our sites (40 extensions, Toshiba phones). I've found that people seem to like the product from demos, but I was wondering how many have some of the gateways in production and if they seem to do the job for the long run. -Jon _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78063.html Thu, 15 May 2008 18:06:11 GMT Hi all, What is maximum number of three party conferences can a quadcore 3GHz system can handle? All the parties a setup with G.711 codec. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78057.html Thu, 15 May 2008 17:22:03 GMT I hate to bring up an old thread, however, I'm implementing SLA as well. I've got SLA working, my tunk executes slatrunk(line1), and my polycom 650 phone rings on the SIP subscribed line (button 1) I'm assuming slatrunk sends the calls to the SIP/station1 SIP device, so the call will always appear on "button 1" on the end device? My question then is how does the receptionist or answering party know which line this ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78053.html Thu, 15 May 2008 16:28:29 GMT Hello in extentions.conf I have the following menu [voicemenu-custom-3] comment = testmenunew alias_exten = 6004 include = default exten = s,1,Answer exten = s,2,Background(thank-you-for-calling) exten = s,3,Agi(agi://10.10.10.155/noaction) exten = s,4,Hangup There is a windows server who manages to flow on the menu for the calling clients, the application runs properly on one server, for ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78051.html Thu, 15 May 2008 16:17:05 GMT Is there a way to force asterisk to ignore the first ring of a call without using Wait() ? When I active *72 call forward on my analog lines from the telco, they always send a single ring and then do the forwarding. Asterisk picks up essentially a dead line and rings the phones which gets really annoying. Thanks. ________________________________ This e-mail, facsimile, or letter and any files or attachments transmitted with ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78050.html Thu, 15 May 2008 16:13:34 GMT Alright guys and gals, I'm a little lost, I'm primarily a SIP/IAX based guy, and have ended up with a Zap installation. Everything was fine with our old provider when we were using PRI, but the new provider screwed up on provisioning and we've been temporarily stuck with a pair of EM Wink T's. Ever since then, we've been dropping 1-2% of all calls (in or out) and even more strange, when a call gets dropped, a phantom call was being generated ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78047.html Thu, 15 May 2008 15:50:17 GMT When I try to use ChanSpy, the following message is sent repeatedly to the console (wrapped for readability): WARNING[32125]: chan_sip.c:3709 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) This appears to happen because SLINEAR (frame type 64 / 0x40) is not considered a native format for the channel I'm trying to spy on, so sip_write() ... http://readlist.com/lists/lists.digium.com/asterisk-users/15/78044.html Thu, 15 May 2008 13:44:37 GMT I have a Wildcard FXO: Wildcard X100P (clone) in my Asterisk (1.4.17) machine and as of late, Caller-ID on it seems to be failing more frequently than not. Sometimes I get "callerid.c:613 callerid_feed: Caller*ID failed checksum" sometimes it fails without even that. In Zapata.conf I have: usecallerid=yes cidsignalling=bell cidstart=ring I'm in Bell Canada land if that makes any difference. Any ...