Re: [asterisk-users] Polycom Productivity Suite
On May 21, 2009, at 5:59 AM, Karl Fife wrote: While I have not needed to do this for myself, I believe you can create this functionality quite easily using Polycom's 'Enhanced Feature Keys' (EFK's). IIRC, EFK's are available in the newest firmware revision 3.1.x and newer. -Karl - Original Message - From: Matt Darnell mattdarn...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 21, 2009 3:04 AM Subject: [asterisk-users] Polycom Productivity Suite Has anyone been able to do the following: 1. Set the phone to automatically record all calls to the USB stick, now you have to press three keys. 2. Put Record on the main screen when a call is active. This would eliminate having to press the 'more' softkey. Thanks, Matt ___ -- 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 ___ -- 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 Yes with EFK in the latest firmwares you are able to change the on screen button layout. I used it to bring a Do Not Disturb button to the main screen of the SoundPoint IP330's. I may just be dense but paired with the Administrator and Developer guides from Polycom it was still rather frustrating getting the EFK working. If needed I could post that portion of sip.cfg to get you started. -- Robin D. Rodriguez Systems Engineer Ifbyphone, Inc. Phone: (866) 250-1663 Fax: (847) 676-6553 rrodrig...@ifbyphone.com http://www.ifbyphone.com ___ -- 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
Re: [asterisk-users] Need to find small footprint asterisk platform
what about http://www.rowetel.com/ucasterisk/ip04.html seems like what you might be after good luck Anthony Plack wrote: Hey all, I have a potential project which calls for a very small form-factor computer like this: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp However, I am needing an FXS port integrated into a small footprint computer. Nothing larger than a WiFi router or gateway device, but the smaller the better, and able to run Asterisk with at least a spare USB port and preferably WiFi on the system (but no necessary). Even a device that could integrate the S100U into the case would be good. Anyone know of a device like this? The AA40 is a bit much for my needs, and the cost is to high. Thanks in advance. ___ -- 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 -- Robin D. Rodriguez Systems Engineer Ifbyphone, Inc. Phone: (866) 250-1663 Fax: (847) 676-6553 rrodrig...@ifbyphone.com http://www.ifbyphone.com ___ -- 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
Re: [asterisk-users] Asterisk multi-cpu
Mike wrote: Hi, I know somebody is going to give me the link to the wiki hardware pages, but I can't find the answer there. I'd like to know if, for an Asterisk only system (nothing else of note running on it), I get a real gain from having 2 CPUs. Does the amount of traffic/SIP registrations/codec translation possible doubles with 2 CPUs? (each quad core E5420 to be precise)? Does it increase by 50%? It is only a marginal increase, or none at all? I wish I could test it myself, but I haven't bought the hardware yet and this will help me decide what I am buying. Regards, Mike ___ -- 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 In my experience I don't see any benefit from dual cpu's on asterisk boxes unless you are doing a lot of transcoding and generally I would suggest trying to avoid transcoding as it generally works out more cost effective in the long run than continually adding hardware. I've found chan_sip to be a limiter long before the hardware is stressed. -- Robin D. Rodriguez Systems Engineer Ifbyphone, Inc. http://www.ifbyphone.com ___ -- 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
Re: [asterisk-users] OT: Accountless, free, skinnable, browser based SIP client wanted
Is your client completely committed to using a web-based softphone and requiring them to make sure they have speakers turned on and a microphone plugged in? I think it's a fair guess that if they have a CC they have a phone which makes some sort of click to call technology more attractive to me at least For example my employer: ifbyphone.com offers products that do this, though there are plenty of others that do as well. Robin Steve Edwards wrote: I have a client that wants to put a phone on their web page for customers to call them via their Asterisk server. ) A keypad is needed to enter credit card details. ) Speed dial buttons like Tech Support, Sales, etc. are a requirement. Actually, passing the SIP address in the HTTP link would work with a bit of arm twisting. ) Free is preferred, but not a requirement. ) SIP is preferred, but IAX may also work. ) Cross platform is preferred, but Windows is the primary user base. ) They want it done yesterday. Gizmocall works fine, but requires their customers to create a gizmo account just to call my client and it does not have speed dial buttons. Any suggestions? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 -- ___ -- 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
Re: [asterisk-users] looking for a link or pdf ot something about opensip/openser and load balancing
this pdf http://www.dialogic.com/products/docs/appnotes/10833_HMP_OpenSER_SIP_an.pdf was enough to get me started with opensips David fire wrote: hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 ___ -- 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
Re: [asterisk-users] 2 asterisk boxes
Jerry Geis wrote: I am trying to setup a second asterisk box to play with console/dsp over sip. My sip.conf on the second box is: [secondbox] type=friend username=secondbox secret=secret disallow=all allow=ulaw allow=alaw allow=gsm host=SERVERIP context=consoledsp The second box is not connecting to my asterisk server. When I startup asterisk and I enter sip set debug I never see anything being displayed... sip show peers on the second box shows: sip show peers Name/username HostDyn Nat ACL Port Status secondbox SERVERIP 5060 Unmonitored 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline] However I never see connection attempts, I dont see anything being logged that its failing to connect. Sip show peers on the server has: secondbox (Unspecified)D 0Unmonitored running sip set debug on the server I never see a connection attempt from the secondbox to look at any error messages why its not connecting. I have done a service iptables stop on the second box. The server is OK as it has phones on it. How do I tell why the secondbox is not connecting to the server??? Thanks, Jerry ___ -- 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 Take a look at http://www.voip-info.org/wiki-Asterisk+config+sip.conf specifically the section labeled Asterisk as a sip client -- Robin D. Rodriguez Systems Engineer Ifbyphone, Inc. Phone: (866) 250-1663 Fax: (847) 676-6553 [EMAIL PROTECTED] http://www.ifbyphone.com ___ -- 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
Re: [asterisk-users] sip and nat
Johanna NIRINA wrote: I'm using asterisk 1.4 . There is some sip clients is behind a NAT : the asterisk server can't send request to these client. I'm looking for a solution to solve that in the server (asterisk) side. (sorry for my english). thanks, johanna _ Découvrez Windows Live Spaces et créez votre site Web perso en quelques clics ! http://spaces.live.com/signup.aspx ___ -- 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 Common solutions include stun or a combination of qualify=yes and/or nat=yes entries in sip.conf http://www.voip-info.org/wiki/view/Asterisk+sip+qualify -- Robin D. Rodriguez Systems Engineer Ifbyphone, Inc. Phone: (866) 250-1663 Fax: (847) 676-6553 [EMAIL PROTECTED] http://www.ifbyphone.com ___ -- 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
Re: [asterisk-users] Call monitor/barge/train
Mark Hamilton wrote: Hi, I’m planning on migrating someone who uses a very mature system. They would be logging in either as AgentLogin() or AQM. The main requirement however, is: The supervisor will have a control panel, where he will see how many of his agents are on call. If they are, he can “right-click” on the agent and get the options Call Monitor (where the super just listens in on the call, or new reps can listenin), Call Train (where the super and agent can talk to each other for training, but the customer doesn’t hear them, or older reps can train newer reps), Call Barge (where everyone can hear everyone else, super agent and caller) How or what can facilitate this? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users looking at the command Chanspy should give you a lot of relevant information. http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy -Robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] implementing an intercom with asterisk
Jonathan Disher wrote: I am looking to replace the phone system at my father's shop with an Asterisk box and some Cisco phones, but one piece of the implementation is tripping me up. He has two buildings (the office, and the shop proper), separated by about 3-400 yards. Currently with the ancient Meridian system installed, there is a paging intercom (to page employees, etc) on a dedicated extension - play a loud tone, then set up a 2 way channel. Anyone got any ideas, hardware wise, on how I might implement this with an Asterisk system? Thanks, and if this isn't appropriate for this list, if anyone has a better destination for the question, Id be quite appreciative. -j with pretty good success I've used some like this http://vikingelectronics.com/products/view_product.php?pid=317 with cheap grandstream ata's -Robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max amount of concurrent calls on and iax trunk
Rosli Sukri wrote: hi, wanted to ask if anybody has experienced setting up two asterisk 1.2 boxes connected via iax trunk. have u guys ever stress tested the trunks i.e how many concurrent calls can a trunk handle and whether codec has any effect on it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users What are the hardware specs of the boxes, and what is the speed of the connection between them? -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Penalties not working properly
Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn’t go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Penalties not working properly
Syed Nasruddin wrote: Hi, Actully the way I want the penalties functionality to behave it is not doing it accordingly. I am right now using ringall. Set penalty 1 for one agent and 2 for secnd agent. All the calls come in and go to first agent#1 having penalty one. But the second call also go to agent#1 and start waiting for it to be free rather it should have gone to penalty two agent#2 I have added call-limit=1 for bot sip accounts. And started the services. Still find the status wrong. nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Tuesday, August 05, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? rrmemory isn't ringall, it won't ring all members. But yes - you can use ringall with penalties. My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? What you are seeing is caused by status NOT IN USE. You have to set call-limit in sip.conf for all your phones, to any value, so that device states work correctly, and queue can know that those phones are busy. Now you probably can see in CLI that queue is sending second call to first agent(s). Regards, Atis Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Very carefully reread the descriptions on penalties and queue strategies on voip-info.org, the first time I tried to do what you want I was confused too, but I assure you if you read about the linear strategy you will find what you need. Robin -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
