[asterisk-users] time on asterisk
I am also having this problem using includes based on time of day, however the restart did not help and when enabled it finds no context with extension 's'. This is for my incoming calls, see below...Any Ideas! [default] include =extensions include = after|18:00-7:29|mon-fri|*|* include = during|7:30-17:59|mon-fri|*|* ;include = after|*|sat|*|* ;include = after|*|sun|*|* ;always on;;; ;exten = s,1,Answer ;off ;exten = s,2,Background(/tmp/afterhours) ;on ;exten =s,2,dial(sip/202sip/203,20) ;exten=s,3,hangup() ;maingreeting [during] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten =s,3,Set(TIMEOUT(DIGIT)=3) exten = s,4,Background(/tmp/maingreeting) exten =s,5,waitexten(5) exten =s,6,dial(sip/201,20|t)) exten =s,7,hangup() exten = 1,1,Goto(dental,s,1) exten =1,2,hangup() exten = 2,1,dial(zap/g2/322) exten = 2,2,voicemail(u322)) exten =2,3,hangup() exten = 3,1,dial(sip/241,20) exten = 3,2,voicemail(u241) exten =3,3,hangup() exten = 4,1,Goto(medical,s,1) exten =4,2,hangup() exten = 6,1,dial(sip/232,20) exten = 6,2,voicemail(u232) exten =6,3,hangup() exten = 7,1,Goto(admin,s,1) exten =7,2,hangup() exten = 0,1,dial(sip/201,20) exten = 0,2,voicemail(u201) exten =0,3,hangup() ;;;Nightmode [after] exten = s,1,Answer exten =s,2,Set(TIMEOUT(DIGIT)=2) exten = s,3,Background(/tmp/afterhours) exten =s,4,waitexten(10) exten =s,5,hangup() exten =5,1,Dial(zap/g1/,20) exten =5,2,hangup() ___ -- 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
[asterisk-users] Adtran supervision problems
I am sending a call down a EM wink trunk to a adtran tsu600 channelbank. The extension is setup like so... exten=799179,1,Dial(zap/g2,20,D(9179)) exten=799179,2,Hangup() It should Dial the Adtran and send some DTMF signals to a telephone on an fxs module in the Adtran. Asterisk is seeing the call answered when the T-1 is picked up by the Adtran not when the ringing phone is answered. This means the digits have already been sent by the time the ringing phone is answered. Does anyone have an idea on how to signal this correctly? ___ --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
[asterisk-users] Cdr reports
I am trying to figure out how long a caller waited in queue for someone to answer versus how long they stayed on the phone after the answer. I am assuming that the duration is the total talk time and that the billsecs are the total time in queue. is this correct? or should i be deducting the billsecs from the duration to get this number? ___ --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
[asterisk-users] Agents being bounced from queues after a call and sometimes randomly...
I am having a serious problem with agents being logged out of the queue after they finish a call. I am using static agents and agents.conf. I am running 2.1.17. Anyone having these problems or could think of anything that would cause them. Jordan Novak Telecommunications Engineer ___ --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
[asterisk-users] Polycom echo problem
I have three polycom 501 that are all hearing echo. The other party sounds fine but you can hear yourself rather well. The volume does help if lowered but that also makes the other party extremely quiet. Is there any way to control the gain of the mic or stop the microphone from picking up so much from the handset. It only happens while you are on the handset. ___ --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
[asterisk-users] Binding to multiple addresses
I have a simliar problem as the port binding question. I have a four port parelell processing NIC, I would like to team them together. Can I do this in asterisk if they are not actually teamed in hardware. I would be binding to several addresses simultaniously. ___ --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
[asterisk-users] addqueuemember recording and reporting
On 6/4/07, Jordan Novak [EMAIL PROTECTED] wrote: I am having a difficult time with the transition from agentcallback login... Here are a few of the isssues, I am logging in using chan_ local ie:local/8000 as the extension I'm not sure if this will solve any of your problems or not, but I've found it's often necessary to use the /n on the end of a local channel to make it work correctly for queues, as in Local/8000/n Call Detail records no longer show agent/ as the dstchannel show agents no longer shows the channels state If I'm understanding correctly, you're no longer using agents, you're using dynamic queue *members*. There's a subtle but important difference. show queues does not show the member Now this is strange. How are you adding the Local/8000/n as a member of the queue? Through AddQueueMember()? Through the manager interface? -Jared I am using local/8111/n now. I am using the manager interface to log in and out. There has to be a way to view the activity in the queue. If show queue and show agents doesn't work this really isn't going to do much for anyone running a real call center. What huge thing am I missing? ___ --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
[asterisk-users] addqueuemember recording and reporting problems
I am having a difficult time with the transition from agentcallback login... Here are a few of the isssues, I am logging in using chan_ local ie:local/8000 as the extension Call Detail records no longer show agent/ as the dstchannel show agents no longer shows the channels state show queues does not show the member Can anybody help? I have a ton of time invested in applications I developed that rely on this. ___ --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
[asterisk-users] Web based call control
Does anyone know if it is possible to use a manager command to answer an incoming call and not consider it answered unitl it is received. Here is an example, I am deivering a call in the dialplan to a home telephone number. I don't want his voicemail to answer and I have no idea how long it will take to go to their home phone voicemail, but I don't want to deliver the call there, I want it to go to the next priority in asterisk. So I was thinking that it would be nice to build a web interface that they could have a button to answer with. This would send a manager command to the server telling it to answer the channel, any thoughts on how to do this. winmail.dat___ --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
[asterisk-users] agentcallback login kicking agents out after call completion.
Has anyone had this happen to them using chan_agent. It does not happen all the time. ___ --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
[asterisk-users] polycom boot server...
I have to re-image one phone, I do not want to setup a small network with DHCP and FTP to get it done. Can I just point the phone at the server manually to try to bypass putting another dhcp server on my network. ___ --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
[asterisk-users] Agents.conf feature replication using addqueuemember
I (have to) would like to move my agents out of agents.conf in preparation for the deprecation of agentcallback login. Everyone I have spoken to is upset about this but the functionality can be accomplished in the dialplan and that is fine by me. I do have an issue with losing the features contained in the agents.conf though. I have to have the ackcall-yes working. All of my agents login on home phones and cellphones. When a call get presented to them there is a possibilty that their voicemail will answer before the queue timeout is reached. I fear that may connect callers to the voicemail when it answers. We currently get around this using the ackcall-yes which will wait for the caller to press pound, which a voicemail will not do and therefore the call will be put back in the queue. Other features that are important are the recording(which can be done on the dialplan side) and the update cdr. Both are important as the Monitor/mixmonitor will not name the file to be assoicated with the agent and only agent related calls. IE: if you have to look up the recording based on extension you will get personal calls as well as queue calls. Updatecdr is invalueable for the same reasons in the call detail records. Is anyone aware of a way to accomplish these things? Are any efforts being made to replicate the missing features? I can deal with the cdr and recordings, but the ackcall=yes is a show stopper!!! ___ --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
[asterisk-users] Queue report statistics
Here is the run down... billsec is talk time duration is wait time dst is the queue extension lastdata is the queue name lastapp will show logins dstchannel is the destination agent disposition is answered or abandoned status Mysql example to show all agent call detail for agent 8000 on queue number 8877... ( I have a bad habit of using like statements, this will work with = if you type better than I normally do, just lose the %) SELECT * FROM cdr WHERE dst LIKE '8877%' AND dstchannel LIKE 'Agent/8000%' ___ --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
[asterisk-users] Queue call distribution
I have noticed that asterisk will only try one interface per queue at a time. Is there any way get get it to dial say three at a time and connect the first one that it reaches. ___ --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
[asterisk-users] Queue application strategy
I am using rrmemory for my queues. I have noticed that the application will only distribute or dial one number at a time. Is there a different strategy that will allow the queue to distribute more than one call at a time? I don't want to use ringall because that would tie up thirteen of my trunks every time it tried to distribute a call. Any thoughts? ___ --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
[asterisk-users] Queue priority
I am running 1.2.11, I have had the queue priority lock the system up twice in the last week. I will forgo how I know this is what caused it, just trust me. I used the weight function in queues.conf to add priority to the queue. Is there another way to do it that will make it more stable? ___ --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
[asterisk-users] queue priority causes crash
I am running 1.2.11, I have had the queue priority lock the system up twice in the last week. I will forgo how I know this is what caused it, just trust me. I used the weight function in queues.conf to add priority to the queue. Is there another way to do it that will make it more stable? ___ --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
[asterisk-users] queue priority
What is the most stable version supporting queue priority. I have had many crashes, I am using 1.2.11 and have set the weight in queues.conf. is there a better way or a patch. I can't seem to find much. Any suggestions? ___ --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
[asterisk-users] wireless desktop phones
I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? ___ --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
[asterisk-users] wireless desktop phones
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone. But of course they want IP. Are there any adpaters
that will give me just enough bandwidth to get it done. The computer
network is all wireless so the phones would have all the bandwidth.
___
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[asterisk-users] Sendmail and exchange for voicemail integration
I am having real trouble getting Asterisk to send to exchange. They are on the same LAN. Does anyone know of a walkthrough for this setup. I have gotten it to work before, but that was to a hotmail account. Jordan Novak ___ --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
[asterisk-users] Manager connection problems
I am wondering how many and how often manager connections can be setup and torn down reasonably. here is the scenerio... I have 10 to 20 agents on two queues one with priority over the other I changed this the day before I also implemented a php program that runs every 8 seconds on an automatic refresh It establishes a connection to asterisk and runs a mysql query to update the database It runs two commands, Agents and QueueStatus During the weekend the system dropped all the calls and logged out all the agents and came back on its own. I am running asterisk without safe or any other scripts I run a Te410p with forthy eight channels to a pbx I can't find one error in any logs, I am unsure where to look in asterisk though. Basically I have a logon and off twice every 10 seconds, is this the problem! I implemented the priorities within queues.conf, does that cause any problems. I would appreciate any feedback!!! Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. 1319 Saint Andrews Street La Crosse,WI 54603 (608) 783-7560 x1299 1-800-666-2833 x1299 ___ --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
[asterisk-users] manager command queue...
...I am having trouble deciphering the returned status line, it seems to return 1-5 as far as I can tell. i am only aware of the status codes produced by ExtensionState, which does not return a 5. I cannot figure out why the codes are diffferent. Can anyone help? Or map the codes for me, i have googled my eyeballs out of the sockets. ___ --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
[asterisk-users] Long call setup times on SIP to zaptel calls
I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak ___ --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
[asterisk-users] Polycom phone locks up, send sip busy messages
I have a soundpoint 501 phone that has locked up twice now. You can make a call but when a call is sent to it, it responds with sip busy messages. You get the same message when the phone is in do not disturb. I reset to defaults the first time and it worked for a week or so and then stopped. The incoming calls are ringing three phones ( dial(sip/1sip/2sip/3 ), often two of them are in do not disturb. I read that I wasn't supposed to register these phone and have them set as static hosts. The interesting thing is that the phone displays missed calls every time asterisk tries to send it a call. So instead of ringing you see the counter fly off the chart. Can anyone give me some insite. Jordan Novak ___ --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
[asterisk-users] Agentcallbacklogin deprecation
I agree with these fella's, this is a piss poor way of fixing it. I only know of one call center that used static agents, mostly because they were sold a peice of crap and they had no idea how to use it the other way. I think you will find the majority of call centers are callback centers. This decision has taken Asterisk out of the realm of providing reasonable call center solutions. VIVA ZAPATA!!! Let's start a revolution! Message: 24 Date: Wed, 20 Dec 2006 22:22:26 +0100 From: Lenz [EMAIL PROTECTED] Subject: Re: [asterisk-users] AgentCallbackLogin() deprecated in 1.4 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; delsp=yes; charset=iso-8859-15 I have been speaking privately to a number of CC integrators and resellers about the AgentCallbackLogin() deprecation issue, and I'd dare say nobody is enthusiastic about it. With all its problems, AgentCallBackLogin is the workhorse of most of today's Asterisk CCs, and my impression is that the proposed solution meets a very lukewarm reception at the moment. Just my euro 0.02 l. On Wed, 20 Dec 2006 17:26:51 +0100, Douglas Garstang [EMAIL PROTECTED] wrote: The other thing is, that show agents doesn't show me which agents are logged in and if I use show queue I can see local channels attached to a queue but no agents. For my point of view there is some functionality lost with the new concept. Funny. I said the same thing in this list about 2 months ago and I got told I was nuts. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it http://queuemetrics.loway.it/ Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. ___ --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
[asterisk-users] Ssh access over a zap channel...
My need to do this through asterisk is simply the ability to provide me access with no additional cost to my customer. It seems like a nice thing to include as long as authentication is done well. I have worked on a dozen or more types of switches and all of them have supported this or had the capabilty through hardware or licensing. I am trying to get around opening and closing the firewall, which at some locations is simply not accessible to me. Jordan Novak ___ --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
[asterisk-users] ssh access using zaptel channel to dial in.
Has anyone done this, or have a thought on how to do it. I forsee it working like this... Dial in to a main greeting, dial an extension using a modem string like 782-,,,##409*. The extension would some kind of modem emulator. I know this compromises security. I was hoping to use an authenticate app in there as well. My main question is using the zap hardware and some kind of dialplan app to accomplish this Jordan Novak ___ --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
[asterisk-users] Desktop application for zap/agent call control
I am looking for a desktop control panel for zap (agent proxies). Does any one know of an application that is similiar to a softphone but controls zap/agents interfaces. I am looking for phone book, transfer, and possibly presence control. And it must be standalone, unlike HUD pro and hudLite. Jordan Novak Senior Telecommunications Engineer ___ --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
[asterisk-users] I am unable to find any included rpms with hudlite...
I am installling on a scratch asterisk running white box linux (fedora) Does anyone know where to find them after the rpm runs. I am looking for ircd and the perl dependancies. The instructions make a ton of assumptions, so I am not sure what is happening here. Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. ___ --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
[asterisk-users] web interface to control zap interface
I am looking for a web interface to control my zap agents. Allowing them to do conferences and transfers. I am familiar with flash operator panel but am unsure of how I would set it up to allow the agent, caller, to dial another number and have a three way conference. I have setup features.conf to do a attended transfer but can't figure out how to make it three-way. Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. ___ --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
[asterisk-users] FXO lines taking several rings to answer, always two
They are in Kewl start now but I have tried groundstart and loopstart. Waht could i be missing that would cause this.I start with a Exten= s,1,answer. I am using three FXS modules on a tp400.___ --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
[asterisk-users] simple dialplan trick I can't figure out (smdi, mwi substitute)
This is what I would like to do... exten =7299,1,Voicemail(u8896) exten =7299,2,Dial(zap/g1/#641299) exten =7299,3,Hangup WhatI would expect to happen is... Incoming call is answered by voicemail... Voicemail app finishes and the next priority starts. This is where the problem lies... After the intial call is ended the process is ended and the voicemail notification string "#641299" is never connected. Can anyone let me know if there is a way to do this. Or even better, a way to poll voicemail to see when a message is recorded, say every sixty seconds or so. ___ --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
[asterisk-users] call center status viewer
Can anyone point me in the direction of a good status viewer for agents. I have looked at the voip-info wiki and saw some good commercial ones. I just need opinions on any products. I am currently using FOP. I am looking for login/out, ringing, hangup and the like. I do strictly monitor agent proxies and not actual devices.___ --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
[asterisk-users] sip agent stuck in queue even after restarts
This is the first time I have used a sip device on a queue. Iogged in under the extension and now I can't logout. No kidding. I have restarted asterisk with persistant agents=off and also done a show agents. It shows no agents logged on and I am still receiving calls. To complicate things I am using Flash operator panel to see logged in agents and it has, at best, been sporadic. I have had no problems for the last six months and now I am in a hole. I beleive that the agent is stuck in the astdb somehow but every attempt I have made to remove it fails. Can any one see what I might be missing? winmail.dat___ --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
[asterisk-users] IVR menu system external database information collection
I would like to ask my callers several questions storing the values in a MSQL or MySQL database. My thought is to provide data to the agents application, like a pop-up. So I am looking to have asterisk collect values but instead of storing them in a berkley database, they can be more accesible via odbc and possibly .Net This makes for a hard Google ;-) I would rather not delve into AGI if possible, I'm a idiot! winmail.dat___ --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
[asterisk-users] voicemail maintenance questions
How is the best way to add,clear mailboxes and change passwords for voicemail. I am guessing you need to remove the conf entries for the mailbox restart asterisk and then add them back in and restart asterisk. Is there a better way?___ --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
[asterisk-users] Voicemail maintenance
Has anyone created a GUI for this. I would like to implement a server specifically for Voicemail using out of band signalling tied to a PBX. I fear the management will be exhaustive though. ___ --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
[asterisk-users] (no subject)
I have two questions. First I am running a t400p with three fxo ports signalling fxs (inbound CO lines). I have six polycom 501's. The problem is the amount of time the call setup takes. I have done this with Mitel phones before with a t-1 and had the same problem. My customer always complains about the call setup time. Am I doing something wrong or is this how it is. It takes up to five seconds to pickup or start ringing the CO. I would be happy to supply fake ringback if anyone knows how to do that. Second Problem is SIP Polycom phone line programming, I have read many contradicting things. How should it be provisioned to allow multiple incoming calls. How many lines,calls per line and the rest of the bull, Iknow loaded question. I am using kewl start on those three lines by the way.___ --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
[asterisk-users] TDM2400P
This is definately a t-1 slipping, check your timing on the spans, also ensure that you have replaced the t-1 cables. I see this most of the time caused by cables that were made by hand. Usually because a solid conductor was used with stranded wire or solid cat5 with stranded rj-45's. If possible use a premade patch cord made of stranded wire. The interupt would be my second step, although this usually makes the t completley unusable, especially using all four ports. Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. ___ --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
[asterisk-users] multiple zaptel cards
I am in need of an additional x100p in one of my servers. It already has a fully loaded tdm400p in it. I can't figure out how to define the other one in zaptel.conf. Which one do I define first,I am guessing it is dicated by the order the drivers are loaded. I am Wiki'ed out!___ --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
[asterisk-users] polycom 501 digitmap
This is really starting to get to me. I have deleted this field in the phones per the wiki. I am trying to get the phones to dial on there own. Is there anyway to get the phone to dial 1-8 after three digits are received and 9 after seven to ten digits. I am willing to wait for a timeout but that doesn't seem to work. Any help is greatly appreciated. Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. ___ --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
[asterisk-users] reloading agents and queues
Is there a manger command that will reload these two configs, something like extensions reload, so it doesn't drop calls in progress. Jordan Novak Senior TelecommunicationsEngineer Logistics Health Inc. ___ --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
[asterisk-users] Flash operator panel
Does anyone know how to switch out the background image? I cannot find it defined anywhere.___ --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
[asterisk-users] Flash operator panel
Can anybody steer me in the right direction? I have installed the fop and have it working okay, first problem is agent logins not changing the state color when an agent logs in. I configured it on two boxes one works the other doesn't, same configs alll the way. The other is more of me not understanding how it works. I only see the buttons that i have programmed and am unable to get the password entry box and can't figure out how to do transfers. Jordan Novak Communications Technician ___ --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
[asterisk-users] ACD Queues Agents logout
Here is what I do... Exten=777,1,AgentCallbackLogin() Yup, thats it! use your agent id and password, and then enter your dialable number. I say dialable number because you can basically dial any phone number. We have agents that call a toll free number and login to their home phones, pretty sweet! This has to be in the right context to allow this though. Jordan Novak Communications Technician ___ --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
[asterisk-users] agentcallbacklogin (logging out of)
You need to create another callbacklogin extension using # as the extension to log into. Logging into extension # logs the agent id out. Jordan Novak Communications Technician ___ --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
[Asterisk-Users] Queue - Log if caller disconnects
I am having the same problem with my IAX clients. I posted some issues that are causing my remote IAX agents to be disconnected due to errors in setting up the IAX stream. I have found that calls will abandon when a dynamic agent is logged into a down phone, the agent obviously cant logout if they cant call the switch back. The caller seems to be disconnected when being transferred to an agent that is logged into a down phone. I am using least recent routing. I had thought that asterisk at very worst would try to transfer to the agent, see the phone down, timeout on rings or not ring at all, and then log the agent out. I am definitely missing something or mis-reading my instructions. Please post your resolution and I will do the same. Jordan Novak Communications Technician ___ --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
[Asterisk-Users] IAX2 Destroying channel to avoid deadlock
I am receiving this message intermittently. It is happening during call setup. My phone is registering correctly. I am also having this problem between Asterisks.Any ideas where this comes from? Jun 29 01:05:04 NOTICE[13192]: chan_iax2.c:1601 iax2_destroy: Avoiding IAX destroy deadlock Jordan Novak Communications Technician ___ --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
[Asterisk-Users] Really need some help on IAX2 destroy to prevent deadlock
Every third or fourth call causes down my iax trunk, or to an IAX phone, causes asterisk to display this message on the console Destroying channel to prevent deadlock. Jordan Novak Communications Technician ___ --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
[Asterisk-Users] Trixbox maunual configuration
I love the added apps installed with trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on the other hand, is nearly impossible to do everything with. Trying to edit the configs manually proves impossible due to the excessive use of includes and macros. It is kind of like watching someone try to bite their own ear off. Has anybody tried to wipe all the configs clean and program the switch manually. Will this interfere with the other apps. I would wipe out extensions.conf, voicemail.conf, IAX.conf SIP.conf queues.conf and agents.conf. I do not want to use the FreePBX again after this. I am not trying to put down FreePBX, I know a lot of people have worked very hard on this. It just over complicates things for me. Jordan Novak Communications Technician ___ --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
[Asterisk-Users] Where has the outbound call directory gone
I have installed 1.2.9.1 and it has no /var/spool/asterisk/outgoing directory. I must have missed some change in this addition when upgrading. Does anyone know where the automatic outgoing call directory has gone? Jordan Novak Communications Technician ___ --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
[Asterisk-Users] music on hold Madplay and Files not working
I am unable to get MOH working, is there a requirement to have a sound device in the machine, none of my servers have soundcards. Jordan Novak Communications Technician ___ --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
[Asterisk-Users] (no subject)
I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal so they are terminating on Toshiba digital phones. Loud crackling even happens from time to time when a Mitel SIP phone is connected to Asterisk B at that location over thye LAN with no layer three routing, but it is consistent on the IAX trunk. There is a lot of Data traffic, but thus should work regardless, I dont think the ping times are the issue. Jordan Novak Communications Technician ___ --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
[Asterisk-Users] IAX crackilng
I apologize about doubling these up, I forgot the subject! I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal so they are terminating on Toshiba digital phones. Loud crackling even happens from time to time when a Mitel SIP phone is connected to Asterisk B at that location over thye LAN with no layer three routing, but it is consistent on the IAX trunk. There is a lot of Data traffic, but thus should work regardless, I dont think the ping times are the issue. Jordan Novak Communications Technician ___ --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
[Asterisk-Users] crackling on IAX between asterisks
I have two IAX trunked *, there are loud crackles and pops, they are dialing out a T-1 and are sip devices, it also drops words, any help or Ideas?___ --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
[Asterisk-Users] Voicemail indication on Mitel 52xx phones
I am using Mitel 52xx dual mode phones in SIP mode. They work excellent, I am however having a problem with Voicemail retrieval. The Mitel Phones have a voicemail button on them. The light lites and clears correctly but I am not able to retrieve the voicemails using this button. In the phones web GUI it has a field for a voicemail server and port under additional servers in the networking tab. Does anyone have a expertise to offer with these phones? Or does anyone have an idea to help me make voicemail pickup or work off an additional port, If thats possible. I know this is far fetched, any other phone system and I wouldnt have asked, but it seems I am never told no with asteriskJ Jordan Novak Communications Technician Logistics Health Inc. ___ --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
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 21, Issue 130
Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, April 22, 2006 8:26 PM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 21, Issue 130 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Sipura SP3000 question (Roshan Sembacuttiaratchy) 2. Re: Sipura SP3000 question (Gonzalo Servat) 3. Re: PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3 ([EMAIL PROTECTED]) 4. Re: Sipura SP3000 question (Rich Adamson) 5. How can I get a recording from a CD to my asteriskdigital assistant (Davi-Ann) 6. Asterisk on FreeBSD + Passive ISDN BRI (Cian Hughes) 7. Re: How can I get a recording from a CD to my asterisk digital assistant (Alberto Sagredo) 8. Re: PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3 (John Novack) 9. Re: How can I get a recording from a CD tomyasterisk digital assistant (Davi-Ann) 10. Re: RE: SPA 3000 - UK Replacement (Wayne) 11. Re: Sipura SP3000 question (Wayne) 12. RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ? (Steven Totaro) 13. RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ? (Steven Totaro) 14. RE: Don't see my post ([EMAIL PROTECTED]) 15. RE: How to restrict simultaneous phone registrations ([EMAIL PROTECTED]) 16. RE: No DTMF ([EMAIL PROTECTED]) 17. RE: Don't see my post (Steven Totaro) -- Message: 1 Date: Sat, 22 Apr 2006 19:17:52 + From: Roshan Sembacuttiaratchy [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura SP3000 question To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Sat, Apr 22, 2006 at 11:19:35PM +1000, RumaTech scribbled: As this part is still in testing, I want all the outgoing calls got to PSTN by default and dial, say 0, to get an outside VoIP line. I would like to do it as part of SP3000 configuration, not as part of * dialplan. Can someone help me? I use the following dialplan within the Sipura: ([2-79]11:@gw0|999:@gw0|112:@gw0|0[12]x.|[*x]xx.:@gw0|#9,:[*x]x.|**) Using this, all emergency numbers go directly to PSTN, all numbers starting with 01 and 02 go via VoIP, and all other numbers go through PSTN. Any number prefixed with #9 is then forced to go through VoIP, with the initial #9 not being passed to Asterisk. Adapt and use. :-) Hope this helps, Roshan -- http://roshan.info Be different, act normal. -- Message: 2 Date: Sun, 23 Apr 2006 05:34:15 +1000 From: Gonzalo Servat [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura SP3000 question To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 On 4/23/06, Roshan Sembacuttiaratchy [EMAIL PROTECTED] wrote: I use the following dialplan within the Sipura: ([2-79]11:@gw0|999:@gw0|112:@gw0|0[12]x.|[*x]xx.:@gw0|#9,:[*x]x.|**) [..snip..] Is this @stuff something new in the SPA3000 dialplan syntax? I have SPA-200x ATAs and I never saw any mention of this in the manual, which makes sense if it's a SPA3k new dialplan feature. Cheers, Gonzalo. -- Message: 3 Date: Sat, 22 Apr 2006 19:42:28 + From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Thanks for the response, I'll ask the client to change batteries, though it is a new phone less than two weeks. is there any reason why the Lanline(Verizon) work and not the Asterisk? The only differences is the Asterisk, Linksys router and the DSL modem. One of these 3 should be interfering. -- Original message -- From: John Novack [EMAIL
[Asterisk-Users] editing the asterisk -addons makefile
Can someone point me to some documentation on how to add app_CBMysql.c to my makefile. I also am a little unsure of the directions on how to compile it. Here is what I am working with . Download and compile app_cbmysql in /usr/src/asterisk/apps or wherever you have the Asterisk source. Run as root: cd /usr/src/asterisk/apps wget http://www.fitawi.com/Asterisk/app_cbmysql.c (site is currently down) Edit the Makefile in that folder using the patch www.fitawi.com/Asterisk/Makefile-cbmysql-patch.txt (site is currently down) Compile Asterisk : run make install in the Asterisk source directory (not the subdirectory apps). In this way you will compile only app_cbmysql.c and not all the other parts of Asterisk. Fitawi.com is out of commission for now. Jordan Novak Communications Technician ___ --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
[Asterisk-Users] WebMeetme defines.php?
I am looking at some directions on how to install and it is asking me to edit defines.php, it states that the file should be located in the source directory, but I cant seem to find it anywhere on my machine. Anyone been thru this? Jordan Novak Communications Technician ___ --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
[Asterisk-Users] WebMeetme Problem Please help!!!
Title: WebMeetme Problem Please help!!! I am running Feodra, I have downloaded the WebMeetMe Program, untar it to /var/www/html/WebMeetMe. I can access teh web page as of now. I cannot for the life of me figure out where defines.conf is. The install tells me it is in /var/www/html/WebMeetMe/lib/ however a complete search of the computer cannot find it anywhere. The /lib/ subdirectory does not exist in the untar'ed folder either. I could understand creating it under the /lib/ directory but I can't see a reason why it wouldn,t be there already. Here is what I have done... Download to /home/ directory extract to /var/www/html/ try to edit defines.php no directory or file Am I missing something crucial here? This is the directory of /var/www/html/WebMeetMe about.php conf_control.php css index.php phpagi_2_14 call_operator.php counter.txt images info.txt I checked the folders under this directory. All I can figure is that I am downloading 1.2 and the instructions are for 1.3. Which apparently is a bad link. ___ --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
[Asterisk-Users] web meetme
Can someone point me to instructions on how to install, I have edited the defines.conf and set up the database. I have apache running and have no clue what to do now. I have NO experience with php based stuff. HELP!!! Jordan Novak Communications Technician ___ --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
[Asterisk-Users] Meetme admin
I have found meetmeadmin, sounds good and all but does anyone have a code snippet or idea on how to do this. I want to allow one person to be an admin, mute, kick, join other users. How do I differentiate the admin from the rest and then allow him to kick join and so forth without interrupting the conference or have the others hear digits being dialed. Jordan Novak Communications Technician ___ --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
[Asterisk-Users] asterisk-stat and webmeetme by areski
Title: asterisk-stat and webmeetme by areski I like to think I am not a complete idiot... ...I have googled till my fingers bled. I cannot figure out how to install these apps. I have figured out the database protion as well as editing defines.php but the web portion is killing me. I am running apache and have done no configuration to that. Its a fedora core 3 box with the latest, within the last week, asterisk and zaptel packages. I can't even figure out how to bring up the web page, I know its pretty sad! Can someone with experience guide my hand it would be much appreciated. ___ --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
[Asterisk-Users] voicemail to email sending problems
Title: voicemail to email sending problems I have a box that will send to my personal pop/web based email but will not send to my exchange server. I have checked the MX record and DNS settings. I know there is something you can do like this to check it but it returns either a -1 or 0 (have no idea what that means) sendmail /mx anyway I can send to a ISP based Mail account outside the company. We have .wav files allowed we also require smtp authentication. We do have an IP that is allowed to accept non authenticated mail from our databases, but I am not sure how to use this address with sendamil instaed of it using the MX record. which is mail.timbucktoo.com instaed of the allowed ip to bounce off of. Should I be formatting the address in voicemail.conf to the allowed IP? so its formatted [EMAIL PROTECTED] Any thoughts. ___ --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
[Asterisk-Users] Callid on T-1 trunk
I am not getting any caller Id with my standard T-1. Is a standard T capable of sending callerid? I dont want to spend time troubleshooting my PBX if Asterisk cant send it down that type of trunk. Jordan ___ --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
[Asterisk-Users] RXgain
I have really cranked up the rxgain on a t-1 trunk in Zapata.conf. It seems to have no effect although I raised it to 7 from zero. I am using a te110p. Any thoughts on why? I have not unloaded he modules and reloaded them as it is during the day. Does this even need to be done to take effect; I did restart the asterisk service. Jordan Novak Communications Technician Logistics Health Inc. ___ --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
[Asterisk-Users] Mandrake zaptel module not found after compiling
I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so I had to modify the 01-devfs.rules Make linux26 Make Make install Everything appears to compile correctly but it says module not found when doing modprobe zaptel Is this a rights issue? Jordan Novak ___ --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
[Asterisk-Users] Mandrake zaptel module not found after compiling
I installed as su, and tried to compile using only make. No problems were reported during compiling but problem persists. Any other ideas? Jordan Novak ___ --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
[Asterisk-Users] Web-ex type solution for use with asterisk
Is there an app or softphone for meetings that displays the hosts screen like webex or intercall. Jordan Novak ___ --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
[Asterisk-Users] hunt groups
What I would like to do is exten = 1000,1,Dial(sip/1000)(zap/g1,97837560) exten= 1000,2,Voicemail(u1000) Basically a follow me app that rings numerous interfaces and allows me to answer or it to time out and go to vmail. I didnt include the time out here as I am hoping someone can tell me where that needs to be. I really dont want to make the caller ring one interface and then the other. Ideally I would be able to press pound after answering so that it didnt continue to ring the other interface. Most of the apps that I saw do this are basically the same as forwarding the extension, any system can do that and I know asterisk is better than that. Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 ___ --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
[Asterisk-Users] choppy recorded sounds in asterisk
I have installed asterisk on numerous servers. Every install was done on Fedora and (White box Linux). I now have zap channels in one of the boxes (T-1). No matter what type of channel I call on (sip or zap) I get some really strange artifacts in the sound, almost like a skip in the playback. It seems to always be in about the same place in the recording. Usually in the beginning of playback. For instance somewhere in the Comedian mail part part of the voicemail greeting it will hick-up, it also happens to Meetme and some other agi/php stuff I am using. Someone please help me make Allison sound as sexy as she is supposed to sound! Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 ___ --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
[Asterisk-Users] hotel vmail and iax trouble
I have two issues... First I am working with a hotel software vendor to include an automated way to turn vmail on and off while clearing it at the same time. The vendor is looking to interface via serial cable as they currently do with Mitel systems. i am willling to work with them on an IP interface but I am not so sure on how to implement it in asterisk. Does anyone know of a way that may be prebuilt, AGI or the like? Basically it works like this, When they check out a customer they want to clear the Vmail and forward it to the front desk, when they check in they want to turn the vmail on and have it completly reset password and all. I do not know an easy way to do this in asterisk due to the fact you will be chasing down several files every time. Second I have another site using teliax, I've set this up before but something is strange on this box. I am getting a message stating that IAX2 channel could not be created and is not implemented error number 66. No clue myself. I wou;d post the configs but I am fairly sure that they are good unless someone can tell me that error is generated because of a config problem. LOL winmail.dat___ --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
[Asterisk-Users] webvmail problems
I have done my make webvmail, what else do I need to do? How do you get to the site? Any help would be appreciated. Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 1-800-666-2833 x299 (608) 783-7560 x299 ___ --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
[Asterisk-Users] webvmail
My question is about webvmail, not nwebvmail. I have never used AMP (seems like cheating). My question is in regards to plain jane Asterisk install. Just like making samples after you compile asterisk you are able to make webvmail. Basically it is a interface into the voicemail system fro the web. I have apache installed on Fedora and am able to bring up the localhost test page. When I try to open vmail.cgi from the browser nothing happens. As I stated earlier I dont know whether this is even what I am looking for. I believe the app compiled correctly as I got no errors. Can anyone point me in the right direction? Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 ___ --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
[Asterisk-Users] web meetme instructions
This has to be the worst documentation I have ever come acrossed. I have found two or three docs on how to install it, but they are all so different and make huge assumption about what packages you have installed and locations of files. Has anyone seen something better, I want to get this working it is quite a cool app. Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 ___ --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
[Asterisk-Users] wake up calls
Does anyone have a way to do wake calls? Jordan Novak Communications Technician Logistics Health Inc. ___ --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
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 20, Issue 13
On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote: Does anyone have a way to do wake calls? Jordan Novak Communications Technician Logistics Health Inc. You could use cron and /var/spool/asterisk/outgoing scripts to dial numbers, etc... Can you elaborate, I am fairly new to Linux and a phone guy to boot. I am looking for a way for the users to set a wake up call for themselves from the phone... Something like... Dial an extension for wakeups The caller is asked to set a time and the number of days for which they want it set. The system then calls at those times, and every ten minutes until it is answered. -- Message: 3 Date: Thu, 2 Mar 2006 13:10:53 -0500 From: Wojciech Tryc [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=iso-8859-1; reply-type=response Your pc has to able to support tagged vlans. The switch on the phone will pass through both tagged and untagged vlans. W - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 02, 2006 11:51 AM Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) And about the 802.1x ? The phones can work as passthrough and force the PC to use 802.1x ? What configuration do we put in the switches? Do we put the switch as access (with 802.1x) or trunk (without 802.1x) ? Thanks Joao Pereira Greg Oliver wrote: It actually depends on the switch model. Some put the port into trunking mode automatically with the sw voi command, and some do not. Hopefully one day Cisco will finally make their own products and become uniform instead of buying several companies and glue'ing them all together to get an ethernet switch that works. At least they got the routers right :) On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote: You don't need switchport mode trunk when using switchport voice vlan.. On 3/1/06, Nicholas Kathmann [EMAIL PROTECTED] wrote: Joao Pereira wrote: Hello to all I would like to know If some of you have already configured an Cisco IP Phone (7940 or 7960) to work in a different VLAN than the PC that is connected through the phone switch? I know that this can be done with the Skinny firmware, but I dont if it works with the SIP firmware. The Cisco technical staff told me that these phones dont support 802.1x but can work as pass-through. This way I can still use the PCs with 802.1x and the phones in the same Ethernet plug. Did someone made it with the Cisco IP phones? What configuration do I need in the phones and in the switch? Thanks Joao Pereira If configuring with Cisco switches, I'm pretty sure they pull the information for which VLAN to operate in from the switch. You have to configure the switchports on the Cisco switch like so: interface fastethernet 0/1 switchport trunk native vlan your data vlan switchport mode trunk switchport voice vlan your voice vlan spanning-tree portfast trunk etc. Thanks, Nicholas Kathmann, CISSP Kathmann Consulting, LLC ___ --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 ___ --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 -- Message: 4 Date: Thu, 02 Mar 2006 18:15:28 + From: Joao Pereira [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) To: Wojciech Tryc [EMAIL PROTECTED], asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1
[Asterisk-Users] monitor outgoing calls in queue / campaings
I use Speed dial keys for setting account codes for the different queues. When an agent dials out I have them hit the speed dial key and the call is registered to that account code. Pretty typical of any call center outbound calling. ___ --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
[Asterisk-Users] Conference bridge dimensioning
We are using an Asterisk box to do conferencing right now. I have had about sixteen active lines in conference and the quality was acceptable. We now have a need for 50 people to conference at one time. Does anyone have enough experience doing this to give me some pointers. Will it even be reasonable to try this? Is the mixing done on the the hardware, I plan on using a quad span t-1 card from Digium? The server is a fedora box with a dual core xeon at 2.0 Ghz and 2 gigs of Ram. Is there a rule of thumb to go by as far as conferencing resources? Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 1-800-666-2833 x299 (608) 783-7560 x299 ___ --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
[Asterisk-Users] Sendmail with exchange
I am using Asterisk to send Voicemail out as Email. I am running into a problem I believe to be caused by the exchange server requiring SMTP authentication. I cannot get the sys admin's to turn it off. Does anyone know enough about sendmail to help me. I am assuming that the default mail client is sendmail. It will also send to other non-SMTP authenticated servers. Your help is much appreciated. Jordan Novak ___ --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
[Asterisk-Users] Virtual Extensions
I have run into two programs offering Virtual users. This allows a person to enter a code and take over any extension, another code is used to release the feature. The two programs are Ipmanager and Scopserv. I hate using GUI's as I have not seen a truly good one. I would like to implement this feature without the GUI. Any Ideas! From what I gather they are using Macro's, of which I have no knowledge. Any help would be appreciated. ___ --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
[Asterisk-Users] PRI QSIG and legacy toshiba intergration
I love it... I buy a half a million dollars worth of Trashiba's finest ... I download Asterisk for free... I now refer to it as legacy 18 months and 300 extensions later! Anyway, I am trying to integrate my dial plans acrossed platforms. PSTNCTX670Asterisk The dialplan I would like to setup, 1xx,2xx,3xx,7xxx to the CTX 670 4xx,6xxx toa remote ctx100 (this is setup using QSIG ISDN on a PRI tie line) now I would like to have 8xxx going to asterisk All of my incoming calls would be handled by the ctx670, mostly on DNIS equipted lines. If the user dials a four digit extension starting with "8" on the CTX 670, how do I transfer that digit string to asterisk? Going backwards, the asterisk user dialing a CTX extension, I plan to handle this with DID digits sent back to the CTX, when I see the Incoming DID digits I can route the call based on DID I receive from asterisk on that particular line group. It will sort work like this... Ext.123 is dialed on asterisk Asterisk picks up a zap channel sends DTMF 123 after the CTX picks up CTX looks at DID and sends it to the destination 123 I think that I need a context with all of my CTX extensions in it! Ultimately I am trying to find out if Qsig can somehow help me, I am more of a phone guy, I need an asterisk guy to exchange help with.. This e-mail message and any attachments may contain information that is privileged and confidential. The information contained in this e-mail is intended only for the use of the addressee; access by anyone else is unauthorized. If this message has been sent to you in error, do not review, disseminate, distribute or copy it. Please immediately reply to the sender by e-mail or by telephone at608-783-7560 X299,then delete the message and any attachments from your system. Thank you for your cooperation. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
