Re: [asterisk-users] Who has the best call recording solution!
Hi Mark, I mentioned this before in a previous post. I created a system using php/mssql (which is the database we use at the office, but clearly could be done with mysql) that records all of the calls in our queues. Works like this: Call comes in and before the queue command, I call MixMonitor to set up the recording (use the bridge option too so you don't waste space by recording the hold music if you have any), and save it using the unique ID, using the gsm format to a general folder. From there, I wrote a php script using deadagi to move it to a directory of the extension that answered the queue call (which you can get via the CDR variables and any others that you manually set) and also updates the database (also renames the file to a better convention). The web script the users access can then either playback their recordings, which generates a call script to dial their extension and listen to the call via the phone, or they can download it. If they download it, it uses sox to convert it to a wav file before sending you to the link to download it. Also for the managers, they can listen to any calls by some filters on the query to the DB. Nice thing, is under the gsm format, we save our recordings for a year (which another script manages those files). While our office is a small call center (about 500 calls a day) currently we have about 63,000 recordings on our server and it is only taking up about 38 gigs of space (on the same server as Asterisk). Most of our calls are about 15-20 minutes long. I know my solution is sort of clunky/buggy (at least in terms of adding on/making changes. It was sort of a prototype that was just pushed into production before I could finalize it) and probably wouldn't be ideal for a large call center, but I wrote it in about a week, maybe two. But clearly if you cannot find a solution that works for your office from something that has already been made, you can build your own pretty easily. I may someday sit down and actually go back and re-write it to put out on the net anyone to use...but we shall see. Kevin Mark Hamilton wrote: Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. When I say best, I mean how they record, convert it to some low-diskspace-consuming format, and then leave it there, until a web-app requests it, and then it’s changed to wav or mp3 and then lets it download, etc. Either that or someone records, then pushes off the recordings to a ‘recordings server’, then when someone requests to listen to it on the box that was recorded, it pulls the relevant recording from the ‘server’, converts it and allows it for download? Something like that.. you get the drift. Basically, I’m looking to record different queues that are hosted. But do not want to compromise too much diskspace, yet want to make it available for download through some web-app for listening (wav or mp3). Thanks, Mark. ___ -- 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 -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- 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] Reg call recording
What I have done for our office is actually built my own interface with php and used our SQL database to store the information. Basically I keep all the recordings in gsm format, and store them however I want. I use MixMonitor and use DeadAGI to run a script to rename the file and move it to the directory for that extension. So in your exmaple, .../recordings/123/[file name] I also used session information from the login page to store the person's extension (which we also have in the DB, but there are other ways to do this) that is looking at the interface so when play want to listen to the call, it will generate a call file and dial their phone and playback the file (works nice if you don't have speakers/headphones). Or they can download it. Downloading it will run a script to convert it to wav. I don't know of a best way to do this. I know if you take the time and put the effort, you can get what you/your company wants if you build your own. Or go with some of the other suggestions made which also work perfectly well. I think for me it took about 2 weeks to fully build/test everything and I was coding it by myself (on top of other responsibilities at work). Kevin Bikrish Amatya wrote: Hi all I am using asterisk as pbx for my company. My company has requirement that all the incoming and outgoing calls should be recorded for all the extensions and should be able to play recorded call on extensions basis, that is , say 123 extension has made what call on the particular date and should be able to play and listen to it. What is the better way to achieve this? Any kind of suggestion is truly appreciated. Bikrish ___ -- 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 -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- 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] Trouble with Polycom phones
No, even with the numerical IP addresses they still had the problem. Kevin Mike wrote: I`m curious: did going with numerical IP addresses fix your problem? Mick -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Smith Sent: Wednesday, June 04, 2008 13:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with Polycom phones Yes, I was using a name instead of an IP address. And if memory servesI *think* it is using TCPprefered...but I could be wrong. Kevin Mike wrote: I have been running into a few issues with Asterisk/polycom and I am running out of ideas. This problem has been ongoing for the last couple of weeks. I will try to be as detailed as I can, but I might leave out a few details. Any suggestions would be greatly appreciated. Now, the phones lose their registration with Asterisk. Are you using a numeric IP address or a name for the Asterisk server in the Polycom config? I had the same issue (only from 2.2 up IIRC) until I put in the numerical IP. Can't explain it, maybe somebody else can. Mick ___ -- 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 ___ -- 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 -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- 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] MiixMonitor filename for queue calls.
Hi Ed,
Glad to see you figured out your problem. I'm not sure what the
differences are between your config and mine, but maybe this will help
others too.
I add and remove my agents from the queue. So my agents.conf file is
just the presistentagens=yes. Also I just run the command in the dial
plan like below which saved mine items just fine. No configurations in
the queue.conf file for the monitor type.
exten = 852,n,MixMonitor(/mercury/recordings/holding/${UNIQUEID}.gsm|b|)
From there, in the hangup extension, I run a php script to take the CDR
record and the file (rename it of course to
queue-extension-callerid-callid-timestamp.gsm), and place it into the
agents folder and the database for our agents/supervisors to review or
download them.
Kevin
Ed Nunez wrote:
Can anyone give me input on the following issue?
I have a queue with MixMonitor enabled.
This is also enabled in agents.conf.
On my extensions.conf, I am setting the monitor filename as fillows,
although I see the filename as desired in the console as I make my
test call, the system is only using the default file name to save the
mixmonitor file (agented + uniqueID)
Agents.conf
[general]
persistentagents=yes
[agents]
maxlogintries=3
musiconhold = default
updatecdr=yes
recordagentcalls=yes
recordformat=wav49
urlprefix=http://pbx.netoneint.com/calls/
savecallsin=/var/calls
agent = 1000,1000,Ed Test1
agent = 1001,1001,Ed Test2
queues.conf
[noi-noc]
monitor-format = wav49
monitor-type = MixMonitor
member = Agent/1001
member = Agent/1000
extensions.conf
exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)
exten = 8484,1,answer
exten = 8484,2,Queue(noi-noc)
Console output
-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1,
MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun 6 15:06:38 2008) in
new stack
-- Executing [EMAIL PROTECTED]:2] Queue(Zap/1-1, noi-noc) in
new stack
-- Started music on hold, class 'default', on Zap/1-1
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing [EMAIL PROTECTED]:1]
Dial(Local/[EMAIL PROTECTED],2, SIP/1658) in new stack
-- Called 1658
-- SIP/1658-087e7610 is ringing
-- Agent/1001 is ringing
-- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2
-- Agent/1001 answered Zap/1-1
-- Stopped music on hold on Zap/1-1
[Jun 6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The
device state of this queue member, Agent/1001, is still 'Not in Use'
when it probably should not be! Please check UPGRADE.txt for correct
configuration settings.
== Begin MixMonitor Recording Zap/1-1
== Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
== Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
== End MixMonitor Recording Zap/1-1
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Re: [asterisk-users] Trouble with Polycom phones
Hi Mike, The odd part, is some of the phones now are not having this problem anymore. Mine phone for example, has been fine since last Saturday (which I had to move it so it of course rebooted ;) ). However, I did change this value today on another couple of phones with this problem still. So we shall see if this helps. Thanks, Kevin Mike wrote: I`m curious: did going with numerical IP addresses fix your problem? Mick -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Smith Sent: Wednesday, June 04, 2008 13:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with Polycom phones Yes, I was using a name instead of an IP address. And if memory servesI *think* it is using TCPprefered...but I could be wrong. Kevin Mike wrote: I have been running into a few issues with Asterisk/polycom and I am running out of ideas. This problem has been ongoing for the last couple of weeks. I will try to be as detailed as I can, but I might leave out a few details. Any suggestions would be greatly appreciated. Now, the phones lose their registration with Asterisk. Are you using a numeric IP address or a name for the Asterisk server in the Polycom config? I had the same issue (only from 2.2 up IIRC) until I put in the numerical IP. Can't explain it, maybe somebody else can. Mick ___ -- 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 ___ -- 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 -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- 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] SIP call recording
Hi everyone, Perhaps I am just mis-reading the documentation, but for call recording, is it possible to record the conversation over a SIP channel? We have call recording preformed on all of our ZAP connections, but I was wondering if it is possible to record (similar to MixMonitor) with a SIP connection. So far, every one I have tried (Record, Monitor, MixMonitor) does not seem to create the file. Asterisk version is 1.2. Thanks, Kevin -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- 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] Question on DeadAGI
I have always had problems getting the script to run during an active
channel through hang up with DeadAGI. I found it best just to use it on
the hang up extension like below: Maybe that is how it is supposed to be
run, but from what I have read and you have, I don't see any flaws.
exten = h,1,DeadAGI(get-usage.php)
Another thing I do is I put a simple verbose statement letting me know
that the script was called, or entered some part of execution.
Kevin
Nhadie Ramos wrote:
Hi,
How can i get the deadAGI to work at this scenario
Basically when someonc calls international, i will get the remaining
balance using AGI get-available.php.
but after the call i would like to get the usage by calling
get-usage.php so i can update users balance, but looking at the debug
it seems the AGI was not called. is there som
exten = _00.,1,AGI(get-available.php)
exten = _00.,n,GotoIf($[${CALLSTATUS} = 1]?70)
exten = _00.,n,GotoIf($[${CALLSTATUS} = 2]?80)
exten = _00.,70,Dial(SIP/[EMAIL PROTECTED])
exten = _00.,n,Hangup
exten = _00.,n,DEADAGI(get-usage.php)
exten = _00.,80,Busy
exten = _00.,n,Hangup
Regards,
Nhadie
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Re: [asterisk-users] Trouble with Polycom phones
Yes, I was using a name instead of an IP address. And if memory servesI *think* it is using TCPprefered...but I could be wrong. Kevin Mike wrote: I have been running into a few issues with Asterisk/polycom and I am running out of ideas. This problem has been ongoing for the last couple of weeks. I will try to be as detailed as I can, but I might leave out a few details. Any suggestions would be greatly appreciated. Now, the phones lose their registration with Asterisk. Are you using a numeric IP address or a name for the Asterisk server in the Polycom config? I had the same issue (only from 2.2 up IIRC) until I put in the numerical IP. Can't explain it, maybe somebody else can. Mick ___ -- 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] Trouble with Polycom phones
JR Richardson wrote: You mentioned this started happening 3 months ago, what happened then? Network changes, equipment changes, traffic increased, new users (downloading allot during the day, surfing porn), wireless interference? The initial problem started when our DS3 was throwing errors. Once that was resolved, it was fine until about a week later when the problems started again...but this time no errors from showing on the DS3. Otherwise, I will try some other suggestions the next time I am back in that office. Thanks again, Kevin ___ -- 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] Newbie Voicemail: Just use one [context] invoicemail.conf?!
Perhaps seeing some of your dial plan (such as the macro, etc) would help not only me, but also others, because maybe I am just not following you. Off the top of my head there are a few things you could do..but again, it depends on how your dialplan is set up and how you access the macro. One way to do this is in in the outbound calling context you could set a variable with the context of the voicemail depending on the extension, then pass that to the marco. Another way is you could set an if statement to get the extension and then go to the proper context (assuming your extensions have some meaning to them). I'm sure there are others methods to do this too. Kevin Lee, John (Sydney) wrote: I was thinking about dividing my users into different groups (contexts) in voicemail.conf so that I could use voicemail show users for [context] to manage them easier. However, I found out that I should not do that because if I am using [macro-stdexten] in extensions.conf, I will need to hardcode the [context] in Voicemail command within [macro-stdexten] and there is no way [macro-stdexten] can know which voicemail context a user is in anyway. As a result, I just go back to put all users in [default] in voicemail.conf. Am I missing anything? ___ -- 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] Asterisk first time user
I guess take this into consideration if time isn't a real factor (however, I'm sure it is). In my experience I found it best to start learning with the configuration files only then use the GUI. The GUI's are very nice and handy, but sometimes I feel they lack what you could do with manually creating your dial plan. Also you learn how to debug / troubleshoot problems by experimenting with it, I found that to be very helpful. Maybe build out your dial plan with the GUI so you can get your office up and running, then make a new context to experiment with doing things manually. Kevin Aaron Stranberg wrote: Thanks for the response, to clarify a bit, I don't mind the hands on installation but after the system is up and running I would like to have a GUI front end that I can dump off to less linux friendly folks for creation of new extensions, voicemail setup etc.. Thanks again for the response. -Aaron On Mon, May 19, 2008 at 3:52 PM, Matt Watson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On May 19, 2008 03:21:34 pm Aaron Stranberg wrote: Folks, We are a small office with remote users less than 20 total phone extensions, and I am looking for some guidance on choosing between asterisknow and a centos/ubuntu or any other os with an asterisk + asteriskgui build out? Looking to get up and going quick with some method of GUI administration that won't require a ton of ongoing linux admin level support. I hit a couple of stumbles going the asterisk + asterisk GUI route (404 errors on ivr page etc..) and am tempted to take the easy path of asterisknow iso and go. Thanks for any pointers, and advance apologies if this had been beat to death. -Aaron IMHO, there is really no way to say this one is best. Each solution might be better at X while the other is better at Y... its very dependent on your situation Though, I gather you'd rather not deal with the actual OS-level, so you are probably best to stick with one of the complete packages like AsteriskNOW, Trixbox (they have a free and paid version), PBX in a Flash, and i;m sure there are many others... I haven't used any of them however so I can't really speak about the pros and cons of them. -- Matt http://www.mattgwatson.ca ___ -- 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 ___ -- 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] Fedora 9 + Asterisk
I almost hate to admit this...but I'm still running Asterisk 1.2 on Fedora 4 :D However, I'm planning on upgrading to 1.4 but it has been working out just fine so far and I just can't find time to upgrade. Otherwise, at least with Fedora I have had no major issues running Asterisk. Most of any items I found for later releases, I was able to apply it in some form on the older release. One would assume that it would work just fine, but then again, if it isn't a production system you are testing on, then just give it a try and find out. Kevin OCG Technical Support wrote: Anyone tried Asterisk with Fedora 9 (recent release)? ___ -- 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] polycom ip330/ip501 second ethernet port
Hi Robert, While I'm not sure how our network compares with yours, we run about twenty 601 phones along with our office workstations (some stations are without a phone). Each station with a phone is connected with the other Ethernet port on the phone so we have one drop to each station. The phones are on a separate VLAN from the rest of the network as well. From the user end, I have not had a report of any problems with the connections, call quality, etc. I would say give it a shot, maybe with a larger network that could change, but for a small office like I'm in charge of, it is working just fine. Kevin Robert McNaught wrote: Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. I apologize for this question not being directly related to asterisk, but since Polycom phones are used a lot with asterisk, it seems a good place to post ;-) Robert McNaught ___ --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] Queue Question
Hi Jeremy,
A few thoughts that come to mind. We have a queue that is open between
certain hours. I have a few checks in place before a caller enters,
first it checks to see if there it is within the time window, then
checks to see if there are any agents log into queue, if any fail they
get our closed message. Sounds like you are trying to do something similar.
Not sure what you have for extension numbers numbers, but you will get
the idea.
Your first friend:
GotoIfTime(time range|days of week|days of
month|months?[[context|]extension|]pri)
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
I don't know how your dial plan is structured. My guess is the after
hours operation is in a separate part of the code from the other. Since
we are just looking at after hours, I would use the reverse on your
time. Because the command jumps when the statement is true. I do not
know what will happen if you say go from 17:00-8:00, but you can try it.
Example:
exten = 800,1,GotoIfTime(8:00-17:00|mon-fri|*|*?NormalOp,900,1) ; Since
this will fail if it is 9pm, it moves on to the next priority in this
exten.
[NormalOp]
exten = 900,1,blah
Next, is your other test. Use the queue agent count function
QUEUEAGENTCOUT(queuename)
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+queueagentcount
If the number is greater then 0, then you move them into the queue, if
not, whatever you want.
Finally, in terms of your other questions about logging the agents in.
You could do the database way. You also could create a log in extension
where you can take their cell number ( caller id) and use the
application AddQueueMember(queuename[|interface][|penalty])
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AddQueueMember
So you should be able to do something like
AddQueueMember(queueName|ZAP/${CALLID(num)})
Anyway hope that helps.
Kevin
Jeremy Mann wrote:
I’m curious if anyone has implemented the following:
Need to setup an on-call queue, that activates after 5PM and
de-activates at 8AM, also that activates/deactivates on demand(I’m
thinking a feature code here). The “agents” need to log in via cell
phones, and when calls come in from outside to the asterisk system,
it’ll need to call the cell phone agents that are active.
I’m thinking that it’s a simple SQL query, to update the agents status
and number, and that asterisk will do a lookup and append that to the
ZAP channel to dial, but interested in any logic someone might be able
to come up with for the dialplan.
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Re: [asterisk-users] AGI and exec Playback
I'm not sure of a way to do it through AGI, but I know you could make the script take the recording, use sox to convert it to the file format you need, then maybe use like a Flash media player to control the playback of the sound file. It is a bit clunky but it was just one of the ideas (the better ones) that came to mind when I was reading this. On our system, I created the option to call your extension with the call and play it back using ControlPlayback, or it converts it to a simple file format (such as wave, or mp3) and you can then download it and use a media player and do what you want with it. Otherwise I'm not sure what you can or cannot control with AGI in reference to playing sound files. Hopes this gives you a few ideas, Kevin Atis wrote: Hello, I'm looking for a way to play sound file, and control the playback trough web interface. Is it possible to use AGI to play a sound file and then by receiving some event stop playing it, and play another file. The catch is that i want to seek to 1st minute, 5th minute, etc - so regular ControlPlayback with intervals wouldn't fit - i have to use sox to create different file and then jump to it. Also - i have read that in asterisk 1.4. there is SendDTMF trough AMI - is it possible to use that for ControlPlayback? Here i would want regular Forward/Backward buttons on web :) Regards, Atis ___ --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] Call still in queue after Reject Signal
rachid wrote: Hi, I have a queue with maxlen=1, and when i make a call, the call enters into the queue, but he doesn't exit from it after a reject signal received from the agent?? please, have you any idea how to remove calls after a reject signal??? Thanks. Rachid ___ --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 Hi Rachid, Could you post some more information such as the dial plain before the call enters the queue. For example, are you trying to have them fall out of the one queue and move on to say another queue? maxlen=1 will only allow 1 call into the queue, it does not control, at least from the notes I have, what happens if the call is rejected. -Kevin ___ --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 file
Paul wrote:
I'm going to top post in this situation.
Kevin - Commands that operate on the channel variables won't help if we
are using a call file. We will have a new channel.
Agreed, I misread and thought he was trying to generate a call file.
-Kevin
This syntax works with asterisk 1.2.x for me:
Application: AGI
Data: say_it.php|call_status_message
I have done other things where a bunch of parameters are stored in
postgres or mysql and the only parameter I pass via the call file is the
record key. The php script receives the key as a parameter and gets
everything else from the db. Something like this:
Application: AGI
Data: inform.php|68456943
Kevin Smith wrote:
Nitesh Divecha wrote:
Hello All,
Is there any way to pass additional parameters while calling AGI from
*.call file?
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php
Something like Data: recordvoice.php?id=3453name=asterisk
Cheers,
Nitesh
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I'm not 100% sure if you can pass it directly, but you can use the set
option in the call file to set local variables within Asterisk and then
pass them to the AGI script. So for your example it would be.
Set: name=asterisk
This will set the variable ${name} in asterisk and depending how your
script was created you should be able to grab the variable to use within
the script. If you are using say the PHP AGI you can use something like
the following:
$var = $agi-get_variable(name);
This will create an array with $var['data'] holding 'asterisk';
Now one more thing I am not sure of is for multiple variables (haven't
tried it yet ;D ). You may have to do it one of two ways.
Set: name=asterisk, id=3453
or
Set: name=asterisk
Set: id=3453
and if those don't work, just format it so you can filter it out with PHP.
Hopefully this will help.
Kevin
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Re: [asterisk-users] .call file
Nitesh Divecha wrote:
Hello All,
Is there any way to pass additional parameters while calling AGI from
*.call file?
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php
Something like Data: recordvoice.php?id=3453name=asterisk
Cheers,
Nitesh
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I'm not 100% sure if you can pass it directly, but you can use the set
option in the call file to set local variables within Asterisk and then
pass them to the AGI script. So for your example it would be.
Set: name=asterisk
This will set the variable ${name} in asterisk and depending how your
script was created you should be able to grab the variable to use within
the script. If you are using say the PHP AGI you can use something like
the following:
$var = $agi-get_variable(name);
This will create an array with $var['data'] holding 'asterisk';
Now one more thing I am not sure of is for multiple variables (haven't
tried it yet ;D ). You may have to do it one of two ways.
Set: name=asterisk, id=3453
or
Set: name=asterisk
Set: id=3453
and if those don't work, just format it so you can filter it out with PHP.
Hopefully this will help.
Kevin
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[asterisk-users] g729 codec
Hi everyone, Simple question that I haven't been able to find a direct answer to. We currently have call recording with our asterisk system. The files, I am assuming since that is the codec we are using, are being recorded in the g729 codec. Is there a way to listen to these calls, say on windows media player or another audio program? Or do I need to convert the files to a different format to listen to them outside of Asterisk? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call waiting notification
Hi Kevin, Thanks, that's what I thought but sometimes you need a second opinion from someone with more experience to get administration off your back about an issue such as this. Kevin Kevin P. Fleming wrote: Kevin Smith wrote: We are running Polycom 601's. I can't seem to find anything to say one way or another on this issue, so I figured I would ask. I have call waiting notification working on the phones when a user is on the phone. However, is it possible to see the notification on the screen or hear it on the line when it is in the dial status, IE I just pick the receiver off the hook and I am about to dial a number. Nope. ___ --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
[asterisk-users] Call waiting notification
Hi everyone, We are running Polycom 601's. I can't seem to find anything to say one way or another on this issue, so I figured I would ask. I have call waiting notification working on the phones when a user is on the phone. However, is it possible to see the notification on the screen or hear it on the line when it is in the dial status, IE I just pick the receiver off the hook and I am about to dial a number. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuration / dialplan problem
There are a few things to look at.
First off, you have a lot of wildcard testing that is probably throwing
the dial plan off. For example, you have the following:
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07.,1,Congestion()
If I left it in this order what would happen? From what I understand it
is nautral to think in that order, but really Asterisk is going to sort
the extensions something like this:
exten = _07.,1,Congestion()
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
So now say you dial 07545865143254/8564, it will go to the Congestion
application every time.
What I would do is comment out the wildcard searches and see if that
resolves the problem. If so, try putting all the wildcard tests in an
include and see if that helps.
Take a look at these to articles as well:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
Also just out of observation, why all the testing? Seems to me you could
streamline that code down a bit more. For example, the 01 and 02 tests.
If you know they are dialing N number of digits, make the test
_01XX, so you know they have to dial a certain amount of digits
to be a valid call. Why send a 4 digit number out your trunk if you know
it isn't going anywhere? If you need to dial '0' then 10 digits, try this:
_01NXXNX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
_02NXXNX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
_07956X,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) 3
etc.
Hopefully that will help,
Kevin
Mark Muffett wrote:
I have my extensions.conf set up as follows:
exten = _Z.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _01.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _02.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _0800.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _0845.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _0870.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _09.,1,Congestion()
exten = _00.,1,Congestion()
exten = _07.,1,Congestion()
(where nn are actually real digits).
I would expect this to let me dial the 07956nn numbers etc while
stopping dialing to other 07... numbers, but it seems to stop dialling
to any 07... number including the 3 specifically listed.
Any ideas?
Thanks
Mark
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Re: [asterisk-users] Polycom related question
John Marvin wrote: Kevin Smith wrote: Here is what the configuration looks like for one of the phones, the other is 284: [283](Empire-Defaults) [EMAIL PROTECTED] [283a](Empire-Defaults) [EMAIL PROTECTED] [283b](Empire-Defaults) [EMAIL PROTECTED] So actually you are trying to use one phone to monitor (receive notifies for) multiple boxes. It looks like the Polycom's have some support a different mwi for each registration, but I'm not sure how well it works. Right. It sort of breaks down like this: (I pray this keeps the formatting) Phone 283 Phone 284 Line 1: VM box 283Line 1: VM box 283 Line 2: VM box 284Line 2: VM box 284 Line 3: VM box 285Line 3: VM box 285 So really one line on the phone is just looking at one mail box, but there are two phones per mail box. You didn't specify what username you specify for each config above, so I don't know if the notifies are going to one registration or to different registrations. The messages button on the phone only seems to show the status of one registration, but the indicator light seems to combine the different results together (and you can't clear the light with the clear button since that only applies to one of the registrations). Of course that assumes that you are sending the notifies to different registrations on the phone -- all bets are off if you are sending them to the same registration (which is controlled by the username value) since Asterisk is treating them as separate phones the notifies will collide with eachother. Well the phone will take the sip extension as the username. And I pass the username of the voice mail box in the to the voice mail function depending on which line it is calling from in the dialplan. That is if I am following your statement about which username I specify correctly. If I am not, then this probably won't make much sense. You would get more reliable behaviour if you did as Rich suggested and just specified something like this for just the [283] config: [EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED] In that case Asterisk sums up the total messages in each of the boxes and the messages button on the phone will show you that total rather than the results for only one of the boxes. I'll give it a try the next time I am in the office and see what happens. The polycom documentation is not very clear on how multiple mwi's are supposed to work, so I'm not sure what the right answer is. AGREED! John Thanks again with the suggestions, I'll let you know the results when I get a chance to try it out. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom related question
Sorry for the late reply, school has started again so I am not in the
office as much. I also remove all the old postings I didn't need and
also deleted mine, so if there were any before this with questions that
I still haven't answered, let me know.
Rich Adamson wrote:
John Marvin wrote:
Rich Adamson wrote:
If you look at the sample configs, you'll find:
[EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of
multiple mailboxes
in the sip.conf.samples for v1.2 stable. That is the only way that I
know of to turn on the mwi for two different phones (eg, extensions).
Is that what you're using and its not working?
I think that is the opposite of what Kevin is trying to do. The above
config is for one phone monitoring multiple voicemail boxes. Kevin
wants multiple (two) phones monitoring the same mailbox, i.e. he is
probably specifying the same mailbox within the config for each of
the phones that will be monitoring that mailbox.
Yes, John is correct, I would like two phones to monitor the same
mailbox. I know Rich you asked what type of phones I am using. They are
both Polycom IP 601's firmware 1.6.2. I have the update for 1.6.7 but I
haven't applied it yet since I want to be sure it isn't Asterisk or a
configuration before I upgraded to potentially more problems.
I'm not sure why there would be any problems with that. Kevin, have
you tried just having one phone at a time do the monitoring, to make
sure there aren't any problems with the phone's config? When one
misses a notification, is it always the same phone that misses it?
It's interesting that the problem is intermittent, it would seem that
if Asterisk doesn't support this that it would only notify one phone
each time and that the results would be consistant.
Yes, all of our phones (pretty much did a cookie-cutter configuration)
monitor their own mail box, this is the only setup with one. Actually a
point of clarification on my first post. The phones are actually will be
watching 3 mail boxes total. 2 phones, covering 3 mailboxes. The phones
do work if I just assign them one box only.
Here is what the configuration looks like for one of the phones, the
other is 284:
[283](Empire-Defaults)
[EMAIL PROTECTED]
[283a](Empire-Defaults)
[EMAIL PROTECTED]
[283b](Empire-Defaults)
[EMAIL PROTECTED]
The 'a' and 'b' are for the other lines that are watching the voicemail.
When they want to check say mail box 285, then press the 3rd soft-button
on the phone and it dials from 283b. The voicemail function checks for
the letter in the callerid and logs them into the correct mail box.
Which looks like this:
exten = 86*,1,Answer()
exten = 86*,n,Wait(1)
exten = 86*,n,gotoif,$[${CALLERID(num):-1}=a]?ServiceMail:NextTest
exten = 86*,n(NextTest),gotoif,$[${CALLERID(num):-1}=b]?PartsMail:SalesMail
exten = 86*,n(SalesMail),VoiceMailMain([EMAIL PROTECTED])
exten = 86*,n(ServiceMail),VoiceMailMain([EMAIL PROTECTED])
exten = 86*,n(PartsMail),VoiceMailMain([EMAIL PROTECTED])
exten = 86*,n,HangUp()
Phones don't monitor mailboxes. One needs to tell asterisk which
phones are to be notified when a voicemail is left, and the sip
statements above are the only ones that I'm aware of to accomplish that.
On many phones, there is only one mwi function. If Kevin has one
extn (eg, 111) on a phone set up with a mwi and then a second extn
(eg, 222) on the same phone set up for mwi, one extn's mwi might turn
the indicator on while the second extn will turn it right back off
again. Since I don't recall Kevin saying what type of phone he's
using, I can only guess that might be the problem.
Correct me if I am wrong, but from what I read, I can have at least one
mwi for each line. I am not really if I can list more or not. Which is
why I did the above with the multiple lines.
Its either that, or, my original comment above regarding the sip
definitions.
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[asterisk-users] Polycom related question
Hi everyone, While this isn't a true asterisk question, I know a lot of people here use Polycom phones. Anyway, I have two Polycom 601 phones that share the same voicemail box. Now it is intermittent, but sometimes both phones will have a notification there is a voice mail, but then sometimes only one will show that there is a voicemail. If the phone that doesn't show there is a voicemail connects to the voicemail box it can get the message, but just no indication. My question is, has anyone else tried doing this and had success? If so is there anything on Asterisk that I need to set or in the configuration for the phones that I may be overlooking? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MSSQL connection
Thanks Tim, That was my first thought as well but then I thought, might as well give it a try. But it is turning into a hassle more then anything. I already have a PHP script wrote to for MySQL so the conversion to MSSQL shouldn't be bad. Thanks, Kevin Tim Panton wrote: On 9 Sep 2006, at 00:42, Kevin Smith wrote: Hi everyone, I am looking to log CDR records to our MSSQL database for further examination on the records. From what I gathered from the wiki I have to choose between FreeTDS and unixODBC. Is there a better choice? Which option would be better in the log run? Also configuration asterisk to use both modules. Any good tips on that, I followed the steps provided by the following pages: http://www.voip-info.org/wiki/view/FreeTDS http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc But this is the error I get: (note: some information has been changed for security, such as 'user' and pass was changed to phone) # isql -v MSSQL-astersik phone phone [S1000][unixODBC][FreeTDS][SQL Server]Unable to connect to data source [28000][unixODBC][FreeTDS][SQL Server]Login incorrect. [][unixODBC][FreeTDS][SQL Server]Login failed for user 'phone'. [][unixODBC][FreeTDS][SQL Server]Cannot open database requested in login 'cdr'. Login fails. [ISQL]ERROR: Could not SQLConnect from odbcinst [MSSQL-FreeTDS] Description = FreeTDS ODBC driver for MSSQL Driver = /usr/lib/libtdsodbc.so Setup = /usr/lib/libtdsS.so FileUsage = 1 from odbc [MSSQL-asterisk] description = Asterisk ODBC for MSSQL driver = MSSQL-FreeTDS server = XXX.XXX.XXX.XXX port= 1433 database= cdr user= phone password= phone tds_version = 7.0 language= us_english Maybe I am just overlooking something or there is something that isn't registering with me that is under my nose. Any help would be appreciated. My guess is it is an error between the keyboard and chair ;). We have just been through this - but with Oracle - and came to the conclusion that we didn't want to tightly couple asterisk with the DB, we felt it could be a performance hit - on both sides - plus it meant allowing ODBC traffic over a network we couldn't secure. In the end we got a script written that reads the Master.csv file, turns the new data into XML and does an HTTP Post of the data to a web service running on the Oracle system which parses the XML and inserts the records in the database. We plan to run the script every few minutes (from cron). Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another (quick) Polycom 501 question
Hi Mike, As far as I know, you need to at least start the dialing (ie New call, speaker, etc) for the digitmap to even come into play. The only settings that I am aware of that you can try to change are dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf. Kevin Mike wrote: Hi all, That's my last one for a while (I hope). How can I (if at all possible) make the 501 turn on the speaker phone as soon as a digit is dialed (if the handset is not lifted)? Sort of like what a normal speakerphone does. The reason I want this is I want the 501 digitmap to be taken into consideration even if the handset isnt lifted and the speakerphone button isn't consciously pressed. For all those users who don't want to press send, but like dialing without lifting the handset (and can't be bothered to press the speakerphone button). Yes I know it's capricious, but we have the users we have... Yes, I have read the admin manual, but couldn't find the info. I am assuming I just don't know what to look for, but that this functionality exists. Mike ___ --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
[asterisk-users] MSSQL connection
Hi everyone, I am looking to log CDR records to our MSSQL database for further examination on the records. From what I gathered from the wiki I have to choose between FreeTDS and unixODBC. Is there a better choice? Which option would be better in the log run? Also configuration asterisk to use both modules. Any good tips on that, I followed the steps provided by the following pages: http://www.voip-info.org/wiki/view/FreeTDS http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc But this is the error I get: (note: some information has been changed for security, such as 'user' and pass was changed to phone) # isql -v MSSQL-astersik phone phone [S1000][unixODBC][FreeTDS][SQL Server]Unable to connect to data source [28000][unixODBC][FreeTDS][SQL Server]Login incorrect. [][unixODBC][FreeTDS][SQL Server]Login failed for user 'phone'. [][unixODBC][FreeTDS][SQL Server]Cannot open database requested in login 'cdr'. Login fails. [ISQL]ERROR: Could not SQLConnect from odbcinst [MSSQL-FreeTDS] Description = FreeTDS ODBC driver for MSSQL Driver = /usr/lib/libtdsodbc.so Setup = /usr/lib/libtdsS.so FileUsage = 1 from odbc [MSSQL-asterisk] description = Asterisk ODBC for MSSQL driver = MSSQL-FreeTDS server = XXX.XXX.XXX.XXX port= 1433 database= cdr user= phone password= phone tds_version = 7.0 language= us_english Maybe I am just overlooking something or there is something that isn't registering with me that is under my nose. Any help would be appreciated. My guess is it is an error between the keyboard and chair ;). Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Well personally I am just glad I wasn't the only one seeing the problem. As much as I don't like the place 100% of the blame on something unless I fully know what is going on, in this case Asterisk, but I couldn't see any solution but a bug. Personally I wouldn't mind testing out the branch, but I know my boss, isn't so trusting. How stable are the SVN branches, at least in terms of justification for taking the system down to install it? Or is there an easier way to test? Thanks, Kevin Kevin P. Fleming wrote: - Richard Scobie [EMAIL PROTECTED] wrote: Dave Fullerton wrote: I just verified it here as well. Running Asterisk 1.2.11 and two polycom I'll throw in a "me too" here, with the addition that it also occurs with "canreinvite=no". There were multiple problems in this area, introduced since Asterisk 1.2.9 was released. We believe that with today's commits in SVN branch-1.2 they are cured, so it would help us greatly if could download SVN branch-1.2 and try it out on your system to see if it solves your issue. I apologize for how this crept into the code base... it should not have happened, and we are taking steps to ensure that future changes in the release branch don't cause regressions like 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
Re: [asterisk-users] Blind transfer 3/4 digits
Dialing a number and transferring a number are two different things. And no offense, you are not really providing a lot of details along with your problem. So you can dial the numbers but not transfer from one to the other. What does the CLI say when you try the transfer? That would provide a lot of information that could clue you in to what is going on. What type of phones are you using? Some phones have the ability to pattern match and wait for a certain number of seconds before sending the number to asterisk. For example. On our Polycom phones a user has 3 seconds (between digits) to enter in 10 digits. This could be where most of your problem is. My guess the problem lies with the Phones, not Asterisk form the information you provided. Kevin Ronald Wiplinger wrote: David Gagnon wrote: Ronald, You seem to be a little bit angry about VoIP. If so, I could give you my old Nortel system. Does this would make you happy? David David, I am not angry about VoIP, but please send my your old Nortel system ! I just do not understand why I can DIAL 601 and 6014, but not use blind transfer. Is the question too difficult? I am sure there is somewhere a switch to say, wait two seconds (as for dialing) before you assume it is a complete number. It is also strange that snom phone can do it correct, because it uses the ok key. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 2 septembre 2006 04:20 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Anthony Rodgers wrote: With respect, the problem is with your numbering plan.. This answer is therefore totally nonsense !!! (With all respect!!!) Both answers have actually not lead to any step further, but to more messages. I use to refer to such answers as NON-ANSWERS. Please only reply if and really only if you know a solution for the problem! Thanks for your understanding. bye Ronald - again, I am not angry at all. WHERE do you see a problem in the numbering plan? I see the problem in ASTERISK, because it does not wait for the last digit!!! Where can I set that it waits for it? The beauty on voip IS that you can have different length and overlapping, bye Ronald CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires me to hit ok If I get a call on 601 and transfer to 6014, than 601 will get the busy signal and I hang up as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye Ronald ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0635-4, 2006/09/01 Tested on: 2006/9/2 ¤U¤È 03:52:00 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Hi Avi, I had a similar problem. Have extension 405 put the call on hold (after the transfer) and then off hold. I am willing to bet it will bring back the audio stream. I posted something similar a few weeks ago and if anyone thought it was a bug, to let me know what information I needed to send in to report it, but no one replied. Anyway, I noticed it happening on the latest release of asterisk. I rolled back my installation so I am on asterisk 1.2.9.1, lib 1.2.3, and zaptel 1.2.6 and that corrected the problem for me. Kevin Avi Miller wrote: Hey guys, I've been trying to change my Asterisk setups to use canreinvite=yes. I'm having a small problem with my Polycom IP501 phones and transferring calls. If a call comes in via my ISDN BRI lines (using chan-capi), I can successfully transfer the call using the Polycom Blind Transfer option (Transfer - Blind - EXT - Send). However, if I try to use the attended transfer method, the call is never connected to the new user. When I hit transfer, the caller gets MOH and I dial the destination ext. Once the person answers, I hit Transfer Now .. the MOH stops for the caller, but both phones are dead. The call is never reconnected successfully. On the console, I see this: -- Called 405 -- SIP/405-0849cba0 is ringing -- SIP/405-0849cba0 answered SIP/401-084a0ba8 -- Attempting native bridge of SIP/401-084a0ba8 and SIP/405-0849cba0 -- Stopped music on hold on CAPI/V4BRI-2/92355400-25 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/401-084a0ba8ZOMBIE' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/401-084a0ba8ZOMBIE' -- Incoming call: Got SIP response 500 Internal Server Error back from 192.168.1.128 == Spawn extension (macro-dial, s, 10) exited non-zero on 'CAPI/V4BRI-2/92355400-25' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'CAPI/V4BRI-2/92355400-25' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'CAPI/V4BRI-2/92355400-25' 405 is the extension I'm trying to transfer the call to. Any advice? I've been searching the list archives and the wiki, but can't find anything specific. Ta, Avi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan or matching
Glad I could help. I agree, these mailing lists are a life saver. I
personally have only been using Asterisk for about 5 months now, in fact
I have never even delt with any PBX's before (complete newbie) but
everyone here is very helpful and I am picking up a lot.
Kevin
David Cook wrote:
Thanks Kevin! That's what is great about these forums. I never thought
of using gotoif() inside ... one of those Doh! moments.
I included your concept in my standard [dial-ld] context with
${EXTEN}:1:3=800, etc. rather than by 2's, (so it doesn't overlap
with 8XX area codes) and select my local loop as the first pick.
dbc.
Kevin Smith wrote:
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten = _18XXNXX,1, NoOP()
exten = _18XXNXX,n,gotoif(${EXTEN}:2:2 = 00 | ${EXTEN}:2:2
= 66 | ${EXTEN}:2:2 = 77 | ${EXTEN}:2:2 = 88)?TRUE:FALSE
exten = _18XXNXX,n(TRUE),Dial()
exten = _18XXNXX,n(FALSE), HangUp()
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Re: [asterisk-users] Dialplan or matching
Hey David,
Yes, it can, you just have to play around with the logic and what you
are comparing and when you can do the comparison.
Try something like this:
exten = _18XXNXX,1, NoOP()
exten = _18XXNXX,n,gotoif(${EXTEN}:2:2 = 00 | ${EXTEN}:2:2 =
66 | ${EXTEN}:2:2 = 77 | ${EXTEN}:2:2 = 88)?TRUE:FALSE
exten = _18XXNXX,n(TRUE),Dial()
exten = _18XXNXX,n(FALSE), HangUp()
I'm sure you can take it from there. You can remove the first line with
the NoOP but I normally feel it is good to give an instruction cycle to
Asterisk (and any program) when jumping to another extension (or
function), is it needed, no, but you never know.
Kevin
David Cook wrote:
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches
sort of like the SPA's can?
Tollfree numbers for example. I can have a line for each combination:
exten = _1800NXX, Dial,
exten = _1866NXX, Dial,
exten = _1877NXX, Dial,
exten = _1888NXX, Dial,
But I want to do is something like this:
exten = _18[0678][0678]NXX, Dial, .
Or to prevent the logic error which albeit small, the above would create:
exten = _18[00,66,77,88:2]NXX, Dial, ..
(representing that the next 2 chars must equal any of '00'.'66','77'
or '88'
Is there any syntax that allows this??
dbc.
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Re: [asterisk-users] Polycom upgrade issue
Doug, Note: Don't take this email serious, I'm just messing with you, but it sure as poop is ;). In version 1.6.x released 18th of July 2005 in section 2.2.1.4, Reset the Factory Defaults "To perform this function on all phones except the IP4000, simultaneously press and hold 4,6,8 and * dial pad keys until the password prompt appears." However, depending on which version you are looking at it may be in a different section. Cheers, Kevin Douglas Garstang wrote: How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED]] Sent: Tuesday, August 15, 2006 11:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom upgrade issue I believe 468* resets the phone but dosent return it to the orig. firmware. Also try to name the files with the phones mac id and see what happens. I am doing this with 1.6.6 and its working fine. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, August 15, 2006 10:07 PM Subject: [asterisk-users] Polycom upgrade issue OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log 0527180621|cfg |4|00|Could not get all 512 bytes of the header. 0527181013|cfg |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006 I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there? Thanks! Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call transfer issues
My guess is I stumped everyone ;)
Anyway, I rolled back asterisk to 1.2.9.1 (same for libpri and zaptel
back one release) and transfers were working again. Now I'm still quite
new to asterisks, I know enough to hold my own, but not enough to know
the full inter workings of it. But here is my thought:
Caller A calls in and talks to Employee B. B wants to transfer to C.
Asterisk sets up the bridge between B and C. B completes the transfer.
Now A and C are connected but there is no audio stream. If C or A puts
the other on hold, and then resumes the call, audio is restored.
By that I would say placing them on hold clears a flag or updates one to
connect the audio stream? Or am I way off on this assumption? Also if
this sounds like a possible bug, what information do I need to include,
or is good to include, when submitting bugs?
Thanks,
Kevin
Kevin Smith wrote:
Hey everyone,
Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk
1.2.10. It has been reported to me when doing an attended transfer the
audio drops out. I ran a few different tests and here is what I noticed.
1. Blind transfers work with no problem.
2. Attended transfers were you transfer the call before the person
picks up works.
3. If the person the call is being transferred to answers and then the
transfer completes, the audio drops.
I noticed in the CLI the following (I replaced the number with XXX's)
-- Attempting native bridge of SIP/989XXX-b76167c8 and
SIP/989XXX-08f956b8
== Parsing '/etc/asterisk/manager.conf': Found
-- Stopped music on hold on Zap/2-1
== Spawn extension (Mercury-Directory-Dialer, 989XXX, 8) exited
non-zero on 'SIP/989XXX-b76167c8ZOMBIE'
-- Executing Hangup(SIP/989XXX-b76167c8ZOMBIE, ) in new
stack
== Spawn extension (Mercury-Directory-Dialer, h, 1) exited non-zero
on 'SIP/989XXX-b76167c8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back
from 64.7.177.103
Now what I noticed is that once the transfer is done, I'm still
connected the the person that called me to do an attended transfer.
However, if I hang up the phone, the call drops. If I place the call
on hold and take them off hold, audio is resumed and everything works
normally.
Here is the conf information
exten = s,1,SetCallerID(${ARG1})
exten = s,n,Set(DST_EXT_NUM=${ARG2})
exten = s,n,gotoif,$[${ARG2}=989XX]?TIME:GOON ;Add test if
hours is the basis for voice mail
exten = s,n(GOON),AGI(VoiceMail.php) ;Test for phone status
exten = s,n,Noop(Testing Completed||Code:${ON_PHONE}${IN_QUEUE})
exten = s,n,Dial(SIP/${ARG2},25)
...VoiceMail choice
exten = h,1,HangUp()
Where I have VoiceMail choice it takes the variables from the AGI
script and decides which voice message to play. But the problem is
happening before that occurs so I don't think it has anything to do
with the problem.
Any ideas to what could be the cause or how to correct it? SIP version
or does the new asterisk build have any new features enabled by
default that the older build would not? Any suggestions or thoughts
would be greatly helpful.
Kevin
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[asterisk-users] Call transfer issues
Hey everyone,
Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk
1.2.10. It has been reported to me when doing an attended transfer the
audio drops out. I ran a few different tests and here is what I noticed.
1. Blind transfers work with no problem.
2. Attended transfers were you transfer the call before the person picks
up works.
3. If the person the call is being transferred to answers and then the
transfer completes, the audio drops.
I noticed in the CLI the following (I replaced the number with XXX's)
-- Attempting native bridge of SIP/989XXX-b76167c8 and
SIP/989XXX-08f956b8
== Parsing '/etc/asterisk/manager.conf': Found
-- Stopped music on hold on Zap/2-1
== Spawn extension (Mercury-Directory-Dialer, 989XXX, 8) exited
non-zero on 'SIP/989XXX-b76167c8ZOMBIE'
-- Executing Hangup(SIP/989XXX-b76167c8ZOMBIE, ) in new stack
== Spawn extension (Mercury-Directory-Dialer, h, 1) exited non-zero on
'SIP/989XXX-b76167c8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back
from 64.7.177.103
Now what I noticed is that once the transfer is done, I'm still
connected the the person that called me to do an attended transfer.
However, if I hang up the phone, the call drops. If I place the call on
hold and take them off hold, audio is resumed and everything works
normally.
Here is the conf information
exten = s,1,SetCallerID(${ARG1})
exten = s,n,Set(DST_EXT_NUM=${ARG2})
exten = s,n,gotoif,$[${ARG2}=989XX]?TIME:GOON ;Add test if
hours is the basis for voice mail
exten = s,n(GOON),AGI(VoiceMail.php) ;Test for phone status
exten = s,n,Noop(Testing Completed||Code:${ON_PHONE}${IN_QUEUE})
exten = s,n,Dial(SIP/${ARG2},25)
...VoiceMail choice
exten = h,1,HangUp()
Where I have VoiceMail choice it takes the variables from the AGI script
and decides which voice message to play. But the problem is happening
before that occurs so I don't think it has anything to do with the problem.
Any ideas to what could be the cause or how to correct it? SIP version
or does the new asterisk build have any new features enabled by default
that the older build would not? Any suggestions or thoughts would be
greatly helpful.
Kevin
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Re: [asterisk-users] Auto retry on Busy
Why don't you just test for the dial status after the dial command
completes? I don't really see why you want something to keep dialing
until it gets through, but this would work.
[something]
1,1,Dial(zap/,sip/, etc/whatever, 10)
1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
1,n(LINEBUSY), Wait(30)
1,n,goto(something,1,1)
1,n(OTHER), do something else
Sure it is pretty rough, but the basics are there. Also you might want
to read this: http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Kevin
Noah Silverman wrote:
Hi,
Does anybody have an easy solution for this.
I want something that will keep trying a busy number every 30 seconds
until it gets through.
I've tried retrydial, but can't get it to work.
Any suggestions?
Thanks,
-N
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Re: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
If I am following you right, for extension matching you need to have a _ in front of the number. So your example should be like this: exten = _949927,1,Goto(mainmenu,s,1) Also I don't know if you did this on purpose or not but N will only match for numbers 2-9, if you want 0-9 you will want to use an X. Otherwise without the _ in front of the number it will not extension pattern match. There are other pattern matching characters too, but you can read them here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns Kevin Mr. Jones wrote: I'm trying to get inbound DIDs working via SIP. I have 20 DIDs coming in via a single SIP profile in sip.conf. I was hoping to have these matched in extensions.conf, so I have setup lines like this: exten=949271,1, Goto(mainmenu,s,1) Unfortunately these aren't getting matched and I'm getting this error: Looking for s in druid-default (domain 949271) SIP/2.0 404 Not Found Any hints or tips? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto retry on Busy
Interesting. I guess unchecked (which my sample had no error checking)
it would lead me to think it would just use up resources. But I suppose
with the correct implementation I could see a use for it.
Kevin
Rushowr wrote:
The reason he might want it is because it's a feature offered by many POTS
and Mobile Telcos. I know that's why I've played with it, the ITSPs and SIP
Termination providers I consult for want to have as many if not more
features to offer than the POTS and Mobile guys.
Cheers,
Rushowr - Sherwood McGowan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent: Friday, August 11, 2006 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto retry on Busy
Why don't you just test for the dial status after the dial command
completes? I don't really see why you want something to keep dialing until
it gets through, but this would work.
[something]
1,1,Dial(zap/,sip/, etc/whatever, 10)
1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
1,n(LINEBUSY), Wait(30)
1,n,goto(something,1,1)
1,n(OTHER), do something else
Sure it is pretty rough, but the basics are there. Also you might want to
read this: http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Kevin
Noah Silverman wrote:
Hi,
Does anybody have an easy solution for this.
I want something that will keep trying a busy number every 30 seconds
until it gets through.
I've tried retrydial, but can't get it to work.
Any suggestions?
Thanks,
-N
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Re: [asterisk-users] Queue Stats
I'm sure there probably is other ways to do this but you could write a script as a cron to use the manager API, filter the data you want, and store it in a database or text file. But depending how often you run it, you may miss some data. Douglas Garstang wrote: Thanks Johann. Yes, I wish they wouldn't reset on a restart. :( -Original Message- From: Johann [mailto:[EMAIL PROTECTED] Sent: Thursday, July 20, 2006 12:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Stats W - Waiting C - Completed A - Abandoned SL - Service level(defined in queues.conf servicelevel value). Percentage of calls answered within the time frame. These numbers reset on reload or restart. --johann Douglas Garstang wrote: Not documented anywhere that I can see. What are the W:, C:, A:, SL: and 'within' fields showing? Is holdtime AVERAGE hold time? hestia*CLI show queues oe_techsupp has 0 calls (max unlimited) in 'rrmemory' strategy (4s holdtime), W:0, C:52, A:11, SL:0.0% within 0s Members: Agent/80014154 (Unavailable) has taken no calls yet Agent/80014109 (Busy) has taken 11 calls (last was 62963 secs ago) Agent/80014150 (Unavailable) has taken no calls yet Agent/80014133 (Busy) has taken 32 calls (last was 1320 secs ago) Agent/80014151 (Unavailable) has taken no calls yet Agent/80014152 (Not in use) has taken 9 calls (last was 5892 secs ago) Agent/80014157 (Unavailable) has taken no calls yet Agent/80014155 (Unavailable) has taken no calls yet No Callers Doug. ___ --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
Re: [asterisk-users] Queue RoundRobin
Hi Santiago, Unless it is a typo on the wiki, I think you want your queue.conf to be like this: member = Agent/@1 member = Agent/:2,1 That way you include group 1, and then include group 2 with consideration of penalty. From the problem you are having it sounds like the agent whose phone keeps ringing is in a lower penalty then the other agent. Are both agents in the same group? If you make the one agent busy does it ring to the next phone? If not, what does the CLI say when it tries to connect the next call to the second phone? Kevin Santiago del Castillo wrote: Hi, I'm setting up a new asterisk for an ecommerce company with cust sup dept. The problem I'm having is with Roundrobin (and rrmemory also): Let's suppose that I have 2 agents logged in into a queue. When a client calls, and both agents are available. It rings the first one, but it doesn't answer the phone. The timeout takes effect and it should start ringing the second agent. But it doesn't. It keeps ringing the first one until it answers the phone Here's my queue.conf: [general] [QueueEN] announce = ann-english strategy = rrmemory timeout = 5 retry = 1 wrapuptime=0 maxlen = 0 announce-frequency = 20 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 [QueueES] strategy = rrmemory timeout = 5 retry = 5 wrapuptime=0 maxlen = 0 announce = ann-spanish announce-frequency = 10 announce-holdtime = once queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-thankyou = queue-thankyou member = Agent/@1 member = Agent/@2,1 The timeout is set too low so the test is faster. Cheers, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP301 and Queues
Hi Julian, If the 301's support ACD log in and log out, they should display a soft button showing the current status of the phone, I know for sure the 601's do. Personally with our 601's I used two of the contact lines and made my own log in and logout buttons and wrote my own script to log our agents in. It doesn't display the status, but I have a section on our intranet page showing the status of all members of a queue that are logged in. So it may not be the answer you wanted, and again I don't have any experience with the 301's to say what they can and cannot do, but there are some workarounds that will come close to the same goal. kevin Julian Varanini wrote: Is there any way to use the polycom phones to log into and out of queues? So the polycom phone could show their current status in that queue? logged in / logged out for example. Thanks Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tough time getting Polycom phones to register after router reboot
If you turn verbose on under the remote console for asterisk does it show any information that phones are trying to register or anything for that matter? Another thing you may want to verify is that the phones are communicating with the server if you aren't seeing anything on verbose. I am assuming the phones have real IPs since you have nat=no. Try using tcpdump from the asterisk server and see if you are seeing anything from one of the phones in question. Kevin [EMAIL PROTECTED] wrote: Strange situation: Had a router issue. After router re-booted most of my Polycom SIP phones re-registered, but some did not. I still cannot get the ones that did not register, to register. All phones are Polycom SIP phones (either 301 or 501). The ones that register show Useragent : PolycomSoundPointIP 1.6.5.0043 whereas the ones that did not register show Useragent : blank, when I do CLI sip show peer 4-digit extebsion number. What is going on with these Polycoms? All phones were registering fine and working for many weeks with no issues. I had the customer check pbx server settings and reboot phones. All sip.conf seting for all phones are equal, nothing is behind nat. I am using asterisk realtime with sigman: nat=no port=5060 host=dynamic qualify=no etc. Above are equivalent to all sip.conf entries for all extensions, and were working for a long time; now some have dropped off registration and are not coming back. Using * 1.2.9 realtime. Anything else I can chedk or do? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom config file location
Stephen, I would check with your polycom reseller. They should have the files you are looking for and you know they will be at least from a creditable source. In terms of setting up your phones for ftp provisioning, you will need to edit the files that you obtain from the reseller, and edit them to match the settings your phones currently have, otherwise it will download whichever files you have. on the server and overwrite the current configuration with the ones on the FTP server. For the most part, the change shouldn't take to long, the files have about 5 to 10 things that you will need to change. Kevin Stephen Murphy wrote: I have deployed 5 Polycom 301 phones manually and I would now like to provision them via my ftp server. My question is: How do I get the current config files the phone is using off the phone? If I do an ftp provisioning all the phones info will be lost true? So basically I need to get the current config files and upload them to the ftp server. Stephen Murphy VP Operations Cell: 604 790 3070 wVoIP: 604 638 8181 web: expansivenetworks.com 501 905 West Pender St Vancouver, BC V6C 1L6 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom, TFTP, and DHCP
Michael, Maybe I am not understanding your question, are you saying that when you configure your phone with a static IP address, you cannot find the boot server and when in DHCP you can? If you are having problems with the phone having a static IP address, make sure it is getting the correct IP, subnet, gateway and DNS. If your DNS is incorrect for example, you won't be able to find the server you entered, since there will be nothing to point the phone where to go. If you are talking about the actual boot server location, that needs to be static as far as I know. It isn't like DHCP addressing where it gets the DNS information from the host. It's a parameter that needs to be set. If your TFTP server is changing IPs I would strongly suggest giving it a static IP. It will make your life a lot easier. Kevin Michael Welter wrote: When I set the tftp address into the IP501 server parameters and boot, the phone says it says it cannot find the boot loader and reuses the previous configuration. When I set the tftp address in DHCP and reboot the phone, it finds the tftp server and loads correctly. My problem is that I don't always have control of the DHCP server. Is there a way to set the phone to find the tftp server on its own? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global variables and AGI
Yes, thanks again for the suggestions. I wrote a few scripts for different things that we needed in the office and by the time I got to that one, I was tired and wasn't thinking straight anymore. I am probably going to just set a dummy variable for now and have asterisk update the global. Down the road we plan on adding a database for call logging, configurations, etc, and I would agree with you Jay, storing the variable there would be the better choice. Thanks again. Kevin Jay Milk wrote: Kevin Smith wrote: Hi everyone, I know that functions like set_variable and get_variable (using php with phpagi) only apply to the channel variable. What I need to do is reset a global variable I have in our system. I have a script that is going to determine when this will happen, but I just have to make it happen. Assuming that I cannot update the variable via the script, it is there a way I can make a call to the system, such as a call file, and place it in the context of the dialplan that I need to change the variable? If so, is there anything special I need in the call file for that to work? Or is there a easier/better way to do this that I haven't thought of. Any suggestions would be helpful. Thanks, Kevin As Timebandit pointed out -- http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set or SetGlobalVar in 1.0.x If most of the interaction with that variable occurs through agi, you might also want to consider storing it outside of Asterisk. I've stored a good number of values in mysql for an asterisk application before. If most of the interaction occurs within the dialplan and/or you're trying to avoid agi, you could also use the asterisk database directly with DBPut and DBGet. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom, TFTP, and DHCP
I would double check and make sure what next-server is referring to. I'm not 100% sure of how these phones are configured besides what the Admin manual has, we use 601's in the office. I do know you need to tell the phone where the FTP server is located. If it doesn't know how to contact the server you won't get anywhere. You can also setup a domain name for the FTP server, but then you need to make sure your DHCP server (or something that handles DNS internally) can handle the DNS requests and point the phone in the correct direction. I would say if possible, get the FTP server configured with a static IP and just point the phones at it. I think in the long run it may save you some headaches. That way, you don't need access to the DHCP server and you know exactly where the TFTP server is located. Domain names are nicer however since if you need to adjust the IP address, you don't have to redo all the phones with the new address, the DHCP server will take care of it, but they are a bit more work. Kevin Michael Welter wrote: Kevin Smith wrote: Michael, Maybe I am not understanding your question, are you saying that when you configure your phone with a static IP address, you cannot find the boot server and when in DHCP you can? The phone uses DHCP to get its IP address. In the phone's server params, I enter the IP address of the tftp server. Without the next-server entry in the DHCP configs, the phone says it cannot find the boot server (and uses the previous configuration). However, when next-server in DHCP is set with the tftp IP, the phone loads its configuration from tftp and boots normally. I'd like to not have to set the tftp address in DHCP, because I don't always have access to the DHCP server. Is there someway to tell the phone to override the DHCP server setting? Is there something I'm missing with the phone's network config? Thanks If you are having problems with the phone having a static IP address, make sure it is getting the correct IP, subnet, gateway and DNS. If your DNS is incorrect for example, you won't be able to find the server you entered, since there will be nothing to point the phone where to go. If you are talking about the actual boot server location, that needs to be static as far as I know. It isn't like DHCP addressing where it gets the DNS information from the host. It's a parameter that needs to be set. If your TFTP server is changing IPs I would strongly suggest giving it a static IP. It will make your life a lot easier. ___ --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] Global variables and AGI
Hi everyone, I know that functions like set_variable and get_variable (using php with phpagi) only apply to the channel variable. What I need to do is reset a global variable I have in our system. I have a script that is going to determine when this will happen, but I just have to make it happen. Assuming that I cannot update the variable via the script, it is there a way I can make a call to the system, such as a call file, and place it in the context of the dialplan that I need to change the variable? If so, is there anything special I need in the call file for that to work? Or is there a easier/better way to do this that I haven't thought of. Any suggestions would be helpful. Thanks, Kevin ___ --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] PHP AGI
Hi everyone, Can someone post an example of how you read in a channel variable from asterisk through PHP. I tried the ones voip-info.org but none of them seem to work, or at least I am not doing something write, but I have no problem setting variables and other functions, just reading variables into my script. The variable I want to read in is a macro argument, and just to be safe, I assign it to a channel variable within the macro. Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP AGI
I have tried both ways (with PHPAGI and without), and neither works I
went back to a real simple test, and that doesn't even work.
Here is the CLI:
- Executing AGI(SIP/9897943713-9e04, VoiceMail.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VoiceMail.php
-- AGI Script VoiceMail.php completed, returning 0
Here is my code
#!/usr/bin/php -q
?php
set_time_limit(0);
require('../phpagi/phpagi.php');
$agi = new AGI();
$agi-answer();
$agi-say_digits(1,2,3);
//$agi-stream_file('welcome.gsm');
?
Both stream file and say digits did not work. The file is located in my
/agi-bin/ directory. The PHPAGI files are located in in the a directory
up called phpagi. So I don't see what is wrong with ../phpagi/phpagi.php
Any ideas?
Thanks,
Kevin
Time Bandit wrote:
Can someone post an example of how you read in a channel variable from
asterisk through PHP. I tried the ones voip-info.org but none of them
seem to work, or at least I am not doing something write, but I have no
problem setting variables and other functions, just reading variables
into my script. The variable I want to read in is a macro argument, and
just to be safe, I assign it to a channel variable within the macro.
Use phpagi : http://phpagi.sourceforge.net/
hth
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Re: [asterisk-users] PHP AGI
Hey guys, thanks for the suggestions, I finally figured it out.
I need to run the script using the CGI version of php or
#!/usr/bin/php-cgi -q...not really sure why, but it all started working,
AGI classes and all.
Thanks again,
Kevin
Time Bandit wrote:
I have tried both ways (with PHPAGI and without), and neither works I
went back to a real simple test, and that doesn't even work.
Here is the CLI:
- Executing AGI(SIP/9897943713-9e04, VoiceMail.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VoiceMail.php
-- AGI Script VoiceMail.php completed, returning 0
Here is my code
#!/usr/bin/php -q
?php
set_time_limit(0);
require('../phpagi/phpagi.php');
$agi = new AGI();
$agi-answer();
$agi-say_digits(1,2,3);
//$agi-stream_file('welcome.gsm');
?
Both stream file and say digits did not work. The file is located in my
/agi-bin/ directory. The PHPAGI files are located in in the a directory
up called phpagi. So I don't see what is wrong with ../phpagi/phpagi.php
Any ideas?
see http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SayDigits
I think you should write it like this : $agi-say_digits(123);
for stream_file, Asterisk will look in /var/lib/asterisk/sounds/
b.t.w., you can tail asterisk log while running your AGI. also, set
your verbosity to something high like 50
or do as Michiel said
hth
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Re: [asterisk-users] PHP AGI
I agree, that's what every example I saw was using. But ya, it's working now so I'm a happy camper :D Time Bandit wrote: Hey guys, thanks for the suggestions, I finally figured it out. I need to run the script using the CGI version of php or #!/usr/bin/php-cgi -q...not really sure why, but it all started working, AGI classes and all. Strange, I run it with standard PHP #!/usr/bin/php -q Well, if it works, then I guess everything is ok :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemails randomly not deleting in 1.2.9.1 ??
We have had this problem too, but just not as frequently as others are reporting. I started to write a PHP script as a workaround to browse all of the mail box folders and remove the txt file that is not needed. However, I haven't tested it fully to make sure it doesn't mess anything else up. It does flag the correct messages and leave the others alone and seems to be fairly dynamic when new folders are made etc. However, it is limited to the default setup (WAV, wav, gsm, and txt) files. I haven't had time to make that dynamic. Also Matt, what other problems have you had with 1.2.9.1. I am just wondering because since 1.2.8 I have noticed a few problems with our phones rebooting, random calls dropping when we do transfers, etc. Were those some of the issues? Does 1.2.7 seem to resolve the issues you have had with the other builds? Thanks, Kevin Matt wrote: It is a bug. It happens when someone is, I believe, leaving a message and you delete a message at the same time (or something along those lines). 1.2.9.1 may have fixed the IAX exploit, but it is way too unstable, and has way too many bugs to be used in production, IMHO. we rolled back to 1.2.7 after running 1.2.9.1 for 4 days and having it crash on us several times a day. 1.2.7 has now been running for over 2 weeks. On 7/7/06, Anthony Davis [EMAIL PROTECTED] wrote: We updated our systems to 1.2.9.1 (from 1.2.4) about 3 weeks ago. About every other day since then, a user will complain that they deleted a voicemail, but that Asterisk continues to tell their phone that they have new messages. They also can no longer delete these messages. A quick check on the server shows that the .WAV and .GSM files were deleted, but that the msg000N.txt file remains. Manually deleting the text file obviously resolves the issue. I've been trying for weeks now to come up with a repeatable way to invoke this behavior, but have failed utterly. However, we have seen this behavior about a dozen times. Anyone else experiencing this? Any suggestions? If it makes any difference our phones are Polycom SoundPoint 501s Thanks, Anthony ___ --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
[Asterisk-Users] Auto NOTIFY
Hey everyone, I wrote in last week about our Polycom phones rebooting. I had a nice theory with it being the PoE switch but that was thrown out the window today when phones even with a power supply rebooted. So my question now points back to Asterisk. Is there any feature on Asterisk that sends a NOTIFY signal to the phones that is automatically enabled? Or is it only manual? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto NOTIFY
Hey Doug, That's what I figured, but correct me if I am wrong. Isn't 1 will always set the phones to reboot on a NOTIFY command regardless of any changes in the configuration file? I thought 0 would means it requires both a notify request and a change in the configuration file. But you are right, I'm out of ideas. Seeing today one phone reboot with a power supply really threw me for a loop. Thanks, Kevin Douglas Garstang wrote: The following command on the Asterisk console will reboot a polycom phone: sip notify polycom-check-cfg ip-addr but in sip.conf, voIpProt.SIP.specialEvent.checkSync.alwaysReboot needs to be set to 1. otherwise... beats the heck out of me! -Original Message- From: Kevin Smith [mailto:[EMAIL PROTECTED] Sent: Friday, June 30, 2006 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Auto NOTIFY Hey everyone, I wrote in last week about our Polycom phones rebooting. I had a nice theory with it being the PoE switch but that was thrown out the window today when phones even with a power supply rebooted. So my question now points back to Asterisk. Is there any feature on Asterisk that sends a NOTIFY signal to the phones that is automatically enabled? Or is it only manual? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto NOTIFY
I upgraded from 1.6.2 to 1.6.6. After which, the problems started to happen. While it isn't a good thing, at least I'm not crazy and someone else is having the problem as well. ;). I also turned on the logger on asterisk with full debug information. Sure it's crazy, but maybe if one of the phones reboot this weekend I can see if there was something sent that verbose isn't showing me. Kevin Douglas Garstang wrote: Ohoh... Kevin, what version of SIP software are you running? One of my Polycom phones just rebooted itself for no apparent reason. -Original Message- From: Kevin Smith [mailto:[EMAIL PROTECTED] Sent: Friday, June 30, 2006 1:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Auto NOTIFY Hey Doug, That's what I figured, but correct me if I am wrong. Isn't 1 will always set the phones to reboot on a NOTIFY command regardless of any changes in the configuration file? I thought 0 would means it requires both a notify request and a change in the configuration file. But you are right, I'm out of ideas. Seeing today one phone reboot with a power supply really threw me for a loop. Thanks, Kevin Douglas Garstang wrote: The following command on the Asterisk console will reboot a polycom phone: sip notify polycom-check-cfg ip-addr but in sip.conf, voIpProt.SIP.specialEvent.checkSync.alwaysReboot needs to be set to 1. otherwise... beats the heck out of me! -Original Message- From: Kevin Smith [mailto:[EMAIL PROTECTED] Sent: Friday, June 30, 2006 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Auto NOTIFY Hey everyone, I wrote in last week about our Polycom phones rebooting. I had a nice theory with it being the PoE switch but that was thrown out the window today when phones even with a power supply rebooted. So my question now points back to Asterisk. Is there any feature on Asterisk that sends a NOTIFY signal to the phones that is automatically enabled? Or is it only manual? Thanks, Kevin ___ --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 ___ --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 601 question
Hey everyone, I know this isn't a direct Asterisk issue, but some of you may know this answer. I recently upgraded the SIP version to 1.6.6 on all of our phones in the office. Everything is working fine, except one aspect. The phones in the office reboot randomly for no apparent reason. I haven't changed anything in the configuration files since the upgrade. The only setting in the sip.conf file that I can think would cause this problem is voIpPort.SIP.specialEvent.checkSync.alwaysReboot=0 Which is to me is fine, I wouldn't want the phones to reboot unless I did change something in the configuration files. Any other thoughts as to what may have caused the phone to reboot? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 601 question
Hey Chris, That is interesting. The ones in the office are all connected using a PoE switch. One would hope that the transformer and support filtering/feedback circuitry would be able to filter or compensate for any power fluctuation the switch encounters. I will have to look into that, and see if there is an overload somewhere along the line. It is making me lean that way, because other phones (same settings) are using the AC adapters in another office. The ones on the adapter have not been having this problem, but they don't use the phone much so they may have never noticed if it did. Thanks for the idea Chris. Kevin Chris Mason (Lists) wrote: Kevin Smith wrote: Any other thoughts as to what may have caused the phone to reboot? the power supplies on these phones are very underrated and any power fluctuation will cause them to reboot. I get it when we are on generator and the A/C cuts in. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username/auth name mismatch
I'm not to familiar with Express Talk, but try removing the username=200 from your sip definition. From your lines menu it doesn't look like you are sending a username to asterisk. The SIP number is probably going reference to the sip context and since you are telling asterisk there is a userame to authenticate, it may not be getting one. Other then that, port-wise things should be correct since you are seeing the error. Kevin sasa wrote: Hi, I have a asterisk/voip newbie and I am sorry if my quetion is banal. I used in my private LAN, Express Talk on Windows XP and Asterisk latest version on Fedora Core 4 , with this configuration in Express Talk Lines menu: Setting for Line: Default Line Settings Full 'friendly' Display Name: port SIP Numeber: 200 Server: 10.0.0.112 Password: mypassword In menu Network: Local SIP Port to Listen on: 5070 Local RTP ports: 8000 My sip.conf: [200] type=friend callerid=port username=200 secret=mypassword host=dinamic context=internal My extensions.conf: [internal] exten = 200,1,Dial(SIP/200,20) ..but in Asterixk log file I have: Registration from 'sip:[EMAIL PROTECTED]' failed for '10.0.0.230 - Username/auth name mismatch and on Express Talk I have: Register attempt for sip:[EMAIL PROTECTED] failed 404 Not found ..where: 10.0.0.112 - asterisk ip address 10.0.0.230 -- express talk ip address ..where is my error ? thanks. Salvatore. ___ --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
[Asterisk-Users] Dial plan question
Hey everyone,
Hopefully this will be simple enough to answer. I have a menu setup like
below:
exten = 850,n,Set(MenuLoop=1)
exten = 850,n,Playback(mercury-prompts/welcome)
exten =
850,n(MainMenu),Background(mercury-prompts/MainMenu-if-you-know-the-ext)
exten = t,1,Gotoif,$[${MenuLoop}=1]?|850|100:|t|2 ;Loops the main
menu twice
exten = t,n,Goto(Mercury-Sales,852,1)
exten = 850,100(FirstLoop),Set(MenuLoop=2)
exten = 850,101(SecondLoop),Goto(Mercury-Network,850,MainMenu)
Basically we want the caller to be routed through the menu twice which
it does very well. I would like to make the code a little easier to
update by using 'n' priorities instead of numbers. The part I am having
trouble is from the 't' extension. I would like to have to say
|850|FirstLoop, but that gives me an error saying it needs a number. I
know if i am in the same extension I can just use the name and it works.
Is this possible, so far things that I have tried such as using
?context|exten|label hasn't worked.
Thanks,
Kevin
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Re: [Asterisk-Users] More Level QueueSystem
Hi Patrick, Let me see if I am following you here. When a caller calls in, obviously you want them to be in the first queue level based on your dial plan. Now, how do you want the caller to reach the next queue? Is the only way a caller going to go to the next queue via a transfer from the level 1 attendant? If so, I would make the dial plan like this: 123,1,Answer() 123,2,Queue(1stLevel,t) 124,1,Answer() 124,2,Queue(2ndLevel,t) 125,1,Answer() 125,2,Queue(3rdLevel,t) This provides a few different things that it looks like you are going for. One, it will allow separation of each queue level. So when the attendant in level 1 needs to transfer to a level 2, they just transfer to the new extension and the caller is moved to the new queue. Also, if say queue 1 is closed, this will prevent callers from gaining access to higher queue levels. Also you can add NoOP statments to record items, or an AGI script as well before the caller enters the queue so you know what happened. The return codes are as follows: 0 means that the queue is full, emtpy (no members present) or doesn't exist. -1 means that caller hung upbut if the call is bridged then it means either of the parties could have stopped the call. 1 I think means the caller entered the queue without a problem. I don't think that will be returned. At least that is how I understood everything. Kevin Patrick Bök wrote: Hi, I am trying to set up a dial plan und I have a few problems to realise some functions. The dial plan should look like this: 123,1,Answer() 123,2,Queue(1stlevel,t) 123,3,Queue(2ndlevel,t) 123,4,Queue(3rdlevel,t) 123,5,Hangup() If a member of the 1stlevel-Queue can answer the call it should be hanged up after finishing. If not, the current member answering the call should be able to transfer the caller to the 2ndlevel-Queue. And so on. How can I check whether it is transfered or hanged up? I do not know how to realise this workflow, the transfer, within the dial plan and I have not found any solution within the Wiki. The next problem I have got with the queue app is the value of the return code: 0 for not being answered -1 for hangup 1 for bridged (does bridge in this context mean the same as transfer???) Would be nice if you could help me about the transfer problem between the queues. Thanks a lot, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Allowing multiple exchanges
Hey Doug, Few things you can do. First off, are the numbers for incoming callers or for when you are making a call? One way that we do it because our numbers change a lot is I have a text file with all the numbers on it. Like below: [localtoolexchange] exten = _342, 1, Goto(whereever) etc.. and then I include them where you need them. Now this is for outgoing, for incoming you just would need to remove the _. Now if it is a range of numbers that you know you can do the following: exten = _[12347-9][2-6789]X, Goto(whereever) The first part will look for 1,2,3,4,7,8, and 9. The second 2,3,4,5,6,7,8,9, and finally X is 0-9. If you have them in a database, I would use the text file method. It is easy to write a script to build a new file and reload it into asterisk. But you also can write the second part of the script with a little more tinkering. Kevin Doug Crompton wrote: What is the best way to include a whole group of exchanges into a dial plan? I want to route local toll free by exchange (first three) and I will have a bunch. Can they be stored somewhere and compared as a group to that position in the dialplan? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring Polycom 501 IP phones via the console
Hi Stephen, I use the 601's but I don't think they are THAT much different that this information won't be helpful or get you in the right direction. What is your network setup like? Are you using NAT or does the phone have a public IP address? Also are you seeing any errors on the CLI of asterisk? I know you said your configurations are local, but are you using a bootserver (which can be local) to grab the files? Things I would check if you are using NAT (I think 2-5 need to be done in the web interface): 1. Make sure your SIP.conf file is configured to use NAT and give it a port to signal on, say 1 for example (which I will use below to, but change to better fit what you would like). 2. Assign the phone an internal address, add port pass thrus for UDP packets 1-100050 (I think should be enough) for that IP. 3. Assign RTP port range to start at 10001 4. Make sure you have a NAT address listed in the phone and you have the signaling port set to 1 and Media start port at 10001. 5. Also if you are using a DNS name for the server (such as server-1.whateva.com) I use TCPperferred for DNS lookups. If you are not using NAT, it pretty much should work out of the box provided it knows where the server is going. Of course, make sure the SIP username and password are correct. I personally used a bootserver and manually changed my configuration files and I got my 601's working in no time. Hopefully something in here will help. Kevin Stephen Bosch wrote: Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call-pickup function in Queue application
Hi Attilla, I'm not sure if there is something like that available or not, but I know there are some alternatives. You can set the time out limit to say 15 seconds, which for me is about 3-4 rings on the phone before it goes looking for the next agent. The other option you can manually remove the interface from the queue via the CLI by the following: remove queue member Interface from queue name However, I'm not sure if that will have an effect on the call...hopefully it will just send the caller looking for the next number. I haven't personally tried it. I know some phones like the Polycom 601 have a buddy watch option. As far as I know, and someone can step in and correct me if I am wrong, that will just show if the person is on the phone or not. I don't think you can pick up on the line. Kevin Attilla De Groot wrote: Hi All, I need a function that I believe isn't available in Asterisk, but I don't know if I'm correct about this. I have a queue and I want agents that are in that queue to have the ability to answer a call in the queue with calling an extention. For example, if I'm an agent and my colleague forgot to logout I could take the call when his phone is still ringing without walking to his desk or waiting for round robin. Can anyone tell me if this already is avalible ? Regards, Attilla ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring Polycom 501 IP phones via the console
Hi Stephen, Sorry if the e-mail is a bit choppy but I figured it would be best to cut/paste answers in. Now again, I am using the 601's so things may be a little different, but for the most part should be similar. No NAT. This is just one Polycom 501 that is dialing out through an Asterisk server with a TDM-400 card in it. I'm not using a bootserver; I figured that with one phone, I ought to be able to just do it locally on the phone. The impression I am getting is that Polycom really doesn't want people configuring the phones that way. The Admin guide contains slightly more than *no* information on how to do that. It just seems like I should be able to enter a few things on the on the phone console and have it working, then fine tune things for larger deployments later. I just want to see the thing work first. I wonder if you are looking at a different guide. The Administrator guide I have (in Section 2.2.2) has a whole list of advantages for using a bootserver. If you are going to use FTP, then you need to make sure the phone has the proper information to access, same with HTTP. Then you just need the proper files up on the location. True, for 1 phone it isn't needed, but I am managing about 20 phones (some in different states and soon more) so it is very handy to have. That's the trouble. So many places to configure! Yes, I know, it took me about two days to get things finally sorted out, but once you get there...you will be like DUH! (Only one line configured for the Polycom in sip.conf, like so: [general] context=default srvlookup=yes [polycom] type=friend secret=welcome qualify=500 ;qualify peer is no more than 500 ms away nat=no ;this phone is not natted host=dynamic;this device registers with us canreinvite=no ;Asterisk by default tries to redirect context=internal;the internal context controls what we can do Okay, above looks fine. Now here may be some confusion. The sip entry isn't for a line...it is just a registration for Asterisk. The 601 for example, one key (which you will see later) can handle 24 calls (which is its max), The 501 can handle 3. But this just verifies the phone has access to the server, the context it belongs to, etc, the number of lines it can use is based on the phone and the available channels on Asterisk. Address: [this is supposed to be the DNS or IP address of the SIP server] Port: 5060 DNS Lookup: UDP only [I set this to UDP only because the internal DNS server we're using here only does UDP] Register: Yes Address is the address of the SIP server. Port: 5060 which is default For DNS, if you can only use UDP that is fine., and of course you want the phone to register. Now I have to set up the lines, so I go back up a level and down into Line 1: ... where I see Display Name: [don't know what this is for] Display name, is caller ID basically. If you have support for caller ID name, that is what it is. I do fill it in, like for example my company's name is on my phone config, but I don't see any reason why you can't leave it blank. I was thinking ahead for if/when we do SS7 or something the name will show up. Address: [what goes here? SIP server address again?] This is a little confusing, but this is the number or extension. For example, a phone number. You also can dial Internet addresses so that is why it is called an address. I believe this is also used later... but for now, I would set this to your extension, even if it isn't used, it is there for when it is. Label: [and here?] One the phone, next to the line keys, this will be the label..such as Line 1, or My Phone, it will show up there. Type: Private [the other option is Shared] I leave it at Private Third Party Name: [and what's this?] According to Polycom, this field must match the registration address value of the other registration which makes up the bridge line...what did I do with it? I left it blank. Auth User ID: polycom [here's where I assumed I had to put the extension name] Yes, however, again I use our phone numbers both in address and here...why? Because it was much easier to code in my opinion. I think if you leave this blank, it will use the address, but I'm not sure, which is why I matched it. Since polycom is your name in SIP you will want that there. Auth Password: [here's where I put the password welcome] Yes Num Line Keys: [left this blank] Calls Per Line Key: [left this blank] Here is what I was talking about earlier. Num Line Keys, is how many keys for numbers. For example, if you set it to 2. On the right of the LCD screen you will see a graphic of a phone in spots 1 and 2 and your contacts (if any) would follow. For starters I would set both to 1. Now, if you change calls per line key to 2, then it is like you have call waiting. You will be on a call and you will hear a beep and see on the phone someone else is calling. After
[Asterisk-Users] Busy Signals
Hey everyone,
A few employees have noticed some problem here and there when trying to
make outgoing phone calls. After it happens, they try again, and are
able to call through.
The dial plan for outbound calling looks like below. Which I know they
are getting to the Congestion part (which explains the busy) but what I
can't seem to figure out is the cause for why they are getting sent there.
exten = s,1,SetCallerID(${ARG1})
exten = s,2,Wait(2)
exten = s,3,Dial(${TRUNK1}/${ARG2})
exten = s,4,Congestion(10)
exten = s,104,Congestion(10)
The log for a call looked like this
May 26 12:21:08 VERBOSE[6997] logger.c: -- Channel 0/4, span 1 got
hangup request
May 26 12:21:08 VERBOSE[16613] logger.c: -- Zap/4-1 is circuit-busy
May 26 12:21:08 VERBOSE[16613] logger.c: -- Hungup 'Zap/4-1'
May 26 12:21:08 VERBOSE[16613] logger.c: == Everyone is busy/congested
at this time (1:0/1/0)
My question is it asterisk having an issue with the PRI or is the PRI
really reporting the number is busy. I know one case like this I was
calling home, and which when I got through to them, they were not even
on the phone. Are there any tests that I can run on the T1 card in the
server to the PRI? Any suggestions would be helpful.
Kevin
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Re: [Asterisk-Users] call queue problems
Yes there is. QUEUE_MEMBER_LIST(queuename)
This should return you a list of comman-separated list of the members in
a queue. After that you would need to format it (if needed) so asterisk
can read it back to you. Of course then you can make some logic
decesions on whether you want to remove the memeber from the queue, etc.
Also you may find this page helpful for things you are looking for
http://www.voip-info.org/wiki/view/Asterisk+functions
Kevin
Dumpolid Exeplish wrote:
Thanks Kevin,
the tip worked like a charm. However, there are newer issues now! Is
there any way of knowing which users are looed in? sometimes, customer
support users forget to login B4 they shutdown their computers (we use
soft phones) and presistentmembers=yes is set in queues.conf so the
users are not logged off automatically . I have an extension on which
I dial to get the count of loged in users. Is there a way to find out
which extensions are currently logged in??
Thanks agai
On 4/24/06, *Kevin Smith* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
What I would suggest doing, since we have a similar setup (where
our 24
support contracts can enter a pin number to be routed to an on call
tech) is I use the function QUEUEAGENTCOUNT(queue name). Since you
said
that the calls should only be routed after the last support person
logs
out, just do a test to see if there is anyone logged in the queue, if
not, send them to the NOC.
example:
exten = s,1,gotoif,$[${QUEUEAGENTCOUNT(124)} 0]?YES:NO
exten = s,n(YES),queue(124) ;Since there are more then 0 people
in your
queue
exten = s,n(NO),queue(123) ; If there less then or equal to 0
You also can run other tests and use logic and's and or's to make the
tests more complex.
Hopefully this will help,
Kevin
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[Asterisk-Users] Polycom Delay
Hey everyone, Hopefully someone can point me in the right direction for this. Currently we have two offices, all using Polycom 601 Revsion E I think. All have the same configurations and firmware versions. The differences: Office A: public IP address. Office B: NAT (router has a static IP) Office A: Same state as the asterisk server (Michigan) Office B: Wisconsin Office A: T1 network to the colo where the asterisk server is located Office B: Wireless connection (2 tower hops I think) (our wireless connection, we are a small ISP) to our backbone to the colo Okay, so calls going to and from office A have no problems at all. Office B is having a bit of a delay (about 5 seconds before the CLI shows the call is even started). The odd part is, it only happens when they are making an outbound call. Incoming calls go directly to them without any problems. Both offices for external calls use our PRI we have installed and all interal are SIP. I think also internal calls are having the same problem, but that I haven't had a 100% sure answer if it is or isn't, but I know for sure the PRI calls are. My question is, does it sound like the phone is causing the problem, or the network being NAT, wireless connection, or both having more to do with the problem. While I know it isn't an answer you can say, hey this is the solution, I would like any input or experience that anyone has had with a problem like this. Thanks Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call queue problems
Hi,
What I would suggest doing, since we have a similar setup (where our 24
support contracts can enter a pin number to be routed to an on call
tech) is I use the function QUEUEAGENTCOUNT(queue name). Since you said
that the calls should only be routed after the last support person logs
out, just do a test to see if there is anyone logged in the queue, if
not, send them to the NOC.
example:
exten = s,1,gotoif,$[${QUEUEAGENTCOUNT(124)} 0]?YES:NO
exten = s,n(YES),queue(124) ;Since there are more then 0 people in your
queue
exten = s,n(NO),queue(123) ; If there less then or equal to 0
You also can run other tests and use logic and's and or's to make the
tests more complex.
Hopefully this will help,
Kevin
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[Asterisk-Users] MoH issue
Hey everyone,
Hopefully I can describe the problem well enough so bear with me.
There are 3 companies that are tied into our asterisk server. Company A
(us) uses the default settings for music on hold. Companies B and C
however, want something different. For them I have when a call comes
into their dial plan it sets the music on hold to their music and that
seems to work. However, here is the problem. Calling out, it still plays
the old on hold music.
Here is the situation, the 3 companies if they call each other us SIP
and don't even touch the PRI, only outgoing calls outside the companies
will do that. So I also would like if B called C, C's music on hold
would be the one heard.
Here is how I started the dialplan.
[Empire-Outbound]
exten = _.,1,Answer()
exten = _.,n,SetMusicOnHold(OrigMusic)
exten = _.,n,Wait(2)
exten = _.,n,Goto(Empire-Outbound2,${EXTEN},1)
[Empire-Outbound2]
include = A-DirectDial ;Direct Dial Context
include = Empire-VoiceMail ;Voicemail context
include = Empire-Wildcard ;Basic calling function
Does switch between contexts reset the moh? Or can I not change the moh
for SIP channels and only on Zap?
Thanks,
Kevin
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[Asterisk-Users] Packet Testing
Hi everyone, On the Polycom 601 phones we are using, the forward feature works very nicely for agents that are out on trips. I was wondering if there is a way to test to see if they have the forward option enabled. When it is enabled the call comes in and gets -- Got SIP response 302 Moved Temporarily response and then it uses the correct outbound macro to forward the call to the number specified. I am wondering if I am able to test for that SIP response or something in the SIP packet that I can grab to test. From what I read online, I didn't see much that would allow me to test for it, but I may have just missed it. Thanks, Kevin ___ --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 issues
Hi everyone, I have been having some problems lately with our PRI and Asterisk, or maybe it is just me. It happens once maybe twice a day, but when some of our customers are calling in, the phone just drops on them. I pulled the information below from the log from one that happened. I notice why it is happening, but I can't seem to figure out a way to stop it from happening. I also notice that it is saying I don't have a D channel defined. I am not sure why it is saying that either. Below are my zapata.conf files. If anyone has any suggestions/ideas it would be greatly appreciated. Thanks, Kevin /etc/asterisk/zapata.conf switchtype=national defaultzone=us context=default signalling=pri_cpe group=1 channel = 1-23 dchannel=24 callerid=asreceived /etc/zapata.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 DEBUG[15151] chan_zap.c: Exception on 19, channel 2 Mar 29 17:08:18 DEBUG[15148] chan_zap.c: Exception on 18, channel 1 Mar 29 17:08:18 DEBUG[15148] chan_zap.c: Got event Alarm(4) on channel 1 (index 0) Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: Alarm (4) on Primary D-channel of span 1 Mar 29 17:08:18 WARNING[24038] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 3: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 3 Mar 29 17:08:18 DEBUG[15151] chan_zap.c: Got event Alarm(4) on channel 2 (index 0) Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 4: Red Alarm Mar 29 17:08:18 WARNING[15151] chan_zap.c: Detected alarm on channel 2: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 4 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 5: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 5 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 6: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 6 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 7: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 7 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 8: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 8 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 9: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 9 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 10: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 10 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 11: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 11 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 12: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 12 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 13: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 13 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 14: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 14 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 15: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 15 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 16: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 16 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 17: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to
Re: [Asterisk-Users] PRI issues
Mike, here is the interrupts (sorry for the formatting) CPU0 CPU1 CPU2 CPU3 0:117 174878327 0 0IO-APIC-edge timer 1: 0928 0 0IO-APIC-edge i8042 8: 0 1 0 0IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 11: 0 0 0 0IO-APIC-edge cpqphp 12: 0 3525 0 0IO-APIC-edge i8042 14: 06264322 0 0IO-APIC-edge ide0 169: 0 704988 0 0 IO-APIC-level cciss0 177: 0 48262108 0 0 IO-APIC-level eth0 193: 0 699441691 0 0 IO-APIC-level wct4xxp NMI: 0 0 0 0 LOC: 174914618 174914617 174914616 174914615 ERR: 0 MIS: 0 Doug, As for how the connection is connected. Yes, it is a PRI from a provider. I switched the span to use the telco for the timing. So now my zaptel.conf looks like this: span=1,1,0,esf,b8zs. I realized that was still 0 and I never changed that after I sent the e-mail. I was using asterisk for testing before we hooked the PRI in and it was one of those overlooked items. Thanks, Kevin Michael Welter wrote: Post your 'cat /proc/interrupts' for us. Kevin Smith wrote: Hi everyone, I have been having some problems lately with our PRI and Asterisk, or maybe it is just me. It happens once maybe twice a day, but when some of our customers are calling in, the phone just drops on them. I pulled the information below from the log from one that happened. I notice why it is happening, but I can't seem to figure out a way to stop it from happening. I also notice that it is saying I don't have a D channel defined. I am not sure why it is saying that either. Below are my zapata.conf files. If anyone has any suggestions/ideas it would be greatly appreciated. Thanks, Kevin /etc/asterisk/zapata.conf switchtype=national defaultzone=us context=default signalling=pri_cpe group=1 channel = 1-23 dchannel=24 callerid=asreceived /etc/zapata.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Mar 29 17:08:18 DEBUG[15151] chan_zap.c: Exception on 19, channel 2 Mar 29 17:08:18 DEBUG[15148] chan_zap.c: Exception on 18, channel 1 Mar 29 17:08:18 DEBUG[15148] chan_zap.c: Got event Alarm(4) on channel 1 (index 0) Mar 29 17:08:18 NOTICE[24038] chan_zap.c: PRI got event: Alarm (4) on Primary D-channel of span 1 Mar 29 17:08:18 WARNING[24038] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 3: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 3 Mar 29 17:08:18 DEBUG[15151] chan_zap.c: Got event Alarm(4) on channel 2 (index 0) Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 4: Red Alarm Mar 29 17:08:18 WARNING[15151] chan_zap.c: Detected alarm on channel 2: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 4 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 5: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 5 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 6: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 6 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 7: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 7 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 8: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable to disable echo cancellation on channel 8 Mar 29 17:08:18 WARNING[24039] chan_zap.c: Detected alarm on channel 9: Red Alarm Mar 29 17:08:18 WARNING[24039] chan_zap.c: Unable
Re: [Asterisk-Users] Confused on Agents and Queues
Hi Matt, We have somewhat of a similar setup here in my office. We have multiple queues to which different agents are a member to anyone of them. Basically what I chose to do was make my own custom log in script. I reference to the voicemail box and use the ID and password to authenticate our users. However, the difference we use the same phone. But you could use both the mail box and the password to authenticate the user (VMauthenicate) and then use AddQueueMember with the caller ID they are calling from. If you need some help going in that direction, feel free to let me know. Kevin Matt wrote: Hi, I'm confused with agents and queues in Asterisk. If I use AddQueueMember() then show queues shows the agents that I have logged into the queue... however the agent ID has to be the extension the agent is sitting at ... kinda useless for stats tracking. If I use AgentCallbackLogin() then show queues shows no agents logged in, but it works and show agents shows the agent logged in. How can I have my agents log in with a unique ID for *THEM* and have the calls ring to whatever extension they are at? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 601 Message Center
As far as I can tell everything is pretty much the same. Below is the debug output for a particular phone I left a voicemail for. Maybe I am missing something that I am just not seeing. Otherwise I'm still not getting a count, but the other notifications are still working. Thanks again, Kevin Here is the phone.cfg section: msg msg.bypassInstantMessage=0 mwi msg.mwi.1.subscribe=[EMAIL PROTECTED] msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=6245* msg.mwi.2.subscribe= msg.mwi.2.callBackMode=disabled msg.mwi.2.callBack= msg.mwi.3.subscribe= msg.mwi.3.callBackMode=disabled msg.mwi.3.callBack= msg.mwi.4.subscribe= msg.mwi.4.callBackMode=disabled msg.mwi.4.callBack= msg.mwi.5.subscribe= msg.mwi.5.callBackMode=disabled msg.mwi.5.callBack= msg.mwi.6.subscribe= msg.mwi.6.callBackMode=disabled msg.mwi.6.callBack=/ /msg nat nat.ip= nat.signalPort= nat.mediaPortStart=/ Here is the sip.conf for *. The Mercury-Defaults, is just some simple rules for the sip that I applied to everyone. But the mailboxes needed to be different for obvious reasons. [9897943727](Mercury-Defaults) [EMAIL PROTECTED] --- -- SIP/9897943727-2689 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing VoiceMail(Zap/1-1, [EMAIL PROTECTED]) in new stack Destroying call '[EMAIL PROTECTED]' -- Playing '/var/spool/asterisk/voicemail/Mercury-Network-Emp/9897943727/unavail' (language 'en') 12 headers, 0 lines Reliably Transmitting (no NAT) to 64.7.177.102:5060: OPTIONS sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK0cf03d0b;rport From: asterisk sip:[EMAIL PROTECTED];tag=as18caf65c To: sip:[EMAIL PROTECTED];transport=udp Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 25 Mar 2006 19:30:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- voip-1*CLI -- SIP read from 64.7.177.102:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK0cf03d0b;rport From: asterisk sip:[EMAIL PROTECTED];tag=as18caf65c To: sip:[EMAIL PROTECTED];transport=udp;tag=88909039-CD314322 CSeq: 102 OPTIONS Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041 Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' -- Playing 'vm-intro' (language 'en') 12 headers, 0 lines Reliably Transmitting (no NAT) to 64.7.177.102:5060: OPTIONS sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK6f26102d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as57826c20 To: sip:[EMAIL PROTECTED];transport=udp Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 25 Mar 2006 19:30:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- voip-1*CLI -- SIP read from 64.7.177.102:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK6f26102d;rport From: asterisk sip:[EMAIL PROTECTED];tag=as57826c20 To: sip:[EMAIL PROTECTED];transport=udp;tag=32013F46-4AA2FB CSeq: 102 OPTIONS Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041 Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/Mercury-Network-Emp/9897943727/INBOX/msg0003 format: wav49, 0x8c77e68 -- x=1, open writing: /var/spool/asterisk/voicemail/Mercury-Network-Emp/9897943727/INBOX/msg0003 format: gsm, 0x8c5c408 -- x=2, open writing: /var/spool/asterisk/voicemail/Mercury-Network-Emp/9897943727/INBOX/msg0003 format: wav, 0x8ca0480 -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en') -- Saving message as is -- Playing 'vm-msgsaved' (language 'en') 12 headers, 3 lines Reliably Transmitting (no NAT) to 64.7.177.102:5060: NOTIFY sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK54a0c577;rport From: asterisk sip:[EMAIL PROTECTED];tag=as5fddfb64 To: sip:[EMAIL PROTECTED];transport=udp Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 4/0 (0/0) --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms voip-1*CLI -- SIP read from 64.7.177.102:5060: SIP/2.0 200 OK Via:
Re: [Asterisk-Users] Polycom 601 Message Center
Hey William, Yes, Mercury-Network-Emp is the context of my voicemail.conf, which is why in the sip it has the @Mercury-Network-Emp so it knows which context to apply it to. Any other ideas? Thanks, Kevin ___ --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 601 Message Center
While I know this is not a true asterisk problem, I figure someone where may know. When you click on Messages and it gives you the count of Urgent, New, etc. How can you make the phone gather that information? For example, my phone shows me there is an e-mail. It also sends an e-mail. Yet, when I click on message before I connect to the contact center, it doesn't have any counts. Here is what part of the phone configuration looks like. msg msg.bypassInstantMessage=0 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=6245* msg.mwi.2.subscribe= msg.mwi.2.callBackMode=disabled msg.mwi.2.callBack= msg.mwi.3.subscribe= msg.mwi.3.callBackMode=disabled msg.mwi.3.callBack= msg.mwi.4.subscribe= msg.mwi.4.callBackMode=disabled msg.mwi.4.callBack= msg.mwi.5.subscribe= msg.mwi.5.callBackMode=disabled msg.mwi.5.callBack= msg.mwi.6.subscribe= msg.mwi.6.callBackMode=disabled msg.mwi.6.callBack=/ /msg Is there anything wrong? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 601 Message Center
Yes, everything is on the same server. Everything works, message indication, e-mail, etc. But the count stays at zero. I would agree with you too Aaron, it was in fact working on my phone. I'm clueless as to what could be causing it. Kevin Aaron Daniel wrote: Just curious, but is your voicemail on the same server that the phone registers with? As long as your mwi is working, it should automatically receive a count of how many messages you have from asterisk. Aaron On Fri, 24 Mar 2006, Kevin Smith wrote: While I know this is not a true asterisk problem, I figure someone where may know. When you click on Messages and it gives you the count of Urgent, New, etc. How can you make the phone gather that information? For example, my phone shows me there is an e-mail. It also sends an e-mail. Yet, when I click on message before I connect to the contact center, it doesn't have any counts. Here is what part of the phone configuration looks like. msg msg.bypassInstantMessage=0 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=6245* msg.mwi.2.subscribe= msg.mwi.2.callBackMode=disabled msg.mwi.2.callBack= msg.mwi.3.subscribe= msg.mwi.3.callBackMode=disabled msg.mwi.3.callBack= msg.mwi.4.subscribe= msg.mwi.4.callBackMode=disabled msg.mwi.4.callBack= msg.mwi.5.subscribe= msg.mwi.5.callBackMode=disabled msg.mwi.5.callBack= msg.mwi.6.subscribe= msg.mwi.6.callBackMode=disabled msg.mwi.6.callBack=/ /msg Is there anything wrong? Thanks, Kevin ___ --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
[Asterisk-Users] Polycom hand/head set echo and Zapata config
Hey everyone, I have been trying to figure this out and I am just getting no where with it. The office is using Polycom IP 601 phones. Everything sounds great in terms of quality on both heads. However, users of the phone are having trouble with their headsets and handsets. Some users are hearing their voices back come back on the phone. If they adjust their volume lower, then it goes away, but then they can hardly hear the customer. I have tried using the noise suppression and cancellation features on the sip.conf for the phone but that hasn't worked either. Has anyone had this before and have you found a way to adjust the phone to cancel the echo? Or am I wrong and it isn't the phone but the zapata.conf for asterisk or the TDM card? Is there a way to boost the volume of the caller without adjusting the phones? For example the rxgain and txgain in the zapata.conf file. Any suggestions would be greatly helpful. Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom - directory dial
Do you mean, say number 444-555- calls in. You want to hit dial for
that number, from say the missed calls list, and have it on add a 9 in
front? If so just do this in extensions.conf
exten = _9NXXNXX,1,Dial(Zap/g1/${EXTEN} ;Takes calls with a 9
exten = _NXXNXX,1,Dial(Zap/g1/9${EXTEN}) ;Takes calls without a 9
Kevin
Bill Gibbs wrote:
This is not an Asterisk specific question but doesn’t anyone know if
you can automatically prepend a 9 on the call lists so clients can
return dial without having to repunch in the number? If you go to
directories now it just shows the number without a 9 (obviously).
Maybe on the Asterisk side??
Bill
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[Asterisk-Users] Agents and agent counts
Hey everyone,
I have noticed a few questions close to the issue I am having but I
haven't seen any that quite match the problem I am seeing.
I have 3 queues. Some members share one queue and some are completely
separate. Some members have a higher penalty then others. I am using
addqueuememeber and removequeuemember for the login and log out and I
verify members with their password for voicemail (that all seems to work
just fine). The problem I am having is that if a member is in a queue on
their own, everything works fine, a call can go into the queue. However,
if 2 members with different penalties are logged in on the same queue,
the test for the number of members in a queue fails. Below is the code
that is failing.
852,5,Set(Queue_Count_Switch=${IF(${QUEUEAGENTCOUNT(sales)}?7:100)})
;Checks to see if there are active agents
exten =
852,6,Goto(Mercury-Sales,852,${Queue_Count_Switch})
;Sends to closed if there are none
exten = 852,7,Queue(sales|tT|||)
Here is what the CLI shows for queue members (note: NUMBER1 and NUMBER2
represent phone numbers that are real. they are different however and
typed in correctly)
saleshas 0 calls (max unlimited) in 'leastrecent' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/NUMBER1queue with penalty 3 (dynamic) (Not in use) has taken
no calls yet
SIP/NUMBER2queue with penalty 2 (dynamic) (Not in use) has taken
no calls yet
No Callers
And here is the CLI output.
- Executing Answer(Zap/1-1, ) in new stack
-- Executing Wait(Zap/1-1, 2) in new stack
-- Executing Playback(Zap/1-1, mercury-prompts/Sales-welcome) in
new stack
-- Playing 'mercury-prompts/Sales-welcome' (language 'en')
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing Set(Zap/1-1, Queue_Count_Switch=100) in new stack
-- Executing Goto(Zap/1-1, Mercury-Sales|852|100) in new stack
-- Goto (Mercury-Sales,852,100)
-- Executing Wait(Zap/1-1, 2) in new stack
-- Executing Playback(Zap/1-1, mercury-prompts/Sales-afterhours)
in new stack
-- Playing 'mercury-prompts/Sales-afterhours' (language 'en')
-- Channel 0/1, span 1 got hangup request
== Spawn extension (Mercury-Sales, 852, 101) exited non-zeroexten =
-- Hungup 'Zap/1-1'
Now what really confuses me is that when only 1 member say, NUMBER1, is
in the sales queue, it works fine. And vice-versa, but as soon as the
other member is in, then it stops working. Now even if they are both at
the same penalty then it still it fails saying we are closed (which is
exten 852,100). I am at a loss as to what could be causing it. Anyone
have any ideas or see if something that may be going wrong? Does the IF
statement return true for anything but 0 and -1 or is it only 1?
Thanks,
Kevin
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[Asterisk-Users] Auto dial feature
Hey everyone, We have a special mail box for certain customers when we are out of the office. Basically they enter a pin number and if it is valid they leave a message and it notifies the on call techs. My question is regarding externnotify for the voice mail.conf. If I enabled that and set up a call file, will it do it for every voice mail box I have on the system? Is there a way I can limit it to just the one voice mail box on the system? If not, what would be the best way to send out the voice mail message that was recorded to our on call techs. I need it to attempt 3 times in two minute intervals. Any suggestions is greatly appreciated. Thanks, Kevin ___ --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] Sound issue
Hey everyone, I know this is a problem with mpg123, but it just started happening and I have no idea why. I haven't changed any of the audio format settings yet. Before tonight, I was able to call, listen to the queues, hear the music on hold, no problems. I added a new context to a dial plan, reloaded and now I get this error. Ouch ... error while writing audio data: : Broken pipe Then asterisk just crashes. I have so far tried to make clean make make install, that didn't work. Replaced my own configuration files with the samples. Still I get the same error. I also noticed that i had a lot of processes of mpg123 running. [EMAIL PROTECTED] asterisk]# pgrep mpg123 2354 2372 2390 2462 2763 2785 2805 2823 2843 2862 2883 2905 2927 2949 2966 2986 3005 8608 8656 9180 9708 I can't seem to get rid of them either. I even restarted the whole server, which asterisk is configured to auto start and they are back. Currently I am running asterisk 1.2.4. The only other thing that I changed today was add odbc support for mssql. But I don't see any correlation there. Yes, I am getting a db error, but that is because the server isn't ready to accept asterisk yet. Any suggestions? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound Issue
Hey Rich and everyone. I tried what you suggested, and it didn't work. I even recomplied everything, moved all of my configuration files out and remade the samples, so as far as I can tell everything is back to day 1. However, it is still pulling in the database information. This is really the only thing I could think of that is causing any problems. Here is a list from the CLI of errors, warnings, etc. [ Booting...Feb 25 23:57:08 NOTICE[18542]: cdr.c:1188 do_reload: CDR simple logging enabled. ..Feb 25 23:57:08 ERROR[18542]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server on . Check debug for more info. Feb 25 23:57:08 WARNING[18542]: res_config_mysql.c:450 load_module: MySQL RealTime: Couldn't establish connection. Check debug. Feb 25 23:57:08 NOTICE[18542]: config.c:863 ast_config_engine_register: Registered Config Engine mysql ..Feb 25 23:57:08 NOTICE[18542]: config.c:863 ast_config_engine_register: Registered Config Engine odbc .Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:265 load_odbc_config: Adding ENV var: INFORMIXSERVER=my_special_database Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:265 load_odbc_config: Adding ENV var: INFORMIXDIR=/opt/informix Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:294 load_odbc_config: registered database handle 'asterisk' dsn-[asterisk] Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:552 odbc_obj_connect: Connecting asterisk Feb 25 23:57:08 WARNING[18542]: res_odbc.c:563 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:597 load_module: res_odbc loaded. ...Warning, flexibel rate not heavily tested! .Feb 25 23:57:08 WARNING[18542]: pbx_dundi.c:4584 set_config: Unable to look up host 'voip-1.sgnwmi-1.mercury.net' Feb 25 23:57:08 WARNING[18542]: chan_mgcp.c:4213 reload_config: Unable to get our IP address, MGCP disabled ..Feb 25 23:57:08 WARNING[18542]: chan_skinny.c:3154 reload_config: Unable to get our IP address, Skinny disabled ..Feb 25 23:57:08 WARNING[18542]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/cdr_tds.so: undefined symbol: tds_free_connection Feb 25 23:57:08 WARNING[18542]: loader.c:554 load_modules: Loading module cdr_tds.so failed! Ouch ... error while writing audio data: : Broken pipe I'm not to familar with removing src files after I compile them (still a little new to unix), but is there a way I can remove the files for unixODBC and freeTDS? Otherwise, I'm lost as to what could be causing the problem. Thanks, Kevin ___ --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 IP601 Question
Hey everyone, I haven't seen an issue quite like mine, so I am hoping anyone who used the Polycom 601's may have an idea. We are going to be switching our office over to Asterisk. All the phones are going to be 601's, I am going to set up a boot server, but for now I am just going to test everything on one phone. My question is I have the phone registered in Asterisk (phone icon on the polycom is black), but I cannot make any calls. I tried to dial the extension shown in the extensions.conf file and I just get a busy signal. However, if I plug in an old budgetone 100 with the same settings, it works just fine. Any ideas? Also, when setting up a boot sever, the phone updates the log entries and there is a cfg file for the mac address of the phone, but during a reboot it cannot connect to the bootserver. Do I need to have the sip.ld, etc, files up there for it to work properly? Any suggestions would be greatly appreciated. Here is the sip.conf file [test] type=friend secret=blahpoly insecure=yes host=dynamic qualify=500 nat=no mailbox=testmailbox callerid=Yourname test conext=local disallow=all allow=ulaw progressinband=no here is the local section of the dial plan. exten = 850,1,Goto(Mercury-Network,850,1) exten = 888,1,VoiceMailMain(@Mercury-Network-Emp) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP601 Question
Hey guys, Thanks for the suggestions. I did find the problem. Looking in the sip debug, I was getting a 407 error, corrected that, then was getting a 404. Which lead me to look at my context and bam...typo, I had conext instead of context. Corrected that and all is well. Thanks again. Kevin C F wrote: What does the dialplan for the Polyocm 601 (the one the phone uses, not Asterisk) look like? You can see if it's a polycom or asterisk thing, by enabling sip debug, and watch what is coming in from the Polycom. if nothing is coming then it's the Polycom doing it. On 2/23/06, Kevin Smith [EMAIL PROTECTED] wrote: Hey everyone, I haven't seen an issue quite like mine, so I am hoping anyone who used the Polycom 601's may have an idea. We are going to be switching our office over to Asterisk. All the phones are going to be 601's, I am going to set up a boot server, but for now I am just going to test everything on one phone. My question is I have the phone registered in Asterisk (phone icon on the polycom is black), but I cannot make any calls. I tried to dial the extension shown in the extensions.conf file and I just get a busy signal. However, if I plug in an old budgetone 100 with the same settings, it works just fine. Any ideas? Also, when setting up a boot sever, the phone updates the log entries and there is a cfg file for the mac address of the phone, but during a reboot it cannot connect to the bootserver. Do I need to have the sip.ld, etc, files up there for it to work properly? Any suggestions would be greatly appreciated. Here is the sip.conf file [test] type=friend secret=blahpoly insecure=yes host=dynamic qualify=500 nat=no mailbox=testmailbox callerid=Yourname test conext=local disallow=all allow=ulaw progressinband=no here is the local section of the dial plan. exten = 850,1,Goto(Mercury-Network,850,1) exten = 888,1,VoiceMailMain(@Mercury-Network-Emp) ___ --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
[Asterisk-Users] Incoming/Outgoing call question
Hey everyone,
I have a more of an opinion question then a technical question. The
asterisk server I am setting up is going to host 3 different businesses.
Each business is in the same building, and on the same network. My
question is regarding calls coming in and going out. We are a small ISP
and have a lot of numbers that are forwarded to our phone system. The
other companies have about 3 to 5 numbers going into their offices. My
question is if there is a good way to test for which number and where to
send it to.
Right now my though process was something like this (keep in mind I
haven't wrote it):
[default]
include = Our-Numbers
include = Business1
include = Business2
[Out-Numbers]
exten = s,1,gotoif,$[${EXTEN}=Number1 |
${EXTEN}=Number2..${EXTEN}=NumberN]?Match:1|:
Is that the best way to test for the number that is being dialed? Or can
you recommend a better way. If anyone has done something similar could
you share how you did this type of a setup? I know I could manually put
in each one, but I think there probably is a better way. If I have to go
that route, then I probably will write a script to generate the file.
Thanks,
Kevin
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[Asterisk-Users] Agent counts
Hey everyone, I am having a little trouble getting this section of the dial plan configured. Does anyone know of a way I can get the number of agents that are currently logged into a queue? My goal is if no agent is logged in the queue, it gives customers the message we are closed depending on the queue they dial in to. Any suggestions would be great. Thanks, Kevin ___ --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
