Re: [asterisk-users] New generic sounds
Philipp Kempgen wrote: Mojo with Horan Company, LLC schrieb: Eric Wieling wrote: The word Dialing... and Calling... As in Dialing 911, please wait... and as in Calling 911, please wait... oooh boy wouldn't I be frustrated if I heard that instead of a ring when I dialed 911? what else is it gonna tell me? Thank you for calling 911. All of our representatives are currently busy. Your estimated hold time is 2 hours and 15 minutes. Thank you for your patience. ... MOH simpsons You have selected 'Regicide.' If you know the name of the king or queen being assassinated, please press 1 now. /simpsons ___ -- 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] AEL includes?
voip*CLI ael reload Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown root token '#include' Asterisk 1.2.14. Old, I know but my boss won't spring for a spare box, and I don't want to upgrade our only production computer. Jay Rodrigo R Passos wrote: Jay, What error? Jay Moore wrote: How do I include a file (not a context) in AEL? #include filename returns an error. Thanks, Jay ___ -- 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
[asterisk-users] AEL includes?
How do I include a file (not a context) in AEL? #include filename returns an error. Thanks, Jay ___ -- 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] Recorded calls skipping
Steve Totaro wrote: Jay Moore wrote: Greetings, List. I'm having a problem where my recorded calls are skipping every 4-5 seconds are so. I can hear the caller (or callee) just fine and then a second or so of silence followed by the person talking again. I'm saving my calls as .gsm files and it's worked fine for the past 11 months. I make sure I remove the recorded files from my Asterisk box and put them onto our fileserver, so it's not an issue of disk space. No other settings have been changed, so I'm not sure why my calls aren't being recorded properly now. Any thoughts? Thanks in advance, Jay You do not mention call volumes or simultaneous calls being recorded. If you are pushing around 70 or so simultaneous calls then you probably have an I/O issue with your hard drive. Although, I received complaints from the phone users about audio chopping before the recording were affected. I assume you are using the monitor app? Thanks, Steve Totaro At absolute maximum, we're probably recording 7-8 simultaneous calls, but most of the time it's 1-2. It's a newer rig, so I'm more inclined to think it's software and not hardware. Unfortunately, it's an older version of Asterisk, but I've had zero problems until the skipping calls, and if it ain't broke, don't fix it, right? :) I restarted Asterisk and it seems to have solved the problem -- for now at least. I'm a rookie when it comes to Asterisk, any suggestions on what to do to if it happens again? Thanks, Jay ___ --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] IVR help, please
Doug Lytle wrote: Jay Moore wrote: Hi list. I'm new to IVRs and trying to set up one that toggles an auto-forward flag on or off for specific accounts. Why don't you post what you've currently written and we'll go from there? Doug Actually, after switching to AEL, I think I finally got it working properly. Thank you for your response, however. Jay ___ --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] IVR help, please
Hi list. I'm new to IVRs and trying to set up one that toggles an auto-forward flag on or off for specific accounts. I'd like to have my users dial an extension and then be prompted to enter the account number. (done) Next I'd like it to jump to the appropriate line in the dial plan that corresponds to the entered account number (if it is valid) and have it play back the current status based on a quick DB query (i.e. - Acct #1234 is currently 'on'). (done) Then I'd like it to prompt the user to Press 1 to turn forwarding on (or 2 for off), but this is where I get stuck. I can't seem to figure out how to do sub menus. I've Googled and checked the wiki, but I can't seem to find exactly what it is I need. Can anyone advise? Thanks in advance, Jay ___ --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] Recorded calls skipping
Once again, my initial message goes ignored or unreceived. Let's try this one more time, shall we? :) Jay Moore wrote: Greetings, List. I'm having a problem where my recorded calls are skipping every 4-5 seconds are so. I can hear the caller (or callee) just fine and then a second or so of silence followed by the person talking again. I'm saving my calls as .gsm files and it's worked fine for the past 11 months. I make sure I remove the recorded files from my Asterisk box and put them onto our fileserver, so it's not an issue of disk space. No other settings have been changed, so I'm not sure why my calls aren't being recorded properly now. Any thoughts? Thanks in advance, Jay ___ --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
[asterisk-users] Recorded calls skipping
Greetings, List. I'm having a problem where my recorded calls are skipping every 4-5 seconds are so. I can hear the caller (or callee) just fine and then a second or so of silence followed by the person talking again. I'm saving my calls as .gsm files and it's worked fine for the past 11 months. I make sure I remove the recorded files from my Asterisk box and put them onto our fileserver, so it's not an issue of disk space. No other settings have been changed, so I'm not sure why my calls aren't being recorded properly now. Any thoughts? Thanks in advance, Jay ___ --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] Call Center Scenario -- take 2
Not sure if my original message made it through. Going to try this again. :) --- Greetings, List. I would like to implement a procedure in my call center but am not sure the best way to implement it. I'm hoping I can describe it here and that I'll receive some feedback and/or suggestions on how to proceed. Here's my situation: My call center fields calls regarding internet access issues for local apartment complexes and businesses. Most of the time, we get a few calls here and there from new tenants unsure how to set up their connection. Every so often, however, there will be some sort of issue (ISP going down, router crashing, etc...) that will leave all users without internet access. When this happens, we get a flood of calls and the girls in my call center can quickly become overwhelmed. What I'd like to do is set up a system whereby incoming calls during a known outage are instead redirected to a recording explaining the issue and the option to have the caller leave a message (a la voicemail). All calls come down our T1 and we are able to identify the incoming account based on its DID. We would need to do this on a per-account basis. My girls would also need to have the ability to toggle on/off the redirection as well as record a message for the caller to hear -- at a moment's notice. Since my girls only field the calls and don't do any actual support (I do that), it'd be ideal if my VM indicator would also let me know if any callers left messages during a known outage. Again, this would be ideal, but most certainly not necessary. So, what say you list? Any suggestions on the most efficient way to do this? I am quite familiar with PHP and not adverse to writing a script to do this for me (I suspect I will have to anyway), but don't wish to reinvent the wheel if something like this already exists. Thanks in advance, Jay ___ --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 Center Scenario -- take 2
I figured as much. I'm not particularly versed in all things Asterisk, however. How would you recommend I go about doing this? I can check the voip wiki for the specifics if you could maybe toss some keywords out. Thanks :) Alex Balashov wrote: Sounds like you could benefit from some custom AGI programming, although it won't be very complicated. But I don't know that there's a business rules engine that does exactly what you're looking for right out of the box. On Thu, 6 Dec 2007, Jay Moore wrote: Not sure if my original message made it through. Going to try this again. :) --- Greetings, List. I would like to implement a procedure in my call center but am not sure the best way to implement it. I'm hoping I can describe it here and that I'll receive some feedback and/or suggestions on how to proceed. Here's my situation: My call center fields calls regarding internet access issues for local apartment complexes and businesses. Most of the time, we get a few calls here and there from new tenants unsure how to set up their connection. Every so often, however, there will be some sort of issue (ISP going down, router crashing, etc...) that will leave all users without internet access. When this happens, we get a flood of calls and the girls in my call center can quickly become overwhelmed. What I'd like to do is set up a system whereby incoming calls during a known outage are instead redirected to a recording explaining the issue and the option to have the caller leave a message (a la voicemail). All calls come down our T1 and we are able to identify the incoming account based on its DID. We would need to do this on a per-account basis. My girls would also need to have the ability to toggle on/off the redirection as well as record a message for the caller to hear -- at a moment's notice. Since my girls only field the calls and don't do any actual support (I do that), it'd be ideal if my VM indicator would also let me know if any callers left messages during a known outage. Again, this would be ideal, but most certainly not necessary. So, what say you list? Any suggestions on the most efficient way to do this? I am quite familiar with PHP and not adverse to writing a script to do this for me (I suspect I will have to anyway), but don't wish to reinvent the wheel if something like this already exists. Thanks in advance, Jay ___ --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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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
[asterisk-users] Call center scenario
Greetings, List. I would like to implement a procedure in my call center but am not sure the best way to implement it. I'm hoping I can describe it here and that I'll receive some feedback and/or suggestions on how to proceed. Here's my situation: My call center fields calls regarding internet access issues for local apartment complexes and businesses. Most of the time, we get a few calls here and there from new tenants unsure how to set up their connection. Every so often, however, there will be some sort of issue (ISP going down, router crashing, etc...) that will leave all users without internet access. When this happens, we get a flood of calls and the girls in my call center can quickly become overwhelmed. What I'd like to do is set up a system whereby incoming calls during a known outage are instead redirected to a recording explaining the issue and the option to have the caller leave a message (a la voicemail). All calls come down our T1 and we are able to identify the incoming account based on its DID. We would need to do this on a per-account basis. My girls would also need to have the ability to toggle on/off the redirection as well as record a message for the caller to hear -- at a moment's notice. Since my girls only field the calls and don't do any actual support (I do that), it'd be ideal if my VM indicator would also let me know if any callers left messages during a known outage. Again, this would be ideal, but most certainly not necessary. So, what say you list? Any suggestions on the most efficient way to do this? I am quite familiar with PHP and not adverse to writing a script to do this for me (I suspect I will have to anyway), but don't wish to reinvent the wheel if something like this already exists. Thanks in advance, Jay ___ --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] les.net losing DID's
This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. Except now, right, pal? Your site is down, you see. A shame, that. ___ --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] les.net losing DID's
Steve Totaro wrote: Jay Moore wrote: This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. Except now, right, pal? Your site is down, you see. A shame, that. Site came right up for me Shush. You're not supposed to side with the spammer. :P That said, it doesn't work for me on either my work or home computers (both on different ISPs). DNS issue, perhaps? Jay ___ --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] MoH mysteriously stopped working
Folks, I have somewhat of a serious issue here. My music on hold mysteriously stopped working. I have made no changes to my Asterisk box in the past month and up until an hour ago, MoH was working fine (as far as I know). CLI: -- Started music on hold, class 'default', on channel 'IAX2/lobby-2' -- Stopped music on hold on IAX2/lobby-2 voip*CLI moh reload voip*CLI 1 class reloaded. == Destroying musiconhold processes == Parsing '/etc/asterisk/musiconhold.conf': Found Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:422 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:504 monmp3thread: Unable to spawn mp3player musiconhold.conf: - [default] mode = quietmp3 directory = /var/lib/asterisk/mohmp3 random = yes I have had .gsm (and only .gsm) files in that directory since day one, and it's always played them just fine. The .gsm files are still in that directory, and transferring them to my computer and playing them works just fine. I have autoload set in modules.conf, and I can't figure out why my music on hold suddenly stopped working. Any thoughts? Thanks in advance, Jay ___ --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] MoH mysteriously stopped working
Peder, Unfortunately, this did not work. Any other thoughts? Jay Peder @ NetworkOblivion wrote: I've had MOH die probably 4-5 times in the last 2+ years and the only way to get it back is to stop * and restart it. Reloading MOH or just doing a regular reload doesn't work. I have to actually do a stop now and then asterisk to get it to work again. * restarts and MOH works fine. No clue why, but I have seen it on multiple versions of *. Jay Moore wrote: Folks, I have somewhat of a serious issue here. My music on hold mysteriously stopped working. I have made no changes to my Asterisk box in the past month and up until an hour ago, MoH was working fine (as far as I know). CLI: -- Started music on hold, class 'default', on channel 'IAX2/lobby-2' -- Stopped music on hold on IAX2/lobby-2 voip*CLI moh reload voip*CLI 1 class reloaded. == Destroying musiconhold processes == Parsing '/etc/asterisk/musiconhold.conf': Found Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:422 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:504 monmp3thread: Unable to spawn mp3player musiconhold.conf: - [default] mode = quietmp3 directory = /var/lib/asterisk/mohmp3 random = yes I have had .gsm (and only .gsm) files in that directory since day one, and it's always played them just fine. The .gsm files are still in that directory, and transferring them to my computer and playing them works just fine. I have autoload set in modules.conf, and I can't figure out why my music on hold suddenly stopped working. Any thoughts? Thanks in advance, Jay ___ --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] MoH mysteriously stopped working
Stephen Bosch wrote: I think it is trying to play mp3 files. Yes, this appears to be the case, I presume(see below). I am not sure why as I've always had .gsm files for playback. mode = quietmp3 Is this mode appropriate when you're using gsm audio files? I could not find any info to tell me otherwise. From what I have read, it simply tells how *loud* to play the file(s). Obviously, something has changed. Are you absolutely sure that: - you had only gsm files in that directory - your system isn't configured to use mp3 MOH files? I can say with 100% confidence that I only had .gsm files in that directory and always *only* had .gsm files. There's been a Windows virus going around again that deletes all the .mp3 files it finds. Is this system running Samba? No, but way to think outside the box there. I had no idea the virus even existed. What happens when you put an Asterisk ready mp3 file in that directory and restart Asterisk? Restarting Asterisk had no effect. Replacing the .gsm files with their .mp3 counterparts fixed the problem. I'm still stumped why Asterisk just decided to stop playing my .gsm files out of the blue. I'm back up and running 'normally' but I'm not satisfied with my 'fix'. Any other ideas would be much appreciated. Again, thanks in advance. Jay ___ --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] Recording calls after queues?
Greetings, List. With my current setup, I record all incoming calls to my queues. My problem is that once a call is transferred out of a queue, recording stops. How can I make it so recording continues even after a call is transferred? If you need me to post any dialplan or conf logic, please ask. Thanks, Jay ___ --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 stats
Jared Smith wrote: On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote: My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. It sounds like you've got quite the queue setup (although I don't quite see why your calls jump out and back into the moh queue). All the of queue statistics you need should be available with careful parsing of the queue log (usually located in /var/log/asterisk/queue_log). You can also trigger custom queue log events from the dialplan by calling the QueueLog() application. In your case, you might want to add a custom queue log entry every time the caller rejoins the moh queue, saying something to the effect of Caller XYZ has rejoined the moh queue for the 10th time or something like that. We had some issues with the announcement message not playing reliably. My fix was to just have them drop out and re-enter the queue. It doesn't seem to have any adverse effects, but if you have any alternative suggestions, I'm more than willing to try them. I've checked my queue log (38megs, yikes) and looked at the queuelog.txt info file for how to parse the lines, but I still have a question. For example, a snippet of my log looks like (line numbers mine): 1) 1185460404|1185460400.334916|queue-ring|NONE|ENTERQUEUE||732 2) 1185460420|1185460400.334916|queue-ring|NONE|EXITWITHTIMEOUT|1 3) 1185460427|1185460400.334916|queue-answer|NONE|ENTERQUEUE||732 4) 1185460448|1185460400.334916|queue-answer|NONE|EXITWITHTIMEOUT|1 5) 1185460454|1185460400.334916|queue-answer|NONE|ENTERQUEUE||732 6) 1185460456|1185460400.334916|queue-answer|SIP/agent3-0a5bc480|CONNECT|2 7) 1185460496|1185460400.334916|queue-answer|SIP/agent3-0a5bc480|COMPLETECALLER|2|40 Here's how I interpret this: 1) Call comes into my ring queue 2) Call exits ring queue due to timeout 3) Call enters answer (moh) queue 4) Call exits answer queue due to timeout 5) Call enters answer queue again 6) Agent 3 picks up the call out of the queue 7) Call ends; caller hangs up So here is my question: In this format: 1|2|3|4|5|6, 1 - ? 2 - ? 3 - queue in question? 4 - agent answering the queue? 5 - queue event? 6 - queue event info? Is that correct? What are options 1 and 2? Times of some sort I'm guessing, but I'm not entirely sure. Thanks for your help, Jay ___ --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 stats
Hopefully that helps clarify things! It does immensely. Thanks a ton! Jay ___ --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] Queue stats
Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller falls into our 'music on hold' queue, a message is played and the caller hears our music on hold while the phones are rung again. 3) After 30 seconds, if the caller is still in our 'moh' queue, they drop out of queue and immediately re-enter the 'moh' queue again until the call is answered or the caller hangs up. How can I find out how many calls are answered out of each queue during certain times (1st shift, 2nd shift, etc...)? Also, I'm curious how I can track how many times a call repeats the 'moh' queue. Thanks in advance, Jay ___ --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] Flash(), Centrex Lines, and 3 way calling
Greetings, List. I have my Asterisk box setup with 8 Centrex lines that were left over from our old PBX system. My boss is asking me to set up Asterisk so that he can flash hook and make an outgoing call on the same line to have a 3 way call. This is what he wants to do: 1) Incoming call on his Centrex line 2) Flash hook and dial a new number (goes out the same line) 3) Flash hook again and all 3 parties are connected. I've tried a bunch of dialplans using the Flash() command, but I can't seem to get it right. Can anyone advise on how I can go about doing this? Thanks in advance, Jay ___ --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] Blind xfer issue -- URGENT!
That's exactly what is happening. The *caller* is hitting #0 and
transferring the *agent* (my operator) to the new number. I don't have
the 'T' flag set [exten = s,n,Queue(queue-answer|t|||20)], so I was led
to assume that the caller could not transfer. Am I wrong?
Jay
Wes Baehr wrote:
It sounds more like the agents are making the transfers...
If a caller were to transfer a call (#0 1555-555-1212), they would be
transferring the AGENT to the that number, not themselves!
Either way, the caller SHOULD be disconnected after the transfer. (Or
perhaps leaked somewhere else into the dialplan they shouldn't be going,
which lets them dial out long-distance.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Thursday, June 21, 2007 6:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind xfer issue -- URGENT!
Use the dialplan show CLI command (show dialplan in 1.2) to show
you exactly what asterisk has picked up, and scan it for aforementioned
leaks.
Rizwan Hisham wrote:
Then i think u should use Atis's idea of using transfer_context
variable...you should set it inside your dialplan and it should be
the first thing you do in your dialplan.
Are you sure there is no leak in your dialplan, because asterisk cant
transfer your caller to an extension it cant find. There must be leak,
check if you are using any wrong extension patterns like _XXX. or
something like that.
On 6/19/07, *Jay Moore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
The way I have my dialplan set up, the callers shouldn't be able to
make
any outgoing calls.
Incoming calls come down my T1:
{zapata.conf}
; T1
group=1
context=incoming_t1
signalling=em_w
channel = 1-24
Which puts them into the 'incoming_t1' context:
{extensions.conf}
[incoming_t1]
#include callcenter/extension_ans.conf
include = answering-service
Which includes my callcenter answering service extensions conf file
and
includes the 'answering-service' context:
{callcenter/extension_ans.conf}
[answering-service]
; Catch all extensions
exten = _X.,1,Set(account=${EXTEN})
exten = _X.,n,AGI(get_cid.php)
exten = _X.,n,Set(CALLERID(all)=${cid}${account})
exten = _X.,n,Set(context=COM)
exten = _X.,n,Set(type=INC)
exten = _X.,n,Set(from=${account})
exten = _X.,n,Set(to=COM)
exten = _X.,n,AGI(create_filename.php)
exten = _X.,n,Set(MONITOR_FILENAME=${filename})
exten = _X.,n,Goto(queue-answer,s,1)
Which then parses all incoming calls (you can see the rest of the
dialplan in my previous message).
I'm not sure what I'm doing wrong. It seems to me I'm doing
everything
properly. Callers should not be able to transfer (no 'T' in the
Queue()
command), and they should not be able to dial any extension.
I'm completely lost here.
Jay
Rizwan Hisham wrote:
I dont know how to solve your transfer problem, but i have an
idea which
you
can use to overcome this abnormality.
You should restrict the callers with context which includes only
your local
office extensions.
I assume all your incoming calls fall in [default] context. what
you should
do is:
[default]
include= localext
exten= _X.,1,Noop(Incoming call received)
[localext]
*This context should include all your office extensions**
This way, caller can only transfer himself within your office
extensions.
I hope you get my point
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
www.axvoice.com http://www.axvoice.com
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[asterisk-users] Blind xfer issue -- URGENT!
Greetings, folks.
I'm having a problem with blind transfers. It seems that, despite not
having the T flag set, callers are able to use the blind transfer option.
Scenario is this:
- Asterisk 1.2.14
- Caller calls into our call center on one of our many phone numbers.
- Call gets placed into queue.
- Operator answers call.
- Caller is able to hit our blind xfer key sequence (#0) and dial any
number.
- Call is placed from our Asterisk box and connected to my operator
(*not* the caller).
I do NOT want this to happen. I *only* want our operators to be able to
transfer calls. I thought I had this set up properly (lowercase 't'
flag), but I apparently was incorrect. I cannot get the transfer to
stick with the caller (i.e. - the caller making free calls on my dime),
but I'm not ruling out that that too is possible.
I need some quick help here. Apparently someone has been making a lot
of long distance calls from our end and I need to immediately figure out
if it's an employee doing something they shouldn't be or a dialplan
issue with Asterisk.
Any help any of you can provide would be great. If you need more info,
please ask.
Thanks in advance,
Jay
---
Relevant dialplan snippets:
{Extensions.conf}
; Catch all extensions
exten = _X.,1,Set(account=${EXTEN})
exten = _X.,n,AGI(get_cid.php)
exten = _X.,n,Set(CALLERID(all)=${cid}${account})
exten = _X.,n,Set(context=COM)
exten = _X.,n,Set(type=INC)
exten = _X.,n,Set(from=${account})
exten = _X.,n,Set(to=COM)
exten = _X.,n,AGI(create_filename.php)
exten = _X.,n,Set(MONITOR_FILENAME=${filename})
exten = _X.,n,Goto(queue-answer,s,1)
[queue-answer]
; 1) Call rings for 15 sec
; 2) Call gets placed into normal queue
exten = s,1,Queue(queue-ring|rt|||15)
exten = s,2,Playback(_test_rec0)
exten = s,n,Queue(queue-answer|t|||20)
exten = s,n,Goto(queue-answer,s,2)
--
{queues.conf}
[queue-ring]
timeout = 15
strategy = rrmemory
leavewhenempty = yes
member = SIP/comcenter1
member = SIP/comcenter2
member = SIP/comcenter3
monitor-format=gsm
monitor-join=yes
context = queue-ring
[queue-answer]
timeout = 30
strategy = rrmemory
leavewhenempty = yes
member = SIP/comcenter1
member = SIP/comcenter2
member = SIP/comcenter3
context=queue-answer
monitor-format=gsm
monitor-join=yes
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Re: [asterisk-users] Blind xfer issue -- URGENT!
The way I have my dialplan set up, the callers shouldn't be able to make
any outgoing calls.
Incoming calls come down my T1:
{zapata.conf}
; T1
group=1
context=incoming_t1
signalling=em_w
channel = 1-24
Which puts them into the 'incoming_t1' context:
{extensions.conf}
[incoming_t1]
#include callcenter/extension_ans.conf
include = answering-service
Which includes my callcenter answering service extensions conf file and
includes the 'answering-service' context:
{callcenter/extension_ans.conf}
[answering-service]
; Catch all extensions
exten = _X.,1,Set(account=${EXTEN})
exten = _X.,n,AGI(get_cid.php)
exten = _X.,n,Set(CALLERID(all)=${cid}${account})
exten = _X.,n,Set(context=COM)
exten = _X.,n,Set(type=INC)
exten = _X.,n,Set(from=${account})
exten = _X.,n,Set(to=COM)
exten = _X.,n,AGI(create_filename.php)
exten = _X.,n,Set(MONITOR_FILENAME=${filename})
exten = _X.,n,Goto(queue-answer,s,1)
Which then parses all incoming calls (you can see the rest of the
dialplan in my previous message).
I'm not sure what I'm doing wrong. It seems to me I'm doing everything
properly. Callers should not be able to transfer (no 'T' in the Queue()
command), and they should not be able to dial any extension.
I'm completely lost here.
Jay
Rizwan Hisham wrote:
I dont know how to solve your transfer problem, but i have an idea which
you
can use to overcome this abnormality.
You should restrict the callers with context which includes only your local
office extensions.
I assume all your incoming calls fall in [default] context. what you should
do is:
[default]
include= localext
exten= _X.,1,Noop(Incoming call received)
[localext]
*This context should include all your office extensions**
This way, caller can only transfer himself within your office extensions.
I hope you get my point
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[asterisk-users] MoH WAY too loud
Hi folks! I'm having a problem where my music on hold is just blaring to my callers. I've tried several different formats (converting using mpg123 and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail. Every file plays way too loud. I did notice that sox has a -v flag for adjusting volume, but danged if I can find documentation online that'll tell me what parameter to pass. Any help any of you can provide would be much appreciated, thanks. Jay PS - What file type should I be using for MoH anyway? I know mp3 is out, but is wav or gsm preferred? Or is there another format I should consider? 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] MoH WAY too loud
Doug, Thanks for the reply. Immediately after hitting send I found exactly what I was looking for. Don't know why I didn't consider doing a 'man sox' earlier. I must be getting senile. ;) That said, I altered my initial .gsm files and made them 75% quieter (-v .25). I replaced my loud files with my newer, quieter files and reloaded res_musiconhold.so to no avail. I confirmed the new files *are* quieter, but Asterisk still plays them extremely loud. Do I need to reload a different module, or perhaps completely restart Asterisk to use these newer files? Thanks, Jay Doug Lytle wrote: Jay Moore wrote: Hi folks! I did notice that sox has a -v flag for adjusting volume, but danged if I can find documentation online that'll tell me what parameter to pass. Doing a 'man sox' does wonders: -v volume Change amplitude (floating point); less than 1.0 decreases, greater than 1.0 increases. May use a negative number to invert the phase of the audio data. It is interesting to note that we perceive volume logarithmically but this adjusts the amplitude linearly. So, this is how I increase the volume on my paging sox paging.gsm -v 4 /var/lib/asterisk/sounds/outx2.gsm 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] Re: asterisk-users Digest, Vol 33, Issue 12
Ok, I'll bite. This is the 4th message like this I've gotten today. I don't speak French but it looks like an autoresponder. If so, why is it replying back to the list, why not on every message sent, and why is it incrementing the issue number? Or am I missing something? Jay [EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] Device not registering after boot
Hi folks. I'm having a problem with a SIP-enabled device that doesn't seem to want to register after it reboots. If I program the device manually via its interface, it registers just fine. However, once I reboot it, it fails to register with Asterisk, despite all the proper information being stored in its memory. I contacted the manufacturer of the device and was told to try this: 1) if the asterix server accepts them, write an IP notify to reset the asterix server to 'SET' just after the device resets. I am not sure exactly what that means. Can anyone help? Thanks, Jay ___ --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] Voicemail mailbox number passed in connection?
I do it by calling my own extension. If it's me calling me, it passes
me direct to VoicemailMain. If it's someone else calling me, it rings
my phone as normal:
exten = 202,1,GotoIf($[${CALLERIDNUM} = 202] ? 5 : 2)
exten = 202,2,Dial(SIP/jay,10,tT)
exten = 202,3,VoiceMail([EMAIL PROTECTED]|u)
exten = 202,4,HangUp()
exten = 202,5,VoicemailMain([EMAIL PROTECTED])
HTH,
Jay
Lutgring, Sam wrote:
Does anyone know how to configure a SIP phone to pass the mailbox number
to the voicemail service when dialing? I would like to press the
message waiting lamp and be prompted for my password instead of mailbox
number. Can this be passed in the set-up call or based on caller-id?
Thanks
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Re: [asterisk-users] voip-info.org status update
You'll have to check the horse-wiki and pray it never goes down. Alternatively, you could get a Cisco horse. While it may cost more, at least you'll have a number you can call for tech support should your horse throw a shoe. The downside being, of course, if you want to modify your horse (e.g. - adding a rear spoiler, tinting its blinders, or adding a saddle with a piece of spinny plastic that makes it look like you're actually walking *backwards*) you'll have to use proprietary parts only purchasable from stables.cisco.com. :( Jay Rob Schall wrote: Of course you should buy a horse. But then there are the questions like. Do I get one like the Budweiser ones? Or just a mule (they can be helpful). What about color? Maybe a spotted one? Will my horse be able to talk to other horses using SIP? Or will it only be able to use IAX? Man, so many decisions if we have to go that way. Paul wrote: If a wiki site about automobiles crashes, should I buy a horse? shadowym wrote: I'm curious what you think that agenda might be? If it is to push the perception of Asterisk as a solid alternative to Traditional PBX's into the mainstream then I am guilty as charged! -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 6:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] voip-info.org status update On Thursday 15 March 2007 12:32 am, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. No UPS? Someone spilled their coffee into it? Something! Obviously you didn't read Google's research paper on drive failures. And aside from that, you're also obviously pushing an agenda with these inciteful comments. - ___ --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] Cannot xfer parked callers
Here's how it's currently working: 1) Call comes in 2) Operator parks call (700) 3) Operator picks up call on another phone (701) 4) Operator tries to transfer to a different phone (we use #0) but the transfer doesn't work. We can transfer initial callers all we want and it works fine. Once a call is parked, however, we can no longer transfer the caller. Any ideas? Thanks, Jay ___ --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] Cannot xfer parked callers
Bruce, I'm running 1.2.14. I am not willing to switch to 1.4 yet due to the stability issue. From what I read on the page you linked, I could not find what version had the supposed fix. I also can't seem to find a later 1.2 version of Asterisk (if one exists). Any suggestions? Thanks, Jay Bruce Reeves wrote: Jay, there is a bug in Mantis regarding this, a change was made to allow native bridging of parked calls. The change has been reverted in a more recent SVN version of 1.2. See http://bugs.digium.com/view.php?id=8804 On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote: Here's how it's currently working: 1) Call comes in 2) Operator parks call (700) 3) Operator picks up call on another phone (701) 4) Operator tries to transfer to a different phone (we use #0) but the transfer doesn't work. We can transfer initial callers all we want and it works fine. Once a call is parked, however, we can no longer transfer the caller. Any ideas? Thanks, Jay ___ --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] Adding 4 more POTS lines
Jim, I have 2 TDM400s in my * box (as well as a T1 card). I use all 8 ports, and aside from some minor echoing during peak periods, it's running smooth as ice. Jay Jim Freeze wrote: Hello I have a working * server with a TDM card and 4 FXO ports. We have 4 lines now and need to add 2 more lines (and possibly two more later). I'm wondering the best upgrade path for this situation. The simplest I can invision is adding another TDM400 card with 4 FXO ports, and use 2 now and the remaining 2 later. Are there success stories with using 2 TDM cards? Any info will be appreciated. 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] Cannot xfer parked callers
Ah, I misread. I'll probably do that and hopefully it'll fix the issue. Thanks! Jay Bruce Reeves wrote: Jay, The proble is in both 1.2.14 and 1.4 the fix mentioned in the bug was added to the svn revisions of both versions. If you are not wanting to switch from 1.2.14 to 1.2 svn the you can edit the features.c file and add the lines mentioned in the notes back to the file, then make and make install. On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote: Bruce, I'm running 1.2.14. I am not willing to switch to 1.4 yet due to the stability issue. From what I read on the page you linked, I could not find what version had the supposed fix. I also can't seem to find a later 1.2 version of Asterisk (if one exists). Any suggestions? Thanks, Jay Bruce Reeves wrote: Jay, there is a bug in Mantis regarding this, a change was made to allow native bridging of parked calls. The change has been reverted in a more recent SVN version of 1.2. See http://bugs.digium.com/view.php?id=8804 On 1/25/07, Jay Moore [EMAIL PROTECTED] wrote: Here's how it's currently working: 1) Call comes in 2) Operator parks call (700) 3) Operator picks up call on another phone (701) 4) Operator tries to transfer to a different phone (we use #0) but the transfer doesn't work. We can transfer initial callers all we want and it works fine. Once a call is parked, however, we can no longer transfer the caller. Any ideas? Thanks, Jay ___ --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
Re: [asterisk-users] 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
IMO, the 480i, by a LONG shot. The 480i is easier to use, looks nicer, has better audio quality, easier to read, and has a great speakerphone. The web-interface is also leagues better than the tripe the Polycom phones have. The only issue I have with the 480i, is that it's a little unintuitive in how to disable the X missed calls option. There's no option in the web-interface (I'm told one is coming, however), so you have to manually edit a .cfg file and send the info back to the phone. Other than that, I have had zero problems with my 480i's, and nothing but frustration with any of the Polycoms I have on hand. HTH, Jay Vikas wrote: I need to provide a 80 people office with VOIP. I want to commit to one vendor Polycom or Aastra. Price of the phones is not a factor in the decision. The quality of the phones is the factor. Some of the features that I am evaluating on are: (arranged in order of priority) 1. Sound quality 2. complete product line with conference phone and receptionist phone (not on Aastra) 3. cordless (not on 501/430) 4. backlit LCD (not on 501/430) 5. Inbuilt POE (not on 501) 6. speaker phone 7. 2 network ports. Which one will you choose ? Vikas ___ --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] Recording queue calls after an xfer?
I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay ___ --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] Recording queue calls after an xfer?
Yeah. 1.2.14. I heard bad things about 1.4 not being all that stable. I'm hesitant to move to it. Jay Julian Lyndon-Smith wrote: 1.2 series ? I think that 1.4 has that fixed. At least, that's what my team leaders are telling me ;) Julian. Jay Moore wrote: I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay ___ --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] Dropping Incompatible Voice Frame
Having a problem here that I can't seem to find a fix for. PSTN call comes in, operator answers, transfers call to a phone behind an IAXy. Caller hears no sound after being transferred. IAXy can hear caller, but not vice versa. Client reads: NOTICE[11342]: channel.c:1950 ast_read: Dropping incompatible voice frame on IAX2/jim-7 of format gsm since our native format has changed to ulaw Not sure how to proceed. Please advise. Thanks, Jay ___ --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] Zap calls
I have 8 Zap channels, 25-32, all of which have their own line.
My zapata.conf file looks similar to:
group=1
context=context_1
signalling=fxs_ks
channel = 25
group=2
context=context_2
signalling=fxs_ks
channel = 26
and so forth for all 8 lines, where each channel has its own group and
incoming context.
The first 4 channels are our primary trunk lines. If we have to make an
outgoing call on a trunk line, how can I have it pick the first
available line of the 4?
My first thought would be to have another group that includes the first
4 channels, and then use that group in the Dial() command like so:
group=9
context=whatever
signally=fxs_ks
channel = 25-28
and
Dial(Zap/g9/${EXTEN},60)
Can I repeat channels like that or will it cause Asterisk to choke? If
I can't do it that way, can someone suggest a way to do it?
Thanks in advance,
Jay
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[asterisk-users] MixMonitor write issue
Greetings, I am using MixMonitor to record my outgoing calls. It seems that MixMonitor will not write to a directory if it doesn't exist (ie - it doesn't create a new directory if needed). I have checked to ensure permissions are properly set, and if I manually create the directory, MixMonitor behaves normally. Rather than send several 'mkdir' commands each time I want to record a file, I was hoping someone knew an easier way to do this. It strikes me odd that directories are created when I record queue calls with 'monitor-join = yes', but can't do the same for outgoing calls. Any help would be much appreciated. Regards, Jay ___ --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] MixMonitor and Queues
Recompiled Asterisk after installing sox and it's still not merging the
two streams into a single recorded file. What am I doing wrong?
Jay
Jay Moore wrote:
Ed,
Thanks for the help. One more question, however. Everything is working
fine with the exception of sox joining the calls. I have sox installed
and monitor-join set to yes in both queues.conf and agents.conf
I installed sox after I installed Asterisk. Do I need to recompile
Asterisk for it to work with sox?
This is the last hurdle I need to overcome (I hope) before I can use my
Asterisk box in a live situation. Any help would be much appreciated.
Regards,
Jay
Ed Nuñez wrote:
In queues.conf you must have the following under the queues you want
to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
this will join the in and out files into one.
In agents.conf
recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls;this is the path where call
will be recorded.
That's all
If you want to change the file name place this in your extensions.conf
on a line prior to sending the call to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones
for 15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which
plays MoH until the call is picked up).
I've tinkered with MixMonitor and I have my queues set up, but I'm not
sure how to combine the two. Ideally, I'd like to only record once
the call comes out of queue (no point in recording hold music, unless
I want to hear people mumble about how lousy a company we are for
placing them on hold ;) )
On a semi-related note, is it possible to determine the extension that
pull the call out of queue before the call is bridged? The reason I
ask is that I'd like to put the receiving extension in the name of the
file that MixMonitor creates. If not, no biggie.
Recap:
Two queues. First rings for 15 seconds then drops into the second.
Second plays music on hold till the call is answered. I want to
record the call when it's pulled out of either queue using
MixMonitor. Bonus points if I can determine the answering extension
before MixMonitor starts (if possible).
Any help would be greatly appreciated.
Thanks,
Jay
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Re: [asterisk-users] MixMonitor and Queues
Well I'll be. That fixed it nicely. I was adding the .gsm extension
myself not realizing that Asterisk did it as well. Removing my addition
fixed the problem.
Thanks a ton!
Jay
Ex Vitorino wrote:
Jay,
I had a similar issue recently... My filename had more than one .
(dot / period)
and the sox version I was using failed to mix files in such conditions...
If that is your case, try:
- Using a filename with no .
- Upgrade sox to the latest version which fixes the funny behaviour
Cheers,
--
Ex Vito
On 12/28/06, Jay Moore [EMAIL PROTECTED] wrote:
Recompiled Asterisk after installing sox and it's still not merging the
two streams into a single recorded file. What am I doing wrong?
Jay
Jay Moore wrote:
Ed,
Thanks for the help. One more question, however. Everything is
working
fine with the exception of sox joining the calls. I have sox installed
and monitor-join set to yes in both queues.conf and agents.conf
I installed sox after I installed Asterisk. Do I need to recompile
Asterisk for it to work with sox?
This is the last hurdle I need to overcome (I hope) before I can use my
Asterisk box in a live situation. Any help would be much appreciated.
Regards,
Jay
Ed Nuñez wrote:
In queues.conf you must have the following under the queues you want
to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
this will join the in and out files into one.
In agents.conf
recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls;this is the path where call
will be recorded.
That's all
If you want to change the file name place this in your extensions.conf
on a line prior to sending the call to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay
Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones
for 15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which
plays MoH until the call is picked up).
I've tinkered with MixMonitor and I have my queues set up, but I'm not
sure how to combine the two. Ideally, I'd like to only record once
the call comes out of queue (no point in recording hold music, unless
I want to hear people mumble about how lousy a company we are for
placing them on hold ;) )
On a semi-related note, is it possible to determine the extension that
pull the call out of queue before the call is bridged? The reason I
ask is that I'd like to put the receiving extension in the name of the
file that MixMonitor creates. If not, no biggie.
Recap:
Two queues. First rings for 15 seconds then drops into the second.
Second plays music on hold till the call is answered. I want to
record the call when it's pulled out of either queue using
MixMonitor. Bonus points if I can determine the answering extension
before MixMonitor starts (if possible).
Any help would be greatly appreciated.
Thanks,
Jay
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Re: [asterisk-users] MixMonitor and Queues
Ed,
Thanks for the help. One more question, however. Everything is working
fine with the exception of sox joining the calls. I have sox installed
and monitor-join set to yes in both queues.conf and agents.conf
I installed sox after I installed Asterisk. Do I need to recompile
Asterisk for it to work with sox?
This is the last hurdle I need to overcome (I hope) before I can use my
Asterisk box in a live situation. Any help would be much appreciated.
Regards,
Jay
Ed Nuñez wrote:
In queues.conf you must have the following under the queues you want to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
thiswill join the in and out files into one.
In agents.conf
recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls ;this is the path where call will be
recorded.
That's all
If you want to change the file name place this in your extensions.conf on a
line prior to sending the call to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones for
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which
plays MoH until the call is picked up).
I've tinkered with MixMonitor and I have my queues set up, but I'm not
sure how to combine the two. Ideally, I'd like to only record once the
call comes out of queue (no point in recording hold music, unless I want
to hear people mumble about how lousy a company we are for placing them
on hold ;) )
On a semi-related note, is it possible to determine the extension that
pull the call out of queue before the call is bridged? The reason I ask
is that I'd like to put the receiving extension in the name of the file
that MixMonitor creates. If not, no biggie.
Recap:
Two queues. First rings for 15 seconds then drops into the second.
Second plays music on hold till the call is answered. I want to record
the call when it's pulled out of either queue using MixMonitor. Bonus
points if I can determine the answering extension before MixMonitor
starts (if possible).
Any help would be greatly appreciated.
Thanks,
Jay
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[asterisk-users] IAX calls not ringing
Greetings folks. I seem to be having a problem where calls made from an IAX device (three single-line phones attached to IAXys) do not play the ring tone when calling out. There's a dial tone when I pick up the phone, and the call goes through just fine, it just doesn't ring. All my SIP phones ring normally, however. Is there an option I need to enable that I'm missing? Thanks, Jay ___ --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] MixMonitor and Queues
Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my queues set up, but I'm not sure how to combine the two. Ideally, I'd like to only record once the call comes out of queue (no point in recording hold music, unless I want to hear people mumble about how lousy a company we are for placing them on hold ;) ) On a semi-related note, is it possible to determine the extension that pull the call out of queue before the call is bridged? The reason I ask is that I'd like to put the receiving extension in the name of the file that MixMonitor creates. If not, no biggie. Recap: Two queues. First rings for 15 seconds then drops into the second. Second plays music on hold till the call is answered. I want to record the call when it's pulled out of either queue using MixMonitor. Bonus points if I can determine the answering extension before MixMonitor starts (if possible). Any help would be greatly appreciated. Thanks, Jay ___ --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] PHPAGI example usage of input.php
Are you including the file extension? Jay Tom Vile wrote: I am trying to get the example input.php working from PHPAGI but it will not playback the letters that I put in because of this error: Nov 15 14:25:22 WARNING[18678] file.c: File /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any format But the file does exist and I see the entries for the key presses that I put in but it will not stream the file back to me using Cepstral. Asterisk 1.2.9 CentOS 4.2 Thanks, Tom Vile ___ --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] PHPAGI example usage of input.php
1) Try giving it an extension (say .gsm) and seeing if that works. Make sure you change both the file and your script. 2) Does the rest of the script work? If you run './test.php', do you get any errors? Jay Tom Vile wrote: There are no file extensions. It is just -rw-r--r-- 1 asterisk asterisk 32 Nov 15 12:52 swift_082da06a422be49e3a475925d9fc50e6 -rw-r--r-- 1 asterisk asterisk7 Nov 15 12:52 swift_6fc422233a40a75a1f028e11c3cd1140 -rw-r--r-- 1 asterisk asterisk 13 Nov 15 12:52 swift_80339585692b0188288da14748213dcc -rw-r--r-- 1 asterisk asterisk 11 Nov 15 12:54 swift_f87b365372c500c76e497087ac7e470a On 11/15/06, Jay Moore [EMAIL PROTECTED] wrote: Are you including the file extension? Jay Tom Vile wrote: I am trying to get the example input.php working from PHPAGI but it will not playback the letters that I put in because of this error: Nov 15 14:25:22 WARNING[18678] file.c: File /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any format But the file does exist and I see the entries for the key presses that I put in but it will not stream the file back to me using Cepstral. Asterisk 1.2.9 CentOS 4.2 Thanks, Tom Vile ___ --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] Quick Q...
Actually, while I was waiting for an answer, I figured out my problem. If I have any further questions, however, I'll be sure to post. Thanks! Jay Dovid B wrote: Post away. - Original Message - From: Jay Moore [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 09, 2006 6:58 PM Subject: [asterisk-users] Quick Q... Before I make any serious gaffes, is this an acceptable place to post PHPAGI questions as well? I can't seem to find a dedicated mailing list for it. If not, any suggestions? Thanks, Jay ___ --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] Quick Q...
Before I make any serious gaffes, is this an acceptable place to post PHPAGI questions as well? I can't seem to find a dedicated mailing list for it. If not, any suggestions? Thanks, Jay ___ --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] File structure question
Tzafrir Cohen wrote: On Thu, Aug 31, 2006 at 03:52:00PM -0500, Jay Moore wrote: I have a question on how I can better organize my .conf files. I have 3 different groups of people who use my VoIP service. Let's call them 'Office', 'Factory' and 'Public'. In my Asterisk directory, I have created three folders: 'office', 'factory' and 'public', inside each of which has a sip.conf and an extensions.conf file with appropriate account and extension information. Say, for example, I need to limit some users of the 'Public' group so they cannot make calls outside the building. Obviously I would create two separate contexts. One for users who can make calls outside the build, and one for users who cannot. I would then assign the appropriate context to each user. Right now, I have each appropriate context defined in the main extensions.conf. What I'd like to do is reduce the clutter in extensions.conf and move each context into the extensions.conf in the appropriate subfolder. How do I tell the main extensions.conf file to include the other extensions.conf files without putting an #include file in a context of its own? I hope what I've explained makes sense. If not, please ask questions and I'll try to answer. #include is a verbatim text include. if extensions.conf has: [main] exten = aaa,1,Line1 #include otherfile.conf exten = aaa,2,Line2 and othererfile.conf has: exten = aaa,2,OtherLine1 [other] exten = aaa,1,OtherLine2 You'll eventually get: main: aaa: 1. Line1 2. OtherLine1 other: aaa: 1. OtherLine2 2. Line2 Right, I guess I was wondering if it's possible to include a file without it being in a context. The goal I wanted to achieve was to have as few contexts in the main extensions.conf file as possible. Jay ___ --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] File structure question
Marco: Ah I see. There's a [general] context. I'm pretty new to this Asterisk stuff and I didn't realize there was a general context that you could do things like global includes. Thanks, I'll give it a shot when I'm back in the office on Tuesday. Peter: No need to be an ass about it, pal. Not all of us are as adept at this as you are. Jay Marco Mouta wrote: So the #include could be made just after the [general] section o extensions.conf? outside of any specific context, i think this was the question. On 9/4/06, Peter Bowyer [EMAIL PROTECTED] wrote: On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Right, I guess I was wondering if it's possible to include a file without it being in a context. The goal I wanted to achieve was to have as few contexts in the main extensions.conf file as possible. Did you try it? It would take... perhaps 30 seconds? A minute if you're a slow typist... Yes, you can do this. #include is a literal text include, as the last poster said. -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --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] File structure question
Peter Bowyer wrote: On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Marco: Ah I see. There's a [general] context. I'm pretty new to this Asterisk stuff and I didn't realize there was a general context that you could do things like global includes. Thanks, I'll give it a shot when I'm back in the office on Tuesday. Peter: No need to be an ass about it, pal. Not all of us are as adept at this as you are. You've still not got it. #include is a general text include - can be used anywhere. Well, perhaps it has to be at the start of a line. Contexts, not even the [general] section which isn't actually a context, has any relevance. It will insert the contents of the included file as though it was in the main file, wherever you put it. You could put the whole of the sip.conf file in an #include'd file. The whole of one context. One and a half contexts. 2 lines out of the [general] section. And so on. All of which, to repeat, could be experienced with a small investment of your time. It really does pay to experiment with the simple things, you find your learning curve is so much flatter than if you ask questions in a vacuum. Peter Perhaps if answering the simple things politely is too difficult for you, you'd be better off not answering at all. Someday, I hope, you'll find that 'simple' is a relative term. Jay ___ --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] File structure question
I have a question on how I can better organize my .conf files. I have 3 different groups of people who use my VoIP service. Let's call them 'Office', 'Factory' and 'Public'. In my Asterisk directory, I have created three folders: 'office', 'factory' and 'public', inside each of which has a sip.conf and an extensions.conf file with appropriate account and extension information. Say, for example, I need to limit some users of the 'Public' group so they cannot make calls outside the building. Obviously I would create two separate contexts. One for users who can make calls outside the build, and one for users who cannot. I would then assign the appropriate context to each user. Right now, I have each appropriate context defined in the main extensions.conf. What I'd like to do is reduce the clutter in extensions.conf and move each context into the extensions.conf in the appropriate subfolder. How do I tell the main extensions.conf file to include the other extensions.conf files without putting an #include file in a context of its own? I hope what I've explained makes sense. If not, please ask questions and I'll try to answer. Thanks much, Jay ___ --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
