Re: [asterisk-users] asterisk-users Digest, Vol 51, Issue 51
On Oct 16, 2008, at 2:36 AM, [EMAIL PROTECTED]
wrote:
I want to call an extension like 8 and invoke an external C
program upon
calling, pass an constant integer like 1 to the C program.
What I have done is:
/etc/extensions.conf:
exten = 8,1,system(/usr/local/src/parallel/fire 1)
exten = 8,n, Dial(SIP/8)
exten = 8,n,Hangup
the C program under /usr/local/src/parallel/fire will wait for the
input, if
it's 1 external LED light will be on, if it's 0 LED light will be off.
I have changed the file ownership and group since my asterisk user is
asterisk (with freepbx):
[EMAIL PROTECTED] parallel]# ls -l fire*
-rwxrwxrwx 1 asterisk asterisk 5882 Oct 16 09:18 fire
-rw-rw-rw- 1 asterisk asterisk 2793 Oct 15 22:25 fire.c
If I run the program separately everything is fine:
...
However if I call to 8 I can see it execute the system command
but it
doesn't output an integer 1 to my 'fire' program.
CLI:
-- Executing System(SIP/10-09a63138, /usr/local/src/parallel/fire
1) in
new stack
Any ideas on this or I shouldn't use System() at all?
I do something similar, e.g.
exten = xxx,1,Answer()
exten = xxx,2,System(/usr/local/bin/door 1000 ${EXTEN:3})
It works great for me using Asterisk 1.4.19. If your app relies on
some elements of the user environment it may not find that. Try just a
simple application that writes the input to a file, e.g. /tmp/output
and see if that works. Maybe the asterisk process can't open your
parallel port? I'm using a USB-based device for digital IO and that
works great.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
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Re: [asterisk-users] One Way Audio Problem
On Oct 13, 2008, at 9:29 AM, [EMAIL PROTECTED] wrote: IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code
On Oct 9, 2008, at 10:40 AM, [EMAIL PROTECTED]
wrote:
Hi I have searched the mailing lists and come across similar threads,
but no actual solution. I am trying to use a Cisco AS5300 as a
gateway for PSTNr. I have been able to configure it to take outbound
calls and send them to the PSTN just fine. Inbound calls however are
rejected by asterisk with 488 Not acceptable here code.
here are the details:
AS5300:
IOS (tm) 5300 Software (C5300-JS-M), Version 12.3(23), RELEASE
SOFTWARE (fc5)
Make sure the context defined the for the cisco exists and matches the
extensions. Try a catch all pattern: exten = _.*,1,Nop(${EXTEN})
I had some random issues and I ended up defining another entry in
SIP.conf for the cisco by by IP address, e.g.
[172.31.2.7]
type=friend
host=172.31.2.7
insecure=very
context=cisco
qualify=2000
dtmfmode=inband
That works for me.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
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Re: [asterisk-users] PoE switch recommendations?
On Oct 6, 2008, at 12:56 PM, [EMAIL PROTECTED] wrote: We've been EXTREMELY happy with the HP 5400ZL series chassis switch. Same here. We have 4 of them and they have worked very, very well. I have 25 polycom phones at present doing PoE from them and everything is working great. They are reasonably priced, come with a lifetime warranty and free software updates. (Unlike with Cisco!) Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_cepstral.so
Has anyone got res_cepstral.so to work with Asterisk 1.4.21.1? It appears to crash Asterisk on my box (kernel 2.6.26 gcc 4.1.2). Tech supports doesn't seem to have any ideas. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreeTDS Versions?
Does any have some good experience with the various freetds variants? Is 0.64 better or worse than 0.82? I know that to use 0.82 you have to use ODBC, since libtds.a is not long installed. Which is more stable? I plan on using it for CDR, realtime and func_odbc. I'm connecting to SQL Server. I've had a few crashes with 0.82, I think, and I haven't used 0.64. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opening Doors with Asterisk!?
On Aug 19, 2008, at 1:44 AM, [EMAIL PROTECTED] wrote: i read a few articles online about the possibility to setup a buzzer door system to PBX using asterisk! I took a somewhat unique approach, based on reading recent postings. We already had intercoms without door relays. So, I bought a Grandstream GXW-4008 (8-port FXS) which supports loop current disconnect (often called CPC). My intercoms need this, and I found out that our Linksys PAP2Ts cannot do this. (Only for incoming calls, which is rather useless here.) To open the relay, I bough a USB-IIRO-8 from ACCES I/O Products. It provides me with 8 relay closures (and 8 inputs, which I'm not currently using.) I wrote a very simple C app to close a relay based on command line parameters (relay number and duration.) I have an extension in Asterisk use the System dialplan command to open the door with our existing door latches through our HAI control system (which provides the delay to keep it open for 6 seconds or until it opens.) It's extremely fast and reliable. I run the phone and relay through the same Cat 6 cable. We are only using 3 doors for now, but I can go up to 8, of course, with this setup. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slightly Off Topic: Cisco Premisys Slimline
Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to when I go off hook), but no dial tone. My config is: controller T1 7/1 framing esf linecode b8zs cablelength short 133 ds0-group 0 timeslots 1 type fxs-loop-start ds0-group 1 timeslots 2 type fxs-loop-start ds0-group 2 timeslots 3 type fxs-loop-start ds0-group 3 timeslots 4 type fxs-loop-start ds0-group 4 timeslots 5 type fxs-loop-start ds0-group 5 timeslots 6 type fxs-loop-start ds0-group 6 timeslots 7 type fxs-loop-start ds0-group 7 timeslots 8 type fxs-loop-start ds0-group 8 timeslots 9 type fxs-loop-start ds0-group 9 timeslots 10 type fxs-loop-start ds0-group 10 timeslots 11 type fxs-loop-start ds0-group 11 timeslots 12 type fxs-loop-start The cisco then sends calls to Asterisk, and that part works great from a PRI. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production
On May 6, 2008, at 10:20 AM, [EMAIL PROTECTED] wrote: I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. I'm running 1.4.19 and it has been pretty stable. Anything before 1.4.19, however, I found was embarrassingly unstable. I'd often get several crashes within an hour. However, since moving to 19 things have been better. I don't run Queues, though, but I do run a custom derivative of Queues that fixed some bugs and greatly enhanced its usability for us. We do tens of thousands of calls per day (mostly inbound) running on under Debian, although I had to upgrade the kernel to 2.6.23.11 in order to get ztdummy to work on my HP DL380. CPU load remains rather low. We are all SIP, no zaptel. I used to run IAX2 between my three servers (one's a backup and for testing, the other handles desk phones and ATAs), but found IAX2 very, very unreliable. It would hang Asterisk, crash, etc. I just replaced it with SIP (and turned off the module) and those problems went away. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One Way Audio After Dial
I've encountered an odd situation with Asterisk 1.4.19 that I can't figure out. If I dial an extension via a Cisco AS5400 with the g option to come back, when I then Dial another extension after that, we don't get audio from the caller. There are no firewalls, no routers, no anything but a network switch between. The calls come in as SIP from the Cisco and terminate on a SIP soft client. I searched for something similar, but everything I found dealt with NATing and the like, which I don't do. Static IPs to static IPs. If I remove the first dial with the g, then everything works just fine. If I call a local SIP soft client, everything works fine (instead something via the Cisco.) If I set canreinvite=no for the Cisco everything works. It seems like the g option should disable canreinvite for that call, so why the difference? Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New generic sounds
On May 1, 2008, at 10:17 PM, [EMAIL PROTECTED] wrote: We're about to do another batch of sounds, and I see by my word count that we have some extra time left over. So, suggestions will be entertained for additional prompts in English, Spanish, or French. The only rules are: 1) the prompts have to be generic to Asterisk. No Welcome to so-and-so's business unless the business is fake and the prompt is funny. 2) The prompt may not be profane. Our professional speakers do have a sense of humor, but there are some things they just will not say. How about: your call is being transfered to your call is being transfered an (to go along with a) supervisor manager incorporated enter the four-digit extension of the person you are trying to reach Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy
On Apr 3, 2008, at 10:32 AM, [EMAIL PROTECTED] wrote: uname -a shows x86_64 and Centos 5.1, 2.6.18-53.1.14.el5 You can try zttest, although I'd bet it will hang. See what's going to the console (or use dmesg.) If it's a lot of rtc errors, then you'll likely need to upgrade your kernel to at least 2.6.23.11. That worked for me. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy
On Apr 3, 2008, at 12:45 PM, [EMAIL PROTECTED] wrote: You can try zttest, although I'd bet it will hang. See what's going to the console (or use dmesg.) If it's a lot of rtc errors, then you'll likely need to upgrade your kernel to at least 2.6.23.11. That worked for me. I'd be surprised if that is the solution - I have been using ztdummy with the RTC hook since 2.6.9 with no problems, and again on 2.6.18-53.1.6.el5 Unless it's a 64-bit issue - I've only ever used 32-bit. I'm using 32-bit as well, but it may depend on the hardware. I'm using HP DL-380s and Debian (both etch and sarge had the same problems.) I did nothing other than upgrade my kernel (and recompile zaptel, of course) and it just started to work. I used to get a lot of those missing interrupt errors to the console before, not now. If zttest hangs, I'd suspect the drivers are loaded. Otherwise, you'll get an error that it can't find a zap device (and playback then works.) Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 44, Issue 104
All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy fixed the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) Which previous versions have you tried? I'll also note that the OP needs to get Zaptel working under Xen, which is probably a different issue than your own. I've tried 2.6.8, 2.6.18-5, 2.6.19, 2.6.21.3 and perhaps more. These are the only ones I recall. I tuned in late and didn't see they wanted Xen support, but I figure others may find it helpful via google. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED] wrote: Sure some others on here may disagree, but I am also over on the trixbox forums, and have often seen talk about the 2.6.9 kernel having interrupt issues, and such that cause asterisk issues. One reason I think they moved forward into the CentOS 5.x stuff, so they got the 2.6.18 kernel, which I am told works much better, and doesn't have the issues the old kernel did. I've also found that I can't get ztdummy working on anything less than 2.6.23.11. Previous versions seem to have a broken RTC. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Mar 19, 2008, at 5:56 PM, [EMAIL PROTECTED] wrote: Anyone? Just a user? I'm just a user, although I also develop things for internal use. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardphone SIP phone costs
On Mar 19, 2008, at 12:16 PM, [EMAIL PROTECTED] wrote: I understand the maximizing pricing and branding aspect of phones but when you look at feature set it just doesn't make sense. And as far as purchasing the phone you can get it without a contract at the same price. When I starting thinking about it, can anyone else see a time when desk phones are replaced by smart phones? Why would a company pay for work cell phone and desk phone when one device could potentially do it all? I know there are issues that need to be considered like safety (911) for one. But can anyone else see where I'm coming from on this. We use Polycom hard phones and Linksys ATAs (now owned by Cisco) since both have good prices, good feature lists and are very configurable. I couldn't see paying $500 for a Cisco phone when I can get a Polycom 601 for $250 that does more than I really even need at. Our old PBX phones were fairly pricey, as I recall. As for why a company would purchase hard phones, several reasons. First, we are replacing many hard phones with computers. We have a custom application and have been moving folks main numbers to use the computer. We can make it ring externally and then they just put their headset on and hit an fkey to answer. The reason to not use a cell, in addition to potentially delaying an emergency response, is reliability. In any kind of emergency, they just don't work. And coverage and dropped calls are a problem, especially in office buildings. However, professionalism is, IMHO, the main reason. Cell phones sound terrible, generally have a huge delay (often with a related echo), they fade in and out, etc. I actively don't deal with companies where their sales people are on cell phones, and I have indeed actually to go with other vendors based on this. If you can't be professional enough to have an office with a real phone, why would I want to trust you''ll support anything you sell? In the grand scheme of things, phone are cheap. With SIP phones, employees can move their phone to another office if they move and just plug it in. Companies can also better monitor employees. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inband SIP DTMF
I've been searching to a solution to this for a while and can't
figure it out, perhaps someone has done something similar.
I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to
my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low
on my lightly loaded switched gigabit ethernet network. One Asterisk
uses Zaptel and a Digium card, and DTMF recognition works great.
However, the same software can't seem to recognize DTMF over SIP inband.
I've tried enabling DTMF debugging, which never prints anything even
with core set debug 10 and core set verbose 10. Any thoughts on
how to get this working? My dialplan snippet is:
exten = 6220,1,Answer()
exten = 6220,n,Playback(high)
exten = 6220,n,Read(TEST||7)
exten = 6220,n,SayDigits(${TEST})
It always says:
-- User entered nothing.
If I send rfc2833 messages from a soft phone, it always works, of
course. I've tried relaxdtmf=yes in sip.conf, as well. I added
dtmfmode=inband to the SIP peer for the Cisco. Nothing. Tones are
generated by phones connected to our PBX that we don't have a problem
with otherwise or even from a cell phone.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
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Re: [asterisk-users] Hardphone SIP phone costs
On Mar 19, 2008, at 2:48 PM, [EMAIL PROTECTED] wrote: My mobile does not sound terrible, does not have echo, does not fade in or out, and the last time I used it to call the emergency services, I got through straight away. I've not had a dropped call for a long time either (going through tunnels on the train, or over Dartmoor excepted) I've never heard a cell phone on the other end that I couldn't tell was a cell phone, even on a good day. They compress the audio so much it's rather obvious. That may vary by carrier, ATT and Verizon being the largest in the US are both pretty awful. A fun test is to call a landline from your cell in the same room and note now long the delay is. I find it long enough to interfere with conversations, people talking over each other (especially when both are on cells from different carriers.) None of the carriers really offer a phone that can do SIP, as far as I've seen. As soon as the iPhone software 2.0 is out, there will be one for that. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
On Mar 19, 2008, at 1:00 PM, [EMAIL PROTECTED] wrote: Am I expecting too much? Perhaps. I think the hardware on which we run Asterisk can be much more reliable than the software, which is often the case. We have a bunch of HP servers with RAID and have never lost anything. A HD may fail, but the RAID keeps it going until we pop a new drive in there. A server class PC with redundant power supplies and RAID is really quit inexpensive now. If you are running on a $1000 box, you can't expect the reliability of dedicated telco hardware. As for Asterisk, reliability has been a concern. Concurrency issues keep cropping up (read bugs.digium.com), especially with the SIP stack. This is particularly the case with buggy clients (soft phones, and under high volume of calls.) However, in fairness, writing heavily threaded code in C is very hard to get right. I think testing could surely be better, perhaps come code reviews and more guidelines for writing threaded code. We had an old hardware system and it wasn't without some issues. We needed to support around 30 call takers and another 50 hard phones. It took us a while in the 90s to get everything working acceptably. Our transition time with Asterisk has actually been shorter. Since we have a highly customized operation, going with a Avaya or Cisco solution would have cost in excess of $500K. With Asterisk, we spent maybe $50K on hardware (including a Cisco gateway, two Asterisk servers and some Polycom phones.) This cost is trivial compared to how much we pend on our yearly phone bill. The great benefit to Asterisk for us was that everything is open source software and thus we can customize it. We wrote a custom app that plugs into Asterisk that handles all of our custom business rules and provides far more capabilities than our old (and very expensive) hardware solution. Since we already had a custom developed desktop application, we could plug in a SIP stack and further customize things to be just what we wanted. I remember talking to a rep from a large reseller and listing our requirements, and he was amazed we could do all we were going on 90s technologies, since their new (and even more expensive) stuff couldn't without lots of consulting. We had just two developers over 6 months go from zero to a full call center solution. On the other hand, if I were to support a small office with 20 people and simple voice mail for mission-critical telecommunications, I'd likely get a hardware solution. They are reliable and not that expensive. Asterisk, for now, and in my opinion, is always going to require more interaction that other hardware solutions. But, it's cheaper and more flexible. You may not care about cheap and flexible, and if not, maybe it's not what you want. I've not tested products like CallWeaver or others. People claim some of these are more reliable, but Asterisk seems more popular. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
Check around on bugs.digium.com. You'll find a number of issues reported that sound similar. I'm hoping that 1.4.19 will fix a lot of stuff, since the release candidates seem much more stable to me. I couldn't keep Asterisk up for more than a few days before on 1.4.18. I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. 1.4.17, which you mentioned, is also very buggy. 1.4.18 fixed many issues. Norman Franke Answering Service for Directors, Inc. www.myasd.com On Mar 18, 2008, at 7:40 AM, [EMAIL PROTECTED] wrote: We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. ___ -- 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 1.4 reliability problems
I believe most of them will be in 1.4.19-rc3 (and in SVN), but I applied patches to 1.4.19-rc2 from: Patches from 11712 and 12098. Plus another one I reported as 12162. Norman Franke Answering Service for Directors, Inc. www.myasd.com On Mar 18, 2008, at 12:11 PM, [EMAIL PROTECTED] wrote: On Tue, 2008-03-18 at 11:05 -0400, Norman Franke wrote: I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. Mind sharing which patches you have applied? ___ -- 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 in the call center - how do you do it?
On Mar 5, 2008, at 5:46 PM, [EMAIL PROTECTED] wrote: If you are running a call centre (large or small) using Asterisk, I'd be interested to know how you log your agents in out: E.g. - Do you use AgentLogin (to force calls onto the agents, perhaps)? - Do you still use AgentCallbackLogin? - If you use AddQueueMember, are you - running it through the agent's phones (i.e. in the dialplan)? - through a manager interface some software (homebrew or otherwise)? - Do you allow agent hot-desking? - if so, how do you determine which agent is logged in at which desk at what time? - how do you deal with authentication, or don't you bother? It'd also be useful if you could tell me what version of Asterisk you're using. And, finally, a pure fishing expedition: - What kind of reporting (if any) do you currently get out of the Asterisk, and are you happy with it? We are a medium sided center, I'd guess, mostly inbound. We don't use the Queue app, since it seemed rather inadequate for us, so we rolled our own solution that does skills-based routing and various other enhanced features (all database driven.) Along with a custom client, we pass custom headers to handle client-server communication. Any agent can log into any workstation and things just work, and our app handles authentication of agents. (We also authenticate the workstations, but that's hard coded into the app.) As for reporting, again, a totally custom developed system that's an extension to what we were using with our old phone switch. On top of that, I've developed a number of web-based applications (using Apache Tapestry) to slice and dice our data for reporting (mostly graphically) that we use a lot. Since it's all quite specific to how we work and our custom solutions, it wouldn't help anyone, I'm sure. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage
On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED] wrote: On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote: I've had it with Dell server garbage.They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver.They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I change underwear. [end rant]. Can anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards? HP DL380 is my baby. Thanks, Steve Totaro Ditto. We've been using HPs for a while without problem. I'm currently using a DL380 (a recent quad processor one) and it screams. -Norman ___ -- 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] Asterisk Realtime unixODBC timeout?
How does one get asterisk to timeout realtime request via res_odbc to unixODBC? I've set timeouts as appropriate for freetds (which unixODBC is using.) However, it doesn't seem to work. It takes over 3 minutes to timeout a connection and queries never seem to timeout, so a channel waiting on a query never terminates. I did notice that res_odbc.c never sets a timeout on the query using something like this: SQLSetStmtAttr( hstmt, SQL_ATTR_QUERY_TIMEOUT, (SQLPOINTER)10, 0 ); Has anyone tested this? Is it just failing for FreeTDS connections? (I don't have anything else.) Norman Franke ASD, Inc. ___ -- 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] Background Noise Elimination
Greetings! We have a somewhat noisy background in our call center, and I'd like to reduce this. Obviously, we could plaster the walls with sound absorbing material, but is there anything we can do in software either using any algorithms for our open source-based SIP library or inside Asterisk itself? Related to this, anyone have a good source for good panels? We are using Plantronics noise canceling headsets, which don't really seem to work all that well. Our ancient system handled noise much better, but I suspect that was partly due to the Dialogic ADPCM algorithm used that just reduced the intelligibility of lower volume noises in general. We are using PCMU direct from the agent's mic to through Asterisk to PRIs, so we don't suffer from compression artifacts. The down side, is that you can make out even very quiet conversations in the background (like 3 agents to one side.) How have people handled this? I'm experimenting with a noise gate that will lower the volume when the agent isn't talking, but that won't help when the agent is talking. Norman Franke ASD, Inc. ___ --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 1.4 Fax
On Jan 1, 2008, at 1:00 PM, [EMAIL PROTECTED] wrote: The most recent versions of t38modem can apparently provide both a SIP and H.323 T.38 to class 1 FAX modem interface for HylaFAX. What it cannot provide is an audio FAX interface. The sipmodem code I am working on will integrate audio and T.38 FAX processing in a single SIP entity. Apparently, indeed. I've been unable to get it to send faxes via a Cisco gateway. (Receive is OK.) The other side always reports errors, so it may or may not work for you. Norman Franke ASD, Inc. ___ --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] Bulk Reverse Phone Lookup
Is anyone aware of a service where we can lookup phone numbers to determine a name and/or name + address available in bulk? We want to look up every number called to our call center, so it will be tens of thousands per day. Services that charge 3 to 5 cents per lookup will get way too expensive very quickly. Thus, I'm looking for a service that can either license a database or provide bulk lookups for maybe $300-$500/mo? Or even license a database for a few grand. Anyone know of something like this? -Norman ___ --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] Voicemail playback on iPhone
On Oct 24, 2007, at 12:25 PM, [EMAIL PROTECTED] wrote: This is semi-related, but I have a Tmobile MDA and I couldn't play the files either. The issue was not a codec issue, it was an email encoding issue. If I sent the message to an email account and it was then downloaded to my desktop via outlook and then forwarded on to my phone, I can listen to them. If I just send it direct to the phone, I see the attachment and it opens in media player, but it won't play. I don't know if you are having codec issues or email encoding issues, but it is a place to look. I have an iPhone and tried several things to get a message to play in an email and I gave up. I ended up mailing a link that then runs the file through a conversion CGI-like deal. Unfortunately, the iPhone also doesn't support many low bandwidth codecs. It does support AMR, but that's about it. I eventually got this working, but not with Asterisk. It's for our legacy voice mail system. -Norman Franke ASD, Inc. ___ --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] Anyone use the Linksys phones? (Zeeshan Zakaria)
Note that the newish SPA962 has 6 appearances and a color screen. I've noticed that the bright color screen does impress people when they first see it. PoE is also very nice and web provisioning was quite easy. I've yet to try a more automated provisioning method on it. I know that getting the polycom's to auto provision wasn't very straight forward. I do provision some the linksys PAP2Ts via HTTP and that works quite well, so I suspect the SPA's to be relatively similar. Norman Franke ASD, Inc. www.myasd.com On Sep 24, 2007, at 7:06 AM, [EMAIL PROTECTED] wrote: Linksys are great phones. I like them but there only problem is limited line appearances. I prefer Aastra over them because Aastra has more lines appearances. They both are good. If you are not planning to have more than 4 lines, then Linksys is a great phone. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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] Redundancy / Failover
Noah, Thanks for the input. I'm thinking the problem with the stop gracefully is that it would confuse the auto fail-over appliance, in that it would either detect the server is dead and hard switch the T1s or keep sending calls there which Asterisk would reject. I'm thinking a better method may be the fail-over switch coupled with some logic in the client and server, perhaps using SIP NOTIFY to inform clients they should disconnect when idle, and reconnect to the specified alternate server. Once everyone is off, then taking that box down and upgrading. Asterisk supports SIP NOTIFY, so that may be the most workable. -Norman Hi Norman - To add to what Edgar said, yes, use linux-ha. It works nicely in combination with DRBD. DRBD uses a dedicated network interface on each box with a crossover cable between the two. It does a block level copy of the entire filesystem, so you have two machines that are identical. The you use the linux-ha heartbeat to monitor the OS and asterisk. If anything goes wrong, it can fail over to the second machine. This is pretty easy to set up with Analog lines. With PRI's you'd need the fonebridge or the FSV-4PFS from http://www.failsafevoip.com Thanks, I wasn't aware of the FSV-4PFS box. Can one switch it remotely (e.g. over the network?) From what I understand, it has its own heartbeat-type monitoring of asterisk. If asterisk fails, it will automatically fail the PRI over to your backup machine. Can you manually force the failover? I think so, but I'm not positive. You can ask the failsafevoip people directly. I've exchanged emails with them before and they are knowledgeable and responsive. It would be nice to have a way to gracefully switch boxes, e.g. all new calls to the backup box, wait until all calls on the primary normally end, and then take server down for an upgrade. If you're using heartbeat, and it's directly monitoring the asterisk process, you should be able to issue a stop gracefully command. That will bring asterisk down when all the calls are complete. Then, heartbeat should fail over to the other machine. Of course, if someone is on a long call and you've already issued a stop gracefully command, your asterisk cluster won't accept any new calls until that long call is finished. - Noah ___ --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] Redundancy / Failover
On Jul 19, 2007, at 5:16 PM, [EMAIL PROTECTED] wrote: On 7/18/07, Norman Franke [EMAIL PROTECTED] wrote: I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. To add to what Edgar said, yes, use linux-ha. It works nicely in combination with DRBD. DRBD uses a dedicated network interface on each box with a crossover cable between the two. It does a block level copy of the entire filesystem, so you have two machines that are identical. The you use the linux-ha heartbeat to monitor the OS and asterisk. If anything goes wrong, it can fail over to the second machine. This is pretty easy to set up with Analog lines. With PRI's you'd need the fonebridge or the FSV-4PFS from http://www.failsafevoip.com Thanks, I wasn't aware of the FSV-4PFS box. Can one switch it remotely (e.g. over the network?) All of my dynamic data is stored in a database (using Asterisk RT to read queue and agent settings.) So, that eases part of the problem. It would be nice to have a way to gracefully switch boxes, e.g. all new calls to the backup box, wait until all calls on the primary normally end, and then take server down for an upgrade. It's impossible to tell what the Ranch Networks box does from their web site. Anyone using it? -Norman Franke www.myasd.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redundancy / Failover
I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. We'll be using queues (modified), which precludes some of the standard redundancy solutions, since the queue needs to know all the agents and calls to effectively decide what to do. (Correct me if I'm wrong as to distributing queues.) As far as I can tell, I need to be able to fail over to a backup server. What's the best way to go about this? Obviously one can just have the clients re-connect to the new server, but with 50 machines, that can be a major pain. Both could have the same network, and just swap network cables and re-login. One could log into both and the second wouldn't send calls to the agent unless it got calls from the main switch in the event of failure. I'm sure there other solutions. I've come across other products like Ranch Networks (their web site is rather uninformative, although it seems promising) and Redfone's foneBRIDGE (which seems it may help.) I'll be using Sangoma cards on the Asterisk box, if that matters. Does anyone have experience they'd like to share on effective ways to do this? -Norman ___ --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
