Re: [asterisk-users] "Your call is not allowed. P U A M I"
Google search reveals a fairly dated reference to the same carrier switch tag message being delivered in a Skype for Business forum thread. https://social.microsoft.com/Forums/en-US/e25b3198-b5a0-4a43-9328-4a1aff5f6ed0/1800-number-dialing-issue?forum=communicationsservertelephony On Thu, Apr 20, 2017 at 3:43 PM, Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Thursday 20 April 2017 at 21:29:58, Steve Edwards wrote: > > > Not an Asterisk question, but... > > > > A bunch of our 8xx numbers started playing this recording when dialed. > Our > > provider (Inteliquent) says it's not them. > > Where are Inteliquent feeding the calls (assuming they connect instead of > playing that message) to? > > Are they a SIP trunk provider, supposedly passing calls to your PBX (in > which > case it's either them or your PBX, so there shouldn't be a lot of > discussion)? > > Does Inteliquent have any record of the calls being placed IN to the 8xx > numbers (if they do, this eliminates the possibility of message being > played > by the callER's service provider)? > > Does it make any difference which carrier you use to make the call? > > > Does anybody know who is playing it and what it means? > > I've certainly never heard (of) it. > > > Antony. > > -- > This email was created using 100% recycled electrons. > >Please reply to the > list; > please *don't* CC > me. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FollowMe / Asterisk 1.4 Question
On Fri, Mar 5, 2010 at 10:33 AM, Cory Andrews ipcbc...@gmail.com wrote: Is there a way to strip the normal features out of FollowMe (call acceptance, etc), and just set followme up to to blind transfer any call to an extension's associated cell number if it is not answered on the extension after 4 rings? Users want followme calls to wind up in their cellphone voicemail and I'm having some issues with ring/answer timing and Asterisk wants to pull the call back into the extension's Asterisk VM box Thanks in advance! -- Cory J Andrews 725 Powell Lane Lewiston, NY 14092 voice - 716.579.6331 email - ipcbc...@gmail.com -- Hey Cory - When I original put together app_followme, the intent was to avoid what you're looking for it to do. :-) I think you're probably better off going the dial plan route as another poster had recommended earlier this week. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having a heck of a time
On Wed, Oct 28, 2009 at 2:09 PM, Robert Grignon rgrig...@fleetone.com wrote: This has been a rollercoaster ride Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers) Where I stand right now, I have a PRI on the gateway and circuit is working I can make calls through the gateway Here is my problem: DAHDI_TEST is not returning anything and DAHDI_MONITOR doesn't work [r...@lin-vgw1 asterisk]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... --- Results after 0 passes --- Best: 0.000 -- Worst: 100.000 -- Average: 100.00, Difference: 100.00 Also if I establish a call and run dahdi_monitor it doesnt look quite like it is supposed to: [r...@lin-vgw1 asterisk]# dahdi_monitor 121 -vv Visual Audio Levels. Use chan_dahdi.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) I'm looking for some ideas here? I could be way off, but I think the fact that it says it's opening the psuedo interface implies that it doesn't see your Sangoma card. You might want to try and check with Sangoma support to see what they have to say about that. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Followme for multiple persons?
JD wrote:
I've got a challenge (or clarification request if I am mistaken) for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take calls
to help out the organization.
So, a formal queue is out. They don't want their home phones or cell
phones to blindly send them callers. They want to take calls when/if
they happen to be free to take a call at that particular moment. Plus,
the queue function can't handle the roll to voicemail problem that all
cell phones have. Plus, they won't have the discipline to log-in/log-out.
Fine, I thought, I'll just use the followme function in Asterisk 1.4. It
rings four numbers at once. It asks the friendly screening question,
allowing a volunteer to press 1 to take the call. Or, they hang up and
perhaps someone else will take it. (Or, if nobody does, it goes to
voicemail.)
Fine and dandy. Or so I thought. The problem is that followme is
designed to assume that it is only going to reach exactly one person.
So, if a phone answers and they press 2 to reject the call: bam,
asterisk stops trying the other three phone numbers.
I am currently trying to educate the volunteers to refrain from
pressing 2 but that is prone to problems. I'd rather that there not be
a reject function at all. Or, making it so that pressing 2 doesn't
really reject the call, it just hangs it up. I could change the audio,
and remap 2 to 9 and hope nobody presses it, but that seems like an
accident waiting to happen.
Does anyone have suggestions?
John
I think you can make the following code mod to have the next in dial plan
step not do anything. If someone has the time, this would probably be a decent
option to add to the application for future versions to make this behavior
optional via an application option parameter.
[r...@btwtechshowdemoc apps]# svn diff app_followme.c
Index: app_followme.c
===
--- app_followme.c (revision 188040)
+++ app_followme.c (working copy)
@@ -724,9 +724,9 @@
if
(!strcmp(tmpuser-yn, tpargs-nextindp)) {
if
(option_debug)
ast_log(LOG_DEBUG, Next in dial plan step requested.\n);
- *status = 1;
+ // *status = 1;
ast_frfree(f);
- return NULL;
+ // return NULL;
}
}
--
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
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Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call
nik600 wrote: Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR and ForkCDR but they don't do what i need, expecially because they fill-in lastdata and lastapp with ResetCDR So, what can i do? Is it better to do some customization to generate a CDR event on each dialplan step or is better to parse the logfile and extract the information needed? I'm using Asterisk 1.4.23.1 We generated a patch for a client probably about a year ago against the 1.4 branch that logged apps for each call, params, and exit status codes into a separate file. Like others have said, it generates a tremendous amount of data and probably does impact performance on very high load servers, but it was very useful to determine EXACTLY what happened with a given call. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Silk for Free
Dean Collins wrote: http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio_codec.html?tk=rss_news any thoughts? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). Cheaper to give away for hopes of proliferation what you've already implemented versus having someone else get theirs proliferated and popular first and then you are strapped with the cost of implementation of someone else's popular and free codec? -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] $20 Bounty
OCG Technical Support wrote: Perhaps if he threw in a paperclip and some tictacs people would respond... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: March 3, 2009 7:37 PM To: Asterisk Users List Subject: Re: [asterisk-users] $20 Bounty On Tue, 3 Mar 2009, Dean Collins wrote: I'll pay anyone a $20 bounty for someone to replicate the USA Asterisk Weather App on Tropo. Wow. $20. cricketcricketcricket :) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'd be willing to wager someone another $20 that he does have a taker on that bounty within the next 48 hours. For those of us that reside in countries where a single cup of coffee can run more than 20% of that complete bounty, we scoff at the opportunity. For others, their family may be able to live for half a week on that money. For better or for worse, it's a global economy now when it comes to this kind of stuff, and I have no doubt that Dean will get a taker on his offer. BJ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Managing the spiralling costs
Vikas wrote: I have been using the inbound 800 services from vitelity. Slowly the usage has been rising and in the month of Jan the bill was for $650. I am currently on a 1.9 cents a minute plan. Am I paying too much ? Some suggestions my team generated to reduce the toll free incoming call bill were: 1. When people call in on the 800 number take the local number they are calling from and then call them back from our unlimited outgoing account from broadvoice. 2. Find a vendor with a better rate. Any idea what we can do to better manage the 800 cost. Thanks for your time, Vikas As far as a better rate, that really depends where your callers are coming from. If they're calling in primarily from off-net areas, you'll find that $0.019/min as a blended rate is actually fairly competitive and your provider is probably losing money on your business! If they're calling in from on-net areas, you may find another provider willing to give you a better per min rate, but my experience has been that the monthly revenue commitment usually starts in around the $5k/month range before the rate comes down below what you're already paying. All that being said, before you go seeking a better rate, make sure you've done a good amount of due dilligence on a different provider to make sure that you're still going to receive the best level of service even though you may have a better rate. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: What do you guys think of this?
Alex Balashov wrote: RE Kushner List Account wrote: The question is, what are you actually paying for as a customer? To discriminate against bits just because they actually use what they are paying for is beyond me. At least a bandwidth cap is easier to understand. You get what you pay for. Speaking as a former sysadmin of an ISP, I would say that the issue is the following: 1) There is a high correlation of network-disrupting levels of traffic and BitTorrent; 2) Unlike some bursty downloads (like your CentOS ISO from an FTP server), BitTorrent traffic has the tendency to be sustained at higher levels for longer periods since the architecture presumes that everyone's a client and everyone's a server and fragments are always moving around. This is what tends to upset oversubscription assumptions that are otherwise functional, and are the only way that the ISP can possibly afford to give you the bandwidth for the price of consumer-grade broadband. I would tend to agree with you that discriminating against types of services and/or traffic through rate-limiting buckets and deep packet inspection is worse than a blanket bandwidth cap. However, you need to keep in mind the other side of the coin; were it not for Torrent, there would not be a need for traffic policing (in the overwhelming preponderance of cases) either way, so it's considered unfair to punish everyone with a bandwidth cap on everything when in reality, it's not a problem if their applications *occasionally* burst to very high levels of throughput. This is different from using up a lot of bandwidth continuously. My ISP doesn't care if I chug down a CentOS ISO tonight at close to my DSL line rate. But if I downloaded them all day long, all day, every day, there would be a problem, but the way to solve that problem isn't by taking away others' freedom to download a CentOS ISO when they feel like it in principle. Have you checked the FTP and/or HTTP mirrors lately for the DVD iso of CentOS? The only place I've been able to find them is on the Torrents themselves. -- Bird's The Word Technologies, Inc. http://www.btwtech.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 and rawplayer
Ade Vickers wrote: -Original Message- Hi Folks, I'm using the rawplayer program to provide my music-on-hold, and it works very well, with one small drawback: every time I reset Asterisk, for any reason, the MoH resets to the beginning of the list. Since MoH isn't used that often, it basically means the same track is played over over again... What I'd like to do is have rawplayer continuously playing away in the background, even if it's playing to itself only, so there's an excellent chance that any caller who will be given the pleasure of my MoH choices, will get a different tune to the one s/he heard last time... Any ideas? Asterisk is v1.4.18.1, running on Ubuntu 2.6.20-15.27-server. I'm still stuck with this, and would appreciate any thoughts... Thanks in advance! Ade. This would probably involve some kind of IPC named pipe or other inter process method of getting the data from pt A to pt B to work. While technically possible, it's not a trivial amount of work to get it going in the codebase. You might be better off with something like streaming MP3 over http or something else like that if you're looking for something with no code modifications. Are you really resetting Asterisk that much that this becomes a problem? If so, why? -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Sangoma Question
Jeremy Mann wrote: Any advise on troubleshooting this: Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: OOF alarm is OFF Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: RED alarm is OFF Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RAI alarm is OFF Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RED alarm is OFF It happens nightly, and I have to reset asterisk to “clear” it. Zap/Dahdi channels wont’ work until I do. The message is that you're losing frame/timing on your circuit. So there's two issues really. The first is that chan_zap/dahdi using Sangoma wanpipe drivers isn't recovering from a red alarm. The second is trying to find out why your provider is dropping frame on you every night. You'll want to consult Sangoma support for the first issue, and call your carrier on the second one. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] make func_realtime work like app_realtime (1.6)
Tilghman Lesher wrote: On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote: Wesley Haut wrote: Yell at me if you will, but I hate func_realtime - it's not very usable nor is it change-friendly (update your database and your dialplan completely breaks). I agree completely. As it stands, the REALTIME() function is nearly completely useless. If Asterisk had better string manipulation functionality, it would be /marginally/ better, though still not much good. A far better approach would be to allow you to specify the specific field you want to retrieve - the same way that you do for a write. /That/ would make the function many times more useful than it currently is. What if I made it work with the HASH() dialplan function, similar to how func_odbc works? Keys are column names, values are the associated field value. +1 to that Tilghman. HASH() wasn't there at the time that we originally put func_realtime together, but it would certainly make it more usable. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] FW: Google Alert - dean collins
Dean Collins wrote: Who is Chris Langford in Huntsville Alabama and is he seeking Digium’s permission in order to report the asterisk mailing lists out onto the internet http://asteriskbizrss.blogspot.com/ http://www.blogger.com/profile/04174728129647374395 What can be done to stop people doing this and making money out of selling ads on these crappy blogsites? Cheers, Dean *From:* Google Alerts [mailto:[EMAIL PROTECTED] *Sent:* Saturday, 27 September 2008 2:38 PM *To:* Dean Collins *Subject:* Google Alert - dean collins Google Blogs Alert for: *dean collins* [asterisk-biz] Philippines http://asteriskbizrss.blogspot.com/2008/09/asterisk-biz-philippines.html By Chris Langford(Chris Langford) I have a friend who is setting up a domestic outbound call center with about 20 agents initially looking for a simple low cost implementation. Email me with reference information and I’ll send you contact details. Regards,. *Dean Collins* *...* Asterisk Biz - http://asteriskbizrss.blogspot.com/ http://asteriskbizrss.blogspot.com/ This as-it-happens Google Alert is brought to you by Google. Remove http://www.google.com/alerts/remove?s=EAWStCkc7lwdmf82QEbYdZMhl=engl= this alert. Create http://www.google.com/alerts?hl=engl= another alert. Manage http://www.google.com/alerts/manage?hl=engl= your alerts. He works for Digium in their sales group. I've known Chris for some time. He's a good guy. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Follow Me app question
Mark Phillips wrote: Hi all, When one uses the follow-me logic to forward calls to lots of phone devices do subsequent calls get routed to the VM (or whatever the 10x is)? In other words, if I want my office, house and cell phones to ring whenever a call comes in and I answer it on my cell, does the next call that comes in when I'm on my cell get sent to VM or does it ring the follow-me group again? It's going to ring the follow-me group again. Follow-me isn't state aware of profiles already in use, although from what you're saying here, I could kind of understand why you'd want such a feature. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Is there a way to encrypt passwords stored in the realtime database?
Igor Hernandez wrote: I was thinking the same thing I believe Tzafrir just alluded to. If the passwords are encrypted in the DB with a public key then...asterisk needs to have the private key stored somewhere to be able to decrypt the values to authenticate the user. In this way there is nothing preventing whoever intrudes your boxes from getting that key and decrypting the values himself. I might be missing something though and if thats the case chime in, I'm interested in this issue. Regards, You are. md5secret simply stores the crypt hash. When it receives the password attempt, it too, is crypted using MD5 algorithm and then the two hashes are compared. Using MD5 crypt hash, there is no way to decrypt the hash. It's a brute force methodology to get the password back if you've lost it. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] OT Astricon/Digium Beach Ball Mailing
Alex Balashov wrote: randulo wrote: On Sat, Jul 19, 2008 at 7:09 PM, Alex Balashov [EMAIL PROTECTED] wrote: I've asked a number of others I know in real life who got the beach balls and all are reported as being fully functional. So this is not a case for the bug tracker? Perhaps a bounty... I've already submitted plastic patches to Beach Ball-rc5-pl5-beta trunk. Hopefully there's no white space involved in that patch. Sorry. Couldn't resist. :-) BJ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] reload stopping EVERYTHING on CLI and causing havoc.
Sherwood McGowan wrote: Mark Hamilton wrote: Hi, Yesterday I made a change in queues.conf and so tried doing a reload app_queue.so in the CLI. (Using 1.4.18). It didn’t seem to do anything, infact all action on CLI stopped. Then, I did a reload. Same thing. After that there was no other way.. because even stop now wouldn’t work, so I did a service asterisk restart And then asterisk kept giving the same thing on prompt “Died successfully” and all that it usually says when you issue a stop now, except it kept showing that on root prompt after doing a service asterisk restart. Did a killall asterisk, and finally it stopped. Then started asterisk service. It was fine. Did a full restart at night, and it was fine. NOW, I wanted to do a reload again today mid-day when in full use, and it still didn’t work, and ALL of the above happened again. -- How do I diagnose what’s causing this? Thanks, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've had this problem before, haven't debugged it. I definitely look forward to hearing what is said about this. Example from my recent experience, I wanted to restart the server and so did pbx0*CLI restart now But nothing happened...system continued to allow calls to take place. I've found that sometimes exiting and reconnecting to the CLI helps, but there have been a couple occasions where NOTHING would allow the server to restart save for a reboot. Even killall asterisk didn't kill the process Sherwood McGowan Sounds like you're waiting on a mutex locked somewhere within the queue infrastructure. You'll want to take a look at DEBUG_THREAD and follow on research behind that. http://lists.digium.com/pipermail/asterisk-dev/2003-November/002384.html -- Bird's The Word Technologies, Inc. http://www.btwtech.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] reload stopping EVERYTHING on CLI and causing havoc.
BJ Weschke wrote: Sherwood McGowan wrote: Mark Hamilton wrote: Hi, Yesterday I made a change in queues.conf and so tried doing a reload app_queue.so in the CLI. (Using 1.4.18). It didn’t seem to do anything, infact all action on CLI stopped. Then, I did a reload. Same thing. After that there was no other way.. because even stop now wouldn’t work, so I did a service asterisk restart And then asterisk kept giving the same thing on prompt “Died successfully” and all that it usually says when you issue a stop now, except it kept showing that on root prompt after doing a service asterisk restart. Did a killall asterisk, and finally it stopped. Then started asterisk service. It was fine. Did a full restart at night, and it was fine. NOW, I wanted to do a reload again today mid-day when in full use, and it still didn’t work, and ALL of the above happened again. -- How do I diagnose what’s causing this? Thanks, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've had this problem before, haven't debugged it. I definitely look forward to hearing what is said about this. Example from my recent experience, I wanted to restart the server and so did pbx0*CLI restart now But nothing happened...system continued to allow calls to take place. I've found that sometimes exiting and reconnecting to the CLI helps, but there have been a couple occasions where NOTHING would allow the server to restart save for a reboot. Even killall asterisk didn't kill the process Sherwood McGowan Sounds like you're waiting on a mutex locked somewhere within the queue infrastructure. You'll want to take a look at DEBUG_THREAD and follow on research behind that. http://lists.digium.com/pipermail/asterisk-dev/2003-November/002384.html Sorry about that. Gave you some 'dated' info there in that link. Try this instead maybe for 1.4... Tips for Asterisk 1.4 Two things you need to make sure you enable when you compile asterisk: DEBUG_THREADS DONT_OPTIMIZE These can be toggled under the Compiler Flags options when you do a make menuselect. Make sure that you have them turned on (I.E. an Asterisk next to them). Then, follow the guidelines for obtaining a backtrace with GDB. These can be found in the file asterisk/doc/backtrace.txt. I would suggest that you issue the following, to make your life easy; script backtrace.txt - This will start a typescript session, which captures the Input / Output from the session into a file called backtrace.txt Once you get into gdb, then issue: set pagination off to turn of paging. Then, get your backtrace, exit out of GDB and come find someone on #asterisk-dev on irc.freenode.org to take a look at your backtrace or the asterisk-dev mailing list. Backtracing a core dump file in /tmp 1. start Asterisk with safe_asterisk 2. enter gdb asterisk core. 3. enter bt while in gdb (or do a bt full) 4. enter thread apply all bt Naturally you'll need to have gdb installed on your system -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] DUDE!!!!! was RE: Dialplan Visualization(Extensions.conf or Dialplan Show)
Steve Totaro wrote: On Fri, Apr 18, 2008 at 11:09 AM, John Signorello [EMAIL PROTECTED] wrote: excuse me... But did you not just post [asterisk-users] OT Nice IBM 1U Server Gets Along w/Old and New Digium Boards Cheap X305 $199 Did you not provide a link to a COMMERICAL entity? Wasn't your a post a unsolicited post, that is, not in response to a question??? There seems to be two standards here. The fact that you do not work for them is immaterial. If your argument is no commercial reference at all, then how do you explain your post? You may have a point although I was more doing a favor or looking out rather than trying to push my wares on someone where they did not fit. I guess I would say it is different because of intent and presentation. I thought about posting on the biz list but then thought it would better serve many Asterisk users. I certainly was clear in the title that it was a server that cost $199 even if someone did not know that OT was short for Off Topic If anyone has any real objection to my occasional postings of this nature on the users list, then I will certainly keep them to myself. It takes more effort to post something that may be helpful than it does to remain silent. Steve, -1 to not posting stuff like that anymore. I think the guys in the office here have already picked up a couple of these machines here collectively to play. They are a very good deal. Is the post a double standard? Possibly, but it is what it is, and I personally prefer to see it rather than not see it. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Best Click-to-call client
equis software wrote: Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? I did a spiel on this at Astricon last year. The slide deck is somewhere around for those interested, but now we also have some code to show for it. :-) Take a look at this developer branch at http://www.asterisk.org/node/48440 and then we've put some pieces together for the Java side of things using Ignite's Realtime API for messaging. http://svn.btwtech.com/svnview/coolvocals/trunk/cti-server/click2call/ http://svn.btwtech.com/svnview/coolvocals/trunk/cti-client/click2call/ Basically the idea here is that there's a servlet that honors requests into it (think AJAX Remote calls from the browser) and then turns around and puts that request into a jabber message that goes to a centralized Servlet that can proxy requests across multiple servers (scalability/LCR/etc) and that in turn launches an Originate call in to the AMI of the machine that was decided would receive the request. Once that hand off is done, the proxy machine that received and directed the original request is now out of the middle of things and jabber messages are sent directly back to the client to signal call progress of the click to call. Is it a shrink wrapped and ready to go package that's completely documented and involves no technical knowledge whatsoever for implementation? U.. no, but that might happen in the relatively near future. :-) What it IS though is solid working code (yes, it has been fully unit tested out and is functional) contributed back to the community so we can all start to make something with it if we so choose. If there's enough interest, I'd certainly entertain opening up a blog site and open up the branch of the Java code for community contributions as well in addition to doing a more detailed tutorial on usage of the code at the upcoming Astricon this year. BJ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] question about queue
With regard to (1), yes, very good point there and certainly reason enough to leave it alone. I had completely forgotten about a use case like that. With regard to (2), I'm pretty sure there's been work done in the recent past to make chan_local more state aware so that this might not be the case any more depending on what version you are using. I might be wrong there, but I know I've got a patch or two hanging around that did make this work. Matt King wrote: Two use-cases where autofill=no is desirable: 1) If it's important that you answer your callers in strict order (i.e. in order to meet estimated wait time commitments etc). 2) If your queue members/agents are local channels (as local channels are always available, so call attempts will be made regardless of who's talking). Kind regards, Matt. BJ wrote / This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. // // That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. // // BJ/ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] question about queue
Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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] FYI about my Mona Vie business venture
I'll give you an A+ for originality after I get done laughing and then we'll still ask you to take this off list. :-) BerkHolz, Steven wrote: Asterisk work does not pay all of my bills, so I have joined up with a company that has a very good payment plan. I have recently become a Mona Vie Independent Distributor. I am not going to go into a sales pitch. This is just an FYI to this opportunity. The company has grown into a Billion dollar company in just 2 years. This company's compensation plan is the best and quickest that I have seen. My brother-in-law has only been in the business for a month and is already making a profit. The first thing that I noticed when researching the opportunity, was that I could find no negative statements about it. The product itself has many health benefits. So far: My knees no longer ache. We are both sleeping better. I literally do not stir once asleep. My restless leg syndrome has not been noticed. I seem to have more energy. The main ingredient is the acai berry. Here is a list of what it is supposed to do: Boosts energy levels Improves digestive function Improves mental clarity/focus Promotes sound sleep Provides all vital vitamins Contains several important minerals Is an extremely powerful free radical fighter Acai has very high levels of fibers Cleanses and Detoxifies the body of infectious toxins Strengthens your immune system Enhances sexual desire and performance Fights cancerous cells Slows down the aging process Promotes healthier and younger-looking skin Alleviates diabetes Normalizes and regulates cholesterol levels Helps maintain healthy heart function Minimizes inflammation Improves circulation Prevents artherosclerosis Enhances visual acuity The income can be made in two ways (actually more, but two primary ways) 1. Reselling the product at a marked up price. This is something that I have no interest in, and do not personally know anyone doing this. 2. Team Commissions. a. You make back 5 percent of the sales that occur below you in your tree. b. You only have to personally sign 2 people. Other people above you will be adding to your tree. c. They call it a binary system, where you only have 2 people directly under you, and any other people that you add go down to the bottom and benefit others as well as yourself. d. I already have two people underneath me and have not personally signed anyone yet, so it is a quick growing tree, even for people that may not be as motivated. e. After a month, My brother-in-law has NO more out of pocket expenses to stay in this system. The money he is earning is paying for his Minimum requirements. The rest is profit. To sign up to be a distributor , which is required to make money, is $54 A case of Mona Vie is $120. A case will last 2 people a month. (you only take 2 ounces a day) This may seem like a lot, but: 1. You will not need to buy any vitamins. 2. My brother-in-law is already making $200 a month, after being in the system for a month, So his cost for the Mona Vie is covered and he is making $80 a month. 3. As more people sign up, the amount he gets back will increase. Anyway, I am not intending this to be pushy or salesy, I just wanted to let my associates, that may be looking for additional income, know about this. Here is the Website, if you are interested in researching this: http://teamvie.blogspot.com/ http://www.monavie.com Also, feel free to Google it. I am very excited with this, both in the health benefits I am already seeing, and the income potential. Please feel free to let me know if this is something that you may be interested in, and I can get you more information. Thank You, Steven B [EMAIL PROTECTED] Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 Please only print this email if it is necessary. Help spread environmental awareness. This e-mail and any files transmitted with it are intended only for the person or entity to which it is addressed and may contain confidential material and/or material protected by law. Any retransmission or use of this information may be a violation of that law. If you received this in error, please contact the sender and delete the material. ___ -- 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 -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] AMD on a SIP trunk...
Add an answer() and a playback of 1 second of silence or something else to make sure the RTP is nailed up. AMD can/will hang if it has no media to analyze. Carlos Chavez wrote: We have an Asterisk server with a small outgoing call center. We use AMD and it usually works very well on Zap channels (E1 PRI). We added a couple of SIP trunks to reduce long distance costs but now AMD gets stuck when the call goes out through the SIP channels. Here is an example call using a SIP line: -- Executing [EMAIL PROTECTED]:1] Set(Local/[EMAIL PROTECTED],2, CIDTEMP=49875calllogId=135514 016566275538) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Local/[EMAIL PROTECTED],2, SIP/juarez-60/6275538|25|C) in new stack -- Called juarez-60/6275538 -- SIP/juarez-60-0892f740 is making progress passing it to Local/[EMAIL PROTECTED],2 -- SIP/juarez-60-0892f740 answered Local/[EMAIL PROTECTED],2 -- Executing [EMAIL PROTECTED]:1] Answer(Local/[EMAIL PROTECTED],1, ) in new stack -- Executing [EMAIL PROTECTED]:2] AMD(Local/[EMAIL PROTECTED],1, ) in new stack -- AMD: Local/[EMAIL PROTECTED],1 016566275538 (null) (Fmt: 64) -- AMD: initialSilence [3500] greeting [2500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] AMD just stops and it takes over a minute until the line is dropped. The same number dialed through Zap works without a hitch. What could be the reason? If I dial the same number without AMD I can talk to the other person so I know the SIP line is fine. ___ -- 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 -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Problem with FollowMe
Mike Coakley wrote:
I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed
the WIKI page on setting it up but I can't seem to get it to work.
Here is my Asterisk version:
pbx1*CLI core show version
Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on
2008-01-10
12:08:48 UTC
Here is a log of when the FollowMe is being called:
NOTE: I've tried to use the AstDB as the WIKI described but when I
looked at the code it didn't seem like that would work so I've hard
coded a phone # for now. But I would prefer to use the AstDB if that
is a workable solution.
-- Executing [EMAIL PROTECTED]:8] FollowMe(Zap/
1-1,2000|a) in new stack
-- Zap/1-1 Playing 'vm-rec-name' (language 'en')
-- Zap/1-1 Playing 'beep' (language 'en')
-- x=0, open writing: /var/spool/asterisk/followme.1201278830.10
format: sln, 0x81c0218
-- User ended message by pressing #
-- Zap/1-1 Playing 'auth-thankyou' (language 'en')
-- Zap/1-1 Playing 'followme/pls-hold-while-try' (language 'en')
-- Started music on hold, class 'default', on channel 'Zap/1-1'
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM1/[EMAIL PROTECTED] cause:
Unknown
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM2/[EMAIL PROTECTED] cause:
Unknown
[Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
[Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec:
Unable to allocate a channel for Local/FM3/[EMAIL PROTECTED] cause:
Unknown
(NOTE: This log was taken just prior to making the changes indicated
below in my Followme.conf file. But the changes reported the same logs
lines, simply with different values (i.e. same errors).
A section of my followme.conf file:
[2000]
context = pstn_inbound
number = 201XXX,5
number = FM1/2000,10
number = FM2/2000,20
number = FM3/2000,30
Here is the relevant section of the macro that calls the FollowMe app:
exten = s,7,GotoIf(${DIALSTATUS} = NOANSWER?:8:14)
exten = s,8,FollowMe(${STATION_EXTENSION},a)
I've tried different context in my FollowMe configuration file but it
doesn't seem to change anything. Any help would be appreciated.
The app_followme that's in 1.4 right now I don't think ever made use of
any assets in AstDB, or at least, not what I've coded into it.
BJ
--
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] Polycom Remotely Cancel Call Forward
Kevin Kiely wrote: Great suggestion, thanks. The boot failed with the mac-phone.cfg removed. I re-touched the file and followed your suggestion. Any way of removing the call forwarding feature via the xml configs? Kevin Kiely wrote: I have a remote user on a Polycom IP Phone who has set call forwarding by accident and is away from the phone. Does anyone know of a way to remotely un-forward the phone? I tried to reboot the phone but that didn't work and removing the mac-phone.cfg caused problems Remove the XML element tag from within mac-phone.cfg that it updated with the forwarding information and then reboot it again. I know there's a way to disable DND on the polycom's via sip.cfg. I'm not sure about call forward. I would need to check the master config file. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Polycom Remotely Cancel Call Forward
Kevin Kiely wrote: I have a remote user on a Polycom IP Phone who has set call forwarding by accident and is away from the phone. Does anyone know of a way to remotely un-forward the phone? I tried to reboot the phone but that didn’t work and removing the mac-phone.cfg caused problems Remove the XML element tag from within mac-phone.cfg that it updated with the forwarding information and then reboot it again. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Agents and AddQueueMember
Rajkumar S wrote: Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use AddQueueMember to add the SIP [Local] channel to queue. The queue logs come like 1197953879|1197953876.34|Auth-Enq|Local/[EMAIL PROTECTED]|CONNECT|3|1197953876.35 Previously it used to come like 1197013076|1197013055.27|Auth-Enq|Agent/1001|CONNECT|21|1197013055.30 Here the problem is that there is no way to find number of calls taken by a person, because there is no agent abstraction here. What is the recommended way to work around this problem ? AddQueueMember(queuename[|interface[|penalty[|options[|membername): You can use the membername option now to provide a member name which will be used in place of what the Interface name is. This option was added for specifically the issue you're pointing out. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - GEOPRIV and location based SIP services
MatsK wrote: Olivier wrote: Hi, I'm wondering whether or not it is achievable to build a web based click2dial application that could automatically detect that a user is connected from office or home. Another option is to directly ask user or let them change default option but having this automatically detected is a bonus. Has anyone tried to build such location based SIP services ? I've read few lines about GEOPRIV which seems to be a building block for location based services but I could make sure if such DHCP extensions are implemented somewhere. Do you think GEOPRIV would help ? Regards Hi Oliver, Linux Journal had an article about timezone handling in asterisk with perlscript for checking the GeoIP database with the IP adr. from the location db. Maybe that could give you a clue how to solve your question. http://www.linuxjournal.com/article/9190 The challange with GEOPRIV is that its rarely used so I would recomend GeoIP, http://www.maxmind.com. The problem with location tracking via IP is that, often, the entity who is the current owner of the IP at the time is not the same entity that has that IP space registered with ARIN. For example, I plugged in the serial side IP of one of the IP T1s in my office, and what comes up is my provider's corporate headquarters in MA, not my office in NJ. There are, however, other ways to do this. You could opt to have the user choose what location they're at and then drop a cookie on the user's browser at that location that will allow the browser to remember what location it's supposed to be. This isn't without its drawbacks either. There are many users that have cookies disabled because of privacy concerns, but at least with this approach the barrier to get it working again is generally in the hands of the user or their administrator. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Definity G3R and MWI
I've just spent the last two hours Googling and searching the Wiki. I'm trying to find if there are any listings of codes for the Avaya Definity G3R, to allow for an Asterisk system to turn on/off a phones MWI that is attached to a G3. We are looking to use an Asterisk system as a voice mail server. I think you're going to need to integrate via the SMDI feature of Asterisk and figure out what the Definity needs as well to work with an SMDI connection. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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 with around 230 SIP connections
Jerry Geis wrote: Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. Would this work? Thanks, Jerry I would have some serious reservations throwing this many clients into an app_meetme room which is the foundation layer for the page functionality. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Annoying PRI Channels Restarting Message
Michael J. Liberatore wrote: Would this be normal? Could this be a problem with the line? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, November 23, 2007 8:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Annoying PRI Channels Restarting Message My guess is that the B channels are in fact bouncing in and out of service and the message is a reflection of it. On Fri, 23 Nov 2007, Michael J. Liberatore wrote: Hi all, i have recently setup a p2p t1 using sangoma t1 cards and asterisk 1.4. Its working great but i am getting an annoying message every little while in asterisk: [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/16 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/17 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/18 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/19 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/20 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/21 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/22 restarted on span 2 [Nov 23 19:17:57] VERBOSE[6487] logger.c: -- B-channel 0/23 restarted on span 2 Basically this goes through all 23 channels and then says its was successfully restarted. the link doesnt appear to be going down because there is nothing in the system log that normally comes up when the link actually goes up or down. this appears to be some asterisk thing. its not affecting calls as far as i can tell and doesnt seem to happen when the channels are in use. Any ideas? Can this be ignored? If so, can i safely disable this by changing it to a debug message in the code? Thats what i did with an annoying message caused by setting ext 700 as a orbit on a snom phone. Check out resetinterval= **resetinterval**: sets the time in seconds between restart of unused channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 or 'never' to disable *entirely*. http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf -- Bird's The Word Technologies, Inc. http://www.btwtech.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] ACD functionality , Skills for agents
Kyriakos wrote: Guys can someone answer how the ACD works when it needs to decide which call to take next from queues with equal weights? Does it take the call with the longest period of watiting or does it work randomly? Whichever thread from the queue that does its processing first is the one that will get the next available member. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] ACD functionality , Skills for agents
Kyriakos wrote: It would be nice to add an option of choosing to answer the call with the longest waiting time, or answer randomly, or round robin, etc... Agreed, but, understand that each queue defined in app_queue is separate. The way the weights work is only by instructing a thread to go into another queue's data space (while holding a mutex lock to make sure multiple threads aren't walking on the same space) and make sure there aren't calls waiting where that queue has a higher weight than the one currently processing before it decides whether or not it can serve up calls to an available member. There is not one large, consolidated, pool of calls waiting for consideration when you are dealing with multiple queues in the current design of app_queue. As a result, true skills based routing with the existing app_queue is, difficult, at best. The queue application does a fairly good job for what most people need for it to do, but when you start getting into these more complex call/queue routing scenarios, you're defining a scope of requirements that the original app_queue just wasn't designed for. Features like queue weight were/are band aids to try to get you closer to the end run goal, but that band aid and others like it has come with its own costs as well (mutex deadlocks,etc) that many people here have complained about in the past. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] How to play Asterisk .raw file
Gary wrote: I used ChanSpy( ) recorded some test conversations. It has .raw extension. What kind of audio file is this? How can I play it? Gary That's SLINEAR. I know CoolEdit, now Adobe Audition can play them. Not sure about Audacity. I've never tried it with that. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] ACD Queue LOG RINGNOANSWER Content 0
Yes. That's supposed to to be the timeout value. In the case where it's 0 are you seeing a call reject or something else? asterisk wrote: In my queue log I see that on the RINGNOANSWER Event I get different content. Some events soe the ring timeout (15000). Other events show 0. Other yet show 1000 Doens anyone know what 0 means? Did it try to ring the phone, but it was busy? Thanks Doug ___ --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 -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] 'a' extension
Peder @ NetworkOblivion wrote: Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears that the called number is 'a' and the calling number is 123. I need to be able to tell that it was a call to x456. Thanks. You could set a CDR variable called origdst when the call starts up and then use the customer CDR format to kick origdst back out into the custom CDR format when the record gets written. This is what we do for this. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Route an incoming call by ANI*DNIS
Dan Casey wrote: does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set(Zap/49-1, DID=1231234*4812*) in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important thing is routing. This shouldn't be necessary. I think with the correct signaling set in zaptel.conf, chan_zap should be doing the parsing work for you. What signaling are you set for now? You might want Feature Group D vs. EM_Wink? ___ --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] DST
Joe Acquisto wrote: My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a. If you've got the files centrally managed, you can update the correct tags in sip.cfg to correct the situation. These are the correct settings for regions affected by the new DST regs: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0 ___ --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] DST
Turbo Fredriksson wrote: Quoting Joe Acquisto [EMAIL PROTECTED]: My thanks to all. Problem resolved with the assistance. Would be nice if you posted HOW it was fixed to... I have this exact same problem at home, but the work phones displays time correctly... If you've got the files centrally managed, you can update the correct tags in sip.cfg to correct the situation. These are the correct settings for regions affected by the new DST regs: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0 ___ --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] FollowMe recorded name filename variable?
Hmm.. I think it should be cleaning it up post-call already. If not, please open a bug on Mantis as that sounds like a bug. On 10/19/07, Anthony Messina [EMAIL PROTECTED] wrote: Is there a variable for the filename that is created by the FollowMe application when a is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Polycom 501 Phones Rebooting
On 9/21/07, Gregory Boehnlein [EMAIL PROTECTED] wrote: Hello, At one of our locations, we have started to see Polycom 501s (running 1.6.7 firmware) randomly reboot. We have taken packet traces of the phones to determine if there is something odd in the Layer 2 or 3 of the network that might cause it, and have not seen anything strange. There are no errors on the ports. This appears to be affecting POE powered as well as AC powered phones. The Polycom Logs for the phones don't seem to provide any clarity. Where should I troubleshoot this next? Greg - Is the phone's app log uploaded back up to the central provisioning server showing anything of value? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ 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] Asterisk w/MS SQL Server 2005
On 9/4/07, Larry Costigan [EMAIL PROTECTED] wrote: I've got an Asterisk switch that is going to run an IVR menu with a database interface that will be doing lookups based on the user entered data and then reading back strings with the appropriate data integrated into the text. I have found quite a bit of data on using MySQL as a database with Asterisk, but I haven't found much about using MS SQL Server with Asterisk... We have a SQL Server 2005 database server that has all the data that is needed for the IVR interface, and it would be great if we could interface directly with it using Asterisk. Does anyone have any suggestions on even attepting this? Another option might be to setup Asterisk to interface with MySQL and then work out the details of exchanging data between MySQL and SQL Server... Any and all help is greatly appreciated!! You'll want to take a look at func_odbc which should probably give you a pretty good conduit to get the data you need to work through the IVR flow. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Problems with Polycom 300/500/600
On 8/31/07, Joe Acquisto [EMAIL PROTECTED] wrote: Any great disadvantage to using polycom 300/500/600 vs the 301/501/601? I recall reading in the release notes of the latest release of the firmware (2.2+) that I believe they've finally stopped supporting the earlier models so it looks like you are reaching or have reached an EOL period on firmware with those models. Aside from that, if you're happy with current functionality of those phones as they stand now, they'll probably be fine for quite some time to come. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Members in 'Unknown' status in output of 'queue show'
On 8/29/07, James FitzGibbon [EMAIL PROTECTED] wrote: Does anyone know what can cause queue members to go into a status of Unknown? pbxtel-01*CLI queue show cshas 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:447, A:20, SL: 91.7% within 60s Members: SIP/1405 (dynamic) (Unknown) has taken no calls yet SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet SIP/1442 (dynamic) (paused) (Unknown) has taken 2 calls (last was 101 secs ago) SIP/1440 (dynamic) (In use) has taken 2 calls (last was 3071 secs ago) SIP/1428 (dynamic) (paused) (Not in use) has taken 2 calls (last was 10818 secs ago) SIP/1404 (dynamic) (paused) (Not in use) has taken 2 calls (last was 2228 secs ago) SIP/1429 (dynamic) (paused) (Unknown) has taken 2 calls (last was 953 secs ago) SIP/1432 (dynamic) (Unavailable) has taken 5 calls (last was 1229 secs ago) SIP/1430 (dynamic) (In use) has taken 2 calls (last was 22744 secs ago) SIP/1435 (dynamic) (In use) has taken 3 calls (last was 13511 secs ago) SIP/1434 (dynamic) (Unknown) has taken 6 calls (last was 9504 secs ago) SIP/1424 (dynamic) (In use) has taken 4 calls (last was 16373 secs ago) SIP/1408 (dynamic) (paused) (Not in use) has taken 2 calls (last was 8685 secs ago) SIP/1203 (dynamic) (In use) has taken 3 calls (last was 16425 secs ago) SIP/1410 (dynamic) (Unknown) has taken 2 calls (last was 8629 secs ago) Callers: 1. Zap/50-1 (wait: 11:15, prio: 0) 2. Zap/36-1 (wait: 0:41, prio: 0) That's just one queue, but I had nearly all my agents just go into Unknown status. This is on * 1.4.10.1. I had this happen once in the past, but couldn't reproduce it in the lab. When this happens, 'ringinuse=no' stops working, because app_queue considers Unknown to be a valid state to dispatch a caller to. So my agents start getting flooded with calls while already on the phone, then the call-limit I've configured in sip.conf kicks in and my console fills up with this: pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to peer '1405' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to peer '1410' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to peer '1405' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to peer '1410' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to peer '1405' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to peer '1410' rejected due to usage limit of 2 I had to restart Asterisk to clear the states - sip reloads, app_queue reloads didn't do anything. Any thoughts as to where to start debugging this? I killed * instead of stopping it so that I got a core file. There is nothing in the log to indicate what went wrong prior to the first instance of ...rejected due to usage limit. Anything else I should gather before submitting a bug? I think we will want to see what state chan_sip is sending into app_queue for it to be called Uknown. What is the last state these channels are in before they go to Unknown in app_queue? -- Bird's The Word Technologies, Inc. http://www.btwtech.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 Manager Proxy - Still required?
On 8/17/07, Andrew Ruthven [EMAIL PROTECTED] wrote: Hi, With more recent version of v1.2 and with v1.4 are things like the AstManProxy still recommended if you want to have a bunch of applications talking directly to Asterisk? If you're looking to have a number of clients monitor events, etc, I'd say that having a proxy in between is still a good thing. The performance of the manager itself is greatly improved since before 1.2 but there are still ongoing (albeit random, sporadic) issues with cpu race and huge memory allocations that still need to get resolved. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Using MSAccess to dial on a Zap line
On 6/27/07, Jason Martin [EMAIL PROTECTED] wrote: Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone system. A middleware program has been written to watch for dial events in a database, then the program calls the Zap station the call rep is at using the manager interface. Is there a better way to do this? The complaint we are getting now is the call rep doesn't want their phone to ring when making a call. Can the manager interface give a phone number to dial on an off hook Zap line? Why not put the off hook zap lines into a meetme room and then as you're dialing lines out join them to the meetme room? That way the zap lines can stay off hook. That also leaves open the ability down the road to have a manager monitor/barge these conversations. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Fwd: Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.
On 6/19/07, Lucian Romi [EMAIL PROTECTED] wrote: In this scenario, how to make asterisk send the invite to SIP/[EMAIL PROTECTED]:5064 instead of Local/[EMAIL PROTECTED] Thanks Asterisk isn't a SIP proxy. As such, you need to use some workarounds to make what you want to do work. One way might be.. In extensions.conf : [default] exten = 101,1,Dial(SIP/[EMAIL PROTECTED]:5064) exten = 101,n,Congestion() -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue problem
On 6/6/07, Elmar Haneke [EMAIL PROTECTED] wrote: Hi, On asterisk 1.4.4 I have an strange effect on agents answering queue calls: If an agent does set current call on hold the phone immediately gets connected to the next incoming call. What might cause this effect? How can it be removed? Set your core debug level to greater than 2 and then try this process again and the go into your log that you've redirect DEBUG messages to and search on changed to state . I think you'll find that by doing what you're doing above, the caller's channel state will change to some kind of ONHOLD state (if SIP - or other channel tech that supports that dev state) but I would think the agent channel itself might go to NOT IN USE. Let us know what you find. This might be worth opening a bug on Mantis based on what you come up with as I would think if you placed a queue caller on hold you wouldn't then want to receive additional calls in from queue. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrated T1
On 5/24/07, Alex Balashov [EMAIL PROTECTED] wrote: On Thu, 24 May 2007, William Moore wrote: On 5/24/07, Jeremy Mann [EMAIL PROTECTED] wrote: Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? Yes, zaptel will create a device node for you. Take a look at the set-hdlc tool in zaptel and the less common channel types in the default zaptel config file (rawhdlc is one, there are also others). Interesting! Is this a relatively recent development? I stand corrected. Is there a way to make PPP-encapsulated T1 work as well? Yes. I've got a client with a TE212P with a PRI channelized on one port and a PPP encapsulated IP T1 on the other. It works fine. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue Problem - Automatic Call Distribution
On 5/16/07, TienSen Chong [EMAIL PROTECTED] wrote: Hi all, I am seeing a strange problem with Asterisk queue. I am not sure if it's my configuration which is wrong or there's something with Asterisk. I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i tried to call the extension number directing to the queue, the MGCP phone is not ringing. However, it is fine to call the MGCP phone directly. The strange thing is after i've called the MGCP phone directly, calling the extension number directing to the queue works fine. I wonder what could be wrong. Any comment and help is very much appreciated. The following is the configuration: queues.conf [queue1] strategy=roundrobin member=MGCP/[EMAIL PROTECTED] I don't think that's a valid interface string as far as app_queue is concerned at present. I'll have to take a look at that. I think a workaround would probably be to define a Local channel as the queue member and then dial to the MGCP phone in the exten you're defining for the Local channel. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call prority (QUEUE_PRO) in the queues
On 4/26/07, gc [EMAIL PROTECTED] wrote: Suppose I have one agent login into two different queue and there are calls waiting in both queues. If the calls in one queue has higher call prority (set QUEUE_PRO to higher value) than the calls in other queue, will the agent get the higher prority call first or the QUEUE_PRO has no effect? We have an Asterisk server( 1.2.17 with CentOS) running as ACD. We are having problem using weight option in the queue. I figure maybe I can use QUEUE_PRO instead. Queue priority will, unfortunately, only cover one queue. It cannot cover and account for priorities of calls from more than one queue. You will want weight for that. What's the problem you were having with it? BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble recording calls
On 4/9/07, ahester [EMAIL PROTECTED] wrote:
Hi all,
I am having the following trouble with recording calls:
When calls come into the support line did number, the call starts to
record on the first queue, but appears to hang up when the call actually
connects to the engineer (ie I see got hangup request on the cli and
then mixmonitor ends.) I am guessing this has to do with the announce
file that is played to the engineer before the call is connected. It
seems that if the call rolls to the next queue because of timeout,
asterisk doesn't even try to record it. (I don't see any mixmonitor on
the cli for the next queue).
I would appreciate any help with this. I have to have all calls
recorded and I have to do announcements so that the callee knows how to
answer the phone.
Thanks,
Andy
The configs are as below:
From extensions.conf:
#after various menu stuff, send to support
exten = 214xxx,13,SetGlobalVar(ORIGIN=support)
exten = 214xxx,14,Queue(support1|tr|||10)
exten = 214xxx,15,Queue(support2|tr|||)
#dial command for sip extensions that are in the queues
exten =
_72XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten = _72XXX,2,Dial(SIP/${EXTEN})
exten =
_73XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten = _73XXX,2,Dial(SIP/${EXTEN})
queues from queues.conf:
[support1]
; Support call queue
announce = 16
strategy = rrmemory
timeout = 15
retry = none
wrapuptime=15
announce-frequency = 0
joinempty = no
leavewhenempty = yes
member = Agent/2008
member = Agent/2009
member = Agent/2014
member = Agent/2015
member = Agent/2017
member = Agent/2018
member = Agent/2019
member = Agent/3520
member = Agent/3521
member = Agent/3522
member = Agent/3524
member = Agent/3529
[support2]
; Support2 call queue
announce = 16
strategy = ringall
announce-frequency = 0
; Added below for testing because the second queue was not even trying
to record
; according to the asterisk console (still doesn't)
Set(MONITOR_FILENAME=support/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
monitor-format = wav
monitor-join = yes
joinempty = yes
member = SIP/72008
member = SIP/72009
You cannot use the MixMonitor app on its own in a callback scenario
because, as you've already discovered, MixMonitor senses the call
transition between the time the agent answers and the calls is then
bridged with the waiting caller and still stop recording.
To fix this, in the 1.4 version of app_queue, there's a
monitor-type=MixMonitor parameter which will use the MixMonitor
appropriately natively in app_queue instead of Monitor.
BJ
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Re: [asterisk-users] ${QUEUESTATUS}
On 4/9/07, Damon Estep [EMAIL PROTECTED] wrote:
There are 6 different ${QUEUESTATUS} variable values defined in asterisk
1.2, I am attempting to make sure I have a full understanting of when they
would be set;
If someone could correct errors with these definitions ot would be
appreciated;
TIMEOUT – the max time specified in the queue command elapsed, only checked
between retries so may not be 100% accurate.
FULL – the number of callers in the queues would exceed the maxlen= value
defined in queues.conf if another caller was added
JOINEMPTY – a call was sent to the queue but the queue had no members, does
not apply when using agentcallbacklogin since there could be unavailable
members defined but not available.
LEAVEEMPTY – the last agent was removed form the queue before alls calls we
handled, remong callers exit with this status, also acts differently when
there are only queue members that are unavaialbe
JOINUNAVAIL/LEAVEUNAVAIL – same as JOINEMPTY/LEAVEEMPTY, except that there
were still queue members, but all were status unavailable (logged out)
So if a queue is made up of only callback agents (agentcallbacklogin) then
the queuestatus will never be joinempty or leaveempty
If the maxlen=0 then there will never be a queuestatus of full
If there is no timeout in the queue command thee wll never be a questatus of
timeout
If there are no callback or static agents joinunavail/leaveunavail will
never apply.
I don't believe that the logic you're describing here with if a
queue is made up of only callback agents then the queuestatus will
never be joinempty or leavempty is right. If a queue is made up of
only callback agents and none of those callback agents are presently
logged in, the status could certainly be joinempty or leavempty.
The rest of it looks pretty good. Thanks for taking the time to
better document this!
BJ
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Re: [asterisk-users] Multi-Level Queue
On 3/31/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: Peder @ NetworkOblivion wrote: I also had a question about acking a call. It appears that acking a call is under agents.conf. I want to specify members as SIP/1234, etc, rather than having users login all the time. I don't want to have to login from my cell, I would prefer it to just know that my cell number is always a member. Is it possible to force an ack of a call if I define members as SIP/? Not directly, no, because channel drivers don't implement call acking (except for chan_agent). However, if you create a context in your dialplan that uses Dial() to call the SIP device with acking turned on, then you can add Local/[EMAIL PROTECTED] as a member of the queue and have those calls run through the Dial() application. ___ Or he could use app_followme which has call acking built in. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue priority
On 3/29/07, Jordan Novak [EMAIL PROTECTED] wrote: What is the most stable version supporting queue priority. I have had many crashes, I am using 1.2.11 and have set the weight in queues.conf. is there a better way or a patch. I can't seem to find much. Any suggestions? Do you have a bug open in mantis with a backtrace of the crash? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue App - Free agent and waiting calls
On 3/19/07, equis software [EMAIL PROTECTED] wrote: Asterisk 1.4 I have strategy= leastrecent and autofill = yes I have 2 agents, one is answering a call and the other is free and have some calls waiting in the queue. Only when the first agent hangup the second agent receive the first call in the queue. It happends some times. I believe there is a bug open in Mantis on this, and I intend to reproduce and start working on it this week to get a resolution. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 cable for Digium T1/E1 Cards
You can use a patch cable, yes. The T1 will look to use pins 1,2,4 and 5 while Ethernet will typically use 1,2,3 and 6 provided you're not using POE or something simliar that requires additional pins. On 3/18/07, Jeronimo Romero [EMAIL PROTECTED] wrote: Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 cable for Digium T1/E1 Cards
Yes. At that point, you're looking for a T1 cross-over. The pinout is as follows: 1 4 RX/Ring/- --TX/Ring/- 2 5 RX/Tip/+ --TX/Tip/+ 4 1 TX/Ring/- --RX/Ring/- 5 2 TX/Tip/+ --RX/Tip/+ 3 3 Shield/Return/Ground 6 6 Shield/Return/Ground On 3/18/07, Jeronimo Romero [EMAIL PROTECTED] wrote: I assume that I would need to cross these pins over if I were going from t1 card to t1 card. Is this correct? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Sent: Sunday, March 18, 2007 7:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards A common Cat5 straight through cable will work fine. T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for signals. A T1 loopback plug would be wired 1 to 4 and 2 to 5. They come in handy for testing T1 cards or for providing a hard loop for the telco. Tom At 05:42 PM 3/18/2007, you wrote: Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 cable for Digium T1/E1 Cards
Correct. Because Ethernet cross-over cables are crossing over 1,2,3 and 6; no 1,2,4 and 5. On 3/18/07, Jeronimo Romero [EMAIL PROTECTED] wrote: So a regular cross over cable wouldn't work? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Sunday, March 18, 2007 10:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards Yes. At that point, you're looking for a T1 cross-over. The pinout is as follows: 1 4 RX/Ring/- --TX/Ring/- 2 5 RX/Tip/+ --TX/Tip/+ 4 1 TX/Ring/- --RX/Ring/- 5 2 TX/Tip/+ --RX/Tip/+ 3 3 Shield/Return/Ground 6 6 Shield/Return/Ground On 3/18/07, Jeronimo Romero [EMAIL PROTECTED] wrote: I assume that I would need to cross these pins over if I were going from t1 card to t1 card. Is this correct? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Sent: Sunday, March 18, 2007 7:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards A common Cat5 straight through cable will work fine. T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for signals. A T1 loopback plug would be wired 1 to 4 and 2 to 5. They come in handy for testing T1 cards or for providing a hard loop for the telco. Tom At 05:42 PM 3/18/2007, you wrote: Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues
If you make a SIP device a queue member, that member will be rung so long as the device state of the SIP interface shows as not in use. With regard to voicemail, are you trying to get a queue call answered by voicemail or is that not your intent? On 3/17/07, Steve Kennedy [EMAIL PROTECTED] wrote: On Sat, Mar 17, 2007 at 11:44:52AM -0700, Steve Edwards wrote: Yes. to which bit? auto-agent (as per resource) or voicemail to an agent? Steve On Sat, 17 Mar 2007, Steve Kennedy wrote: A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Follow-Me Application
On 3/16/07, Kevin Kiely [EMAIL PROTECTED] wrote: I am having an issue with the follow me application in 1.4 The application description (below) indicates that if the specified followmeid profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority. That's not happening for me and the execution terminates not continuing to the next priority in the dialplan. Can anyone confirm this? Thanks, Kevin exten = 502,1,Followme(cell|s) exten = 502,2,Playback(goodbye) exten = 502,3,Hangup -- Executing [EMAIL PROTECTED]:1] FollowMe(SIP/2101-b6e17f60, cell|s) in new stack [Mar 16 23:29:34] WARNING[10814]: app_followme.c:954 app_exec: Profile requested, cell, not found in the configuration. == Spawn extension (from-sip, 502, 1) exited non-zero on 'SIP/2101-b6e17f60' [Description] FollowMe(followmeid|options): This application performs Find-Me/Follow-Me functionality for the caller as defined in the profile matching the followmeid parameter in followme.conf. If the specified followmeid profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority. Options: s- Playback the incoming status message prior to starting the follow-me step(s) a- Record the caller's name so it can be announced to the callee on each step n- Playback the unreachable status message if we've run out of steps to reach the or the callee has elected not to be reachable. Returns -1 on hangup I can't confirm it just now but I can certainly fix it if you post a bug on bugs.digium.com about it. :) Thanks! -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow me on multiple numbers..
On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone answering machine messes it all up. I have call screening setup so the call gets answered by the cell phone answering machine and it never accepts the call. I would appreciate if someone can help me with the setup. With the 1.4 codebase, you'll want to take a look at app_followme -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue announcing hold sequence instead of hold time
All - Next step here would probably be to open a bug on bugs.digium.com with a full VERBOSE/DEBUG log along with associated config files so we can troubleshoot this and fix it if there's a problem. Thanks. On 3/9/07, Drew Gibson [EMAIL PROTECTED] wrote: Trevor G. Hammonds wrote: From: Drew Gibson Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected (estimated hold time is less than 2 minutes ...). Now the caller gets an announcement of their sequence in the queue (Your call is now first in line ...). I believe that the only changes I have made to queues.conf and agents.conf is the addition of the context= statement and editing the list of agents. Has anyone else seen this? What am I missing? regards, Drew Drew, This has been normal behaviour for as long as I can remember. The caller hears the estimated time until they are next in line, then they hear the 'next in line' announcement. Sincerely, Trevor Hammonds Hi Trevor, I should have given a better example. Like Rob Schall, the 2nd, 3rd, 4th, etc callers all get a sequence number rather than an estimated time. Rob, are there any common elements in our configs, like t or H options that might be getting in the way? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk queue and agents
What version of Asterisk is this the r number on the 1.4 branch? I'll try and reproduce the condition here. Also - if you could post into that bug on Mantis a full DEBUG/VERBOSE log and what it looks like when you do show queues when one of these agents is on the phone, that'd be real helpful. Thanks. On 3/7/07, Hall, Eric M. [EMAIL PROTECTED] wrote: BJ Here is the sip.conf file. Hints work great. The only problem is the queue is sending calls to an agent that's on the phone. [general] rtcachefriends=yes videosupport=yes port=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) context=sip ; Send unknown SIP callers to this context allow=g729 allow=h263 ; H.263 is our video codec allow=h263p ; H.263p is the enhanced video codec ;allow=g711 ;allow=all ;allow=ulaw ;allow=gsm nat=1 host=dynamic type=peer qualify=yes notifyringing=yes Subscribecontext=sip call-limit=300 notifyhold = yes limitonpeer = yes notifyringing = yes; Notify subscriptions on RINGING state (default: no) notifyhold = yes [56405] ;Eric Test type=friend ; friend means this device takes and makes calls username=1 ; Username on device callerid=Eric Test Phone 56405 secret=56405; Password for device host=dynamic ; This host is not on the same IP addr every time context=sip ; Inbound calls from this host go here [EMAIL PROTECTED]; Activate the message waiting light if this canreinvite=no; Leave this alone for now; see archives for details nat=1 qualify=yes Subscribecontext=sip notifyringing=yes call-limit=300 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, March 07, 2007 10:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents I don't think this is a bug. From UPGRADE.txt: * Queues depend on the channel driver reporting the proper state for each member of the queue. To get proper signalling on queue members that use the SIP channel driver, you need to enable a call limit (could be set to a high value so it is not put into action) and also make sure that both inbound and outbound calls are accounted for. Example: [general] limitonpeer = yes [peername] type=friend call-limit=10 Please test with that and report your findings, and if it's still not working find us on IRC as we'd like to take a further look and see what might be wrong. BJ On 3/7/07, Hall, Eric M. [EMAIL PROTECTED] wrote: Looks like it's a bug http://bugs.digium.com/view.php?id=9172nbn=3 I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and report back to the list. Eric Hall -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz (Ta^3) Sent: Wednesday, March 07, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents Have a question for the group If I have an agent is on the phone outside of the queue should that person still get queue calls ? Doing a show agents online I see Available however show hints I see inuse. There is a ringinuse feature for SIP devices on 1.4.X which is what you are looking for. -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk queue and agents
Ok. One more thing - how are you logging the agent in? With AgentLogin or AgentCallBackLogin? Additionally, how did you get on that call 56405 to your cell? Was it directly to the SIP device or via the agent channel that the represents that SIP device? BJ On 3/8/07, Hall, Eric M. [EMAIL PROTECTED] wrote: Sorry Forgot to tell you I was on exten 56405 called to my cell. I then called into the Queue with another cell and this is the output. Also forgot to include the show queue voipgw*CLI show queue dayton has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: agent/56432 (Unavailable) has taken no calls yet agent/56422 (Unavailable) has taken no calls yet agent/56426 (Unavailable) has taken no calls yet agent/56424 (Unavailable) has taken no calls yet agent/56429 (Unavailable) has taken no calls yet agent/56427 (Unavailable) has taken no calls yet agent/56425 (Unavailable) has taken no calls yet agent/56411 (Unavailable) has taken no calls yet agent/56428 (Unavailable) has taken no calls yet No Callers masion has 1 calls (max unlimited) in 'fewestcalls' strategy (0s holdtime), W:0, C:0, A:2, SL:0.0% within 0s Members: agent/564321 (Unavailable) has taken no calls yet agent/564221 (Unavailable) has taken no calls yet agent/56405 (paused) (Not in use) has taken no calls yet agent/56423 (Unavailable) has taken no calls yet agent/56421 (paused) (Not in use) has taken no calls yet agent/56420 (Unavailable) has taken no calls yet agent/56416 (paused) (Not in use) has taken no calls yet Callers: 1. SIP/208.70.216.73-09780030 (wait: 0:12, prio: 0) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Thursday, March 08, 2007 7:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk queue and agents Asterisk SVN-branch-1.4-r58243 Voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] voipgw*CLI show agents 56416(Jenifer Henry) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56420(Ran Dodds) not logged in (musiconhold is 'default') 56421(Talena Huffman) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 56423(Manager) not logged in (musiconhold is 'default') 56422(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564221 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56432(ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 564321 (ANNE RIDDIOUGH) not logged in (musiconhold is 'default') 56426(HEATHER PRICE) not logged in (musiconhold is 'default') 56424(BEV BATTAGALLIA) not logged in (musiconhold is 'default') 56429(JOE FERRAU) not logged in (musiconhold is 'default') 56427(MICHELLE CLOUSE) not logged in (musiconhold is 'default') 56425(PATTY ARMSTRONG) not logged in (musiconhold is 'default') 56411(DOREEN BUNDY) not logged in (musiconhold is 'default') 56428(VICKI SHANKS-NORTH) not logged in (musiconhold is 'default') 56405(Eric Hall) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 16 agents configured [3 online , 13 offline] If you tell me how to do a full DEBUG/VERBOSE I will be happy to send you one. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Thursday, March 08, 2007 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents What version of Asterisk is this the r number on the 1.4 branch? I'll try and reproduce the condition here. Also
Re: [asterisk-users] RE: Coaching in asterisk
There's a lot more than just app_chanspy.c changes required to get the full functionality backported to 1.2. On 3/8/07, Wai Wu [EMAIL PROTECTED] wrote: You must be talking about Chanspy. It is included in 1.4. Has anyone tried to compiled for 1.2x? -Original Message- From: [EMAIL PROTECTED] on behalf of Dean Collins Sent: Thu 3/8/2007 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RE: Coaching in asterisk Yep, it's called Whisper Check in voip-info.org I think I've read stuff about it there. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, 8 March 2007 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk queue and agents
I don't think this is a bug. From UPGRADE.txt: * Queues depend on the channel driver reporting the proper state for each member of the queue. To get proper signalling on queue members that use the SIP channel driver, you need to enable a call limit (could be set to a high value so it is not put into action) and also make sure that both inbound and outbound calls are accounted for. Example: [general] limitonpeer = yes [peername] type=friend call-limit=10 Please test with that and report your findings, and if it's still not working find us on IRC as we'd like to take a further look and see what might be wrong. BJ On 3/7/07, Hall, Eric M. [EMAIL PROTECTED] wrote: Looks like it's a bug http://bugs.digium.com/view.php?id=9172nbn=3 I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and report back to the list. Eric Hall -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz (Ta^3) Sent: Wednesday, March 07, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk queue and agents Have a question for the group If I have an agent is on the phone outside of the queue should that person still get queue calls ? Doing a show agents online I see Available however show hints I see inuse. There is a ringinuse feature for SIP devices on 1.4.X which is what you are looking for. -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REMOTE CRASH FIX
On 3/2/07, Mike Lynchfield [EMAIL PROTECTED] wrote: Please note that we are available to fix the current REMOTE crash that affects Asterisk/openpbx/trixbox and crashes these systems via a malformed packet please contacts use if you need a hand to patch your systems. And you'll do it for free too? How gracious of you! If you were charging money, I'd say you belong on the -biz list, but while you're being so gracious, maybe your resources would like to volunteer for some bug marshalling tasks too? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues and LOCAL for members
On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote: Hi, I have an queue stored in relatime and defined members called through LOCAL/ I found out that if I call the members through the LOCAL think the queue statistics is not updated. Any idea, or isnt possible to call members with LOCAL channel. There's been some efforts to have Local channels as viable queue members. I'm not quite sure that I understand your issue. Can you post some more details possibly in a bug on bugs.digium.com ? Thanks. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue cmd option 'i'
On 1/15/07, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Monday, January 15, 2007 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue cmd option 'i' On 1/15/07, James Fromm [EMAIL PROTECTED] wrote: Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should ignore call forward requests from queue members and do nothing when they are requested. Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding should be ignored. Using Queue(customerservice|i) causes Asterisk to crash when sending the call to the member with forwarding enabled on their SIP device. Am I misinterpreting what this option does? You're not misinterpreting. If it crashes, please file a bug at bugs.digium.com. Thanks. I wonder how this could actually work? If Asterisk sends an INVITE to a phone, and the phone responds with 'Moved Temporarily', and Asterisk sends the INVITE again, isn't the phone just going to send 'Moved Temporarily' again? It doesn't send the Invite again, and it doesn't send a new call request (might be INVITE, might be whatever other channel tech is at the requested forwarded exten). -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue cmd option 'i'
On 1/15/07, James Fromm [EMAIL PROTECTED] wrote: Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should ignore call forward requests from queue members and do nothing when they are requested. Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding should be ignored. Using Queue(customerservice|i) causes Asterisk to crash when sending the call to the member with forwarding enabled on their SIP device. Am I misinterpreting what this option does? You're not misinterpreting. If it crashes, please file a bug at bugs.digium.com. Thanks. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over 200 queues, anyone?
On 1/5/07, Lenz [EMAIL PROTECTED] wrote: I think we are going to do it if we get big problems with those many queues. From what I'm seeing, the biggest problems seem to be related to agents, so maybe we can have a try at using straight terminals instead of agents. l. Being somewhat familiar with the innards of app_queue, I wouldn't personally want to set any client of mine up with 200 queues on one instance of app_queue. Like another list member mentioned earlier, I think the risk for deadlock is probably too great. You start adding queue weights into the mix and I shiver to think of what might happen. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
On 1/4/07, Douglas Garstang [EMAIL PROTECTED] wrote: Richard, We have underscores all over the place in our config files, including others in queues.conf. I don't think that's the murder weapon. I think, in general, queues are one of Asterisks biggest features, and also one of it's shakiest. The reload, which is run from a script, caused a reload on 3 servers that are supposed to be redundant, and each crapped it's pants in a slightly different manner. The first stopped processing all queue calls (ie calls would lockup), the second core dumped, and the third seemed ok until you did another 'reload app_queue.so' where it would tell you that the previous reload was not finished yet. Someone made a post yesterday about doing 200 queues on Asterisk. I don't envy what he is about to endure. Doug - There was some bug fixing done on app_queue post 1.2.9.1 to try to accomodate some possibly shaky memory management on linked lists that occurred during a reload. You may want to look at upgrading to latest 1.2.X or backporting those changes and see if the issue still exists. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Avaya S8700
On 12/1/06, Tomer Horn [EMAIL PROTECTED] wrote: Michel R Vaillancourt wrote: Tomer Horn wrote: Hello list, I am curious here if anybody here got an experience connecting Avaya to Asterisk using H323 / T1. I am completely lack of experience with Avaya and I wanna know if anybody here has connected Avaya to Asterisk using H323 and managed to stabilize it. Google provides mixed comments regarding the matter. The purpose of Asterisk on this matter is to provide outgoing calls from the Avaya through Asterisk, so features such as MWI and stuff are not necessary for me. Thanks, Tomer. I have done it with a Definity G3. It was actually pretty straight forward. Have you done it with H323 or T1/E1 ? I've done it both ways to a G3R and an 8700. The h.323 gateways from T1 on carrier interconnect side to an 8700 via h.323 signaling group are actually pre-1.2 Asterisk (still!) and the folks using it are very happy with it. The only gotcha I would warn you about would be packetization between the Avaya and Asterisk on the RTP side of things. If you don't get that right, you won't get good sound quality on the calls. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium through Octasic
On 11/30/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 23 November 2006 11:44, Heidi Mendoza wrote: We're looking at using 4 or 8 port T1 cards with echo cancellation and are evaluating brands to go with. We know that Sangoma has excellent solutions especially when it comes to echo. But we still have to hear about actual performance of a Digium card using the same Octasic DSP echo canceller. Excellent performance. I had an A104d which was giving some very odd audio artifacting, Sangoma replaced the card but did not test the original to ensure that the card was indeed defective or that the problem was solved with the replacement. I haven't put the replacement in service yet, as I had a TE407P on order and it arrived first. :-) After dealing with the crap that the TE406P was, the TE407P is *heaven*. Highly recommended. Ditto here as well. The TE412P and TE212P have been rock solid in deployments I've put them in to. Kudos to the Digium folks for getting it right here. They've got a great product that I wouldn't hesitate to recommend with this product line. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: unable to get channel lock BAD BAD BAD
On 11/16/06, Tim Uckun [EMAIL PROTECTED] wrote: 98% of the people here don't use Trixbox. I don't think this is something with trixbox. I asked the person having the same problem as me if he was using trixbox to see if that would narrow down the realm of the problem. Anyway the error message is in the asterisk log. Googling around I see that other people have posted the problem and nobody has gotten an answer. Steve provided an answer on this guys earlier in the thread. Basically what's happening is that Asterisk is getting a SIP message to do something with a call whose channel lock is still being held by something else within Asterisk. It could be a bad application, it could be any number of other things. The only reason you're seeing such an error now is because before chan_sip would go into race state/infinite loop trying to acquire a lock that possibly could never be acquired (because we don't know who has the lock and for what reason). Our (my) thought in adding it earlier this summer was that in the case where a SIP channel was in bad shape and locks were being held, it was probably better for us to act like we never received the SIP message to the client, log the issue, and hopefully by the time the sender of the message went for a retransmit (since SIP msgs in * only come in via UDP and when we don't respond, one must retransmit assuming the original msg never got there), the lock condition would clear itself and things could process as normal. It's funny this thread came up today though as I have a client who's actually never seen this msg before on any of their production systems and all of a sudden it came up this evening on one of them and actually brought chan_sip to a halt. Is there something environmental that's come to light of late that might be causing this? It certainly begs further investigation in my mind. For those that are getting this msg, a first step at troubleshooting would be to try to figure out what application the given channel whose lock cannot be acquired is in by way of looking at the logs or (if you're lucky enough for it to function with the system in this state) a show channels. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue - how to provide a caller ringing tone when some agent become available
The functionality you're describing isn't currently available in app_queue, but it probably could be done without too much trouble. What does everyone else think about this kind of functionality? Is it useful? Not useful? Thoughts on ringing going on for maybe a longer period of time when the current caller is the next caller in queue but they ringnoanswer to the agent trying to be reached? On 11/15/06, David Hajek [EMAIL PROTECTED] wrote: Hi, I'm a little bit stuck with Queue app. I'm putting callers into the queue and have them hear music on hold when all (static) agents are busy. This is easy. But when agent become available I want the caller to hear a ringing tone (with message that his call has been routed to the support representative). Is this somehow doable? Thanks, David ___ --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 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple queue_log files based on queue - is it possible??
On 10/25/06, Christopher Aloi [EMAIL PROTECTED] wrote: Hello List, Question: Has anyone been able to create multiple queue_log files in /var/log/asterisk for multiple queues? We are designing a multi-tenant system and separating the log files would be useful, instead of dropping all queue actions into one file. Is it possible this is a user configurable option I am missing? No It isn't user configurable. If you want to split them externally you can, but there nothing native to do that at this point. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7971G-GE SEP{MAC}.cnf.xml
On 10/25/06, Kelvin Williams [EMAIL PROTECTED] wrote: I have been forced to introduce a Cisco 7971G-GE into my network, because it has a pretty screen. I have wasted nearly three days fighting with the thing based upon the information on voip-info.org and a few other forums. Asterisk is reporting a 401 Unauthorized. Which typically means bad username/password combination. Unfortunately, all of the usernames and passwords I see in the config file, are as they appear in the sip.conf. Is it possible to get further debug information from Asterisk regarding the 401—ie. What Asterisk doesn't like about it? The chances that the information I have obtained from voip-info.org or the other forums that discuss the 7971G-GE and Asterisk could be incomplete. Would anyone happen to have a working configuration for the 7971G-GE (running SIP70.8-0-4SR1S) they would care to share, or allow me to purchase. I really need to get the child who signs my paycheck his pretty screen. Downgrade the firmware. 8.0.2SR1 is reported to be working at multiple location and I have a 7970 back in my office that is working as well. Multiple folks have reported issues with the later versions of the firmware. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom boot error
On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote: Hi Jesse, Do you know if latest bootrom (3.2.2) and firmware (2.0.1) loads up onto Polycom IP500's? Or are they only for the later models? Do you know if you can still use TFTP for these software updates? They are compatible with the new bootroms and firmwares according to Polycom's release notes, but you cannot use the HTTPS provisioning on the IP500 I believe. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom boot error
On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote: Sweet thanks for that, is there any reason not to go to version 2.0.1 now? I know people were concerned initially because you cant go back but is there a reason to go back if I have a few Polycom IP 500's? We've got clients running 501's on 2.0.1 with a good amount of success. Of those that we've upgraded to date, we didn't really have a reason to rollback with any of them. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipSupply? [Semi-Urgent]
On 10/13/06, Matt [EMAIL PROTECTED] wrote: Hi, Does anyone know what is going on with voipsupply? My sales guy hasn't been online in several days, their 800 number is fasy busy, as are their direct lines. And the canadian store website is down. What the heck is going on? There was a pretty severe early season snow storm up their way today. Lots of trees and lines are down. I'm wondering if this is related to that. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
On 10/11/06, Dinesh Nair [EMAIL PROTECTED] wrote: On 10/11/06 21:15 Joseph said the following: I quits on my as well, when I try to make a second call. There is a bug report on it: http://bugs.digium.com/view.php?id=7972 this seems like a configuration error within FreePBX and isnt really a bug in asterisk. It might be a config issue, but I think you'd agree that a config issue should never segfault the app, and in that respect, we're going to need to do something to get this fixed. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center requirements
On 10/5/06, Todd Houle Asterisk [EMAIL PROTECTED] wrote: Hi Guys- While I played a little with Asterisk a year or so ago, I'm getting ready to roll out a project now that I think is perfect for it. My friend with with a commercial solution he has been very unhappy with and is thinking of replacing it with Asterisk. Below are his requirements. Anything here jump out as a problem? I'm thinking of purchasing a few Digiium card - not sure which we need yet... and finding a box to run it on. The only part I'm not sure is how to address is having the client record auto-appear on screen when the call comes in. I did see plug ins for recording the calls...Is asterisk the best solution for this? thanks Todd Begin forwarded message: From: A. Pathuri [EMAIL PROTECTED] Date: October 2, 2006 2:51:32 AM EDT To: Todd Houle [EMAIL PROTECTED] Subject: Call Center requirements Todd, Here is the brief doc you requested. The process that we need is pretty simple... We get a bunch of DID (Direct Inward Dialing) numbers from SBC. As we get a client, we assign them a DID #. They forward their existing phones to their DID number when their lines are busy or after hours. The DID # is programmed into the telephony system so we can program the caller ID, and enter the appropriate script to pop up when that number comes through. When a call comes in, I would like to have all calls automatically recorded without any of the call agents having to press a record button for each call. We also current have conference call functionality where we can connect one caller to another caller (used when the ER needs to speak to a doctor). Ideally also, I would like the recorded calls to sort by client and store in the appropriate clients folder, which then can be automatically zipped and sent via email to the clients inbox at any desired interval. We are also developing a web-based app where the details of each call can be entered ( a sort of call log) so the clients can also log into a web interface and see the details of each call (currently, most clients get their call logs via fax in the am and at midnight). It would be great if somehow, the caller ID on the server/astericks can automatically pull up the appropriate clients profile from our web app, so the details can be entered into the correct profile. Otherwise, for each call that comes in, the call agent has to pull up the clients profile while the caller is on the phone, before s/he can take down the details of the call. This is really rough, but I hope it gives the basic idea. We can discuss in further detail once you take a look at this. Ofcourse, as well it would be great to be able to setup a co-location in India utilizing the same infrastructure. There are a number of ways to do this, but given the application it appears to be (medical), and additional requirement not mentioned here (and quite possibly the most important) is HIPPA compliance with regard to security of who has access to what information. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unable to call ATT audio conference bridge
On 10/4/06, asterisk-user [EMAIL PROTECTED] wrote:
Hello,
Can someone help me with this please?
Attached is the log file.
thank you
Original Message
Subject:[Fwd: asterisk-users Digest, Vol 26, Issue 166]
Date: Fri, 29 Sep 2006 10:31:21 -0400
From: asterisk-user [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
I tried by adding answer() to the dial plan but the problem still exists.
I am not sure if I am doing this right.
Attached is the log file from asterisk while making the call to the conf
bridge after adding answer()
Could you please let me know if you find anything out of this log file?
thanks for your help.
I'll assume from your macros set your using AMP or FreePBX. That
being the case, add the following to your from-internal-custom
context.
exten = _1800XXX,1,Answer()
exten = _1800XXX,2,Dial(Zap/g1/${EXTEN},240)
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Re: [asterisk-users] Queue AddQueueMember()
On 9/28/06, Douglas Garstang [EMAIL PROTECTED] wrote: All, I've recently been told that the AgentCallBacklogin() application is buggy, and I should not use it. Apparently I should use AddQueueMember() instead. I see though that AddQueueMember() does not take the location to call back as an argument. We have remote agents that are available via PSTN access only. With AgentCallBackLogin() they can enter their PSTN phone number, and Asterisk will call them back at that number when they get a queue call. Can AddQueueMember() do that? Is AgentCallBackLogin() going to be deprecated at some point? Will AddQueueMember() be improved to match the call back functionality of AgentCallBackLogin()? AgentCallBackLogin() is deprecated beginning with 1.4. You can use AddQueueMember() in combination with the Local/ channel to do what you're looking to do above. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unable to call ATT audio conference bridge
On 9/28/06, asterisk-user [EMAIL PROTECTED] wrote: Hello, I have a problem with asterisk and trying to see if someone can help me fix the issue... Problem: I couldn't join ATT's Tele Conference bridge directly without their customer service interaction. Instead of getting the automated prompts to join the conference, it takes me to the customer support and then I got to give them the bridge number and pincode to add me into the conference call. The reason given by ATT was that their conference system is unable to identify our tone. This happens only with ATT conference bridges... not sure what the problem is. This problem started after I installed trixbox on a new hardware. Previous setup with [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] did not have this issue and I even switched back to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (a different box) and called the same conf bridge... that worked fine. I am running trixbox with the following versions: asterisk - 1.2.9.1 zaptel - 1.2.8 libpri - 1.2.3-1.349 using zap over a 8 channel pri Thanks in advance. ATT's IVR to collect the passcode is coming through as early media and since you haven't signaled to the phones that the phone is answered they're probably not letting you send DTMF through the bridge that isn't technically supposed to be there yet. Put an Answer() in your dial plan prior to sending the call out to the Dial() application to reach the bridge for these types of calls and this generally fixes your problems caused by someone else not signaling correctly. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue failover and wrap time
On 9/25/06, Michelle Dupuis [EMAIL PROTECTED] wrote: I have a asterisk box with some queues for a call center and need help on two points: 1. I have a scenario where if a queue has no agents logged in, an inbound call should immediately failover to the failover destination for that queue. However, this does not seem to be working in that, even if no agents are logged in, the call goes into the queue. Is there a config option I'm missing (or did I misunderstand how the failover works?) 2. I have the wrap time set ideally for agents, but sometimes they want to pickup the next call in queue before the wrap time expires. Is there a way for agents to grab the next call? 1) joinempty=strict 2) Not at the preseent time, no. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe
On 9/21/06, Denis Galvão [EMAIL PROTECTED] wrote:
bweschke, is there any news about using astdb to store the numbers to
be dialed?
This is related to this note on bug http://bugs.digium.com/
bug_view_advanced_page.php?bug_id=5574:
(0035684)
shmaltz - reporter
11-02-05 15:01
Also thinking about it a bit more, it would be very usefull if the
settings in followme.conf would allow for an entry that points to the
astdb, something like this:
number = family/key
number = family/key
This will allow to use the dialplan to update the number values in
followme.conf, like this:
exten = _*5X.,1,Set(DB(FM${EXTEN:2:1}/${CALLERIDNUM})=${EXTEN:3})
In which case users can call in *51 follwoed the number to follow,
and the DP will add that to FM1, for the second number they would
dial *52 and that would be added to FM2, and so on. In followme.conf
you would then have:
number = FM1/8143
number = FM2/8143
and so on.
And if an entry in the astdb is empty app_followme will ignore it.
Maybe I'm pushing it, but this feature (since it can all be done in
the dialplan without this app) might not have value if one can't use
it because s/he can't have her/his users call in to change the number
values in followme.conf. In which case it means not being able to use
this.
Denis,
There was some discussion around this feature in app_followme in the
IRC chat rooms and it was decided that for at least the 1.4 release of
app_followme, the group wanted to keep it simple and not have this
feature.
We may add this in the future, but I can tell you that I do plan to
realtime enable the application for the 1.6 cycle which probably
gives you more or less the same functionality.
BJ
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Re: [asterisk-users] CURL
On 9/21/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 21, 2006 at 08:41:37AM -0700, Elpidio Ramos wrote: Ok, after requesting information to digium (no answer yet) and being informed that asterisk-dev is *NOT* a support hot line, I am trying in this list to see if someone has information on this regard. I know this is not a support area so I am only trying to get some clues. I have asterisk be and I am trying to use the CURL function (or application?). It is not available when I try it even though it is documented. Does anyone knows if there is a way to load it as a function/application inside asterisk? if so, is there code to download/compile to get it working inside asterisk? Any clue will be highly appreciated. (I keep trying digium support). Do you actually have the application and function curl? If you do not have it, it may be because libcurl (or its development package) was not availble. You may need a package of the sort of curl-devel or libcurl-dev installed on your system. You do indeed need libcurl-dev in addition to libcurl in order ot have support compiled in. If you don't have libcurl-dev on your system, the configure process will not find the required header files to build the modules. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe
On 9/22/06, C F [EMAIL PROTECTED] wrote: BJ, I believe that asteiskdb is before realtime. It does not give the same functionality, since asterisk apps can only update asteriskdb thru the DP, and built in commands. There was some discussion around this feature in app_followme in the IRC chat rooms and it was decided that for at least the 1.4 release of app_followme, the group wanted to keep it simple and not have this feature. We may add this in the future, but I can tell you that I do plan to realtime enable the application for the 1.6 cycle which probably gives you more or less the same functionality. Doesn't the new func_realtime allow you to read/write realtime values through DP functions? I believe that it does. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-dev] To bweschke regarding app FollowMe
On 9/21/06, Denis Galvão [EMAIL PROTECTED] wrote:
bweschke, is there any news about using astdb to store the numbers to
be dialed?
This is related to this note on bug http://bugs.digium.com/
bug_view_advanced_page.php?bug_id=5574:
(0035684)
shmaltz - reporter
11-02-05 15:01
Also thinking about it a bit more, it would be very usefull if the
settings in followme.conf would allow for an entry that points to the
astdb, something like this:
number = family/key
number = family/key
This will allow to use the dialplan to update the number values in
followme.conf, like this:
exten = _*5X.,1,Set(DB(FM${EXTEN:2:1}/${CALLERIDNUM})=${EXTEN:3})
In which case users can call in *51 follwoed the number to follow,
and the DP will add that to FM1, for the second number they would
dial *52 and that would be added to FM2, and so on. In followme.conf
you would then have:
number = FM1/8143
number = FM2/8143
and so on.
And if an entry in the astdb is empty app_followme will ignore it.
Maybe I'm pushing it, but this feature (since it can all be done in
the dialplan without this app) might not have value if one can't use
it because s/he can't have her/his users call in to change the number
values in followme.conf. In which case it means not being able to use
this.
Denis,
There was some discussion around this feature in app_followme in the
IRC chat rooms and it was decided that for at least the 1.4 release of
app_followme, the group wanted to keep it simple and not have this
feature.
We may add this in the future, but I can tell you that I do plan to
realtime enable the application for the 1.6 cycle which probably
gives you more or less the same functionality.
BJ
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] Channel kept busy when creating ssh tunnel via AGI
On 9/20/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I have a problem with Asterisk AGI command. I wrote a script which launches a shell command. If I launch a normal command for example like ll /tmp/tmp.txt, the AGI command launches the shell commands and then exits. The problem is when I launch THIS command to create an ssh tunnel in background: *ssh -f -N -l asterisk -R 2050:localhost:22 192.168.0.1* The tunnel command above works well if launched via shell but if I launch it using the AGI script, it opens the tunnel but leaves a (SIP or ZAP) channel in use (I checked it typing SIP/ZAP SHOW CHANNELS). The channel closes only when I kill the tunnel process. After killing the process Asterisk console shows: -- AGI Script tunnel.py completed, returning 0 Is there anybody who knows why the channel remains busy? The intent of an AGI script is to have a script/executable that interacts with the channel. As such, the channel will hang around waiting for input from, providing feedback to, and the eventual completion of such script/executable. If you don't want such behavior, you might want to take a look at monitoring a specific channel event in the Asterisk manager and then starting off your script upon the receipt of such an event through the manager. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress a server with a tool
On 9/20/06, nik600 [EMAIL PROTECTED] wrote: hi is there any software usable to simulate work on an asterisk server? I'm interested in it to evaluate the level of currently calls that a server can support For SIP, see http://sipp.sourceforge.net/ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote: Doug Lytle wrote: Steve Totaro wrote: I am trying to connect a Definity G3 to an asterisk system. I had it working OK with the exception of the caller ID on the Definity handsets just He wants to know if your Definity is an S, SI or an R? Doug I am not sure, it is a refrigerator sized unit with three cabinets. The manual was printed in 1988 and is ATTT Definity Communications System Generic1 and Generic3. If you need more, I can go to the remote system and provide more details. I REALLY appreciate you helping on this. Sounds like a G3R. How are you signaling between the two? PRI? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote: BJ Weschke wrote: On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote: Doug Lytle wrote: Steve Totaro wrote: I am trying to connect a Definity G3 to an asterisk system. I had it working OK with the exception of the caller ID on the Definity handsets just He wants to know if your Definity is an S, SI or an R? Doug I am not sure, it is a refrigerator sized unit with three cabinets. The manual was printed in 1988 and is ATTT Definity Communications System Generic1 and Generic3. If you need more, I can go to the remote system and provide more details. I REALLY appreciate you helping on this. Sounds like a G3R. How are you signaling between the two? PRI? Yes. I have two ISDN communication processor links setup which I later read on google was not supported. Now I cannot remove the new entry. I just get identifier not assigned What you'll want to do in order to set this up, is build a trunk group in DSA that consists of the two T1's, and then you'll want to build a signaling group that uses isdn signaling and then (probably 3-4 pages over in DSA) you'll want to assign all the channels of those T1s as members of that signaling group. On the asterisk side you'll want to setup the trunks as pri_net. Put a T1 crossover cable in between, if there's no carrier loops involved, and then you're done. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Firmware
On 9/13/06, Forum [EMAIL PROTECTED] wrote: Unfortunately they pointed me back to Polycom and I have not yet heard back from them. Can somebody post a link to download sip2.0.1? Your reseller should not be pointing you back to Polycom if they are a certified reseller. The download is available to all certified resellers via Polycom's extranet. However, there isn't a publicly available link. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rack for Asterisk with TDM2400 Digium board
On 9/12/06, Antoine Megalla [EMAIL PROTECTED] wrote: Hi, I have a client who wants a call center with 16 analog FXO modules. I offered him a solution with a 1U or 2U rack and Digium TDM2400 card. I know that there mother board compatability issue with the Digium cards, so can anyone suggest a barebone 1U or 2U server (I prefer the SuperMicro Superservere series) with a Mother board that is compatible with the Digium TDM2400 card (which is a Full Length PCI card). I wouldn't recommend anything less than a 4RU machine from SuperMicro with the size of the TDM2400 card. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use PauseQueueMember
On 9/8/06, gc [EMAIL PROTECTED] wrote:
After using PauseQueueMember in my dialplan. I used 'show agents' cli to
show the agent status. It is still show that agent available. Here is the
output from asterisk console:
5156597 (Agent1 ) available at '[EMAIL PROTECTED]' (musiconhold is
'default')
5156598 (Agent2 ) not logged in (musiconhold is 'default')
Although Agent1 is indeed in pause mode.
Here is my dialplan:
exten = 881112,1,PauseQueueMember(|Agent/${CALLERIDNUM})
exten = 881112,n,Playback(vm-goodbye)
exten = 881112,n,Hanup
Am I doing somthing wrong?
I am using asterisk 1.2.9.1
Agents are agents. Not queue members. You need to do show queue
in a queue where that agent is a member to see them paused.
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