Re: [asterisk-users] What version to upgrade to...?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 12/11/2011 10:59 PM, Mike Diehl wrote: Should I go to 1.8.x? Or all the way up to 10.x? This is a production system and I can't afford to be testing code. The 1.8 series is the current LTS release. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFO5gQWCFu3bIiwtTARArU9AJ9/ZWb5uyjqjBFKqyjZa4X1+2fC+wCfVHP1 KY1D7w1siMJtCd1Ktxffwy4= =PbAV -END PGP SIGNATURE- -- _ -- 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] Custom Dialplan
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 08/05/2011 04:32 AM, Richard Zulu wrote: I would like to import my dialplan into freepbx+asterisk since I am switching to that...how can I create my own custom dialplan in freepbx? I'm not sure why you'd want to... freepbx is anathema to custom dialplans. That said, I believe you end up naming your extensions.conf file to extensions_additional.conf and freepbx will pick it up when it starts. It's been a long, long time since I've dealt with freepbx -- in fact I went the other way: from freepbx+asterisk to pure asterisk. When I was using freepbx that was the solution you seek. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFOPcAxCFu3bIiwtTARAkjKAKCPCgcoaRyPNs7BXhge7xxcy7C2qQCdF6hx 2Bwz/YEUSbKFsfzD9V0xX6Q= =W2Dn -END PGP SIGNATURE- -- _ -- 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
[asterisk-users] Voicemail not acting as documented.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 In http://svnview.digium.com/svn/asterisk/branches/1.8/CHANGES?view=markup, line 180 states: Voicemail now runs the externnotify script when pollmailboxes is activated and notices a change. My voicemail.conf configuration for my LDAP vm storage is thus: externnotify = /opt/asterisk/bin/mwi.pl pollmailboxes = yes pollfreq = 30 The script is called whenever I leave a voice mail as well as when I listen to the voicemail via the voicemail() and voicemailmain() applications. When I listen to a voicemail using an email client the script is not called. My impression from that line in the CHANGES document is that it should. Is there some other parameter required to get this to fire or am I reading more into that sentence from the CHANGES document than is actually there? Thanks. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFOMa2yCFu3bIiwtTARAsbIAJ9v2QrOxPqJno14nAwo2Gxqd8l7agCgmy6/ EVysWQqaOEXUW6+vxbnahVE= =cfLA -END PGP SIGNATURE- -- _ -- 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] Voicemail not acting as documented.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 07/28/2011 02:42 PM, Barry L. Kline wrote: Is there some other parameter required to get this to fire or am I reading more into that sentence from the CHANGES document than is actually there? Sorry for replying to my own post, but I've done some more investigating. I glanced through the source for app_voicemail and am beginning to wonder if there need be a physical SIP device configured to use that mailbox for the mailbox to be polled. Is that the case? This Asterisk installation is acting as a VM server for a legacy phone system and none of the VMboxes are actually connected to a SIP phone on this box. Can this be the source of my problem? Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFOMb2YCFu3bIiwtTARAoKrAKCeglvi3UEibUjuAmL4exhec71qZQCePJG4 hbsyA9EsMbuaPlyMut9PabM= =IJo6 -END PGP SIGNATURE- -- _ -- 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] Voicemail not acting as documented.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I found what I believe to be a bug and have submitted it: https://issues.asterisk.org/jira/browse/ASTERISK-18207 Please correct me if I'm wrong. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFOMdzTCFu3bIiwtTARAimrAJ92+GFM1d1WjI5ne7BET2WbBLkuEgCfXs+7 niQbzxp+9BOOy0SyGZkerqU= =bB0q -END PGP SIGNATURE- -- _ -- 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] Call to *2*999... : IP-phone configration
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 06/21/2011 08:37 AM, Jonas Kellens wrote: At the moment, I don't really know what I'm looking for. So if anyone knows how to do it in a Cisco, Grandstream, Yealink or Snom IP-phone I can find out myself what settings to look for in other IP-phones. On a Polycom phone you'd be looking for the 'digitmap' to make the adjustments. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFOAMPhCFu3bIiwtTARAtfWAJ0YD3MYaYHCeFabf6QdTvc8oRCnFQCfbgtk 8Mfp7Pmo/G4FTsueqWNpdEc= =6eo/ -END PGP SIGNATURE- -- _ -- 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] Upgrade and recompilation
On 02/01/2011 12:34 PM, Harel Cohen wrote: As one with theoretical knowledge in programing, but never on Linux, I can understand terms and code structure but I don’t know: 1. What shell commands (e.g. ./configure, make, make install etc.) should I run to recompile Asterisk (same version)? 2. What shell commands should I run if I want to apply a change to source code? 3. Is there a general guide on how to upgrade Asterisk? Read the README file included with the source. Barry -- _ -- 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] Follow-me to my answering machine :-(
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bryan Jacobs wrote: I wonder if all the cell providers let you do this? I presume you mean turn off voice mail. I don't know, but the first time I called Verizon to have it done the gal I spoke with said it couldn't be done. So I said thanks, called in again, got another rep and he said no problem. In less than five minutes I was good to go. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFL2yNWCFu3bIiwtTARArEgAJ9TMJK0qgu/GkapCgjK+zPT+crHaACfQ03X BbTtSecEA2Ahuiqwws+2l10= =hjFW -END PGP SIGNATURE- -- _ -- 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] Follow-me to my answering machine :-(
Bryan Jacobs wrote: I can't just call the car - the car is my cell phone DID with a bluetooth kit. I did this same thing you're attempting. I have a desk set at home, a Polycom in my office and my cell phone all being called at the same time. I called Verizon and had them disable voice mail on my cell phone so that the only voice mail system I use is my Asterisk box. I no longer give out my cell phone number but only my home phone number and allow Asterisk to do all of the heavy lifting. Oh, and I set the caller*ID outbound to the caller*ID of the inbound call so I can still see who it is. Barry -- _ -- 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] Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 DHAVAL INDRODIYA wrote: where as I am using Asterisk 1.6.0.5 and my machine is using *safe_asterisk* script asterisk running Why are you using such an old version in the 1.6.0 branch? 1.6.0.25 is current, upgrade to there and then worry about the problem if it recurs. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLoSMvCFu3bIiwtTARAvDNAJ4ql+42gKH20vMAJLNsYVxqqOhMjgCfRuF9 R6QAJbu5ZSHmJVSkO7UErmY= =VYx9 -END PGP SIGNATURE- -- _ -- 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] Getting verbose or debug tracing in Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tim Culhane wrote: Here is my output of 'sip show peers' user1/user110.41.3.12 D N 10434Unmonitored user2/user210.41.3.12 D N 65293Unmonitored user3/user3(Unspecified)D N 5060 Unmonitored user4/user4(Unspecified)D N 5060 Unmonitored 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 0 offline] So, does this mean the registration worked? What is the difference between monitored and unmonitored? Tim user1 user2 have registered. user3 user4 have not Unmonitored means that you have not specified qualify=yes in the peer configuration. Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLjoj2CFu3bIiwtTARArj4AKCh99NSCRHISUuNv/G72zGERoj8fwCfXpIv nuD43cWZ3m9k8TxFDhx/vdo= =Zptj -END PGP SIGNATURE- -- _ -- 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] 911, location
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. If you're using POTS lines to make the call to 911 they'll know the location, if the POTS lines come into the building that you're calling from. Are you saying that these lines are located in a different location? Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR UnWTQQ1anTXtDqfk54QVj/k= =LtAE -END PGP SIGNATURE- -- _ -- 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] Really Silly Question From Total Newbie
UIT DEVELOPMENT wrote: Sorry for what might seem as really silly questions, but I am not sure how to proceed. Thanks in advance for any insight that you folks can provide! Hello Mike. Welcome to the wonderful world of Asterisk. Before you sludge through a GUI and all the attendant bad habits that can produce, I suggest that you download what we consider to be the Bible of Asterisk. The infobot on IRC says: thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org; Download that and read the first few chapters. It will make your Asterisk experience a lot more enjoyable and will help you understand what you're doing. This list, and the IRC channel #asterisk, are good resources when you finally get to the point where you're stuck and need some help. Regards, Barry ___ -- 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] Can't get G.729 to work...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ben Schorr wrote: I’ve got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I’ve set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows “(ULAW)” (G.711) as the codec in use. How about in sip.conf? Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLJ9csCFu3bIiwtTARAnDfAJ9QL8xqGZYgeHyFwhX7Ebz+h7UVYQCdEt5k af0y9vqC4WV8CmdAN0D0ASE= =++1O -END PGP SIGNATURE- ___ -- 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 Queue Dialplan
Daniel Stefanus wrote: Hi, I want to reconfigure my asterisk dialplan.I have a problem.I have 4 agents in a queue.How is the configuration for the asterisk dialplan if I want to have only 4 agents maximum who can receive the phone,so if the fifth caller try to entering the queue they will be noted by my IVR that all our agents are busy?Thank you so much for this millis,it really helpful especially for a newbie like me. Best Regards, Daniel What do you want to have happen? Normally you put the caller into the queue and when one of your agents become available the caller will be sent to him. If you don't want to put the fifth caller into the queue then I'd suggest looking at the GROUP* functions to keep count of the number of callers. Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR Prompt Recording
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 David Gibbons wrote: This may belong on -biz, but does anyone have experience with a decent and cheap IVR/prompt recording house? Are decent and cheap mutually exclusive? A nice *sounding* lady would be nice... you can keep any burly voice studios to yourself :) I use Allison. www.theivrvoice.com -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLJrn8CFu3bIiwtTARAoFmAKCfrzh1vXWv3isJdPjEGFbEHYLUvwCfSsCd piy49OgEVTFou+fizd25bmo= =g2iu -END PGP SIGNATURE- ___ -- 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] Queues without agent login
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 jonas kellens wrote: Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? Make the phones members of the queue. In queues.conf: [MY_QUEUE] member = SIP/1234 member = SIP/5678 etc. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLBBYACFu3bIiwtTARAg5oAKClAtJ98LaSXnjCDBx4xlRcLQ9l/wCgoeI+ BAi7wu2nQ6vNPZSaLCDB4DA= =egpe -END PGP SIGNATURE- ___ -- 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] max call duration
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 B.Masoud @ SH wrote: How can I set a maximum call duration on a ZAP channel? Look at the parameters on the Dial application. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLAqoTCFu3bIiwtTARAiULAJ9E3g1x5lY5yspsXVKgz3yAFFOAqgCfV9Fy GnifFRJRrv98EWIgzK+RvKw= =UT+T -END PGP SIGNATURE- ___ -- 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] Database postgresql not able to start
James Texter wrote: I found that on a clean boot, I could not connect to Postgresql either. In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the module, and that seems to work. After bootup, cdr_pgsql.so is able to connect immediately. This sounds as though you have Asterisk starting before PostgreSQL. If you're using CentOS or RHEL or other RHEL-inspired distro look at /etc/init.d/asterisk and /etc/init.d/postgresql. Compare the line in each that looks like: # chkconfig: 2345 xx yy The 'xx' is the start priority. If the number is lower in the asterisk file than it is in the postgresql file then that's your problem. You need PG to start before Asterisk. man chkconfig for further details on what you can do. Barry ___ -- 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] Libpri-1.4.10.2 Released
Karl Fife wrote: Perhaps there's an arcane way to query lipbri the older releases from the CLI? Can anyone speak to that? Quick and dirty: strings /usr/lib/libpri.so That's CLI, tho' not the one you're talking about. Barry ___ -- 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] Request for Review: Building Queues with Asterisk
Leif Madsen wrote: Please review and let me know how it goes for you! Where is it? Barry ___ -- 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 configure softphones in asterisk
aster...@opensourcesolution.in wrote: Hi, hi all.Iam new to VOIP so plz forgive me on asking stupid questions. I have installed Asterisk on Centos 5.3, and dowloaded X-LITE softphone on two windows machine. now i want to start from very basic scenario, i want to make two X-Lite phones communicate through asterisk. guys plz plz tell me what r those impt files in asterisk so that both softphone (X-Lite) should start communicating.any support and guidance will be highly appreciated. Regards, Pawan Would you please stop sending this and just do the research. May I suggest: Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org Everything you need for this is in this book. ___ -- 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 know AMI status
velusamy velu wrote: Dear All, I have installed Asterisk 1.6.1.9 to use Bridge Application in AMI. After inatallation I have tried to connect the AMI via telnet. But it didn't connected. I used netstat to know the listening socket. But it was not available. How to start the AMI server socket. Please any one help me... Did you make the necessary changes to manager.conf? Barry ___ -- 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] IMAP voicemail using subfolders fails.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Darrick Hartman wrote: I don't think that Maildir or a database backend solution (such as Exchange) suffers from this same limitation. Maildir makes sense, but the text I quoted in an earlier message is now no longer part of the imapstorage text. I moved to that last evening and we'll see how it works out today. I would be more interested in knowing how sensitive this would be to latency if using an IMAP server that isn't on the same device as the Asterisk server (or perhaps even a remote IMAP server)? - From what I can see, the file is downloaded locally and then played. So I don't think that latency is an issue. I'm not 100% on that but based on what I read in the docs that's the general feeling that I get. Obviously the best solution if I want to be sure is to convert our inbound to Maildir format. Glad you made it home okay from the flight! Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFK4EgQCFu3bIiwtTARAsO9AJ9Mw6H1uGaVDdotKGlJ7hUWDMfwEACfRhDf 5so8No4MOtdOC93piRff8jI= =4C8D -END PGP SIGNATURE- ___ -- 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] IMAP voicemail using subfolders fails.
Kevin P. Fleming wrote: It's not present in the current 1.4 doc/imapstorage.txt file, or any later version. I don't even know why the storage format would matter, since that would be very specific to the IMAP server that is managing that folder. Hmmm http://markmail.org/message/up3rfmdk2kjf6r7y is a link that contains the contents of a README file that looks like it came from Digium. About half-way down is: -- Mailbox Format -- Mailboxes should use the mbx mailbox format. The mbox format does not support concurrent access to mailboxes, which can cause deadlock or strange behaviors. You can convert mailboxes from mbox to mbx using mailutil: Perhaps that came from a different product? I think that I'm going to just go ahead and implement IMAP VM and see what happens. Thanks very much Kevin! Regards, Barry ___ -- 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] Astricon
Randy R wrote: I missed the first part of this, but has anyone said: not all the presentations were recorded. Hi Randy. Yes, that was mentioned. Actually, three of the four tracks were videotaped IIRC. Barry ___ -- 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] Astricon
Darrick Hartman wrote: It would be great if you could make more of the talks available to those that attended the conference. I know there were a few times where two interesting talks happened at the same time. I have to agree John, I'd love to see the videos of the sessions that I missed. It's either that or I need to figure out how to clone myself next year. BTW, you did a great job with Astricon! It was my first one and it won't be my last. Barry ___ -- 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] IMAP voicemail using subfolders fails.
Kevin P. Fleming wrote: That's a bug; the IMAP folder prefix should not be used in construction of prompt names to be played back. Please open an issue on issues.asterisk.org reporting this problem ... Done: https://issues.asterisk.org/view.php?id=16104 On a side note Kevin, my INBOX is in mbox format. The 1.4 imapstorage.txt file used to contain an admonition to use maildir format for storage. I can't seem to find that warning in the 1.6 branch. Is this no longer a problem? Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IMAP voicemail using subfolders fails.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I am running 1.6.0.15 and am trying to get IMAP storage working. I have had no trouble doing so, except that I wish to create a subfolder in my account for voicemail, such that I have: #voicemail/ INBOX Old Family Friends Work I can set IMAPFOLDER=#voicemail.INBOX in voicemail.conf and successfully get voicemail to work as expected. Messages appear in INBOX and are deleted if removed from the phone. If, however, I attempt to change folders with option 2, I get the following error: file.c:950 ast_streamfile: Unable to open vm-#voicemail.INBOX (format 0x4 (ulaw)): No such file or directory Clearly, app_voicemail is looking for vm-INBOX and is building the voicemail prompt file name based upon the voicemail folder. I attempted to symlink vm-INBOX.gsm to vm-vm-#voicemail.INBOX.gsm but that didn't help, either. I have tried using the combination of: IMAPPARENTFOLDER=#voicemail IMAPFOLDER=INBOX but in this case the VM system can't find the messages and the voicemail app simply dies. So that isn't the right incantation. Surely, it's possible to do what I'm looking to do, isn't it? So my questions are: How do I configure app_voicemail to use IMAP subfolders? and I have used '/' as the delimter as well as the '.' character. Am I using the wrong one and if so, what is the correct one? TIA, Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFK3NotCFu3bIiwtTARAgXoAJ48uOG0DgJzjM2mtnjbqbZMJ/nWrQCfYxLJ yH55gWUN5iBp2+bagDHIklA= =oJXN -END PGP SIGNATURE- ___ -- 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] IMAP voicemail using subfolders fails.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Barry L. Kline wrote: I know that it is bad form to reply to yourself, but here is the current state of affairs: file.c:950 ast_streamfile: Unable to open vm-#voicemail.INBOX (format 0x4 (ulaw)): No such file or directory I'm thinking this is a bug. I just changed the delimiter to '/' and then the file that * attempted to open was vm-#voicemail/INBOX. I created a subdirectory vm-#voicemail and linked vm-INBOX.gsm to vm-#voicemail/INBOX.gsm and now I get my proper prompt for changing folders. Either I'm attempting to do something funky with my configuration or the code needs to look for vm-INBOX.gsm regardless. Any suggestions? Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFK3PD2CFu3bIiwtTARAqmuAJ9pSMfXlznArL7i/sUftbrarIKnowCbBB7E ywJVO3DMyWQlwG21KWqTGIU= =+V7p -END PGP SIGNATURE- ___ -- 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] IMAP voicemail using subfolders fails.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John A. Sullivan III wrote: I can't help you directly but I can share my experience with folders. I intentionally did not set up the folder structure in IMAP as recommended in the documentation. To my pleasant surprise, when the folders were needed (e.g., a user moves a voice mail via the voicemail application to friends, etc.), they were created on the fly, i.e., Asterisk created them within the IMAP folder system. I am using 1.6.1.6 with Zimbra as the backend - John Thanks, John. I didn't see that in the docs. I am going to do what you suggested and just let Asterisk put things in the root directory. Did you perhaps use the INBOX or are you using a custom folder? Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFK3PGCCFu3bIiwtTARAnSMAJ4m4PJnV73DsJEU6Y7X2hzJ6u+r0QCeM+76 4U9Gk9wGlUCB0Xp/UpYpmzQ= =Ys8S -END PGP SIGNATURE- ___ -- 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] IMAP voicemail using subfolders fails.
John A. Sullivan III wrote: I'm using the INBOX - John I'm throwing in the towel and going that route as well. Now that I'm not trying to swim upstream things are working well. Thanks John. Barry ___ -- 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] dahdi dies with No more room in scheduler
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 James Lamanna wrote: This is with dahdi 2.2.0 and asterisk 1.6.0.10. Any ideas on this issue? Check to see if this is a bug that has been fixed in 1.6.0.10. I think the current is 1.6.0.15 and there has been significant bug fixes since your version. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKyhoUCFu3bIiwtTARAsUjAJ44sqcqVk3VtrecJVVgjr3LlwtTJgCeP4qH jxsq2fuwqEb+qlrxziOEye4= =CcCX -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Voicemail to email transcribed
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 C F wrote: I have seen lots of companies offering this as a service and have used phonetag.com in the past. They work very nicely, however I have a customer that is not interested in paying $30-$40 a month but would rather buy the software. I have googled and googled all I can come up with are companies that do it as hosted. Does anyone on the list know of software that can transcribe an email/voicemail sent to it and then forward it to the end user? TIA We've talked about this on the VoIP users conference and the feeling is that most likely there is a human on the back end who is doing the transcription, if it is to be at all accurate. Anyone who has worked with the various speech-to-text software, such as ViaVoice or Dragon, knows that training is the key to really accurate transcription. That, and a good quality audio signal. The variations in audio quality you get in voicemail is probably too great to do this all with software only. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKuO3KCFu3bIiwtTARAqWbAJwOhC3REHwQphVpXrG+XfGKwq3ccwCfRoO4 nOv6mEfTV4rQst89YU7/Wpg= =HpnP -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] CDR to Postgres Centos
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ABBAS SHAKEEL wrote: Can i know which querry is executed to insert record to database... i am asking this because of [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:309 pgsql_log: Failed to insert call detail record into database! [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:310 pgsql_log: Reason: ERROR: syntax error at or near ) at character 17 [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:311 pgsql_log: Connection may have been lost... attempting to reconnect. [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:314 pgsql_log: Connection reestablished. [Sep 1 12:46:09] ERROR[19498]: cdr_pgsql.c:320 pgsql_log: HARD ERROR! Attempted reconnection failed. DROPPING CALL RECORD! Which version of Asterisk are you using? Did you create the PG database for Asterisk? Have you confirmed that you can connect to it using the CLI psql with the appropriate credentials? There are a few steps ahead of where you are before we worry about this particular problem. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKnRL4CFu3bIiwtTARAnwNAJ9+CiWdtq17DRSqelNl7bsN5pS32gCeIn+l VNyWYBauMOBvVMhyGUeP/Pk= =G9NP -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] CDR to Postgres Centos
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ABBAS SHAKEEL wrote: but when i execute this ./configure --with-postgres=dir where postgresql is installed it gives an error for missing an pg_config file . i searched the PC but it really dont exists. but database server is fine and working OK! what to do in this situation You should have the following packages installed on your Asterisk system postgresql-libs postgresql-devel postgresql If the database is on the same box, include: postgresql-server If you want to hit the database from the dialplan for any reason, include: postgresql-odbc Once you install these, be sure to rerun ./configure Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKm8rOCFu3bIiwtTARAijbAJ4vt0DVZJYUPRhPrNpXm2KEngRmxACgn24T aHtpBzyGhPBmw8a4veqdLhQ= =TI+m -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.6.0.13 with realtime DB , issue with MWI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 laurent schweizer wrote: how can I indicate to asterisk that number of message has changed and that he need to do an update. I'm not sure what effect realtime DB has on it but did you notice the voicemail.conf parameters: pollmailboxes pollfreq ? Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKlm4DCFu3bIiwtTARAkv8AJ4yCegZaKwJYdxMKQDqa+dON/aZNwCdHWCW zuGvLPLuPBEVXXFB4k4OTnA= =5QPC -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] how does wrapuptime work in queue.conf
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andy Kuo wrote: Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Andy -- wrapuptime is the number of seconds that Asterisk waits between the time a agent hangs up with a caller and the next time that Asterisk sends a call to the newly-available agent. Wrap up time gives the agent a few moments to complete his last call and prepare for the next. What you need to do is use Playback() for your advertisement, then Queue() the call. Otherwise it acts just as you said, provided an agent is available. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKltbjCFu3bIiwtTARAjE0AKCGFEchqYoGWyaeHqlIH+iNyzBKygCgqibn X/gSnE7W7EHnwiUpRC1FLRs= =pdMh -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] how does wrapuptime work in queue.conf
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Andy Kuo wrote:
Hi Barry,
Thank you for the hint, but I forgot to mention that we have a few
advertisements, and we want the callers to listen to only one at a
time, and in a round robin or random order. Using Playback() doesn't
seem to serve that purpose. Is there any better way to achieve that?
Use the RAND function to generate or pick a filename.
exten = Set(advert=advert${RAND(1,10)})
exten = Playback(${advert})
That of course assumes that your advertisements are in files named
advert1.xxx through advert10.xxx (where xxx is wav,sln,etc)
Barry
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Re: [asterisk-users] Multiple user registration ...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mauro Sergio Ferreira Brasil wrote: We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -, but also receive INVITE requests on all devices if someone calls this user - yeah... here the thing gets creepy. The demand is quite simple: let a user registrate with multiple devices using the same SIP user on such way that if someone call him, all these registered devices will ring and the first to take the call will be the lucky one. Instead of trying to make Asterisk do this unnatural act, why not register each device with a separate id, then use the dial function to call all of them? e.g.exten = 122,1,Dial(SIP/1SIP/2SIP/3) You could use some creating scripting to decide which devices to ring based on the dialed extension. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKlU65CFu3bIiwtTARAu0DAJ4szfX1dp/BNZojIKhgIL/tIhkjvQCeLXCf A+Dys6+LgrNhL/zQpU8Vuwk= =1Y6q -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Multiple user registration ...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mauro Sergio Ferreira Brasil wrote: I totally agree with you that this is an unnatural behavior, but I have to agree as well with our commercial staff because their vision was naturaly translated from our telephony world (we don't have a different ID - telephone number - to each phone we have home, right ?). Well, our phones at home are probably analog and can be connected in parallel. Unfortunately, VoIP phones are a different matter and need to be identified individually. I guess I don't get the problem your commercial side is having with this concept. You can produce the same result doing things within the constraints of SIP using the features built into Asterisk. Doing what you want may be possible with a bunch of contortions, but it's going to be an unnatural act fraught with tons of unexpected behavior. If you do get it working the way you describe you'll likely be doing so because of a side-effect behavior in a GIVEN version of Asterisk. The moment you change versions, the side effect may or may not be the same and you may find yourself in the same trouble. I can't offer anything more to help you except to wish you the best of luck. You're going to need it. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKlWonCFu3bIiwtTARAu1WAJ0eS2Eh6n6Tici9eDA82UIesuozNACaA9yi jT8u2aZfUHcSXGvJnc1FDEI= =VQhJ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Show queue-name near the callerId
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Hash: SHA1
Thalassoline - Service technique wrote:
I want to show the QueueName to my Queue Member.
I try to find the solution to show the QueueName near the callerid of
each call.
Can somebody help me ?
Set the CALLERID(name) variable prior to sending the call to the Queue.
e.g.:
exten = s,1,Set(CALLERID(name)=${QueueName}${CALLERID(name)});
exten = s,n,Queue(${QueueName});
Adjust to taste...
Barry
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Re: [asterisk-users] Monitor-join not joining files in the queues.conf file
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dáibhéad Antoine O'Reilligh wrote: Have I forgotten anything? Do you have 'sox' installed on your asterisk box? Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKiyB7CFu3bIiwtTARAkGOAKCjYeHxu9WDvOunloEjiVVQrb1UQQCcD0yM 55z/x/5rhL+C8DDBEN1Jgjg= =ygsx -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to compile 1.6.0.12
I tried patching my 1.6.0.10 source to 1.6.0.12 and during compilation (make) I get a string of errors in chan_sip.c that start: chan_sip.c: In function \u2018handle_incoming\u2019: chan_sip.c:18669: error: expected expression before \u2018\u2019 token chan_sip.c:18674: error: \u2018ret\u2019 undeclared (first use in this function) chan_sip.c:18674: error: (Each undeclared identifier is reported only once chan_sip.c:18674: error: for each function it appears in.) chan_sip.c:18697: error: expected expression before \u2018==\u2019 token chan_sip.c:18856: error: invalid storage class for function \u2018process_request_queue\u2019 chan_sip.c:18856: warning: no previous prototype for \u2018process_request_queue\u2019 Thinking that my patch puked (although I got no errors), I downloaded the 1.6.0.12 tarball and went through the usual ./configure cycle, same problem. Is this just me or are others having this difficulty? Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] ForkCDR and setting the account info?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 J. G. wrote: I've been Googling all morning and searching voip-info.org http://voip-info.org but not quite finding what I'm looking for. I've read that you can modify the billing/account information on a CDR via AGI but I can't find an example or a how to. I'd like to then assign specific accounts in the CDRs. Possible? Look up the CDR function. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKeyj/CFu3bIiwtTARAgozAKCPGThtSJzrqs2vlMMvEsfzICDIkgCcD1oE 9oKNSFEDD06hZQ5qa/T9FQI= =YlEU -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Vyatta routers solving NAT problems
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tarek Sawah wrote: First of all it acts like a firewall and a router.. compared to Cisco routers it has good ACL and firewall policies that can be used and written very well.. second it's easy to setup third my question is has anyone tested it ? and what are their openion regarding this? Tarek -- not only tested it but have it deployed in three different businesses. I have had no trouble whatsoever with Asterisk Vyatta. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD4DBQFKeCk5CFu3bIiwtTARAg7WAJjpXf/fvgcqh1AaZk5TAe0kalk/AJ4h07BK KKeo1MpuvfN9PmolAepbzg== =OaFh -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] AstLinux 0.6.7 released
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Darrick Hartman wrote: upgrade-run-image check http://mirror.astlinux.org/firmrware Note the typo: firmrware The working command is: upgrade-run-image check http://mirror.astlinux.org/firmware -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKdtRmCFu3bIiwtTARAovZAKCouLWC/qtSgBteTJjWKYQH9wYn2gCZAZ9Z lQl7oQER0rf6sGDVMv+NZgY= =BSLs -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Free Fax for Asterisk -- benchfax utility hangs.
I'm trying to install my first channel of FAX. When I run the benchfax utility, I get to various stages and then the program simply hangs. There is no excessive CPU utilization and the benchfax program readily responds to ^C. Sometimes I get fairly far into the test, other times it hangs almost immediately. The version of benchfax version I'm using is 1.0.7. I heard tell of a newer version, 1.0.10, but I have been unable to find it. I'm running CentOS5, on an IBM x3250, , Asterisk 1.6.0.10, with dual Xeon 1.6G processors. In my last test, I got this far (the farthest along so far): [r...@corp-asterisk ~]# ./benchfax benchfax version 1.0.7 Use the '-l' option to see license information for software included in this program. Running test using CCITT FAX test page, US letter size, 204x196 resolution, MMR encoding, ECM enabled and V.17 (14.4kbps) modem NOTE: Each individual test could take 20 seconds or longer; be patient. Test run 1 for flavor 'i686' used 632 milliseconds of CPU time. Test run 2 for flavor 'i686' used 604 milliseconds of CPU time. Test run 3 for flavor 'i686' used 656 milliseconds of CPU time. Test run 4 for flavor 'i686' used 207 milliseconds of CPU time. Test run 5 for flavor 'i686' used 624 milliseconds of CPU time. Test run 1 for flavor 'pentium3m' used 232 milliseconds of CPU time. Test run 2 for flavor 'pentium3m' used 668 milliseconds of CPU time. Test run 3 for flavor 'pentium3m' used 651 milliseconds of CPU time. Test run 4 for flavor 'pentium3m' used 641 milliseconds of CPU time. Test run 5 for flavor 'pentium3m' used 649 milliseconds of CPU time. Test run 1 for flavor 'pentium-m' used 194 milliseconds of CPU time. Test run 2 for flavor 'pentium-m' used 640 milliseconds of CPU time. Test run 3 for flavor 'pentium-m' used 673 milliseconds of CPU time. Test run 4 for flavor 'pentium-m' used 251 milliseconds of CPU time. Test run 5 for flavor 'pentium-m' used 617 milliseconds of CPU time. Test run 1 for flavor 'pentium4m' used 613 milliseconds of CPU time. Test run 2 for flavor 'pentium4m' used 664 milliseconds of CPU time. Test run 3 for flavor 'pentium4m' used 644 milliseconds of CPU time. Test run 4 for flavor 'pentium4m' used 628 milliseconds of CPU time. Test run 5 for flavor 'pentium4m' used 650 milliseconds of CPU time. Test run 1 for flavor 'prescott' used 749 milliseconds of CPU time. Test run 2 for flavor 'prescott' used 672 milliseconds of CPU time. Test run 3 for flavor 'prescott' used 737 milliseconds of CPU time. Test run 4 for flavor 'prescott' used 665 milliseconds of CPU time. Test run 5 for flavor 'prescott' used 649 milliseconds of CPU time. Beginning test run 1 of 5 for flavor 'nocona'... Is there a newer version of this software and can anyone give me an indication if they, too, have had this problem and what they did to get past it. Thanks! Barry ___ -- 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] Free Fax for Asterisk -- benchfax utility hangs.
Kevin P. Fleming wrote: You say it has stopped at different points as you've run different tests; can you verify that? If it always stops at 'nocona', that would mean one particular type of problem, but if it stops at various different places, then that would point to something very different. Hello Kevin. Thanks for the reply. I have three examples that follow. The first I ended after 11 minutes. I got as far as nocona, then the hang. The next run had me down to c3 in 8 minutes, before hanging there. I let it run for another 8 minutes or so. Finally, I let the thing run for 24 minutes and only got to nocona again. What is a reasonable time for this entire test suite to finish, if it completed normally? Barry [r...@corp-asterisk ~]# time ./benchfax benchfax version 1.0.7 Use the '-l' option to see license information for software included in this program. Running test using CCITT FAX test page, US letter size, 204x196 resolution, MMR encoding, ECM enabled and V.17 (14.4kbps) modem NOTE: Each individual test could take 20 seconds or longer; be patient. Test run 1 for flavor 'i686' used 658 milliseconds of CPU time. Test run 2 for flavor 'i686' used 632 milliseconds of CPU time. Test run 3 for flavor 'i686' used 413 milliseconds of CPU time. Test run 4 for flavor 'i686' used 188 milliseconds of CPU time. Test run 5 for flavor 'i686' used 202 milliseconds of CPU time. Test run 1 for flavor 'pentium3m' used 633 milliseconds of CPU time. Test run 2 for flavor 'pentium3m' used 636 milliseconds of CPU time. Test run 3 for flavor 'pentium3m' used 651 milliseconds of CPU time. Test run 4 for flavor 'pentium3m' used 676 milliseconds of CPU time. Test run 5 for flavor 'pentium3m' used 645 milliseconds of CPU time. Test run 1 for flavor 'pentium-m' used 632 milliseconds of CPU time. Test run 2 for flavor 'pentium-m' used 639 milliseconds of CPU time. Test run 3 for flavor 'pentium-m' used 187 milliseconds of CPU time. Test run 4 for flavor 'pentium-m' used 481 milliseconds of CPU time. Test run 5 for flavor 'pentium-m' used 644 milliseconds of CPU time. Test run 1 for flavor 'pentium4m' used 183 milliseconds of CPU time. Test run 2 for flavor 'pentium4m' used 676 milliseconds of CPU time. Test run 3 for flavor 'pentium4m' used 654 milliseconds of CPU time. Test run 4 for flavor 'pentium4m' used 689 milliseconds of CPU time. Test run 5 for flavor 'pentium4m' used 611 milliseconds of CPU time. Test run 1 for flavor 'prescott' used 670 milliseconds of CPU time. Test run 2 for flavor 'prescott' used 340 milliseconds of CPU time. Test run 3 for flavor 'prescott' used 484 milliseconds of CPU time. Test run 4 for flavor 'prescott' used 701 milliseconds of CPU time. Test run 5 for flavor 'prescott' used 662 milliseconds of CPU time. Beginning cache-load run for flavor 'nocona'... real11m29.829s user0m16.895s sys 0m3.393s [r...@corp-asterisk ~]# [r...@corp-asterisk ~]# time ./benchfax benchfax version 1.0.7 Use the '-l' option to see license information for software included in this program. Running test using CCITT FAX test page, US letter size, 204x196 resolution, MMR encoding, ECM enabled and V.17 (14.4kbps) modem NOTE: Each individual test could take 20 seconds or longer; be patient. Test run 1 for flavor 'i686' used 648 milliseconds of CPU time. Test run 2 for flavor 'i686' used 680 milliseconds of CPU time. Test run 3 for flavor 'i686' used 674 milliseconds of CPU time. Test run 4 for flavor 'i686' used 625 milliseconds of CPU time. Test run 5 for flavor 'i686' used 633 milliseconds of CPU time. Test run 1 for flavor 'pentium3m' used 634 milliseconds of CPU time. Test run 2 for flavor 'pentium3m' used 640 milliseconds of CPU time. Test run 3 for flavor 'pentium3m' used 634 milliseconds of CPU time. Test run 4 for flavor 'pentium3m' used 636 milliseconds of CPU time. Test run 5 for flavor 'pentium3m' used 647 milliseconds of CPU time. Test run 1 for flavor 'pentium-m' used 623 milliseconds of CPU time. Test run 2 for flavor 'pentium-m' used 616 milliseconds of CPU time. Test run 3 for flavor 'pentium-m' used 626 milliseconds of CPU time. Test run 4 for flavor 'pentium-m' used 679 milliseconds of CPU time. Test run 5 for flavor 'pentium-m' used 668 milliseconds of CPU time. Test run 1 for flavor 'pentium4m' used 448 milliseconds of CPU time. Test run 2 for flavor 'pentium4m' used 669 milliseconds of CPU time. Test run 3 for flavor 'pentium4m' used 631 milliseconds of CPU time. Test run 4 for flavor 'pentium4m' used 645 milliseconds of CPU time. Test run 5 for flavor 'pentium4m' used 652 milliseconds of CPU time. Test run 1 for flavor 'prescott' used 683 milliseconds of CPU time. Test run 2 for flavor 'prescott' used 658 milliseconds of CPU time. Test run 3 for flavor 'prescott' used 662 milliseconds of CPU time. Test run 4 for flavor 'prescott' used 662 milliseconds of CPU time. Test run 5 for flavor 'prescott' used 660 milliseconds of CPU time. Test run 1 for flavor 'nocona' used
Re: [asterisk-users] Free Fax for Asterisk -- benchfax utility hangs.
Kevin P. Fleming wrote: If you can, run it again under 'strace', capture the output, compress it with gzip or bzip2 and email it to me directly; that may give us a clue what it was doing. Requested strace output sent. Barry ___ -- 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] AGI to announce temperature from weather.com XML file
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Trevor Hammonds wrote: I am hoping someone on the list has an example of a lightweight AGI script that I may modify to either read the simple text file and set a dialplan variable to the current temperature, or hopefully a more-sophisticated one which will parse the XML file to set the dialplan variable. I think that the 'FILE' function will do what you're looking for. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKXyZsCFu3bIiwtTARAg/rAJ9D6RQQ2N51GNU8sWHbxPyJM82U1ACgn4bR c55n6BEUTuMPRSsRgETeE9w= =YLth -END PGP SIGNATURE- ___ -- 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] DEVICE_STATE() and Asterisk 1.6.0.10
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mark Michelson wrote: Thanks for the config info. I have a couple of suggestions for fixes. 1. Try changing the type in [basic-options] from friend to peer. I've found that device state reporting for outbound calls (from the perspective of the phone) tends to be more accurate with this type. 2. If for some odd reason number 1 either doesn't sound appealing to you or doesn't work, then try moving the limitonpeers=yes option from your [basic-options] section to the [general] section. No, neither of these ideas actually make any real sense to me, but they are based on behavior that I have witnessed with my Asterisk setup in my office. Mark Michelson I'll give these a try and see if they help. At this point, I'd be willing to slaughter a goat and place its entrails on the keyboard if I thought it would help. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKX0w5CFu3bIiwtTARAsy8AKCDbPMDZJ98v1HuL/KLDuQsayI84ACfX4OI Jw5YOgQllm1+wbq2wThh4Wg= =eE0m -END PGP SIGNATURE- ___ -- 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] Phantom CallerID on transfers
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Danny Nicholas wrote: Hi Gang, Running Asterisk 1.4SVN using Polycom 501 phones. Just enabled CallerID and for the most part it works as good as you’d expect anything to from the phone company to. Except: on about 1 out of 10 transfers, instead of getting a callerid of “joe cool 100” or “abc company 205-555-1212” I just get “asterisk”. Was the caller ID information presented to you from the telco? It appears to me that you're getting the default if none was provided. BK -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKXfqnCFu3bIiwtTARAvFfAKCUQLIYOzB1sxtnk4bKEA9ZPY9BogCdGHCa 5N+DCZPPuQxZPw6ZIaQ+0Go= =W5vJ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I must be missing something here but I can't figure out why I can't get DEVICE_STATE() to give me anything other than NOT_INUSE. I have two extensions: and 6668. I used 6668 to make a call to yet another phone, so I know that it's busy. I then use to call 6668 and in the dialplan have a noop to see what DEVICE_STATE() is returning for both extensions. I get: [Jul 15 17:20:43] -- Executing [6...@sip-deskset:1] NoOp(SIP/-08636430, SIP/6668 has state NOT_INUSE) in new stack [Jul 15 17:20:43] -- Executing [6...@sip-deskset:2] NoOp(SIP/-08636430, SIP/ has state NOT_INUSE) in new stack 6668 is configure so that I get this: * Name : 6668 Secret : Set MD5Secret: Not set Context : sip-deskset Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 99 Busy level : 1 Dynamic : Yes Callerid : Matts SIP 6668 MaxCallBR: 384 kbps Expire : 2016 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 192.168.1.70 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Transport: UDP Def. Username: Barry's IP450 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing : No 100 on REG : No Status : OK (14 ms) Useragent: PolycomSoundPointIP-SPIP_450-UA/3.1.3.0439 Reg. Contact : sip:6...@192.168.1.70 Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs - From what I have read, with 'Busy Level = 1' I should be seeing BUSY returned from the DEVICE_STATE() call, yet I don't. What is the super-secret sauce required to get Asterisk to return the correct state? TIA, Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKXkmMCFu3bIiwtTARAvoNAJ4nGmm2U27XTs+E2O1Eze5bDTEVZgCcDDQm 4JvWx6zOlAl8ZnFyFxu8lfo= =uhHc -END PGP SIGNATURE- ___ -- 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] DEVICE_STATE() and Asterisk 1.6.0.10
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mark Michelson wrote:
You need to set a call-limit for the SIP peer. Device state calculation for a
SIP peer is predicated on both the call-limit and busylevel. Let's say that
you
were to have a call-limit of 2, but no busylevel set. These are the device
states reported for the peer based on the number of calls currently handled:
Hi Mark. Thanks for your explanation of these parameters.
I should have posted my configurations. I double-checked the contents
of sip.conf and I have this. The 'subscribecontext' was added for
testing, per the other reply I got for my question.
;
; Settings common to all devices on our system
;
[basic-options](!)
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
qualify=yes
;
; Standard desksets here
;
[lan-deskset](!,basic-options)
context=sip-deskset
notifyringing = yes
notifyhold = yes
limitonpeers = yes
call-limit=99
[6668](lan-deskset)
secret=mysecret
callerid=Matts SIP 6668
username=Barry's IP450
call-limit=32
busylevel=1
subscribecontext=hint-context
My hint-context is:
[hint-context]
exten = 6668,hint,SIP/6668;
I'm still not getting anything other than NOT_INUSE from DEVICE_STATE.
Here is the CLI output:
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:1]
NoOp(SIP/-0955ecc8, SIP/6668 has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:2]
NoOp(SIP/-0955ecc8, SIP/ has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:3]
ExecIf(SIP/-0955ecc8, 0?Busy(10)) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:4]
Dial(SIP/-0955ecc8, SIP/6668) in new stack
And here is sip show inuse:
corp-asterisk*CLI sip show inuse
* User name In use Limit
6668 1 32
6667 0 99
1 99
* Peer name In use Limit
6668 1/1/0 32
6667 0/0/0 99
0/0/0 99
For completeness, here is the dialplan that's producing this:
exten = 6668,1,NoOp(SIP/${EXTEN} has state ${DEVICE_STATE(SIP/${EXTEN})});
exten = 6668,n,NoOp(SIP/ has state ${DEVICE_STATE(SIP/)});
exten = 6668,n,ExecIf($[${DEVICE_STATE(SIP/${EXTEN})}=BUSY]?Busy(10));
exten = 6668,n,Dial(SIP/${EXTEN});
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iD8DBQFKXlwaCFu3bIiwtTARAkRpAJ4+2WF9qrIwrC3Kdpwd0YAOm/5S1wCfUR1T
CtI9kZNQYpW2Sv6uFNud7Jo=
=9Zp/
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Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Philipp Kempgen wrote: Just to be sure: Do you have hints configured for the extensions? See http://das-asterisk-buch.de/2.1/blf-leds.html (The text is in german but there are many examples in extensions.conf and extensions.ael syntax. Zurück = Previous, Weiter = Next) Hi Philipp. I did configure the hints (see my other reply) and things are still not working properly. Thanks for your suggestion! Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKXlxZCFu3bIiwtTARAhpLAJ9D8z4Mbhg9ACt62sTR46UQApcQMACdGH48 Uy14CTE7pKMb8qffF+wfiow= =d5oG -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking timeout fails
John A. Sullivan III wrote: Hello, all. I'm having a nasty problem with call parking in Asterisk 1.6.1.1 that smells like a bug. When the call returns, it seems to be returning to a | delimited extension and failing. Here is the output from the console: Hi John. I've just run into the same problem on 1.6.0.10. Have you heard any more about this problem? TIA, Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking timeout fails
Jonathan Thurman wrote: This was fixed in the 1.6.1 SVN, and I would guess that it was also fixed in the 1.6.0. SVN log: r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines Fix call parking callback. Pipes - Commas. You will have to create a patch against the 1.6.0 source, but you could start by looking at the patch in this issue: https://issues.asterisk.org/view.php?id=15162 Please note again that that patch was against 1.6.1.0. -Jonathan Thanks very much Jonathan. I'll figure out how to make this patch against 1.6.0.10. Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking timeout fails
Barry L. Kline wrote: I'll figure out how to make this patch against 1.6.0.10. That was a trivial fix. I hope that they permanently add that patch to the 1.6.0.x series. Thanks again Jonathan. Barry ___ -- 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] Extension status as XML for an Aastra 57i
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jeremy Winder wrote: I'm in the process of converting our current hybrid key system to Asterisk and Aastra 57i phones. One of the features that seems to be a show stopper for almost everyone in the office is the inability to see who is on the phone. Can someone point in the right direction to setup an XML app on the phone so they can press a soft-button and get a list of extensions and their statuses? I know I can use BLF and the line 2-4 buttons; but there are a lot more then 3 other people working here and I'm planning on using those of parking lots. Any help will be greatly appreciated as I'm an Asterisk noob learning as fast as I can. If you'd like a more generalized approach you can install an Openfile server and use the Asterisk plugin. That'll give you an internal IM server which will show the status you seek. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKSnZnCFu3bIiwtTARAveUAKCACUaxvBLfYHtuhojOZW1o/aOVkACdH3EQ ceTOXQOXlENUTGWhevxIrXc= =rG0d -END PGP SIGNATURE- ___ -- 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] Extension status as XML for an Aastra 57i
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Barry L. Kline wrote: If you'd like a more generalized approach you can install an Openfile server and use the Asterisk plugin. That'll give you an internal IM server which will show the status you seek. Sorry, not 'openfile' but 'openfire'. http://www.igniterealtime.org/projects/openfire/index.jsp -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKSntACFu3bIiwtTARAp4NAJ9/SdVaNXlc6nNq3LB8A5ss/X2q8wCcDNd0 Cq6E39xouEePRfGgqZULYzo= =kbf5 -END PGP SIGNATURE- ___ -- 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] res_cepstral, register existing Cepstral licenses.
Kevin P. Fleming wrote: There is no need; your existing Cepstral-supplied licenses will continue to operate, and will be added to any Digium-supplied licenses you purchase and activate. Hi Kevin. That didn't work. If I use 'swift -n Allison-8kHz -o test.wav Hello, test and play the resulting wav file, life is good. If I use SayText(Hello, test) I get to hear the this voice is unlicensed message before the rest. So it appears as though I DO need to worry about this: corp-asterisk*CLI cepstral show licenses Cepstral Licensing Information == Allison Voice Enabled: no Total licensed ports: 0 Which then begs the question... how do I migrate the licenses over? Thanks! Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_cepstral, register existing Cepstral licenses.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have a license for Allison-8kHz and two concurrent port licenses that I purchased from Cepstral at the end of last year. I just got around to installing to my * 1.6.0.10 machine. I've decided that the best way for me to integrate the two would be res_cepstral, which I downloaded and installed. Everything is fine, except the register program, which is looking for a license key sent from Digium. I'm going to end up buying more ports from Digium but I'd like to also use the existing voice/port licenses that I currently have. Is this possible? Is there anyway to migrate the licenses to the Digium implementation of Cepstral? TIA, Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKQ9RICFu3bIiwtTARAjg4AKCriHJ0F18y7HJvbby9FjbCWL72OQCfW3Cy SipgbgQvZb93O3u4ecsxxCY= =9bF7 -END PGP SIGNATURE- ___ -- 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] res_cepstral, register existing Cepstral licenses.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kevin P. Fleming wrote: There is no need; your existing Cepstral-supplied licenses will continue to operate, and will be added to any Digium-supplied licenses you purchase and activate. Thanks Kevin. So I shouldn't worry about this? corp-asterisk*CLI cepstral show licenses Cepstral Licensing Information == Allison Voice Enabled: no Total licensed ports: 0 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKQ/M8CFu3bIiwtTARAui1AJ48Jf/E0gVGrITS32SHiZSCIXfczgCeMIyi 4M9g7qv/k7Vrxy/mJA4j3kA= =5YbR -END PGP SIGNATURE- ___ -- 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] Minimizing downtime during updates
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Karl Fife wrote: I discovered that after running make, you can run 'make sounds' before shutting down the service. This cuts all of the download time from the install process minimizing service downtime to a fraction of what it would othewise be. Ahhh... another helpful tidbit from the Karl Fife experience! Thanks Karl -- I had always wondered about the download time but hadn't taken the time to research a workaround. Your tip will save me some time! BTW, I just implemented my first system using the Polycom config system you spoke about on VUC. I appreciate you taking the time to do that. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKQM0XCFu3bIiwtTARAqPvAJ9reHF+Uczdjo/PE9WofYlTcBqnvQCfWq05 oOL6r2FhK17AoQ0CDtdP8oI= =uQt9 -END PGP SIGNATURE- ___ -- 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] Multiple Outgoing Lines: extensions.conf (Ioan Indreias)
Clara Chan wrote: Loan, Thanks for your help in this matter. Having never used astdb before, can you point me to an example on this?? Thanks hugely, Clara Clara -- You need to read the book. In it you'll find examples. Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) It's downloadable at http://www.asteriskdocs.org Barry ___ -- 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] howto store local exchange prefixes ?
sean darcy wrote:
Maybe I've not explained this correctly. I know, or can look up, the 40+
local exchanges that are local. I can parse the dial EXTEN to determine
the exchange. I can check the exchange against a DB. I want to determine
which exchanges are local. I do not want to store an exchange dialed
by a user.
I didn't explain myself very well.
My Asterisk system sits between the PSTN and a legacy PBX. Asterisk
answers the call and among other things, prompts for an extension
number. I needed to know if the extension entered is valid before
sending the call on to the old PBX. I simply have a lookup subroutine
to validate the extension.
My code for looking up the validity of their entry is:
exten = _[123]XX,1,Verbose(1,${CALLERID(all)} requested extension
${EXTEN});
exten = _[123]XX,n,Gosub(validate-extension,s,1(${EXTEN}));
exten = _[123]XX,n,Goto(extension-${GOSUB_RETVAL});
exten = _[123]XX,n(extension-FOUND),Verbose(1,${CALLERID(all)} xfer to
${DB(${DB_IWATSU_EXTENSIONS}/${EXTEN})} at extension ${EXTEN});
exten = _[123]XX,n,macro(bridge-to-iwatsu,7${EXTEN});
exten = _[123]XX,n(extension-NOTFOUND),background(pbx-invalid);
exten = _[123]XX,n,WaitExten(5);
The lookup, which will initialize the AsteriskDB if necessary, is:
;
; This subroutine's purpose is to check the validity of an extension.
;
; Parameters:
; ARG1 = Extension to check
; Returns:
; FOUND or NOTFOUND
;
[validate-extension]
exten = s,1,Verbose(1,Checking validity of extension ${ARG1});
;
; Let's check to ensure that the database is loaded. We'll do
; that by looking for extension 399, the Iwatsu master phone.
;
exten = s,n,GotoIf(${DB_EXISTS(${DB_IWATSU_EXTENSIONS}/399)}?search:load)
exten = s,n(load),DBdeltree(${DB_IWATSU_EXTENSIONS}); Clear all
existing entries
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/120)='Rikki')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/121)='Terri')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/122)='CorpConf')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/123)='Linda')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/124)='Kim')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/125)='Nancy B')
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/126)='Wayne')
...
;
; Extension 399 is the master extension for the Iwatsu
; and should always show up. It is used for testing
; the validity of the database in the dialplan.
;
exten = s,n,Set(DB(${DB_IWATSU_EXTENSIONS}/399)='MASTER')
;
; Search here
;
exten =
s,n(search),GotoIf(${DB_EXISTS(${DB_IWATSU_EXTENSIONS}/${ARG1})}?found:notfound)
exten = s,n(found),Return(FOUND);
exten = s,n(notfound),Return(NOTFOUND);
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Re: [asterisk-users] Add Monitor application to call suppresses audio
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David Backeberg wrote:
5) exten = s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
6) exten = s,n,Goto(s-${DIALSTATUS},1);
What is the 6 for?
What is the goto supposed to do?
Hi David.
The '6' is in case I get a CHANUNAVAIL or other error back from the
Dial command. If the call is connected then I never get to '6'.
I have determined that the only calls I seem to be having trouble
monitoring are the ones sent to my answering service. If I terminate
the call to my cell phone, my home POTS line, a POTS line here in the
office or even to the inbound PRI at the office, things work fine. I
can even record calls to the answering service's published number. It's
just when I go to the number assigned to us that there is trouble and
I'm currently chasing down the owner of that service to see exactly what
I'm dropping into there.
Thanks!
Barry
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Re: [asterisk-users] howto store local exchange prefixes ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 sean darcy wrote: I've looked at the Berkeley DB. That works pretty well, if the exchanges are all stored. But it looks like the exchanges have to be entered 1 by 1 from the CLI. And can only be reviewed, corrected, or deleted from the CLI. I haven't found any simple frontend for the DB. I do this be writing a dialplan which adds those entries. The first entry checks to see if the DB has been initialized and if so, skips to the lookup. Otherwise it loads each into the database before the lookup. It's very easy to write a quick script to generate the dialplan code. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKGxKvCFu3bIiwtTARAuZLAJ4uZw/76Pefz9y1fdQRCS03qFGmOACfeR98 fD8binGxr+c9mMriW6GXjC0= =ETQ2 -END PGP SIGNATURE- ___ -- 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] Bridging INBOUND PRI to OUTBOUND PRI fails withMonitor()
Danny Nicholas wrote: You should try Answer before Dial on the Monitored call. Bridging can be very unhappy. Hi Danny. Already done earlier in the dial plan, when the call first comes in but before it gets routed to the part that I showed. Thanks for looking though! Barry ___ -- 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 Productivity Suite
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Robin Rodriguez wrote: still rather frustrating getting the EFK working. If needed I could post that portion of sip.cfg to get you started. Please do! Just having the example could be helpful for those of us preparing to tackle this kind of project. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKFWD1CFu3bIiwtTARAvsvAKCPGs1PHJjbRhackHkCKsb5vi4JJACeKJHj 2gPE1PAyVTekMHhJzURF8+M= =NBDL -END PGP SIGNATURE- ___ -- 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] Bridging INBOUND PRI to OUTBOUND PRI fails withMonitor()
Danny Nicholas wrote: To clarify: Inbound - Answer Outbound - Answer (again) Dial. Hmmm... that seems like it would be from the department of redundancy department but I gave it a try, both before and after the Monitor() command with the same result... it fails. Thanks! Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bridging INBOUND PRI to OUTBOUND PRI fails with Monitor()
I wrote a note earlier about this problem but have done quite a bit more debugging. Now I'm stuck at what to do next. I have inbound calls being answered by our Asterisk box, which then dials our answering service and bridges those calls. The inbound and outbound are both PRIs. The answering service takes our calls on a PRI. If I don't use the Monitor() application, things work find and have been for a few thousand calls. If I add the Monitor() application, no audio ever gets passed from the caller to the answering service. I have noted the following things while testing with Monitor(): 1) If I have it call my cell phone instead of the service, it works fine. 2) If I have it call my home phone instead of the services, it works fine. 3) I tried calling another number (in another state) that I know terminates into a PRI and it worked fine. 4) If I call the service without Monitor(), it works fine. Throw in Monitor() and it's virtually guaranteed not to work. My dial plan and debug output for both the working and failing call is at http://www.pastebin.ca/1429504 . Things start to diverge around lines 28-31 and 68-72. Can anyone tell me what I can do to further trace this problem? Thanks in advance for anything you may be able to offer. Barry ___ -- 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] Add Monitor application to call suppresses audio
This is getting really interesting. I had a chance to do some testing last night. To recap, here is what I'm attempting to do: Caller -- INBOUND_PRI -- Asterisk -- OUTBOUND_PRI -- AnswerService The caller dials our number, * picks the call and offers some choices. If the caller needs to speak to a human, * dials the answering service and then bridges the calls. It works flawlessly in this scenario. If I do Monitor() or MixMonitor() on the channel before dialing the outbound call, 99.9% of the time I get no audio to the caller. Now for the interesting part: If I substitute my cell phone number (xxx-) for that of the answering service (1-877-...) everything, including Monitor(), works just fine. The same goes for calling my home number instead of the cell phone. I don't know what my cell phone and home phone terminates into when I dial those numbers but if I call my answering service I know that they have a T1/PRI. I have done a PRI DEBUG SPAN 4 (where 4 is the outbound PRI) and repeated the tests, capturing the output. Nothing in them jumps out at me as yet but I'll keep looking at them. Does this offer anything suggestive David? Barry ___ -- 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] Add Monitor application to call suppresses audio
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Sorry for the delayed response, I was out of the office. David Backeberg wrote: But there's not the native bridging status on the calls with recording enabled, where as the native bridging report fires on the recording-less dialplan. A clue perhaps... So you say call 1 with recording made a file, and the call connected with voice. And call2 with recording made a file, but the customer didn't hear the voice? Yes. In this case I'm using an outside Asterisk server to dial back in and act like the customer. The first time there was a longer delay in making the connection, but I was able to eventually hear the audio. The second attempt gave no audio to the customer side. What happens if you use MixMonitor() instead? Are you mixing these calls back together afterward? My recollection is that monitor makes a call in two halves, one for sender, and for receiver, and then you have to multiplex the halves back together afterwards. Are you doing the multiplex step? MixMonitor() doesn't act any differently... same no audio condition. Monitor does indeed act as you say but if you add an ',m' option to the call it will do the merge the files back together. I'm going to make my cell phone the target and see what the callee (the answering service) is hearing. Thanks for your comments David. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKEdAtCFu3bIiwtTARAgnnAKCn1tQKTT8/orBRRhsZ/EjgQ/0U9gCeJOXg yvDYr2t/iSG40J+7H4XLOf0= =KlGM -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Add Monitor application to call suppresses audio
I have an application where we receive calls on an inbound PRI. After
hours, our Asterisk box dials our answering service on an outbound PRI
and then bridges the caller to the answering service. The flow looks
like this:
(CALLER)INBOUND_PRI -- CONTEXT -- GOSUB(Incoming) --
GOSUB(bridge-to-anssrv) -- DIAL(answering_service) --
OUTBOUND_PRI(service)
This has been working fine for months without so much as a burp. What I
need to do is record these calls.
If I insert a Monitor() prior to dialing the outbound call, I get no
audio in the recording and the caller hears no audio. Occasionally it
works (perhaps 1 out of 5 times) but most of the time the caller can't
hear the callee, and vice versa.
The fully working code looks like this:
1) exten = s,n(place),Verbose(4,Dialing answering service);
2) exten = s,n,Playback(vrec_prompts/this-call-may-be-recorded);
3) exten = s,n,Set(GROUP()=ANSSVC);
4) exten =
s,n,Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)});
5) exten = s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
6) exten = s,n,Goto(s-${DIALSTATUS},1);
If I insert
exten = s,n,Monitor(wav,${CALLFILENAME},m);
before the dial command on line 5, I'm virtually guaranteed that the
call will fail and no audio will be passed.
I'm using Asterisk 1.6.0.9, LIBPRI 1.4.10, and DAHDI 2.1.0.4.
Can anyone shine any light on why this problem is occurring?
TIA,
Barry
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Re: [asterisk-users] Add Monitor application to call suppresses audio
David Backeberg wrote: I don't know why recording is breaking your calls. My guess is something is screwed up with your PRI configuration. Are you getting alarms in your logs from dahdi? Not a peep, either with or without using the monitor command. I've been using this system for around four months during which time it has performed flawlessly, running through 20K calls. You should try to reproduce the problem on demand by generalizing your dialplan, change the number of the answer service to the number of your cell phone, and run some calls through. Done. I've been recording calls with 1.6.0 series using MixMonitor() and haven't been having problems, making me think the recordings step is coincidental. Crank up the verbosity, run some calls through and tell us what's happening. To avoid wrapping, I've posted the results from my tests to this link: http://www.pastebin.ca/1422291 The first call is the dialplan I have been using which works perfectly. The second call, which worked, was the first attempt to use Monitor(), after having restarted Asterisk. The third call is another attempt at getting a recording. It, and any subsequent call, fails miserably. I'm open for any suggestions. Thanks very much David. Barry ___ -- 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] Compact, fanless appliance?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dean Collins wrote: Technoco vdex40 is probably outside of your pricepoint but you might want to consider them My experience with the vxex40 was not great. This was about six months ago. I'd not recommend one unless I could ensure that I got a money-back guarantee and gave the thing a thorough test-drive before deployment -- ESPECIALLY if you're going to hook POTS lines to it. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ/zKHCFu3bIiwtTARAk9UAJ9J5DLgoWUgAqqCXwsX2S2ZD8UnRwCglm2C 23dBCPDYKZQUp1M8ARZ3STc= =8otO -END PGP SIGNATURE- ___ -- 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] Compact, fanless appliance?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jeff LaCoursiere wrote: What were the issues? There were tons of issues involving the generated configurations (such as not being able to use attended or unattended transfers) but the worst issue was that the box would not hang up POTS lines. We had four lines plugged into this and slowly each would be stuff off-hook. These four lines were in a hunt group and eventually you'd not be able to get an inbound call. We found out when someone called us on a cell phone to let us know that our number had been busy for hours. The only way to clear the problem was to reboot it and we ended up doing that multiple times per day. We engaged the vendor who sold us the box who put us in touch with the manufacturer of the box who put is in touch with the gang who wrote the interface. No one could figure this out. We even tried swapping boxes, to no avail. The net result was that the distributer took it back and refunded our money. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ/z2dCFu3bIiwtTARAjNXAJ9bZwyK9e9TRYLL9kxmpzFBET6m8QCdGEed vELcwFD3Pj6YBiCdaH213eY= =HVAM -END PGP SIGNATURE- ___ -- 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 sudden crash
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andrew Nowrot wrote: This had happened twice so far. Does anyone know what is causing this.? Start by upgrading to 1.6.0.9, then if it continues you can start tracking it down. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ+KjICFu3bIiwtTARAgiQAJ0WEGJNroqfnfpQWnUABt4Uh4KguQCgrE23 sGSrNv9+nKyaT08PC0Izc0g= =e5bf -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Who has the clever Polycom upgrade system?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I remember someone wrote a great document concerning Polycom server provisioning that provided a way to ensure that updates to the firmware did not overwrite customizations. I'll be damned if I can remember where I saw it. It may have been discussed during a VUC session or may have been on this list. Either way, I'm unable to google my way to it. Can anyone point me in the right direction? Thanks! Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD4DBQFJ9iSWCFu3bIiwtTARAlH2AJjcCtRPi9dyqwY0p2AqCZelgskIAKCVeuSV ++7hraanUhxNBF2RvIMmRg== =Hg/L -END PGP SIGNATURE- ___ -- 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] Who has the clever Polycom upgrade system?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kevin P. Fleming wrote: It's easy; just don't edit the files that come with the firmware! Hi Kevin. That's the model I currently use. The one I'm interested in is linked in Darrick's post below. It's an interesting approach. Thanks for replying! Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ9i+2CFu3bIiwtTARAj0wAKCP8K5NnEnHawuA5q5k0Aq1bKBe8QCeLBz7 e/xvLf7N0Ofl1Ic7Q9/ZT5A= =yDXt -END PGP SIGNATURE- ___ -- 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] Who has the clever Polycom upgrade system?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Darrick Hartman (lists) wrote: That would be Karl Fife, of the famous Karl Fife experience. http://kfife.com/voip/ That's what I'm looking for. Thanks Darrick! Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ9i/KCFu3bIiwtTARAnuQAJsHx/fRb/n6EnEj0pco1eY0wgEcugCcDTTY NrTQOrDBYfVbpqyO6LMkIW0= =GaAB -END PGP SIGNATURE- ___ -- 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] module load chan_dahdi.so gives several WARNING-messages
jonas kellens wrote: I have 2 questions about the following output on the Asterisk CLI : Jonas Please do not hijack threads. Please start a new message. There is a good chance that many people didn't read your message because it's in reply to another, unrelated one. Barry ___ -- 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 Database
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Sriram wrote: 1. I need to store the CallerId of the PSTN caller with his language preference so that next time he is played the prompt in his language that he chose the first time.What would be better - storing his number in the Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the dial plan and quering it everytime to see the callers record ? how many records can AstDB handle safely ? In my case the total records wont exceed 20,000 since there are many repeat callers ? 20K records? While I'm not sure exactly how many records AstDB could handle it would seem to me that 20K would be a high number. My inclination would be to use a full database... perhaps you'd like to store more about that callerID than just the caller's preferred language. Using a real DB would certainly make that easier. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ7bRRCFu3bIiwtTARAqTvAJ4jS0/kZeHo33+w9gjZ88dYB3SeDACgg2+t LhVIBsPzxyQ/g542/NjMo8U= =d+JZ -END PGP SIGNATURE- ___ -- 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] Should I go for Asterisk 1.6 ?
--[ UxBoD ]-- wrote: I am going to be building a new home Asterisk server this weekend (Dual core Intel Atom 2GB RAM) and would like to ask whether it would be worth starting fresh with a 1.6 install instead of the 1.4 one I have at the moment ? I do not have a complicated dialplan as it only serves a couple of number and three extensions. For inbound and outbound I am using the IAX2 protocol instead of SIP. Any thoughts or help would be most gratefully accepted. I have been using 1.6.0.x now for a while with minor (non-critical) issues. If you already have a working Asterisk system, so you don't need to replace things right NOW, why not load up 1.6? That way you can learn about the changes between 1.4 and 1.6 at your leisure. For example, if you're still using Zaptel then you will need to learn the minimal changes required to go to DAHDI. Once you are done testing your server will be at the current level and can drop in as a replacement. Barry ___ -- 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 process ended
Adrien Lemoine wrote: Maybe someone experienced something similar and can drive me in the resolution ? You have given no information about your hardware, OS, Asterisk version or what you need to do to recover the system (e.g. reboot, just restart Asterisk, etc) so no one is going to be able to do much to offer help. Barry ___ -- 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-beginner : cannot make phonecalls using Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 jonas kellens wrote: I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI : /Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)/ /Verbosity is at least 5/ /asterisk*CLI / Nothing is displayed... it stays that way... Jonas. Is there a Send button on that phone? It sounds to me as though the phone is still waiting for more digits. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ437BCFu3bIiwtTARAs73AJ9spwpr7ULu6VyimPPoDIPnzFK6JQCbBEDO bQ0m2dROkUEkdtwCHtbHTBI= =4Zmk -END PGP SIGNATURE- ___ -- 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] chan_zap.so missing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 markus wrote: Hello everyone! I installed Asterisk following the instructions of the book Asterisk: The Future of Telephony. (very nice book) However, I failed. I installed zaptel, libpri and asterisk (in this order). If you are using Asterisk 1.6.x then you need to use DAHDI, not Zaptel. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJttR6CFu3bIiwtTARAqrjAJ9UsxkBxhylB5iuBw9ph794mj/ohACgkRyX lXV6JzAaJwj70ZWIHJLSUZ8= =2UlA -END PGP SIGNATURE- ___ -- 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] Compiling to use IMAP: how?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ken D'Ambrosio wrote: So: what/how do I need to install to meet this dependency? Did you run configure again after installing the missing components? Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJq/enCFu3bIiwtTARAg2xAJ9Yi8eOlw/bvtkmk/j/EoftD8KxaQCePqWa 4Mara5nHWJIqz6tigUqq2D8= =Hvxg -END PGP SIGNATURE- ___ -- 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] changing /etc/dahdi/system.conf
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brandon B. wrote: At the top of my /etc/dahdi/system.conf file is this line: # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 -- do not hand edit OK, so how do I adjust the timing source and LBO numbers, and echo cancellers if I'm not supposed to edit this file? I had the same question when I ran dahdi_genconf. My answer: ignore that message. You'll only want to run dahdi_genconf once, which creates that file. Then edit it to your heart's content. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJpwQkCFu3bIiwtTARAgY6AKCEWrmsKFQdWbdRh9NgjCakJJk9qwCgioSB 5ufZXOMuUBalAxWPvhfauTE= =r10C -END PGP SIGNATURE- ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jared Smith wrote: While I personally believe it's a bug, it has been in Asterisk for a very long time, and I know from teaching Asterisk training classes that there are *many* *many* people abusing this in their dialplans. I'd be quite hesitant to change this behavior without some very large warning signs. I think that the appropriate time is during an upgrade to a new version. Even from 1.6.0 to 1.6.1 would be okay, given that the behavior change is documented in the upgrade.txt document. Doing it from a .05 to a .06 release can certainly catch many off-guard. BK -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJpWAfCFu3bIiwtTARAp1AAJoDgKg1o0UPHg/0uGXesOVMZyP+0wCfXzbY XWUUOuxPwKdWG2xsbEGV2PY= =6+mm -END PGP SIGNATURE- ___ -- 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] HDD FULLL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 David @ULC wrote: When I am trying to delete voice logs, [r...@vicidialnow monitor]# rm * -r -f -bash: /bin/rm: Argument list too long [r...@vicidialnow monitor]# Argument list too long is coming as a road block. Now way to forcefully delete files ? Use: cd /path/to/monitor find . -type f | xargs rm -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJo+4sCFu3bIiwtTARAmtlAJ9ZSHjMUTFogxjV1+R3SVai46PxtQCgifkJ m18j5pNazt3YBytO3rUV/NU= =djs8 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gosub behavior change =1.6.0.5 to 1.6.0.6
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Here's one that may be of interest to any upgraders. If you rely on the
behavior of gosub you may want to make note of this change.
I have an incoming call context:
exten = _,n,GoSub(incoming,${EXTEN},1(${EXTEN}));
that is supposed to gosub into the incoming extension at priority 1.
Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the
requested extension wasn't present in the incoming context.
When I upgraded to 1.6.0.6 this behavior changed and I would simply get
an error on the console that a matching extension was not found, and the
dialplan would simply stop. It was easy enough to add:
[incoming]
exten = _,1,Goto(i,1)
to restore the previous behavior (I'm looking at four-digits from a PRI)
which I should probably have done anyway.
I don't know if this is a bug or WAD but just wanted to mention it.
Barry
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Re: [asterisk-users] Caller ID replacement
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 David Ruggles wrote: global variables that link the cell phone #'s and extensions and have this done somewhat automagically. Load your cross-reference in AstDB and do the lookup that way. If the cell number exists in the database, replace the callerID with the extension number. If it doesn't exist then it must be from someone else so don't change the callerId. BK -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJlFGwCFu3bIiwtTARAlRSAJ48FS53xS4u0eIeJ63VrZulPZxMMQCffFHw 7riqdRkR6vq5tGT9Z78FpiQ= =SuKH -END PGP SIGNATURE- ___ -- 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 1.6 dahdi only?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Gladden wrote: Is 1.6 so cutting edge that I should not expect to find complete documentation (yet)like I seem to be expecting very easily? Most of what is applicable to 1.4 is applicable to 1.6. I'm running 1.6 without any hiccups -- YMMV. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJf0s4CFu3bIiwtTARApXZAJ9kse5IimuCkzFG7FqlmQRzbxOlGgCfY8wA CeGjEgTSVagAovNT/TaNjDM= =z1O2 -END PGP SIGNATURE- ___ -- 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] AMI interface problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mark Michelson wrote: Thanks for pointing this out. I have located the erroneous code and have fixed it in subversion, revision 161490. The next rc of 1.6.0 will not have this bug. Mark -- This bug still exists in the recently-released 1.6.0.3. I just went from 1.6.0.3-rc1 (which had the bug) to 1.6.0.3 with the same problem. Reverting back to 1.6.0.1 makes the problem disappear. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJZ7nxCFu3bIiwtTARAtK8AKClSd5lEd0V1jxz7nsqwYEr1LUwmwCfbVRQ +keIC2kqp4sZ1jFwy2jQq6E= =PqM1 -END PGP SIGNATURE- ___ -- 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] AMI interface problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tilghman Lesher wrote: Mark erroneously assumed there would be another release candidate, which there was not. So while it's not in 1.6.0.3, it will be in the 1.6.0.4 release, when that occurs. Thanks Tilghman. I wait with bated breath. Best regards, Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJZ+rlCFu3bIiwtTARAoHRAJ9gjZl8D2QuNOfrBprnPbohaCj5qgCgotCY tL6Cq+q+skp28mLAhbfZXlE= =3xZD -END PGP SIGNATURE- ___ -- 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] Outbound fax issues
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mikel Lindsaar wrote: What does putting ww at the front do? Each w makes Asterisk wait a 1/2 second before sending the DTMF to dial. (It may be a 1/4 second each 'w') Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJUOFkCFu3bIiwtTARAgyWAKCt/LOdJNdL6763OWXP/K3AuHyOJACfZQuO 0kJs1pvYx9rYIoLOys4eQTs= =cTza -END PGP SIGNATURE- ___ -- 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] top posting again [was: Re: CDR Design]
Bill Andersen wrote: In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! ./bill Not when you take the time to properly trim your reply it's not. BK ___ -- 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
