RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: Ok I had a chance to test web-meetme 3.0.1 and I have few comments here - the Makefile for CBmysql lacks procedure that verifies existence of /var/lib/asterisk/sounds/conf-recordings directory where the conference records should reside. You are right that this should be documented at least, and part of the make install process ideally. I had to go through .php files to find out where they are supposed to be and create the directory manually. Strange enough, the recording still does not work and the main web interface lack any support for the record files (I would expect some link in the past conference list). There will be a link if the conference is recorded. I received a report of the recording option not working just this weekend and I started Looking for the cause today. I was out of town for a week, otherwise I would have gotten a chance to respond earlier. What version of Asterisk are you using? I've had recording working with SVN before 1.4, the 1.4 betas and currently 1.4.1. - Active Directory integration works fine, but we should be able to gather email addreess for the participant from AD, too (avoid using the sql users table if web-meetme was configured to use AD). Actually this is still a big mystery to me - how do I add participants to the conference using the web-interface? It must be done via the web interface as otherwise we have no information about the participant except of his channel number. I've never user the sql option for the user/participant. It was contributed by another user of the suite. Depending on the technology the caller used to call into the conference you should have their Caller-id number and possibly their Caller-id name. What additional Information would you like to see? It is very promising project but it needs - a better documentation Contributions welcome. There is a new How-To up on SF that covers the installation on a step by step basis. I've tried to comment the configuration files to make it clear how each setting works. Some features have been contributed to the project, and I am sorry to say that beyond making sure they integrate cleanly, I have not taken enough time to document their setup and use. I guess I should ask for supporting documentation before merging the changes/features. - fix the conference recording backend I hope to have this resolved this week. - clear the confusion with users/email addresses/mail notifications. More details about what you would like the system to do please... If all that works, it would be just perfect... Thanks, Ondrej Thank you for the feedback. I am surprised almost daily how many people have found it useful. I did not really expect it to be as popular as it has become, and I am more than happy to try and address any problems. Thanks, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk VS Cisco,Avaya,Siemens,....
Hi all, We are an ISP in Switzerland using only Asterisk boxes for VoIP and we are looking for others companies using Asterisk too to prove to our clients that Asterisk is a stable solution used by other big companies. Thanks a lot for your help, Thomas Deillon Telecom Engineer Smart-Telecom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 sluggish keys: found the problem!
I have learned the hard way that using old configs with new firmware is asking for trouble. It is much better to keep your custom configurations in a MAC specific overrides file and replace the sip.cfg and phone1.cfg files completely. This doesn't guarantee that you won't have problems, but it's a lot easier to troubleshoot an overrides file with a dozen items in it than to sift through big, customized sip.cfg files. Where can I find documentation on how to setup an override file using the phone's MAC? I see a (MAC)-phone.cfg file the phone uploads has something about overrides in it, but it looks like settings that the phone re-reads... Any help appreciated! Thanks. -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Make an iso image
Dears Can anyone guide me .. I want to put my asterisk system on an iso image like trixbox .how can I do that ,I am using centos 4.4 final Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Make an iso image
On Tue, Apr 17, 2007 at 10:17:25AM +0300, Khaled Chehab wrote: Dears Can anyone guide me .. I want to put my asterisk system on an iso image like trixbox .how can I do that ,I am using centos 4.4 final Trixbox is implemented using kickstart installation. A different method: http://www.mondorescue.org/ -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sending an SMS via Asterisk?
I've been googling and reading a lot, but I'm not getting any closer to
getting an SMS sent via Asterisk.
Prior to switching to asterisk, I used sms_client on an ISDN line to
dial one of two Swisscom SMS centers: 0900900941 or 0794998990.
My dialplan looks like this:
exten = 0900900941,1,Goto(smsmotx,${CALLERIDNUM},1)
exten = 0794998990,1,Goto(smsmotx,${CALLERIDNUM},1)
; outgoing SMS
[smsmotx]
exten = _X.,1,Set(smsFrom=${CALLERIDNUM})
exten = _X.,n,SMS(${smsFrom},,${EXTEN},${CALLERIDNAME}) ; Create an SMS
exten = _X.,n,SMS(${smsFrom}) ; Send queued SMS
exten = _X.,n,Hangup()
When I attempt to send an SMS using smsq, Asterisk appears to be
behaving normally, a call is made etc., but the SMS never arrives ...
What am I doing wrong? Let me know what diagnostics I need to provide
if anyone wants to take a closer look.
thanks
/Per Jessen, Zürich
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Re: [asterisk-users] agents and music on hold with autoanswer..
If you want to be able to run accurate reporting, you should tell the agents that they must log out whenever they are unavailable to answer (...) you're right; I'm going to make this new rule (fortunately we have few agents only and it'd be easy to do that) thank you so much for your reply bye bye MAS! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No of Calls
Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Nokia E60 firmware update break SIP
On Mon, 16 Apr 2007 20:14:40 -0700, Martin Joseph wrote: The phone no longer registers with asterisk, although it displays the little icon as though it has, and it doesn't even seem to try to pass calls to asterisk... So, I would avoid 3.06330904 20-11-06 RM-49 i've got an E61 running the same firmware revision and it works fine and dandy with asterisk 1.2.17. one thing you may want to do is to delete all your SIP profiles in the phone and reconfigure it from scratch. upgrading firmware from 2.x to 3.x broke something which wasnt forward compatible. we had similar issues, but deleting all profiles and reconfiguring from scratch fixed it. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
http://site.asteriskguide.com/bandcalc/bandcalc.php On Tue, 17 Apr 2007 11:54:28 +0400, Arun Kumar [EMAIL PROTECTED] wrote: Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // Phone: +44 (0) 845 869 2749 SIP: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. thanks On 4/17/07, -- [ UxBoD ] -- [EMAIL PROTECTED] wrote: http://site.asteriskguide.com/bandcalc/bandcalc.php On Tue, 17 Apr 2007 11:54:28 +0400, Arun Kumar [EMAIL PROTECTED] wrote: Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // Phone: +44 (0) 845 869 2749 SIP: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
Try this one then as it is closer to the value you are getting http://www.asteriskguru.com/tools/bandwidth_calculator.php On Tue, 17 Apr 2007 12:22:34 +0400, Arun Kumar [EMAIL PROTECTED] wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. thanks On 4/17/07, -- [ UxBoD ] -- [EMAIL PROTECTED] wrote: http://site.asteriskguide.com/bandcalc/bandcalc.php On Tue, 17 Apr 2007 11:54:28 +0400, Arun Kumar [EMAIL PROTECTED] wrote: Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // Phone: +44 (0) 845 869 2749 SIP: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // Phone: +44 (0) 845 869 2749 SIP: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
- [ UxBoD ] -- wrote: Try this one then as it is closer to the value you are getting http://www.asteriskguru.com/tools/bandwidth_calculator.php When I do it here, the asteriskguru one comes out as less bandwidth than the asteriskguide one. (remembering that the guru one states total in both directions and the guide one states total in one direction). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card). thanks On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card) b'coz these are test server. what else I can use for timing. thanks On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GETVARIABLE and IAX
Hello, I'm a developer writing FastAgi scripts to do Asterisk call handling. One of the functions provided by these scripts is to collect call data and write it into our own custom CDR tables. For SIP-based calls, for example, we find it useful to capture the SIP call ID so that we can use it to marry together the CDR and a wireshark log of the SIP and RTP packets involved in the call. I now have a requirement to do the same thing for IAX-based calls. Is there an IAX equivalent to GETVARIABLE(SIP_HEADER(Call-ID)) that would return the 15-bit Call Number(s?) for the IAX streams involved in the call? Are there any IAX-specific variables at all that can be accessed via GETVARIABLE (or some alternative method)? Best wishes, Dominic Fox ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio Problems - Operating System??
Darren Nay wrote: Now my question. I've heard through the grapevine that the Operating system running Asterisk can make a big difference in performance. I am currently running SuSE Linux Enterprise Server 10.A friend of mine actually talked to someone at Digium about this specific problem and they told him -not- to run SuSE. Is this correct? Dunno about correct, but FWIW I'm using SUSE 10.3Alpha1. Haven't noticed any SUSE-specific problems. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk VS Cisco,Avaya,Siemens,....
Thomas Asterisk is several years proven with many tens of thousands of testimonies. Ours include deployments on all seven continents and some very large ones as well. Please contact me offline if I can assist specifically. Successful deployment though depends on many factors not least the competency of the person installing. We have had to put together quite a few rescue packages where the customer was left with an installer that had disappeared. Your choice of Interface will be an important decision for the customer on a day-to-day basis and you need to consider the desktop tools such as softphone, online messaging, fax and sms if you are to compete directly with the incumbants. Where you will always exceed the incumbants is flexibility. You may wish to look at OutCall that we release recently as source and see if it will assist you in integrating with clients exisitng legacy softwares, not just Outlook but possibly proprietary as well. http://outcall.sf.net http://www.bicomsystems.com/home/C/P/731/143_3564/ Steve www.bicomsystems.com - Original Message - From: Thomas Deillon To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, April 17, 2007 8:50 AM Subject: [asterisk-users] Asterisk VS Cisco,Avaya,Siemens, Hi all, We are an ISP in Switzerland using only Asterisk boxes for VoIP and we are looking for others companies using Asterisk too to prove to our clients that Asterisk is a stable solution used by other big companies. Thanks a lot for your help, Thomas Deillon Telecom Engineer Smart-Telecom -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with queue announcements under high call volumes
On Mon, 2007-04-16 at 18:18 -0400, Matthew J. Roth wrote: [snip] Apr 16 14:30:01 WARNING[19451] file.c: Failed to write frame [snip] Just a wild guess because I don't really have an idea what is causing this but are your ulimit settings high enough? Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail - digits/1F does not exist in any format
Philippe Lindheimer wrote: I've seen this before, in an ISDN card (can't recall which one) that defaults the incoming language to german. How does this work? my misdn.conf has 'language=en'. Since you don't have german, it defaults to english files but voicemail still runs through the german logic (e.g. 1F for femail). Are you saying that the voicemail got left with language=de (somehow), and that it's looking for German language-files when I try to play it back? I reported a bug against this, it was silently killing the call - no error handling. Yep, that's what happens. The playback just stops. I suggested that they check if the desired language is installed and if not, that within the app the 'temporarily change' the language to english so that it doesn't go off looking for sound files that are not there. I can't recall the bug number - but they didn't feel it was a reasonable approach ... different opinions I guess, they decided the behavior was accetable. I'll have a look around for that. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
Install zaptel and only enable the ztdummy module. As long as you are not running in a VM, this will supply you the timing that you are looking for. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Arun Kumar [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Thomas Kenyon [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 4:54:47 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] No of Calls how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card) b'coz these are test server. what else I can use for timing. thanks On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
Steve Totaro wrote: You could buy one of those X100P clones for ~$20 shipped and use that for timing (and also an added FXO port), or a bare TDM400P with no modules for ~$100 and have the option of adding modules for future upgrades. I thought that if yo uused a bare TDMP400P, that you needed a modified zaptel to enable timing on it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No of Calls
Thomas Kenyon wrote: Steve Totaro wrote: You could buy one of those X100P clones for ~$20 shipped and use that for timing (and also an added FXO port), or a bare TDM400P with no modules for ~$100 and have the option of adding modules for future upgrades. I thought that if yo uused a bare TDMP400P, that you needed a modified zaptel to enable timing on it. Yes and no, as Tzafrir Cohen pointed out, you can define timing only when loading the module or you can modify the zaptel source. I would do both personally. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
Just saw this article this morning: http://www.engadget.com/2007/04/16/vonage-no-workaround-were-pretty-much-screwed/ What happened to their workaround, whereby they route all of their traffic to someone else, who takes cares of LCR and ENUM? I don't understand how that wouldn't indemnify Vonage. On 4/13/07, Salvatore Giudice [EMAIL PROTECTED] wrote: My wife's name is Nanae... =) The VoIP patent stuff is something that needs to be talked about more. VoIP is really going to suffer in the years to come because of patents. Might make a good topic for a whitepaper at a conference of speaking engagement. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702)979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Friday, April 13, 2007 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit I love this thread, especially when it came to the chicken boner part of the discussion - brings back NANAE with a smile - and I'm glad no one found it off-topic, I think it's well worth talking about (the suit, not the chicken boners) as this may have an effect on some of what we do. I hope to talk a little about it on the Asterisk Users Conference today at 12:30 EDT if anyone wants to. Otherwise, it's about features.conf and whatever else comes up. For info, see http://x2z.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon-Vonage Lawsuit
Where can I see the actual statement cited. The whole Vonage statement is in fact riddled with such holes, making it hard to figure out exactly what's going on Thanks, Steve Matt wrote: Just saw this article this morning: http://www.engadget.com/2007/04/16/vonage-no-workaround-were-pretty-much-screwed/ What happened to their workaround, whereby they route all of their traffic to someone else, who takes cares of LCR and ENUM? I don't understand how that wouldn't indemnify Vonage. On 4/13/07, *Salvatore Giudice* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: My wife's name is Nanae... =) The VoIP patent stuff is something that needs to be talked about more. VoIP is really going to suffer in the years to come because of patents. Might make a good topic for a whitepaper at a conference of speaking engagement. -- Salvatore Giudice [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702)979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Wilson Pickett Sent: Friday, April 13, 2007 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit I love this thread, especially when it came to the chicken boner part of the discussion - brings back NANAE with a smile - and I'm glad no one found it off-topic, I think it's well worth talking about (the suit, not the chicken boners) as this may have an effect on some of what we do. I hope to talk a little about it on the Asterisk Users Conference today at 12:30 EDT if anyone wants to. Otherwise, it's about features.conf and whatever else comes up. For info, see http://x2z.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using meetme like call
Rob Schall wrote: One easy way to get close to this affect: Create a group dialDial(SIP/1000SIP/1001) then have a dynamic meetme room generating extension. This way, you can put them on hold for a brief second, dial that extension, create a room, then transfer them into it. This keeps the number of conference rooms to a min, while letting you create them on the fly for when you need more than 3 people on a call. Rob Thanks Rob, another way (I think): I make a standard 2 way call (2000 to 2001), if other user (2002) call 2000 or 2001 and the DIALSTATUS is busy using channelredirect I put the three user in one conference. I think this is MY solution... Now I try! -- Pasqualotto Enrico Netspin srl mail: [EMAIL PROTECTED] cell: 347 3292620 web: www.netspin.it smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Whats this about!
[Apr 17 09:14:45] WARNING[11234]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 Hi all i ned to know what the above warning is trying to say. I have a slight idea that its about some audio conversion, maybe. but can anybody tell me for sure whats it about? -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] master.csv interpretation
Try this: http://horanappraisals.com/asterisk/total_account_codes/ Thanks, Jeremy Download and save videos directly from youtube http://downloadandsaveyoutubevideos.info/ On 4/3/07, Adrian Marsh [EMAIL PROTECTED] wrote: Anyone know of any tools for interpreting master.csv call logs? (Excel is kind of basic) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whats this about!
This msg shows up whenever i start asterisk on my machine using the following command /usr/sbin/asterisk -c its shown 3 times, everytime [Apr 17 11:11:37] WARNING[27872]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 .[Apr 17 11:11:37] WARNING[27872]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Apr 17 11:11:37] WARNING[27872]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 its the same everytime i dont know y its shown 3 times On 4/17/07, Astawerks [EMAIL PROTECTED] wrote: what were you doing when you seen that message? Astawerks VoIP Hardware sales and consulting http://www.astawerks.com 614-495-1400 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Rizwan Hisham *Sent:* Tuesday, April 17, 2007 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Whats this about! [Apr 17 09:14:45] WARNING[11234]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 Hi all i ned to know what the above warning is trying to say. I have a slight idea that its about some audio conversion, maybe. but can anybody tell me for sure whats it about? -- Regards Rizwan Hisham Software Engineer -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/762 - Release Date: 4/15/2007 4:22 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/762 - Release Date: 4/15/2007 4:22 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRAM
So who is going to be the first person to roll out an Asterisk server with PRAM memory? http://hardware.slashdot.org/article.pl?sid=07/04/17/0155210 At least it will take care of the worrying issue that Flash memory only has so many re-writes in it's lifetime for all the appliance Asterisk builders Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation image001.gif Description: image001.gif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problems in Fedora 6
On Tue, 2007-04-17 at 13:47 +1200, Aaron Martin wrote: If I manually run the following commands: modprobe zaptel modprobe wctdm ztcfg -vvv asterisk -vvvc Then SOMETIMES asterisk will work perfectly with the zap channels, allowing both incoming and outgoing calls as per my dialplan. I realize this is a bit of a shot in the dark, but with FC6 and a TDM-400 31B (3 FXS, 1 FX0), I found that I had to also load the wctdm24xxp module. This really shouldn't be needed, and lsmod shows it isn't actually used, but my card wouldn't work correctly without it. --Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] peers are using wrong contexts
Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, default is always being used. The output of sip show peers shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the default context. Please, could anyone help me resolve this. Thanks in advance. This is a part of users.conf [951XX] callwaiting = yes cid_number = 951XX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XX secret = 00 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Whats this about!
what were you doing when you seen that message? Astawerks VoIP Hardware sales and consulting HYPERLINK http://www.astawerks.com/http://www.astawerks.com 614-495-1400 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham Sent: Tuesday, April 17, 2007 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Whats this about! [Apr 17 09:14:45] WARNING[11234]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 Hi all i ned to know what the above warning is trying to say. I have a slight idea that its about some audio conversion, maybe. but can anybody tell me for sure whats it about? -- Regards Rizwan Hisham Software Engineer -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/762 - Release Date: 4/15/2007 4:22 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/762 - Release Date: 4/15/2007 4:22 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue report statistics
Here is the run down... billsec is talk time duration is wait time dst is the queue extension lastdata is the queue name lastapp will show logins dstchannel is the destination agent disposition is answered or abandoned status Mysql example to show all agent call detail for agent 8000 on queue number 8877... ( I have a bad habit of using like statements, this will work with = if you type better than I normally do, just lose the %) SELECT * FROM cdr WHERE dst LIKE '8877%' AND dstchannel LIKE 'Agent/8000%' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using meetme like call
hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because for the users is the simplest (I think). The problem is that when user call one extension that isn't available or not responding the first user remain in the room for all work day. :( There's a way to make ring two phone and enter in the conference in the same time? Thank Enrico. -- Pasqualotto Enrico Netspin srl mail: [EMAIL PROTECTED] cell: 347 3292620 web: www.netspin.it smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
I'm sure they are exploring all options. Eventually, it's just a matter of time until the investors start with the class action lawsuits. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, April 17, 2007 8:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit Just saw this article this morning: http://www.engadget.com/2007/04/16/vonage-no-workaround-were-pretty-much-scr ewed/ What happened to their workaround, whereby they route all of their traffic to someone else, who takes cares of LCR and ENUM? I don't understand how that wouldn't indemnify Vonage. On 4/13/07, Salvatore Giudice [EMAIL PROTECTED] wrote: My wife's name is Nanae... =) The VoIP patent stuff is something that needs to be talked about more. VoIP is really going to suffer in the years to come because of patents. Might make a good topic for a whitepaper at a conference of speaking engagement. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702)979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Wilson Pickett Sent: Friday, April 13, 2007 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit I love this thread, especially when it came to the chicken boner part of the discussion - brings back NANAE with a smile - and I'm glad no one found it off-topic, I think it's well worth talking about (the suit, not the chicken boners) as this may have an effect on some of what we do. I hope to talk a little about it on the Asterisk Users Conference today at 12:30 EDT if anyone wants to. Otherwise, it's about features.conf and whatever else comes up. For info, see http://x2z.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Audio Problems - Operating System??
I have had this issue and my problem was my motherboard timing. i changed it to diffrent brand and that fixed the issue. BTW i dont use SUSE, i use CENTOS but i dont think it has anything to do with it. Message: 17 Date: Mon, 16 Apr 2007 15:34:47 -0600 From: Darren Nay [EMAIL PROTECTED] Subject: [asterisk-users] Audio Problems - Operating System?? To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hey All, I've been using Asterisk for a couple years now, but have always had some unsolvable audio problems. I get audio stuttering and popping quite often. Even if I have just one call up! The server is a Dual Duo-Core 3.0ghz Xeon Processor PC with 4 GB or ram. It just seems to me that this should NOT be happening. The server resources are nearly 98% idle. I've tried using the SLN audio file format, which does reduce the CPU usage when playing audio files, but it didn't help the audio quality. I've also tried putting my audio files on a RAM Drive and still have the same problem. I've also slimmed my asterisk system down to load only the modules that I am using via modules.conf. Now my question. I've heard through the grapevine that the Operating system running Asterisk can make a big difference in performance. I am currently running SuSE Linux Enterprise Server 10.A friend of mine actually talked to someone at Digium about this specific problem and they told him -not- to run SuSE. Is this correct? Has anyone else had any experience similar to this? I'm just wondering if Digium just wanted to push Asterisk Business Edition running on rPath on him, or if there really are some conflicts with SuSE that may cause audio instability. If so then it definitely would explain a lot regarding my poor audio quality problems. I would be happy to hear thoughts that any of you might have. Thanks so much! Darren Nay [EMAIL PROTECTED] -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070416/b3716352/attachment-0001.htm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using meetme like call
Hi Enrico,
you can achieve this with the G option of Dial command
Here is a quick dialplan snippet
[from-internal-custom]
exten = 4002,1,Noop(MeetMeTest Creating MeetMe ${CALLERID(num)})
exten = 4002,n,Answer()
exten = 4002,n,Set(_MEETMEROOM=${CALLERID(num)})
exten = 4002,n,Dial(SIP/XX||G(meetme-custom^s^1))
[meetme-custom]
exten = s,1,MeetMe(${MEETMEROOM},dAxqa)
exten = s,2,MeetMe(${MEETMEROOM},qdx)
When the call is estabilished, call legs are sent to meetme-custom,s,1
(caller) and meetme-custom,s,2 (called)
I used the callerid as dynamic MeetMe room
Then have a look at 'a' option of MeetMe to solve your problem related
to hangup
Hope it helps
Regards
Enrico Pasqualotto ha scritto:
hi all, I have a little question about meetme in Asterisk.
One of my client ask me that all call can, if is necessary, become
conference for 3-4 user during conversation.
I think that are 2 way for make this:
1- all call (instead if the users are only 2) are conference
2- using n-way call
(http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO)
I decide to implement the first way because for the users is the
simplest (I think).
The problem is that when user call one extension that isn't available or
not responding the first user remain in the room for all work day. :(
There's a way to make ring two phone and enter in the conference in the
same time?
Thank Enrico.
--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino
Tel: +39 011 678 100
Fax: +39 011 678 275
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[asterisk-users] Voicemail files permission
I'm using asterisk 1.2.14 When asterisk stores voicemail messages in /var/spool/asterisk/voicemail/default/EXTENSION/INBOX files are created with: -rwx-- 1 asterisk web-aster 6690 Apr 17 16:08 msg0002.WAV -rwx-- 1 asterisk web-aster 6732 Apr 17 16:08 msg0002.gsm -rw--- 1 asterisk web-aster274 Apr 17 16:08 msg0002.txt -rwx-- 1 asterisk web-aster 65324 Apr 17 16:08 msg0002.wav I needed the files to have modes 660. I tried setting up umask in the script that starts asterisk and that did not help. After some searches I found that the apps/app_voicemail.c sets a define about this: #define VOICEMAIL_FILE_MODE 0600 that is used in: if ((ofd = open(outfile, O_WRONLY | O_TRUNC | O_CREAT, VOICEMAIL_FILE_MODE)) 0) But no matter what I set in there I always get the files created the same way. Any ideas? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . - signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
Heres a bunch of Vonage stuff here http://www.vonage-forum.com/article3032.html Astawerks VoIP Hardware sales and consulting http://www.astawerks.com AASTRA 9133i $124.10 614-495-1400 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, April 17, 2007 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit Where can I see the actual statement cited. The whole Vonage statement is in fact riddled with such holes, making it hard to figure out exactly what's going on Thanks, Steve Matt wrote: Just saw this article this morning: http://www.engadget.com/2007/04/16/vonage-no-workaround-were-pretty-mu ch-screwed/ What happened to their workaround, whereby they route all of their traffic to someone else, who takes cares of LCR and ENUM? I don't understand how that wouldn't indemnify Vonage. On 4/13/07, *Salvatore Giudice* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: My wife's name is Nanae... =) The VoIP patent stuff is something that needs to be talked about more. VoIP is really going to suffer in the years to come because of patents. Might make a good topic for a whitepaper at a conference of speaking engagement. -- Salvatore Giudice [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702)979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Wilson Pickett Sent: Friday, April 13, 2007 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit I love this thread, especially when it came to the chicken boner part of the discussion - brings back NANAE with a smile - and I'm glad no one found it off-topic, I think it's well worth talking about (the suit, not the chicken boners) as this may have an effect on some of what we do. I hope to talk a little about it on the Asterisk Users Conference today at 12:30 EDT if anyone wants to. Otherwise, it's about features.conf and whatever else comes up. For info, see http://x2z.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/762 - Release Date: 4/15/2007 4:22 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/762 - Release Date: 4/15/2007 4:22 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using meetme like call
That defiantly makes sense. And it probably could be one less step than mine as well. :) Enrico Pasqualotto wrote: Rob Schall wrote: One easy way to get close to this affect: Create a group dialDial(SIP/1000SIP/1001) then have a dynamic meetme room generating extension. This way, you can put them on hold for a brief second, dial that extension, create a room, then transfer them into it. This keeps the number of conference rooms to a min, while letting you create them on the fly for when you need more than 3 people on a call. Rob Thanks Rob, another way (I think): I make a standard 2 way call (2000 to 2001), if other user (2002) call 2000 or 2001 and the DIALSTATUS is busy using channelredirect I put the three user in one conference. I think this is MY solution... Now I try! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using meetme like call
hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because for the users is the simplest (I think). The problem is that when user call one extension that isn't available or not responding the first user remain in the room for all work day. :( There's a way to make ring two phone and enter in the conference in the same time? Thank Enrico. -- Pasqualotto Enrico Netspin srl mail: [EMAIL PROTECTED] cell: 347 3292620 web: www.netspin.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: Ok I had a chance to test web-meetme 3.0.1 and I have few comments here - the Makefile for CBmysql lacks procedure that verifies existence of /var/lib/asterisk/sounds/conf-recordings directory where the conference records should reside. You are right that this should be documented at least, and part of The make install process ideally. I had to go through .php files to find out where they are supposed to be and create the directory manually. Strange enough, the recording still does not work and the main web interface lack any support for the record files (I would expect some link in the past conference list). There will be a link if the conference is recorded. I received a report of the recording option not working just this weekend and I started Looking for the cause today. I was out of town for a week, otherwise I would have gotten a chance to respond earlier. What version of Asterisk are you using? I've had recording working with SVN before 1.4, the 1.4 betas and currently 1.4.1. *** Update *** Recordings are tied to a moderator joining the conference at this time. I may need to change that based on feedback/requests to do so. *** Update *** - Active Directory integration works fine, but we should be able to gather email addreess for the participant from AD, too (avoid using the sql users table if web-meetme was configured to use AD). Actually this is still a big mystery to me - how do I add participants to the conference using the web-interface? It must be done via the web interface as otherwise we have no information about the participant except of his channel number. I've never user the sql option for the user/participant. It was contributed by another user of the suite. Depending on the technology the caller used to call into the conference you should have their Caller-id number and possibly their Caller-id name. What additional Information would you like to see? It is very promising project but it needs - a better documentation Contributions welcome. There is a new How-To up on SF that covers the installation on a step by step basis. I've tried to comment the configuration files to make it clear how each setting works. Some features have been contributed to the project, and I am sorry to say that beyond making sure they integrate cleanly, I have not taken enough time to document their setup and use. I guess I should ask for supporting documentation before merging the changes/features. - fix the conference recording backend I hope to have this resolved this week. *** See update above *** - clear the confusion with users/email addresses/mail notifications. More details about what you would like the system to do please... If all that works, it would be just perfect... Thanks, Ondrej Thank you for the feedback. I am surprised almost daily how many people have found it useful. I did not really expect it to be as popular as it has become, and I am more than happy to try and address any problems. Thanks, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using meetme like call
One easy way to get close to this affect: Create a group dialDial(SIP/1000SIP/1001) then have a dynamic meetme room generating extension. This way, you can put them on hold for a brief second, dial that extension, create a room, then transfer them into it. This keeps the number of conference rooms to a min, while letting you create them on the fly for when you need more than 3 people on a call. Rob Enrico Pasqualotto wrote: hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because for the users is the simplest (I think). The problem is that when user call one extension that isn't available or not responding the first user remain in the room for all work day. :( There's a way to make ring two phone and enter in the conference in the same time? Thank Enrico. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trigger a wake-up call from the shell?
I have set up a script that ensures certain services are up on my Asterisk box (Trixbox 2.0). I would like it to trigger a wake-up call if certain conditions aren't meant. How might I accomplish this from the shell? -- Donovan Niesen Customer Contact Services www.yourccsteam.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] peers are using wrong contexts
Is this Asterisk 1.4.x? from samples/extensions.conf... ; ; User context is where entries from users.conf are registered. The ; default value is 'default' ; ;userscontext=default ; Is this any help? regards, Drew dima wrote: Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, default is always being used. The output of sip show peers shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the default context. Please, could anyone help me resolve this. Thanks in advance. This is a part of users.conf [951XX] callwaiting = yes cid_number = 951XX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XX secret = 00 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer via CTI
I used autodial to allow a user to make a call by clicking on a web directory and placing a call file into the Asterisk outgoing directory. That works perfectly for me. What if I want to click on the web directory and transfer my existing call? Is there a comparable interface? Thank you. Phil New York ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trigger a wake-up call from the shell?
On 4/17/07, Donovan Niesen [EMAIL PROTECTED] wrote: I have set up a script that ensures certain services are up on my Asterisk box (Trixbox 2.0). I would like it to trigger a wake-up call if certain conditions aren't meant. How might I accomplish this from the shell? Take a look at call files. They allow you to generate a call from * to a phone and then do whatever you want with the other end (play a message, connect you to a tech, etc.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM24 Cards
Hi List; For TDM24 Cards, what it means that it support 6 FXS and/or FXO modules for a total of 24 lines? Does it means that this card can be divided to 6 modules (FXO or FXS) where each module will support 4 ports? Also, when it syas in the characteristics that: Zero (0) FXS modules (green) Six (6) FXO modules (red) Or: One (1) FXS module (green) Zero (0) FXO modules (red) What it means by that? Lastly: how I can take a decision to use TDM2410E or TDM2406E or TDM2401E or TDM2433E?? Regards Bilal Ghayad ITS Functional Consultant Mobile: 009659849460 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail files permission
On Tue, 17 Apr 2007, Gustavo Felisberto wrote: I'm using asterisk 1.2.14 When asterisk stores voicemail messages in /var/spool/asterisk/voicemail/default/EXTENSION/INBOX files are created with: -rwx-- 1 asterisk web-aster 6690 Apr 17 16:08 msg0002.WAV -rwx-- 1 asterisk web-aster 6732 Apr 17 16:08 msg0002.gsm -rw--- 1 asterisk web-aster274 Apr 17 16:08 msg0002.txt -rwx-- 1 asterisk web-aster 65324 Apr 17 16:08 msg0002.wav I needed the files to have modes 660. I tried setting up umask in the script that starts asterisk and that did not help. After some searches I found that the apps/app_voicemail.c sets a define about this: #define VOICEMAIL_FILE_MODE 0600 that is used in: if ((ofd = open(outfile, O_WRONLY | O_TRUNC | O_CREAT, VOICEMAIL_FILE_MODE)) 0) But no matter what I set in there I always get the files created the same way. Any ideas? Yes. in app_voicemail.c, you've already found the #define, but you might also want to add: fchmod (txtdes, 0660) ; after line 2615. ie. before the comment line: /* Now play the beep once we have the message number for our next message. */ You also need to edit app.c: round about line 600, change: others[x] = ast_writefile(recordfile, sfmt[x], comment, O_TRUNC, 0, 0700); into: others[x] = ast_writefile(recordfile, sfmt[x], comment, O_TRUNC, 0, 0660); Enjoy! (I did this to enable a web based php voicemail system to work without requiring it to be suidperl - I'm guessing you're doing sometime similar!) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
2007/4/16, Stephen Bosch [EMAIL PROTECTED]: It's not entirely clear to me why people continue to cling to the idea that Asterisk should handle faxing also. What's the benefit? Hylafax is great, and you can even use it on the same machine. The benefit, I guess, is to save a dedicated line and not changing incoming fax numbers, as you cannot port them individually. But you're right to point it also has drawbacks ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with queue announcements under high call volumes
Patrick wrote: [snip] Just a wild guess because I don't really have an idea what is causing this but are your ulimit settings high enough? Patrick, We have the maximum number of open file descriptors set to 65536. Are there any other resources that you would suggest raising the limit for? Thanks for the suggestion, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
In my case, I have a single PRI coming into a single port PRI card on my Asterisk box. My old fax numbers (prior to our switch to PRI) are DIDs on that PRI. Using IAXModem+HylaFAX, I can recieve faxes without having seperate POTS lines for faxes, or an external fax board, or a multiport T1 card. Adding new fax numbers involves assigning a new DID...so adding additional fax lines doesn't cost anything (again, IAXModem is free...) --TS Olivier [EMAIL PROTECTED] 04/17/07 2:19 PM 2007/4/16, Stephen Bosch [EMAIL PROTECTED]: It's not entirely clear to me why people continue to cling to the idea that Asterisk should handle faxing also. What's the benefit? Hylafax is great, and you can even use it on the same machine. The benefit, I guess, is to save a dedicated line and not changing incoming fax numbers, as you cannot port them individually. But you're right to point it also has drawbacks ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What are service activation codes ?
Hello, What does http://www.voip-info.org/wiki/view/Asterisk+vertical+service+activation+codes exactly mean ? Is there anything richer with call forwarding, call back on no answer, etc ... Have european countries standardized such codes ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
2007/4/16, Stephen Bosch [EMAIL PROTECTED]: It's not entirely clear to me why people continue to cling to the idea that Asterisk should handle faxing also. What's the benefit? Hylafax is great, and you can even use it on the same machine. The benefit, I guess, is to save a dedicated line and not changing incoming fax numbers, as you cannot port them individually. But you're right to point it also has drawbacks ... saving the dedicated line is the biggest one, the next notch is what about when you have busy periods of 5min a month, do you get a second dedicated fax machine, fax line, hunt group etc ? we just give out line 2 in our hunt group as the fax number, have autodetect on the lines in asterisk and switch to the fax extension (channel bank with faxmodem and hylafax) when a fax comes in. this gives us an incoming fax pool as well with no extra line cost. This actually works fairly well through a channel bank even with all the extra d/a a/d conversion going on. We plan to try the soft fax channels as well but have not tried that yet. if someone has problems (pretty rarely), then we tell them the extension of the faxmodem to dial, we also have a real fax machine on another extension we have them try calling. That always works and the extra switching does not impact the faxes themselves. The biggest issue is remote does not talk to usr faxmodem, but will talk to a real fax machine, nothing about the asterisk detection and switching usually matters. we have never done it yet but the faxes could use all the lines at night for mass faxouts or certain lines or whatever since its all connected that way already anyway. lots of the benifits are not possible with a dedicated fax line. as for the drawbacks you could make a lot of the same arguments about voip in general but people still use it anyway. forgetting about voip and virtual channels and asterisk for a minute, why is it so hard to find a faxmodem that actually works as reliably as a physical fax machine ? Should be simple, but I guess it will eventually lead to it just becoming software only when the hardware manufacturers can't get it right. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with queue announcements under high call volumes
ulimit and ulimit -n Matthew J. Roth wrote: Patrick wrote: [snip] Just a wild guess because I don't really have an idea what is causing this but are your ulimit settings high enough? Patrick, We have the maximum number of open file descriptors set to 65536. Are there any other resources that you would suggest raising the limit for? Thanks for the suggestion, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BSNL caller ID (India)
Tzafrir, Can you Please let me know if the zapata.conf below is correct, or do I have to change something. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: tzafrir cohen [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 3:38:16 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) Yes below is the zapata.conf [trunkgroups] [channels] context=incoming usecallerid=yes cidsignalling=dtmf cidstart=ring hidecallerid=no callerid=asreceived callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A200 [slot:2 bus:4 span:1] group=0 signalling = fxs_ks channel = 1 group=0 signalling = fxs_ks channel = 2 group=0 signalling = fxs_ks channel = 3 group=0 signalling = fxs_ks channel = 4 Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 2:40:32 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Hmmm... are you sure you have configured your system to get callerid from the PSTN? callerid=asrecieved in zapata.conf. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with queue
Thanks Philipp, I tried making it 5000, and it worked. Once again thank for your help. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: philipp kempgen [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 5:58:22 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Problem with queue I waited for almost 5 minutes but still did not receive the call. Regards, Sanjay Rajdev - Original Message - From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 5:36:45 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Problem with queue Sanjay Rajdev wrote: Regards, Sanjay Rajdev Tha i did because i dont want any call to get disconnected. Can you let me know what can be the problem doing so. - Original Message - From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 4:35:42 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Problem with queue Sanjay Rajdev wrote: I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But the agents phone never rings again for that Call and therefore the caller goes on an infinite wait and listen the wonderfull on hold music. :) Here are few more observations. If I reload the asterisk it ring again for one time. OR If the agent relogin then also it rings for one more time. OR If the caller disconnecs and callback again, it will ring one more time. Here is the agent.conf [general] persistentagents=yes multiplelogin=no [agents] autologoff=150 wrapuptime=6 6/60/60 = 16,67 *hours*! Use something like 5. ---cut--- ; Define wrapuptime. This is the minimum amount of time when ; after disconnecting before the caller can receive a new call ; note this is in milliseconds. ---cut--- Sorry, it's milliseconds. But even 60 seconds is probably quite long. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending an SMS via Asterisk?
You haven't set the a or p variable or whatever
Description *SMS(queuename|[a][s])*
*SMS(queuename|[s]|number|message)* *deprecated*
a answer, i.e. send initial FSK packet. s act as service centre talking to
a phone.
On 4/17/07, Per Jessen [EMAIL PROTECTED] wrote:
I've been googling and reading a lot, but I'm not getting any closer to
getting an SMS sent via Asterisk.
Prior to switching to asterisk, I used sms_client on an ISDN line to
dial one of two Swisscom SMS centers: 0900900941 or 0794998990.
My dialplan looks like this:
exten = 0900900941,1,Goto(smsmotx,${CALLERIDNUM},1)
exten = 0794998990,1,Goto(smsmotx,${CALLERIDNUM},1)
; outgoing SMS
[smsmotx]
exten = _X.,1,Set(smsFrom=${CALLERIDNUM})
exten = _X.,n,SMS(${smsFrom},,${EXTEN},${CALLERIDNAME}) ; Create an SMS
exten = _X.,n,SMS(${smsFrom}) ; Send queued SMS
exten = _X.,n,Hangup()
When I attempt to send an SMS using smsq, Asterisk appears to be
behaving normally, a call is made etc., but the SMS never arrives ...
What am I doing wrong? Let me know what diagnostics I need to provide
if anyone wants to take a closer look.
thanks
/Per Jessen, Zürich
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Re: [asterisk-users] What are service activation codes ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier wrote: Hello, What does http://www.voip-info.org/wiki/view/Asterisk+vertical+service+activation+codes exactly mean ? Is there anything richer with call forwarding, call back on no answer, etc ... Have european countries standardized such codes ? Regards You need the ETSI standards. These include all of these and more are available from www.etsi.org - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRiUmFEtP/KMNOfRbAQLmjAgAiCqjpLO7H+JOCT234OrnhA0B0qy4WFqd ihSnhiixtOYeMCE54tA5WBxe2ht3kJ0eJZboEkvRbKLEonfI94noCzp83RYitzBo dT29xrYozzkKxDdKRX3LG2ZWOV4rkdrgc3YedJGYv5Vt+d5Sz6+IsgLF8fVEtm4J RKyTm6vcX2pwjo3klxh+esLiixQZEkb6LaEiZO2cPPBbPnbP2I9GF8tpwPusWjUh +PS4lmiAZT5kWD/OfMz6+slMrjbxjdAcWQr79w1slbn7y45trhfjQv8dnNdj5KAV 3vwIyoIuppWXMNIq0TaHblHBKCezORTizY9hweWxDiinyIs+vG/BbA== =QTyt -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BSNL caller ID (India)
Tzafrir, I am sure about both of them in my zapata.conf. I am on Asterisk 1.4.2 and the zapata.conf is in /etc/asterisk directory with all other asterisk configuration files Do you have any other idea which can help me finding out what is wrong. Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: Sanjay Rajdev [EMAIL PROTECTED] Sent: Wednesday, April 18, 2007 1:15:31 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Tue, Apr 17, 2007 at 03:38:16AM +0530, Sanjay Rajdev wrote: Yes below is the zapata.conf [trunkgroups] [channels] context=incoming usecallerid=yes cidsignalling=dtmf cidstart=ring Are you sure about those two? hidecallerid=no callerid=asreceived This is correct, of course. My typo. callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A200 [slot:2 bus:4 span:1] group=0 signalling = fxs_ks channel = 1 group=0 signalling = fxs_ks channel = 2 group=0 signalling = fxs_ks channel = 3 group=0 signalling = fxs_ks channel = 4 Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 2:40:32 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Hmmm... are you sure you have configured your system to get callerid from the PSTN? callerid=asrecieved in zapata.conf. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connection between Asterisk - Cisco 2851
Hello from Greece. I have an assignment related to Cisco. Specifically, i have to connect an asterisk server (VoIP) to a Cisco Router 2851 in a way that the router will handle some calls. I have not found any manuals that would be helpful to me. On the other hand, i found some manuals for connection between asterisk and Cisco Call Managers. I would like to know if the call manager is a part of the router or is an extra element. I would appriciate if you could help me. Thanks in advance, Aggelos ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problems in Fedora 6
On Tue, Apr 17, 2007 at 09:51:02AM -0600, Greg Woods wrote: On Tue, 2007-04-17 at 13:47 +1200, Aaron Martin wrote: If I manually run the following commands: modprobe zaptel modprobe wctdm ztcfg -vvv asterisk -vvvc Then SOMETIMES asterisk will work perfectly with the zap channels, allowing both incoming and outgoing calls as per my dialplan. I realize this is a bit of a shot in the dark, but with FC6 and a TDM-400 31B (3 FXS, 1 FX0), I found that I had to also load the wctdm24xxp module. This really shouldn't be needed, and lsmod shows it isn't actually used, but my card wouldn't work correctly without it. My guess: timing. Remove all the calls to ztcfg from modprobe.conf, use a proper init.d script, and you won't need to modprobe anything except wctdm . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connection between Asterisk - Cisco 2851
It's easy to create SIP trunks between * and CCM, but you can also create them between cisco voice gateways and asterisk too. Here is an example of a simple dial peer that routes inbound calls with specific destination numbers to an Asterisk server: dial-peer voice 1 voip description ## Inbound VoIP dial-peer to Asterisk ## destination-pattern 88.. voice-class codec 1 session protocol sipv2 session target ipv4:192.168.0.16 dtmf-relay rtp-nte If you don't need to use call damager for some other reason, then it's not required to use a Cisco router as a good quality PSTN gateway. Good luck! Hello from Greece. I have an assignment related to Cisco. Specifically, i have to connect an asterisk server (VoIP) to a Cisco Router 2851 in a way that the router will handle some calls. I have not found any manuals that would be helpful to me. On the other hand, i found some manuals for connection between asterisk and Cisco Call Managers. I would like to know if the call manager is a part of the router or is an extra element. I would appriciate if you could help me. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Default lenguage
Hi to all! i have installed asterisk 1.4.2 and asterisknow from the digium svn repository, when i was installing i select using menuselect utility the spanish voice lenguage pack. everythink is ok but i dont know how or where to tell asterisk to use the spanish as the default lenguage... i check on /var/lib/asterisk/sounds and i have the es directory with all the voices in spanish thanks in advanced! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problems in Fedora 6
On Tue, 2007-04-17 at 23:25 +0300, Tzafrir Cohen wrote: My guess: timing. Remove all the calls to ztcfg from modprobe.conf, use a proper init.d script, and you won't need to modprobe anything except wctdm . On my system at least, there is nothing in modprobe.conf at all regarding zaptel or any of its accompanying modules. All the module loading is done by direct modprobe commands in the init.d script, which in turn is automatically created by make install. Then there is the /etc/sysconfig/zaptel file (read by the init.d/zaptel script) that declares which modules to load. All I know is, when I declared only wctdm in there, the Digium card did not work. I had to add wctdm24xxp as well. The symptoms of not working were simply that the zap module would not load; there were no zap commands in the console. --Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.16 - No Caller ID
Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.5 txgain=5.5 group=1 pickupgroup=1-4 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel=1 channel=2 -- Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Connection between Asterisk - Cisco 2851
Thanks for your help..But i dont know yet whether is CCM embeded on cisco 2851 or it is an extra element? Furthermore, what do i have to define in asterisk in order to send some outbound calls to the router? thanks Από: [EMAIL PROTECTED] εκ μέρους Lee Pedder Αποστολή: Τρι 17/04/2007 23:46 Προς: Asterisk Users Mailing List - Non-Commercial Discussion Θέμα: Re: [asterisk-users] Connection between Asterisk - Cisco 2851 It's easy to create SIP trunks between * and CCM, but you can also create them between cisco voice gateways and asterisk too. Here is an example of a simple dial peer that routes inbound calls with specific destination numbers to an Asterisk server: dial-peer voice 1 voip description ## Inbound VoIP dial-peer to Asterisk ## destination-pattern 88.. voice-class codec 1 session protocol sipv2 session target ipv4:192.168.0.16 dtmf-relay rtp-nte If you don't need to use call damager for some other reason, then it's not required to use a Cisco router as a good quality PSTN gateway. Good luck! Hello from Greece. I have an assignment related to Cisco. Specifically, i have to connect an asterisk server (VoIP) to a Cisco Router 2851 in a way that the router will handle some calls. I have not found any manuals that would be helpful to me. On the other hand, i found some manuals for connection between asterisk and Cisco Call Managers. I would like to know if the call manager is a part of the router or is an extra element. I would appriciate if you could help me. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] internal sounds of asterisk / freePBX
Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Querying channel variables via the Manager API
Hello list, we are developing a new application that uses the Manager API in order to find a set of channels where variables are set in a predefined way. To do this, we currently send a Status command to obtain all available channels and then query them all, one by one, fot the status of a certain dialplan variable. As you can imagine, this gets rapidly pretty tedious as the number of active channels on a server grows and requires a lot of round-trips to and from the Asterisk server. I was wondering if there are more efficient ways to get: 1. a variable as set on all channels 2. the complete list of channel variables for one channel, using standard manager response block and not reverting to an execute CLI command show channel Local/[EMAIL PROTECTED] Anybody has ideas/hints on how to make all this a bit less cumbersome? Best regards, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
Can anyone recommend a VoIP provider who supports LNP? I need to move to a new provider for inbound calling and I want to keep my current numbers. My current provider is a gaggle of retards. Any recommendation? I need a service that is reliable. TIA, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommendations for a voip provider who supports LNP?
(sorry about the repost. I accidently had an unrelated subject in the original) Can anyone recommend a VoIP provider who supports LNP? I need to move to a new provider for inbound calling and I want to keep my current numbers. My current provider is a gaggle of retards. Any recommendation? I need a service that is reliable. TIA, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] internal sounds of asterisk / freePBX
It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR datasets
Hello list, I have been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] internal sounds of asterisk / freePBX
HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for a voip provider who supports LNP?
On 4/17/07, Salvatore Giudice wrote: (sorry about the repost. I accidently had an unrelated subject in the original) Can anyone recommend a VoIP provider who supports LNP? I need to move to a new provider for inbound calling and I want to keep my current numbers. My current provider is a gaggle of retards. Any recommendation? I need a service that is reliable. TIA, SG have you considered teliax.com ? check your numbers for LNP at the bottom left. I have been playing with voip for only about a month, but no complaints with teliax svc so far. -baji. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfercapability DIGITAL
Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] internal sounds of asterisk / freePBX
If that's what your phone is setup. Are you even using a SIP phone? What does the PEER context contain? Also, while Asterisk 1.2 and CALL WEAVER are basically the same (besides that fact that CALL WEAVER is trying to fully support faxing and Asterisk/Digium refuse to support correctly faxing) they do not share sound files. So if you are indeed using CALL WEAVER and their sounds you shouldn't be asking about that here. On 4/17/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR datasets
I'd be glad to test the software, however I'm sure you'll find that many people would be unwilling to provide their CDR (especially large ones) because chances are it would contain alot of personal/unidentifiable information. On 4/17/07, Lenz [EMAIL PROTECTED] wrote: Hello list, I have been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] internal sounds of asterisk / freePBX
On 4/17/07, Carlos Jerónimo wrote: HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks it can be if you have sound files with one of the following extensions : au / alaw / al / pcm / ulaw / ul / mu if you have .sln or .wav files then you are not allowing the necessary codec. adding the following lines may help : allow=slin allow=gsm allow=g726 allow=gsm allow=ilbc allow=g723 allow=g729 I don't know what damage, if any, could result from : allow=all - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] internal sounds of asterisk / freePBX
i use xlite and kphone in a diferent pc's. i can phone well. the problem is internal asterisk sounds. I think i not use Call Weaver, what is call weaver, i search at google but i'm was confused. i hope more help's. thanks 2007/4/17, Andrew Joakimsen [EMAIL PROTECTED]: If that's what your phone is setup. Are you even using a SIP phone? What does the PEER context contain? Also, while Asterisk 1.2 and CALL WEAVER are basically the same (besides that fact that CALL WEAVER is trying to fully support faxing and Asterisk/Digium refuse to support correctly faxing) they do not share sound files. So if you are indeed using CALL WEAVER and their sounds you shouldn't be asking about that here. On 4/17/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by
Re: [asterisk-users] Querying channel variables via the Manager API
Hi Lenz, Why not do it the same way as you do the queue log (for queuemetrics)? i.e. have a program which captures all events (or certain events), logs them to a MySql database and then you just query the database. There are many packages out there that run an Asterisk manager command at frequent intervals to obtain what they want. To me (as you have found this is very inefficient). I have an open source package called Asterisk Event Monitor that has a python script (ah.. I think you like perl better? but I like python) that connects to the Asterisk Manager and logs all events to a MySql table. There are PHP scripts that show the events and display the status of sip users, zap channels,etc. but you don't need that. The python script is similar to your qloader script in that my script grabs all the events and writes them to a MySql database. Anything you need to see is thus more efficiently extracted from the database table. any question about it , holler.. the code is here: http://www.micpc.com/eventmonitor earl On Tuesday 17 April 2007 17:32, Lenz wrote: Hello list, we are developing a new application that uses the Manager API in order to find a set of channels where variables are set in a predefined way. To do this, we currently send a Status command to obtain all available channels and then query them all, one by one, fot the status of a certain dialplan variable. As you can imagine, this gets rapidly pretty tedious as the number of active channels on a server grows and requires a lot of round-trips to and from the Asterisk server. I was wondering if there are more efficient ways to get: 1. a variable as set on all channels 2. the complete list of channel variables for one channel, using standard manager response block and not reverting to an execute CLI command show channel Local/[EMAIL PROTECTED] Anybody has ideas/hints on how to make all this a bit less cumbersome? Best regards, l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] internal sounds of asterisk / freePBX
The codecs are correct, so you are having other type of problem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR datasets
Yes, that's why I'm asking if anybody is willing to voluntarily share a CDR file :) I can promise that those files will not be given to third parties and this is a private project I run on my spare time, but this is surely a critical matter for most businesses. l. In data Wed, 18 Apr 2007 00:22:32 +0200, Andrew Joakimsen [EMAIL PROTECTED] ha scritto: I'd be glad to test the software, however I'm sure you'll find that many people would be unwilling to provide their CDR (especially large ones) because chances are it would contain alot of personal/unidentifiable information. On 4/17/07, Lenz [EMAIL PROTECTED] wrote: Hello list, I have been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: wrong values in duration and billsec in CDR
On 3/23/07, C F [EMAIL PROTECTED] wrote: On 3/22/07, Tomislav Parcina [EMAIL PROTECTED] wrote: C F wrote: So, how to solve this problem? Get an ISDN line, or maybe just VoIP. This really isn't answer to my question ;) Why not? FXO is answered as soon as you go off hook. There is no real way it will work on FXO, unless you get an ISDN or all VoIP lines. Actually some telcos use polarity reversals to signal answer and hangup states. That's what answeronpolarityswitch and hanguponpolarityswitch parameters in zapata.conf. Julian J. M. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Querying channel variables via the Manager API
Hi Earl, I was looking for something completely stateless - the problem with the approach you propose is that sometimes an event gets lost, or not logged correctly, and an engine like QueueMetrics' contains way too much code to handle such cases, even though Asterisk's logging facilities are usually quite reliable I was thinking that there must be a way to tell Asterisk give me a complete dump of all the available channel information including variables, as it seems a reasonable query to me :) - of course there are possible real-life workarounds, like caching a channel's status and variables and avoiding querying it again and again unless a timer has expired, but this is of course more complex and a bit less reliable than just asking for a fresh snapshot. BTW, I personally love Python, though I'm personally more fluent with Perl, after 12+ years using it :) Thanks for your offer anyway - I'll check it! l. In data Wed, 18 Apr 2007 01:09:54 +0200, Earl Terwilliger [EMAIL PROTECTED] ha scritto: Hi Lenz, Why not do it the same way as you do the queue log (for queuemetrics)? i.e. have a program which captures all events (or certain events), logs them to a MySql database and then you just query the database. There are many packages out there that run an Asterisk manager command at frequent intervals to obtain what they want. To me (as you have found this is very inefficient). I have an open source package called Asterisk Event Monitor that has a python script (ah.. I think you like perl better? but I like python) that connects to the Asterisk Manager and logs all events to a MySql table. There are PHP scripts that show the events and display the status of sip users, zap channels,etc. but you don't need that. The python script is similar to your qloader script in that my script grabs all the events and writes them to a MySql database. Anything you need to see is thus more efficiently extracted from the database table. any question about it , holler.. the code is here: http://www.micpc.com/eventmonitor earl On Tuesday 17 April 2007 17:32, Lenz wrote: Hello list, we are developing a new application that uses the Manager API in order to find a set of channels where variables are set in a predefined way. To do this, we currently send a Status command to obtain all available channels and then query them all, one by one, fot the status of a certain dialplan variable. As you can imagine, this gets rapidly pretty tedious as the number of active channels on a server grows and requires a lot of round-trips to and from the Asterisk server. I was wondering if there are more efficient ways to get: 1. a variable as set on all channels 2. the complete list of channel variables for one channel, using standard manager response block and not reverting to an execute CLI command show channel Local/[EMAIL PROTECTED] Anybody has ideas/hints on how to make all this a bit less cumbersome? Best regards, l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queues
Is there anyway to setup a queue with only one agent (device) which is always logged in. So when a call hits that queue the device will ring (if not already on a call) or will be put in the queue if the call is already in place? Thanks Miles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question regrading IAX
I have a server that only handles the inbound and outbound call and passes everything to the second server using IAX. Sometimes it so happen that a call comes in on the First machine, this machine forward to the second machine as an inbound call using IAX, now the second machine decides that this is an outbound call request so it forward it back to the first machine to make the outbound call. Is it possible once the second machine has decided that this is a outbound call, to intimate the first machine to directly make the outbond call without traversing to the second machine and coming back. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change to Sip in a phonecall, when the user registers meanwhile
Hello Folks! Is it following possible with Asterisk: User1 is known to Asterisk with a Sip Phone and a Mobile Number. If there is an ingoing phonecall for User1 Asterisk always tries to connect the call to the Sip Phone. BUT: if the Sip Phone is not registered, Asterisk calls the Mobile Number. And know the complicated part: If the user registers while Asterisk is ringing at the Mobile Number, Asterisk should cancel the call to the Mobile and invite the Sip Phone. How could I do this!? Thank you very much for every help! Kalle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Querying channel variables via the Manager API
Hi! I was thinking that there must be a way to tell Asterisk give me a complete dump of all the available channel information including variables In Asterisk 1.4: show application DumpChan Cheers, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TM Malaysia E1 PRI signaling
Anyone configured a E1 PRI in Kuala Lampur, Malaysia with TM Malaysia? What signaling did they provide, framing, formatting? primary-4essLucent 4ESS switch type for the U.S. primary-5essLucent 5ESS switch type for the U.S. primary-dms100 Northern Telecom DMS-100 switch type for the U.S. primary-dpnss DPNSS switch type for Europe primary-net5NET5 switch type for UK, Europe, Asia and Australia primary-ni National ISDN Switch type for the U.S. primary-ntt NTT switch type for Japan primary-qsigQSIG switch type primary-ts014 TS014 switch type for Australia (obsolete) Jason Aarons Consultant http://www.dimensiondata.com/na http://www.dimensiondata.com/na 904-338-3245 cell For urgent issues notify your Project Manager, for 24x7 support contact the Dimension Data NOC at 800-974-6584. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [Asterisk-java-users] asterisk-java.org up again with bonus article on Local/ channels
robert home wrote: does any one know what happened to www.asterisk-java.org or when it'll be back We had problems with the IN NS records at PSI. The problem is fixed now though it might still take a few hours for the changes to propagate. I am sorry for any inconvinience this outage may have caused and have provided a bonus article on Local/ channels to say sorry. The article include a nice diagram on how using Local channels and Originate relates to the events you see on the Manager API. =Stefan P.S. If you still encounter problems please contact me off-list and I'll have a look if I still missed anything. -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues
You can have the agent login once and newer log out. You can certainly set up your asterisk box to persit the login over the reload and the restart. persistentagents=yes Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group (FIG) may transmit information that is confidential and privileged information of Featherstone Informatics Group (FIG). Unless you are the intended addressee, you may not use, copy or disclose to anyone this communication or any information transmitted by this communication. If you have received such communication in error, please advise the sender by e-mail and/or telephone and destroy this communication immediately. This communication and any information transmitted by this communication may also be considered protected health information as defined under the Health Insurance Portability and Accountability Act and its related regulations (a.k.a., HIPAA) or any other similar state law. Please exercise due care and ensure that you comply with its contractual and legal obligations. - Original Message - From: Voip Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 18, 2007 5:23:18 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] queues ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues
Is there anyway to setup a queue with only one agent (device) which is always logged in. So when a call hits that queue the device will ring (if not already on a call) or will be put in the queue if the call is already in place? Sure, in queues.conf you can add many type of members (not just agents) like SIP or Local channels. So you don't need to use AgentLogin/CallBackLogin ej. [recepcion] musicclass = default monitor-format = wav49 strategy = ringall timeout = 15 retry = 2 autopause = no maxlen = 3 context = voicemail setinterfacevar = yes announce-frequency = 15 periodic-announce-frequency = 0 announce-holdtime = yes announce-round-seconds = 10 joinempty = strict leavewhenempty = strict eventwhencalled = yes eventmemberstatus = yes ringinuse = yes timeoutrestart = no member = SIP/9001,1 member = SIP/9005,2 -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Recommendations for a voip provider who supports LNP?
I need a straight origination/termination provider on a per minute charge plan. I would like to avoid a monthly subscription-based provider. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Tuesday, April 17, 2007 6:15 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Recommendations for a voip provider who supports LNP? On 4/17/07, Salvatore Giudice wrote: (sorry about the repost. I accidently had an unrelated subject in the original) Can anyone recommend a VoIP provider who supports LNP? I need to move to a new provider for inbound calling and I want to keep my current numbers. My current provider is a gaggle of retards. Any recommendation? I need a service that is reliable. TIA, SG have you considered teliax.com ? check your numbers for LNP at the bottom left. I have been playing with voip for only about a month, but no complaints with teliax svc so far. -baji. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfercapability DIGITAL
Have you tried: exten = s,n,SetTransferCapability(DIGITAL) ? []'s MM -Original Message- From: robert boardman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 17 Apr 2007 23:17:13 +0100 Delivered: Tue, 17 Apr 2007 19:15:09 Subject:[asterisk-users] Transfercapability DIGITAL Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/cgi-bin/imail.cgi?+_u=levelz_l=1,1176848736.557345.22480.arrino.hst.terra.com.br,4235,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR datasets
How large is large for you? []'s MM -Original Message- From: Lenz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 17 Apr 2007 23:46:28 +0200 Delivered: Tue, 17 Apr 2007 18:45:47 Subject:[asterisk-users] CDR datasets Hello list, I have been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/cgi-bin/imail.cgi?+_u=levelz_l=1,1176847280.613054.25939.caneria.hst.terra.com.br,3716,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trigger a wake-up call from the shell?
donovan,
by wake up call, I am assuming you have some condition that will trigger
a call not an actual 'wake me (a human) up call'...
here is what I set up to remind me to remind my son to take his
singulair pill. at 5;30pm.. I created a cron job to kick this shell
script off... thankfully, he doesn't require it anymore...
this uses channel Zap/2 my POTS line, to call out..
daveC
= the script =
#!/bin/sh
cd /home/dc/asterisk
LOGFILE=`pwd`/singulair.log
echo === ${LOGFILE}
echo ${0}: Started at `date`${LOGFILE}
chown asterisk *.call
# lets see it in the logfile
ls -l /var/spool/asterisk/outgoing ${LOGFILE}
echo -n 856-111-9876 ${LOGFILE}
cp -p /home/dc/asterisk/DJSingulaire856.call
/var/spool/asterisk/outgoing/2.call;
# lets see it in the logfile
ls -l /var/spool/asterisk/outgoing ${LOGFILE}
echo Done${LOGFILE}
= the .call file =
#
# This is a sample file that can be dumped in /var/spool/asterisk/outgoing
# to generate a call.
#
# Comments are indicated by a '#' character that begins a line, or follows
# a space or tab character. To be consistent with the configuration files
# in Asterisk, comments can also be indicated by a semicolon. However, the
# multiline comments (;-- --;) used in Asterisk configuration files are not
# supported. Semicolons can be escaped by a backslash.
#
# Obviously, you MUST specify at least a channel in the same format as you
# would for the Dial application. Only one channel name is permitted.
#
#Channel: Zap/1
Channel: Zap/2/8561119876
#
# You may also specify a wait time (default is 45 seconds) for how long to
# wait for the channel to be answered, a retry time (default is 5 mins)
# for how soon to retry this call, and a maximum number of retries (default
# is 0) for how many times to retry this call.
#
MaxRetries: 2
;RetryTime: 60
;waitTime: 30
#
# Once the call is answered, you must provide either an application/data
# combination, or a context/extension/priority in which to start the PBX.
#
##Context: dialout-alert-MyCell #doesn't quite work
Context: dialout-alert
Extension: s
Priority: 1
#
# Alternatively you can specify just an application
# and its arguments to be run, instead of a context
# extension and priority
#
#Application: VoiceMailMain
#Data: 1234
#
# You can set the callerid that will be used for the outgoing call
#
Callerid: DJ Singulaire (856) 778-0811
#
# You can set channel variables that will be passed to the channel.
# This includes writable dialplan functions.
#
#Set: file1=/tmp/to
#Set: file2=/tmp/msg
#Set: timestamp=20021023104500
#Set: CDR(accountcode|r)=blort
#Set: CDR(userfield|r)=42
= extensions.conf ===
[dialout-alert]
exten = s,1,Answer
exten = s,n,Playback(tt-weasels) (always get a kick out of this!)
exten = s,n,Playback(custom/take-your-singulair)
Donovan Niesen wrote:
I have set up a script that ensures certain services are up on my
Asterisk box (Trixbox 2.0). I would like it to trigger a wake-up call
if certain conditions aren't meant. How might I accomplish this from
the shell?
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Re: [asterisk-users] internal sounds of asterisk / freePBX
Try getting rid of all those macros etc.. so you can see what's going on, something simple like: exten = 500,1,Answer() exten = 500,n,Playback(beep) exten = 500,n,Hangup() Then dial 500 from your soft phone and see what happens. On 4/17/07, EWV2 [EMAIL PROTECTED] wrote: The codecs are correct, so you are having other type of problem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Christopher T Aloi -- ___ --Bandwidth and
