[asterisk-users] freepbx asterisk
dear i am a neewbi for asterisk, plz tell me or if any link is there where i can understand how asterisk, freepbx, web-meetme, dahdi all these tools works and how they are related. plz help me. regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 on Solaris/sparc
On Wednesday 18 July 2012, Jeremy Kister wrote: I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. The system itself is happy and phone calls (between two parties) seem fine. Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file. It sounds as though you may have run into an obscure issue with the default filesystems of Solaris and Linux having diametrically-opposed design philosophies with regard to caching policy. The following is a bit of an over-simplification, but here goes anyway. Solaris is built for robustness: it doesn't even return from a write to disk until it has verified that the data was written successfully. Linux is built for speed: it caches everything it possibly can, serves reads straight from cache and never commits anything to disk unless it's about to run out of RAM or a shutdown is requested. If you write a program that uses temporary files a lot, you can test on Linux on a scrapper and find it blisteringly fast -- only for it to slow to a crawl when you deploy on Solaris. This is because under Linux, short-term temporary files can be written to cache, read from cache and deleted from cache, all without ever seeing oxide -- but Solaris, unless instructed otherwise, will insist to write the whole lot to disk anyway. If this is what's causing your problems, you will have to do some low-level tweaking. But it *is* fixable. -- AJS Price Engines Ltd. DDI: 01283 707058. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] freepbx asterisk
On 20-07-12 09:15, neo nortan wrote: dear i am a neewbi for asterisk, plz tell me or if any link is there where i can understand how asterisk, freepbx, web-meetme, dahdi all these tools works and how they are related. plz help me. http://www.asteriskdocs.org/ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel is rsrvd and does not turn off
Hi Kevin. Thanks for the reply, can probably be just that. I'll contact the support that made the driver. In fact I sent the email to see if anyone has gone through a similar situation and give tips. Thank you, Rodrigo Lang. 2012/7/19 Kevin P. Fleming kpflem...@digium.com On 07/19/2012 03:49 PM, Rodrigo Lang wrote: I tried to shut down the channels with the command channel request hangup Khomp_SMS/B0C2-0 (Khomp_SMS/B0C2-0 the channel is locked), but nothing happens. I can only release these channels when I restart asterisk. You will probably need to ask the person(s) who made the channel driver you are using, since it's not part of Asterisk itself. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to work around asterisk ss7
Hello, Thanks for the suggestion. I have looked into some documentation. I have a doubt here will asterisk and libss7 support incoming and outgoing message? On Wed, Jul 18, 2012 at 5:40 PM, Bharat Lalcheta bharatlalch...@gmail.comwrote: You can use asterisk 1.6+ and libss7 for this functionality. Any Digium or Sangoma card working ok on this setup. Currently i am using it on both of them. On Wed, Jul 18, 2012 at 5:14 PM, Ashish Agarwal ashisha...@gmail.com wrote: Hello, Can someone give me an understanding about E1 with ISUP on CCS 7 signalling? Is it possible with asterisk + digium card and how Regards, Ashish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bharat Lalcheta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Ashish Agarwal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to work around asterisk ss7
messages you mean SMS ? If its SMS then thats not covered here in any of the stacks !! Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Fri, Jul 20, 2012 at 6:44 PM, Ashish Agarwal ashisha...@gmail.comwrote: Hello, Thanks for the suggestion. I have looked into some documentation. I have a doubt here will asterisk and libss7 support incoming and outgoing message? On Wed, Jul 18, 2012 at 5:40 PM, Bharat Lalcheta bharatlalch...@gmail.com wrote: You can use asterisk 1.6+ and libss7 for this functionality. Any Digium or Sangoma card working ok on this setup. Currently i am using it on both of them. On Wed, Jul 18, 2012 at 5:14 PM, Ashish Agarwal ashisha...@gmail.com wrote: Hello, Can someone give me an understanding about E1 with ISUP on CCS 7 signalling? Is it possible with asterisk + digium card and how Regards, Ashish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bharat Lalcheta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Ashish Agarwal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] freepbx asterisk
On Fri, Jul 20, 2012 at 6:21 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 20-07-12 09:15, neo nortan wrote: dear i am a neewbi for asterisk, plz tell me or if any link is there where i can understand how asterisk, freepbx, web-meetme, dahdi all these tools works and how they are related. plz help me. http://www.asteriskdocs.org/ Regards, Patrick You might get a bit of mileage out of http://www.voip-info.org/ wiki as well Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel is rsrvd and does not turn off
I checked with the plate holder and really is a driver error. Thanks, Rodrigo Lang. 2012/7/20 Rodrigo Lang rodrigoferreiral...@gmail.com Hi Kevin. Thanks for the reply, can probably be just that. I'll contact the support that made the driver. In fact I sent the email to see if anyone has gone through a similar situation and give tips. Thank you, Rodrigo Lang. 2012/7/19 Kevin P. Fleming kpflem...@digium.com On 07/19/2012 03:49 PM, Rodrigo Lang wrote: I tried to shut down the channels with the command channel request hangup Khomp_SMS/B0C2-0 (Khomp_SMS/B0C2-0 the channel is locked), but nothing happens. I can only release these channels when I restart asterisk. You will probably need to ask the person(s) who made the channel driver you are using, since it's not part of Asterisk itself. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Rodrigo Lang -- Rodrigo Lang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T.38 (PRI) Fax Debugging
1) Does anyone know of any software to debug the g711cap audio files Asterisk's res_fax generates? Google has not been very helpful. 2) These files are in WAV format, but my Windows Media Player cannot play them. The Linux file command reports RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz Does anyone have a solution? Is it a bug? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Emails
Has anyone been able to make an html template for the voicemail emails. We would love to customize them beyond just plain text. We have dome some Google searches and have not been able to come up with much. I believe that Switchvox has customized the voicemail email into html. Has anyone ever tried this? Thanks, /Josh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 (PRI) Fax Debugging
On 07/20/2012 09:48 AM, Eric Wieling wrote: Neither of your questions relate to T.38 :-) Maybe you meant T.30 instead. 1) Does anyone know of any software to debug the g711cap audio files Asterisk's res_fax generates? Google has not been very helpful. That will depend a lot on what you mean by 'debug'. We have found them useful to determine whether the audio path was 'clean' or not. For example, loading them into Audacity and looking at a spectral plot will tell you whether there were any extreme spikes, which usually are indicative of packet loss. You can also use them to look at the timing of the protocol interactions (delay between transitions between sending and receiving). If your goal is to actually demodulate the audio into data and then interpret the T.30 transactions, I'm not aware of any easily available tools for doing that. Commetrex (the vendor of the FAX stack in res_fax_digium) has one that they use for helping to analyze problems, but I don't believe they make it available outside their company. There might be something in spandsp, but I haven't looked. 2) These files are in WAV format, but my Windows Media Player cannot play them. The Linux file command reports RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz Does anyone have a solution? Is it a bug? I have never attempted to open them with Windows Media Player, so can't be of much help there. I know that Audacity will open them and play the audio properly, because that's what we used when we developed this 'audio capture' feature. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 (PRI) Fax Debugging
Indeed, I did mean T.30. I've opened the files in Audacity, they look clean. I'm trying to figure out why I sometimes get FAXOPT(statusstr)=remote channel hungup, FAXOPT(error)=HANGUP, and FAXOPT(status)=FAILED when the fax appears to have been received correctly. I'm also trying to figure out why sometimes the fax fails in the middle of the page. The local interface is PRI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, July 20, 2012 3:01 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] T.38 (PRI) Fax Debugging On 07/20/2012 09:48 AM, Eric Wieling wrote: Neither of your questions relate to T.38 :-) Maybe you meant T.30 instead. 1) Does anyone know of any software to debug the g711cap audio files Asterisk's res_fax generates? Google has not been very helpful. That will depend a lot on what you mean by 'debug'. We have found them useful to determine whether the audio path was 'clean' or not. For example, loading them into Audacity and looking at a spectral plot will tell you whether there were any extreme spikes, which usually are indicative of packet loss. You can also use them to look at the timing of the protocol interactions (delay between transitions between sending and receiving). If your goal is to actually demodulate the audio into data and then interpret the T.30 transactions, I'm not aware of any easily available tools for doing that. Commetrex (the vendor of the FAX stack in res_fax_digium) has one that they use for helping to analyze problems, but I don't believe they make it available outside their company. There might be something in spandsp, but I haven't looked. 2) These files are in WAV format, but my Windows Media Player cannot play them. The Linux file command reports RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz Does anyone have a solution? Is it a bug? I have never attempted to open them with Windows Media Player, so can't be of much help there. I know that Audacity will open them and play the audio properly, because that's what we used when we developed this 'audio capture' feature. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Emails
Have you tried to insert the HTML code directly into the body? Il 20/07/12 19:53, Josh Hopkins ha scritto: Has anyone been able to make an html template for the voicemail emails. We would love to customize them beyond just plain text. We have dome some Google searches and have not been able to come up with much. I believe that Switchvox has customized the voicemail email into html. Has anyone ever tried this? Thanks, /Josh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Emails
Yes and you just get html code in the email rather than the html format. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi Sent: Friday, July 20, 2012 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Emails Have you tried to insert the HTML code directly into the body? Il 20/07/12 19:53, Josh Hopkins ha scritto: Has anyone been able to make an html template for the voicemail emails. We would love to customize them beyond just plain text. We have dome some Google searches and have not been able to come up with much. I believe that Switchvox has customized the voicemail email into html. Has anyone ever tried this? Thanks, /Josh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Emails
On 07/20/2012 04:35 PM, Josh Hopkins wrote: Yes and you just get html code in the email rather than the html format. app_voicemail does not send MIME-encapsulated emails, it sends raw email. Unless you could add a Content-Type header to the message (which app_voicemail doesn't allow you to do), the email client that receives the HTML is going to treat it as plain text, which is what you saw. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Emails
You can change voicemail.conf to use a delivery option other than sendmail. Other than that, show us your body line from voicemail.conf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Hopkins Sent: Friday, July 20, 2012 4:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Emails Yes and you just get html code in the email rather than the html format. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi Sent: Friday, July 20, 2012 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Emails Have you tried to insert the HTML code directly into the body? Il 20/07/12 19:53, Josh Hopkins ha scritto: Has anyone been able to make an html template for the voicemail emails. We would love to customize them beyond just plain text. We have dome some Google searches and have not been able to come up with much. I believe that Switchvox has customized the voicemail email into html. Has anyone ever tried this? Thanks, /Josh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Emails
On Fri, Jul 20, 2012 at 12:53 PM, Josh Hopkins j...@prorivertech.comwrote: Has anyone been able to make an html template for the voicemail emails. We would love to customize them beyond just plain text. We have dome some Google searches and have not been able to come up with much. ** ** I believe that Switchvox has customized the voicemail email into html. Has anyone ever tried this? Thanks, /Josh What about changing 'mailcmd=' to a shell script that rewrites the email in the format you want before sending it to sendmail? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Emails
On 07/20/2012 04:48 PM, Warren Selby wrote: On Fri, Jul 20, 2012 at 12:53 PM, Josh Hopkins j...@prorivertech.com mailto:j...@prorivertech.com wrote: Has anyone been able to make an html template for the voicemail emails. We would love to customize them beyond just plain text. We have dome some Google searches and have not been able to come up with much. __ __ I believe that Switchvox has customized the voicemail email into html. Has anyone ever tried this? Thanks, /Josh What about changing 'mailcmd=' to a shell script that rewrites the email in the format you want before sending it to sendmail? That is most likely the best way to accomplish it; it would need to receive the already-composed email on stdin, then parse it, modify it, and regenerate it before sending it to the real mailer. This could be done using standard email libraries in many scripting languages. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
