Re: [asterisk-users] Vitelity Setup
Hi Alejandro, I removed the registration and tried as like yours, even inbound calls are not landing, anyways let me check with vitelity support. Hi Stephan, I am not using any SBC. As i said let me check with their support. Thanks for all the views comments. Regards, On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? I don't, but looks interesting. We should probably move this thread to the -biz list :) j -- _ -- 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] Vitelity Setup
On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Alejandro, I removed the registration and tried as like yours, even inbound calls are not landing, anyways let me check with vitelity support. In the Vitel web app you ust set the routing method to the IP of your pbx, maybe that's what's happening I'm pretty sure they check that the outbound calls use the same IP. Hi Stephan, I am not using any SBC. As i said let me check with their support. Thanks for all the views comments. Regards, On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? I don't, but looks interesting. We should probably move this thread to the -biz list :) j -- _ -- 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] Vitelity Setup
yes I did that, even then i am not able to make outbound and inbound as well. On Thu, May 24, 2012 at 12:42 PM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Alejandro, I removed the registration and tried as like yours, even inbound calls are not landing, anyways let me check with vitelity support. In the Vitel web app you ust set the routing method to the IP of your pbx, maybe that's what's happening I'm pretty sure they check that the outbound calls use the same IP. Hi Stephan, I am not using any SBC. As i said let me check with their support. Thanks for all the views comments. Regards, On Wed, May 23, 2012 at 10:48 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote: On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: [...] Just wanted to point out that after experiences with dozens of termination providers, I rate Vitelity pretty low. We still use them for US termination, which seems fine and relatively low cost. Thanks for the detailed input. How do you rate Gafachi? It took us a bit to understand the line model but we plan to use them massively... do you have any experience with Gafachi? I don't, but looks interesting. We should probably move this thread to the -biz list :) j -- _ -- 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] Transfer call issue
Is anybody else experiencing this problem ? -- Thanks, Phil - Original Message - Hello, a client attempted to transfer a call today which failed and returned the channel back to her. When this happened on the console we saw: Got OK on REFER Notify message the version that we are running is 1.8.9.2. Are you aware of any none issues please with this version as I could not find anything in Jira ? -- Thanks, Phil -- _ -- 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] No caller id when using cadence with DAHDI
Thanks for your input. I failed to mention my setup: Centos 5.8, Asterisk 1.8.11.1, libpri 1.4.12, DAHDI 2.5.1 I have a rhino r1t4 connected to 2 channel banks (adit 600). Also a digium B410P for connection to PSTN. Unfortunately rhino drivers don't compile against DAHDI 2.6.1 so I cannot test if the problem is solved in that version. For the time being I have removed cadence specification from any calls to Dial or Queue for FXS. Cheers, Panos On Wed, May 23, 2012 at 7:31 PM, Shaun Ruffell sruff...@digium.com wrote: On Wed, May 23, 2012 at 07:13:01PM +0300, Roeften wrote: Hello everyone, Just thought to let you know of a weird issue in Asterisk 1.8.? + Dahdi 2.6.? (and 2.5.?). When you specify any cadence in an app (Dial, Queue) then caller id does not work. For instance with the default cadences (everything commented out in chan_dahdi.conf) : Dial(DAHDI/54) caller id works Dial(DAHDI/54r1) caller id does not work (even for r1) I just found this issue did not have time to investigate further. Can anyone else verify that this is true for tonezones other than 13 (gr) which I am using? Could you retry with DAHDI-Linux 2.6.1? If you had previously tested with 2.6.0 and you are using a Digium analog card you might be hitting the issue that was fixed with r10481 wctdm24xxp: Shorten RINGOFF debounce interval from 512ms to 128ms [1]. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10481 -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] Vitelity Setup
On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- 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] Vitelity Setup
If I were troubleshooting this, the next thing I would do is verify connectivity on the relevant ports – more plainly, make sure that there's not a firewall rule with unintended consequences somewhere between your asterisk and your ISP. Otherwise, as Alejandro suggests – check with Vitelity support. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass a...@p2ee.org wrote: On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: yes I did that, even then i am not able to make outbound and inbound as well. That's weird. Guess you're gonna have to place a detailed ticket to them. It sounds like a network problem to me but without any detailed info it's hard to say. Maybe you can try sip set debug in the console for the IP and see if you can get an idea of what is happening at the packet level. We use Vitel, Skype SIP (we recently eliminated this one), and now Gafachi and they all seem to work per there set-up instructions right away. -- Alejandro -- _ -- 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] Detecting Fax Tones over IAX2
On 05/23/2012 08:41 PM, Cody Harris wrote: Hello All, I use IAX2 as the incoming connection from my DID provider. For whatever reason, this works best for me, SIP connections lag very frequently and only have about a 50% success rate for incoming calls (they get dropped mysteriously). I'm trying to implement a fax/voice switch. I have faxdetect=both in my sip.conf, and when I use sip, it works well. However, from what I can tell, there's no such option for IAX2 connections. Any ideas on what I can do here, or am I out of luck? It's quite hard to provide suggestions since we don't know what version of Asterisk you are using. However, in Asterisk 10, there is a channel-agnostic FAX detection function that can be applied to any channel type, so at a minimum that is one way to solve your problem. -- 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] Detecting Fax Tones over IAX2
- Original Message - On 05/23/2012 08:41 PM, Cody Harris wrote: Hello All, I use IAX2 as the incoming connection from my DID provider. For whatever reason, this works best for me, SIP connections lag very frequently and only have about a 50% success rate for incoming calls (they get dropped mysteriously). I'm trying to implement a fax/voice switch. I have faxdetect=both in my sip.conf, and when I use sip, it works well. However, from what I can tell, there's no such option for IAX2 connections. Any ideas on what I can do here, or am I out of luck? It's quite hard to provide suggestions since we don't know what version of Asterisk you are using. However, in Asterisk 10, there is a channel-agnostic FAX detection function that can be applied to any channel type, so at a minimum that is one way to solve your problem. BUT, even if fax is detected on an IAX2 channel, the only reason would be to change dialplan logic accordingly correct? There is no T.38 equivalent within IAX2, which means the OP will be handling faxes over a clear VoIP channel. The information here is of utmost relevance: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf --Tim -- _ -- 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] twenty thousands (20, 000) users, which asterisk and how many servers?
AsteriskNOW is a GUI on top of Asterisk; it does not change the ability of the system to handle call load. I thought the AsteriskNOW GUI was now a FreePBX clone. If so, every call now uses a perl script to make the call. This is considerably more overhead than a dial-plan written in native asterisk code. For the 20,000 calls, I would use Opensips for the SIP and Asterisk for audio playback, transcoding, voicemail, fun. Adrian -- _ -- 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 debug logs
Dear list, I have a project where I have: Asterisk 10 --AudioCodes -- E1-- Provider AudioCodes supports T.38 and passes the faxes through E1 to the provider. From what I read, Asterisk 10 has the most stable(full) T.38 among other releases. My Question: Can I somehow see in the logs if T.38 packets sending and see somehow its debugs? Or I should just be better off with capturing sip data through tcpdump? -- Regards, Arstan Jusupov -- _ -- 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] Detecting Fax Tones over IAX2
On 05/24/2012 09:44 AM, Tim Nelson wrote: BUT, even if fax is detected on an IAX2 channel, the only reason would be to change dialplan logic accordingly correct? There is no T.38 equivalent within IAX2, which means the OP will be handling faxes over a clear VoIP channel. The information here is of utmost relevance: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Absolutely correct. -- 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 debug logs
On 05/24/2012 09:54 AM, Arstan wrote: Dear list, I have a project where I have: Asterisk 10 --AudioCodes -- E1-- Provider AudioCodes supports T.38 and passes the faxes through E1 to the provider. From what I read, Asterisk 10 has the most stable(full) T.38 among other releases. Asterisk 10 has T.38 gateway support, but you won't be using it here because your AudioCodes device will be performing that function. Outside of gateway support, the T.38 functionality in Asterisk 1.8 and Asterisk 10 are very close to identical. My Question: Can I somehow see in the logs if T.38 packets sending and see somehow its debugs? Or I should just be better off with capturing sip data through tcpdump? This will depend on what you are asking the Asterisk 10 system to *do* with T.38. Are you sending FAXes from it, or receiving FAXes into it, or something else entirely? -- 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 debug logs
I am sending and receiving fax. I have an issue where sending and receiving is intermittent. Provider is claiming that It doesn't always receives t.38. So I thought if I could see if Asterisk is sending and receiving t.38 as it should be. Oh yeah, I am using ATA with t.38 support which is connected to a physical fax machine. Sent from my iPhone On May 24, 2012, at 11:04 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/24/2012 09:54 AM, Arstan wrote: Dear list, I have a project where I have: Asterisk 10 --AudioCodes -- E1-- Provider AudioCodes supports T.38 and passes the faxes through E1 to the provider. From what I read, Asterisk 10 has the most stable(full) T.38 among other releases. Asterisk 10 has T.38 gateway support, but you won't be using it here because your AudioCodes device will be performing that function. Outside of gateway support, the T.38 functionality in Asterisk 1.8 and Asterisk 10 are very close to identical. My Question: Can I somehow see in the logs if T.38 packets sending and see somehow its debugs? Or I should just be better off with capturing sip data through tcpdump? This will depend on what you are asking the Asterisk 10 system to *do* with T.38. Are you sending FAXes from it, or receiving FAXes into it, or something else entirely? -- 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] T.38 debug logs
Thanks Kevin, updtl debug is what I am looking for, I guess. Arstan Sent from my iPhone On May 24, 2012, at 11:25 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/24/2012 10:19 AM, Arstan Jusupov wrote: I am sending and receiving fax. I have an issue where sending and receiving is intermittent. Provider is claiming that It doesn't always receives t.38. This is very confusing. In your diagram, you show the connection to the provider being an E1. T.38 would never appear on an E1. So I thought if I could see if Asterisk is sending and receiving t.38 as it should be. Oh yeah, I am using ATA with t.38 support which is connected to a physical fax machine. You didn't include this in your diagram either. It sounds like you are just passing T.38 *through* Asterisk, between an ATA and the AudioCodes gateway. In that case, 'updtl debug' on the Asterisk CLI will show you the UDPTL traffic flowing through Asterisk. -- 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] Detecting Fax Tones over IAX2
I'm running on 1.8 as of now On May 24, 2012 11:00 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/24/2012 09:44 AM, Tim Nelson wrote: BUT, even if fax is detected on an IAX2 channel, the only reason would be to change dialplan logic accordingly correct? There is no T.38 equivalent within IAX2, which means the OP will be handling faxes over a clear VoIP channel. The information here is of utmost relevance: http://hylafax.sourceforge.**net/docs/fax-over-voip.pdfhttp://hylafax.sourceforge.net/docs/fax-over-voip.pdf Absolutely correct. -- 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 -- _ -- 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] Detecting Fax Tones over IAX2
Sorry I hit send by mistake (touchscreens, sigh) I've had good success with faxing over voip, I'm not expecting it to be perfect, and my provider (voip.Ms) is planning on t.38, but I'm looking for an interm solution. Audio faxing has worked every attempt both sending receiving (5 and 5). Should I update to asterisk 10 for this? Thanks, Cody On May 24, 2012 11:00 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/24/2012 09:44 AM, Tim Nelson wrote: BUT, even if fax is detected on an IAX2 channel, the only reason would be to change dialplan logic accordingly correct? There is no T.38 equivalent within IAX2, which means the OP will be handling faxes over a clear VoIP channel. The information here is of utmost relevance: http://hylafax.sourceforge.**net/docs/fax-over-voip.pdfhttp://hylafax.sourceforge.net/docs/fax-over-voip.pdf Absolutely correct. -- 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 -- _ -- 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] Detecting Fax Tones over IAX2
On Thursday 24 May 2012, Cody Harris wrote: I'm trying to implement a fax/voice switch. I have faxdetect=both in my sip.conf, and when I use sip, it works well. However, from what I can tell, there's no such option for IAX2 connections. Any ideas on what I can do here, or am I out of luck? Why not just get an extra inbound number from your provider, and use that for your faxes? Saves a lot of fart-arsing around. (Providers are now beginning to issue numbers beginning with a 0, thus requiring the STD code to be dialled even for a call within the same town. Would be ideal for a fax line.) -- 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] Detecting Fax Tones over IAX2
I had considered this, however, I was trying not to buy another DID. It may end up being the best solution. On May 24, 2012 12:26 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 24 May 2012, Cody Harris wrote: I'm trying to implement a fax/voice switch. I have faxdetect=both in my sip.conf, and when I use sip, it works well. However, from what I can tell, there's no such option for IAX2 connections. Any ideas on what I can do here, or am I out of luck? Why not just get an extra inbound number from your provider, and use that for your faxes? Saves a lot of fart-arsing around. (Providers are now beginning to issue numbers beginning with a 0, thus requiring the STD code to be dialled even for a call within the same town. Would be ideal for a fax line.) -- 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 -- _ -- 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] talkoff problem - relaxDTMF is off
About a month ago, we switched our PRIs from being run through a Nortel Meridan system to an Asterisk based PSTN gateway using a TE210P card. Since the cut over I have been getting reports of DTMF tones being heard by my internal users when on calls to/from the PSTN. I have confirmed via logging that the gateway machine is detecting what it thinks is DTMF and regenerating it I have relaxDTMF turned off. Asterisk version 1.8.12.0 (also happened on 1.8.7) Dahdi version 2.5.0 libpri version 1.4.12 Any suggestions? A small sample from the logs. [May 24 11:10:48] DTMF[8188] channel.c: DTMF begin 'D' received on DAHDI/i1/NXXNXX-37e [May 24 11:10:48] DTMF[8188] channel.c: DTMF begin passthrough 'D' on DAHDI/i1/NXXNXX-37e [May 24 11:10:48] DTMF[8188] channel.c: DTMF end 'D' received on DAHDI/i1/NXXNXX-37e, duration 80 ms [May 24 11:10:48] DTMF[8188] channel.c: DTMF end accepted with begin 'D' on DAHDI/i1/NXXNXX-37e [May 24 11:10:48] DTMF[8188] channel.c: DTMF end 'D' detected to have actual duration 64 on the wire, emulation will be triggered on DAHDI/i1/NXXNXX-37e [May 24 11:10:48] DTMF[8188] channel.c: DTMF end 'D' has duration 64 but want minimum 80, emulating on DAHDI/i1/NXXNXX-37e [May 24 11:10:48] DTMF[8188] channel.c: DTMF end emulation of 'D' queued on DAHDI/i1/NXXNXX-37e -- _ -- 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] use of Read cmd with AGI
Hello Steve, it's working fine, thanks for your suupport. thanks,Kamlesh Date: Tue, 22 May 2012 10:36:20 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] use of Read cmd with AGI Un-top-posting... From: alejandro.belt...@setcolombia.com Hi, try some like this: [PERL snippet using get_data AGI command] On Tue, 22 May 2012, Kamlesh Kumar wrote: I tried it but it doesn't work. beep file gets played, and when I enter any digit(s), it doesn't get stored in $keys variable. 1) Does enabling AGI debugging on the Asterisk console shed any clues? 2) Try reducing your AGI script to the bare minium. 3) Post the full source of your AGI and the Asterisk console log with AGI debugging enabled. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
[asterisk-users] extension status using AMI
Hi, I'm using AMI to get the extension status but always get -1 i.e. extension not found. #!/usr/bin/php -q ?phpinclude_once (phpagi-2.14/phpagi.php); include_once (/phpagi-2.14/phpagi-asmanager.php); $agi = new AGI(); $as = new AGI_AsteriskManager(); $exten = $agi-request['agi_extension'];$as-connect(localhost, user, passwd);$status = $as-ExtensionState($exten,'context',1); $status1 = $status['Status']; $agi-verbose(Extension status is .$status1);? Always return Extension status is -1 Thanks,Kamlesh -- _ -- 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] Asterisk MixMonitor starts recording 44 bytes file
Hello All, I have installaed asterisk 10.4 in my machine. Now suddenly MixMonitor application starts generating 44 Bytes of Recording file. Is this new tye of Bug? Help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.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] Asterisk MixMonitor starts recording 44 bytes file
- Original Message - From: Jayesh Labade jayesh.lab...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 24, 2012 3:10:29 PM Subject: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file Hello All, I have installaed asterisk 10.4 in my machine. Now suddenly MixMonitor application starts generating 44 Bytes of Recording file. Is this new tye of Bug? Help me.. Best Regards, Jayesh Labade e-mail: jayesh.lab...@gmail.com It's hard to tell from your email, but if I am correctly guessing what your problem is, what you are saying is that you are deliberately invoking mixmonitor in a call, but once the recording is finished, only 44 bytes are in the recorded file and when you play it you obviously don't get any audio. Generally this sort of problem means that you didn't actually record any audio and it isn't actually a bug, it's just that you are trying to record something that isn't possible to record. I've seen this happen with locally bridged analog phones and also with some cases involving directmedia with SIP. In order to be able to comment specifically on your problem though, I'd need to know what kind of channels you are using, specific details about them (like SIP directmedia settings and such) and what your dialplan looks like. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://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] twenty thousands (20, 000) users, which asterisk and how many servers
Thanks for all for the help and kindly reply. One last point that will help me alot: I am thinking to have 4 Servers running Asterisk and 2 Servers to be for database. The load to be distributed on the 4 Asterisk Servers with ability to be redundant (using any redundancy technique). The 4 Asterisk Servers to take the configuration from the Database Server and actually because there is 2 Database servers, then it will be redundant to each other (in case one database failed, the other will take over). My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Special thanks for the advise. Regards Bilal - Hi All; I need to use Asterisk for 20 000 users, so which asterisk version to be used? Is there asterisk version that supports 20,000 users on one hardware machine? Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to handle 20 000 users, and concurrent calls 2000? Or I need multiple servers, how much? If I am going to use multiple servers (until now I do not know how much, and I do not know if the barrier will be the asterisk software or the hardware), then do I have to use special SIP proxy or I have to use load balancer)? In this case, I have to use asterisk Database (so all the servers will read/write from the database)? What about AsteriskNow, can it support? AsteriskNOW is a GUI on top of Asterisk; it does not change the ability of the system to handle call load. Modern versions of Asterisk can easily handle 2,000 simultaneous calls, even with media (non-transcoded) passing through the server. We have a community member who has improved chan_sip in Asterisk 10 (and later) to be able to handle 10,000 simultaneous calls. Handling 20,000 registrations is probably more of a concern for Asterisk at this point; I've never heard of anyone attempting to handle that many on one system. In spite of all this, though, the other advice you've received in this thread is sound: even if a single system can handle the load, doing so is asking for a major problem if that system experiences a failure. You'd be much better off to at least split the load across two machines, both of which should be large enough to handle the entire load when necessary. -- 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] twenty thousands (20, 000) users, which asterisk and how many servers
My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Hi Bilal, You want to look the Asterisk realtime configuration features if you want to run your configuration from a database rather than configuration files. This should point you in the right direction and get you started: http://www.voip-info.org/wiki/view/Asterisk+RealTime It should be noted that if you're wanting to use AsteriskNow (which relies on FreePBX for its gui configuration features), then Asterisk realtime configuration will not work as it is not compatible at this time. Other web gui's might work, but I am not familiar with them. FreePBX's sentiment on the subject is shared here: http://www.freepbx.org/trac/wiki/AsteriskRealtime -John On 05/24/2012 05:46 PM, bilal ghayyad wrote: Thanks for all for the help and kindly reply. One last point that will help me alot: I am thinking to have 4 Servers running Asterisk and 2 Servers to be for database. The load to be distributed on the 4 Asterisk Servers with ability to be redundant (using any redundancy technique). The 4 Asterisk Servers to take the configuration from the Database Server and actually because there is 2 Database servers, then it will be redundant to each other (in case one database failed, the other will take over). My question is: Is it really possible to have the asterisk configuration in the database server instead of having it in conf files? HOW? I am asking this because what I noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that whatever I do configuration in the GUI, then the configuration will be generated in the conf files, so Asterisk will read from the conf files and not from the database directly. Is it right or I am confused and there is something else? If there is a method to let the configuration to be taken from the database (and not from the configuration), then HOW? Because even in AsteriskNow, the configuration will be generated in a conf files. Special thanks for the advise. Regards Bilal - Hi All; I need to use Asterisk for 20 000 users, so which asterisk version to be used? Is there asterisk version that supports 20,000 users on one hardware machine? Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to handle 20 000 users, and concurrent calls 2000? Or I need multiple servers, how much? If I am going to use multiple servers (until now I do not know how much, and I do not know if the barrier will be the asterisk software or the hardware), then do I have to use special SIP proxy or I have to use load balancer)? In this case, I have to use asterisk Database (so all the servers will read/write from the database)? What about AsteriskNow, can it support? AsteriskNOW is a GUI on top of Asterisk; it does not change the ability of the system to handle call load. Modern versions of Asterisk can easily handle 2,000 simultaneous calls, even with media (non-transcoded) passing through the server. We have a community member who has improved chan_sip in Asterisk 10 (and later) to be able to handle 10,000 simultaneous calls. Handling 20,000 registrations is probably more of a concern for Asterisk at this point; I've never heard of anyone attempting to handle that many on one system. In spite of all this, though, the other advice you've received in this thread is sound: even if a single system can handle the load, doing so is asking for a major problem if that system experiences a failure. You'd be much better off to at least split the load across two machines, both of which should be large enough to handle the entire load when necessary. -- 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] hangup not detected?
Here is the output from the cli: dozer*CLI core show channels Channel Location State Application(Data) DAHDI/5-1s@DB_LOOKUP:24 Up Swift(Schedule for employee 1 active channel 1 active call 1528 calls processed dozer*CLI core show channel dahdi/5-1 -- General -- Name: DAHDI/5-1 Type: DAHDI UniqueID: 1337821128.1363 LinkedID: 1337821128.1363 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) Rings: 1 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 15 Frames in: 3967 Frames out: 15882 Time to Hangup: 0 Elapsed Time: 20h56m23s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: DB_LOOKUP Extension: s Priority: 24 Call Group: 0 Pickup Group: 0 Application: Swift Data: Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX Blocking in: (Not Blocking) Variables: READSTATUS=TIMEOUT return_id= MAX_REPEAT=4 ODBCSTATUS=SUCCESS ODBCROWS=1 COUNTER=2 AAA_OUTPUT=Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX.. data=Thursday, May 24th, 2012, you are scheduled at XX id= ODBC_FETCH_STATUS=SUCCESS ~ODBCFIELDS~=id,data ODBC_ID=903 ID_VALIDATED=AAA_VALIDATE_EMP_NUM(27,) account_id= read_length=7 get_param2=E get_param1=27 validate_func=AAA_VALIDATE_EMP_NUM truck_text=employee number readprompt=AAA/enter_employee_number comp_num=27 BACKGROUNDSTATUS=SUCCESS CDR Variables: level 1: dnid= level 1: dst=4 level 1: dcontext=default level 1: channel=DAHDI/5-1 level 1: lastapp=Swift level 1: lastdata=Schedule for employee number : Thursday, May 24th, 2012, you are schedu level 1: start=2012-05-23 17:58:48 level 1: answer=2012-05-23 17:58:54 level 1: duration=75383 level 1: billsec=75377 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: accountcode=27_EMP level 1: uniqueid=1337821128.1363 level 1: linkedid=1337821128.1363 level 1: userfield=2885 level 1: sequence=1363 Since the 'lastapp' variable is 'Swift', this would indicate that the cepstral wrapper is having a problem, correct? Justin Killen From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Tuesday, May 22, 2012 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup not detected? Okay, the next time it gets in this state I'll gather that information. Justin Killen From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, May 21, 2012 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup not detected? On Fri, May 18, 2012 at 12:00 PM, Justin Killen jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com wrote: I have and automated call-in dispatch system where hundreds of people call in daily for 2-3 minutes each. The extension is set up to get their information, then text-to-speech the dispatch information (via odbc). It then loops 5 times then ends the call. These calls are being handled by an 8 port analog digium card. Sometimes though, I see calls via 'core show channel dahdi/1-1' that have a time of 16 hours. I'm not sure if this is a result of dahdi missing the hangup, ODBC timing out, or TTS failing for some reason. When a channel gets in this state, the call doesn't seem to progress through the dialplan, they always display the TTS line. Doing a 'dahdi destroy channel 1-1' doesn't seem to be effective - the only way I've been able to clear the calls is to do a 'dahdi restart' and/or restart the asterisk service. For TTS I'm using cepstral with the Swift wrapper. Here is a snippet of my dialplan: Can you post the CLI output of a call that gets hung? I'd like to see where it's hanging on. Also, as a work-around to attempt to solve the symptom and not the underlying issue, you could maybe setup a cron job that runs once every ten minutes that checks for stale calls using AMI, and then hangs up any calls up that are over 10 minutes long? Using the AMI Hangup command? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.comhttp://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:
Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file
- Original Message - From: Jayesh Labade jayesh.lab...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 24, 2012 4:10:29 PM Subject: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file Hello All, I have installaed asterisk 10.4 in my machine. Now suddenly MixMonitor application starts generating 44 Bytes of Recording file. Is this new tye of Bug? Help me.. Best Regards, Jayesh Labade Jayesh, Is this machine x86? There was a bug that was recently fixed and should show up in 10.5. https://issues.asterisk.org/jira/browse/ASTERISK-19727 Regards, Michael -- _ -- 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] extension status using AMI
Why don't you use AMI? There's are phpami project if you google. Sent from my iPhone On May 25, 2012, at 1:51 AM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hi, I'm using AMI to get the extension status but always get -1 i.e. extension not found. #!/usr/bin/php -q ?php include_once (phpagi-2.14/phpagi.php); include_once (/phpagi-2.14/phpagi-asmanager.php); $agi = new AGI(); $as = new AGI_AsteriskManager(); $exten = $agi-request['agi_extension']; $as-connect(localhost, user, passwd); $status = $as-ExtensionState($exten,'context',1); $status1 = $status['Status']; $agi-verbose(Extension status is .$status1); ? Always return Extension status is -1 Thanks, Kamlesh -- _ -- 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] hangup not detected?
Looks like Swift() (whatever that is) is not returning ? On 24 May 2012 23:07, Justin Killen jkil...@allamericanasphalt.com wrote: ** ** ** Here is the output from the cli: ** ** dozer*CLI core show channels Channel Location State Application(Data) DAHDI/5-1s@DB_LOOKUP:24 Up Swift(Schedule for employee 1 active channel 1 active call 1528 calls processed dozer*CLI core show channel dahdi/5-1 -- General -- Name: DAHDI/5-1 Type: DAHDI UniqueID: 1337821128.1363 LinkedID: 1337821128.1363 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) Rings: 1 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 15 Frames in: 3967 Frames out: 15882 Time to Hangup: 0 Elapsed Time: 20h56m23s Direct Bridge: none Indirect** **Bridge: none -- PBX -- Context: DB_LOOKUP Extension: s Priority: 24 Call Group: 0 Pickup Group: 0 Application: Swift Data: Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX Blocking in: (Not Blocking) Variables: READSTATUS=TIMEOUT return_id= MAX_REPEAT=4 ODBCSTATUS=SUCCESS ODBCROWS=1 COUNTER=2 AAA_OUTPUT=Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX.. data=Thursday, May 24th, 2012, you are scheduled at XX id= ODBC_FETCH_STATUS=SUCCESS ~ODBCFIELDS~=id,data ODBC_ID=903 ID_VALIDATED=AAA_VALIDATE_EMP_NUM(27,) account_id= read_length=7 get_param2=E get_param1=27 validate_func=AAA_VALIDATE_EMP_NUM truck_text=employee number readprompt=AAA/enter_employee_number comp_num=27 BACKGROUNDSTATUS=SUCCESS ** ** CDR Variables: level 1: dnid= level 1: dst=4 level 1: dcontext=default level 1: channel=DAHDI/5-1 level 1: lastapp=Swift level 1: lastdata=Schedule for employee number : Thursday, May 24th, 2012, you are schedu level 1: start=2012-05-23 17:58:48 level 1: answer=2012-05-23 17:58:54 level 1: duration=75383 level 1: billsec=75377 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: accountcode=27_EMP level 1: uniqueid=1337821128.1363 level 1: linkedid=1337821128.1363 level 1: userfield=2885 level 1: sequence=1363 ** ** ** ** ** ** ** ** ** ** Since the ‘lastapp’ variable is ‘Swift’, this would indicate that the cepstral wrapper is having a problem, correct? ** ** Justin Killen -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Justin Killen *Sent:* Tuesday, May 22, 2012 8:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] hangup not detected? ** ** Okay, the next time it gets in this state I’ll gather that information.*** * ** ** Justin Killen -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Sent:* Monday, May 21, 2012 1:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] hangup not detected? ** ** On Fri, May 18, 2012 at 12:00 PM, Justin Killen jkil...@allamericanasphalt.com wrote: I have and automated call-in dispatch system where hundreds of people call in daily for 2-3 minutes each. The extension is set up to get their information, then text-to-speech the dispatch information (via odbc). It then loops 5 times then ends the call. These calls are being handled by an 8 port analog digium card. Sometimes though, I see calls via ‘core show channel dahdi/1-1’ that have a time of 16 hours. I’m not sure if this is a result of dahdi missing the hangup, ODBC timing out, or TTS failing for some reason. When a channel gets in this state, the call doesn’t seem to progress through the dialplan, they always display the TTS line. Doing a ‘dahdi destroy channel 1-1’ doesn’t seem to be effective – the only way I’ve been able to clear the calls is to do a ‘dahdi restart’ and/or restart the asterisk service.* *** For TTS I’m using cepstral with the Swift wrapper. Here is a snippet of my
Re: [asterisk-users] Deleting OLD Voicemails
El mié, 23-05-2012 a las 11:42 +0200, Danny Dias escribió: Can i delete like this: rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.* You can make that without problems Is that ok? will this break something? Yes, that's ok regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe http://cipher.pe -- _ -- 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] Deleting OLD Voicemails
On 5/23/12 2:42 AM, Danny Dias wrote: Can i delete like this: rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.* Is that ok? will this break something? that's ok no it shouldn't break anything. however if you're going to delete some of the messages. you have to renumber all the messages so that they are consecutive otherwise the voicemail application may give you grief. A little doubt here, once the user hears the voicemail using the phone, the message is automatically moved to Old folder, is that right? yes -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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
