Re: [asterisk-users] Deleting OLD Voicemails
You can delete old files, it won't break anything. Also to prevent saving files in multiple formats, edit voicemail.conf and change format parameter under general. -- Mehmet Avcioglu meh...@activecom.net On May 23, 2012, at 1:03 AM, Danny Dias wrote: Thanks Jason, But how to delete them? there are a lot of old voicemails, but i don't want to break the app_voicemail. 2012/5/22 Jason Parker jpar...@digium.com On 05/22/2012 04:54 PM, Danny Dias wrote: There are 4 files for each voicemail: msg.gsm msg.txt msg.wav msg.WAV That is perfectly normal. The .txt file is metadata that contains things like caller ID and duration. Asterisk will also save voicemails into every format you have specified in voicemail.conf. -- _ -- 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 -- www.danntel.net sip:danny4...@thesipschool.com sip:dann...@opensips.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
[asterisk-users] Transfer call issue
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
[asterisk-users] Unable to execute 'dahdi_scan /etc/asterisk/dahdi_scan.conf'
Hi Can anyone help me with this error Unable to execute 'dahdi_scan /etc/asterisk/dahdi_scan.conf' i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call reached the destination but no voice is coming from destination my voice reflects back thanks -- _ -- 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
Hi, thanks for your answers... Can i delete like this: rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.* Is that ok? will this break something? A little doubt here, once the user hears the voicemail using the phone, the message is automatically moved to Old folder, is that right? Many thanks! 2012/5/23 Mehmet Avcioglu meh...@activecom.net You can delete old files, it won't break anything. Also to prevent saving files in multiple formats, edit voicemail.conf and change format parameter under general. -- Mehmet Avcioglu meh...@activecom.net On May 23, 2012, at 1:03 AM, Danny Dias wrote: Thanks Jason, But how to delete them? there are a lot of old voicemails, but i don't want to break the app_voicemail. 2012/5/22 Jason Parker jpar...@digium.com On 05/22/2012 04:54 PM, Danny Dias wrote: There are 4 files for each voicemail: msg.gsm msg.txt msg.wav msg.WAV That is perfectly normal. The .txt file is metadata that contains things like caller ID and duration. Asterisk will also save voicemails into every format you have specified in voicemail.conf. -- _ -- 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 -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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 -- www.danntel.net *sip:danny4...@thesipschool.com* sip:dann...@opensips.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] Deleting OLD Voicemails
Please check out the scripts located in contrib/scripts Regards Hans On 2012-05-23 11:42, Danny Dias wrote: Hi, thanks for your answers... Can i delete like this: rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.* Is that ok? will this break something? A little doubt here, once the user hears the voicemail using the phone, the message is automatically moved to Old folder, is that right? Many thanks! 2012/5/23 Mehmet Avcioglu meh...@activecom.net mailto:meh...@activecom.net You can delete old files, it won't break anything. Also to prevent saving files in multiple formats, edit voicemail.conf and change format parameter under general. -- Mehmet Avcioglu meh...@activecom.net mailto:meh...@activecom.net On May 23, 2012, at 1:03 AM, Danny Dias wrote: Thanks Jason, But how to delete them? there are a lot of old voicemails, but i don't want to break the app_voicemail. 2012/5/22 Jason Parker jpar...@digium.com mailto:jpar...@digium.com On 05/22/2012 04:54 PM, Danny Dias wrote: There are 4 files for each voicemail: msg.gsm msg.txt msg.wav msg.WAV That is perfectly normal. The .txt file is metadata that contains things like caller ID and duration. Asterisk will also save voicemails into every format you have specified in voicemail.conf. -- _ -- 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 and the media path
Hi Jared Kevin, Thanks for taking the time to answer my questions. I wonder if I could just be reading the tcpdump incorrectly? I'm still seeing rtp streams (and Jared, I have modified the dial string to remove the L).. Here's a screenshot of what I'm seeing in wireshark. I really appreciate the suggestions. Screenshot: http://dl.dropbox.com/u/4156401/Screenshot%20from%202012-05-23%2007%3A39%3A51.png pcap: http://dl.dropbox.com/u/4156401/trace3000.pcap Thanks David On Wed, May 23, 2012 at 7:41 AM, David Wessell da...@ringfree.biz wrote: Hi Jared Kevin, Thanks for taking the time to answer my questions. I wonder if I could just be reading the tcpdump incorrectly? I'm still seeing rtp streams (and Jared, I have modified the dial string to remove the L).. Here's a screenshot of what I'm seeing in wireshark. I really appreciate the suggestions. Thanks David On Mon, May 21, 2012 at 6:08 PM, Jared Geiger ja...@compuwizz.net wrote: A2billing usually stays in the media path due to the dialstring parameters that it uses to cut a call off when the balance would reach $0. To get Asterisk to step out of the media path, I had to change dialcommand_param_sipiax_friend and dialcommand_param to |60|S(14400) which lets all calls go to 14400 seconds. The default uses the L parameter. You need to use the S parameter instead. However the S parameter doesn't like very large numbers in Asterisk 1.4 so I've just hard set mine. ~Jared On Mon, May 21, 2012 at 5:18 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/21/2012 03:45 PM, David Wessell wrote: More specific on sip.conf In sip.conf I have a trunk specified for the SIP provider, and a trunk specified for the PBX itself. Do I need to specify directmedia=yes on both sides? Yes, it has to be set on both peers involved in the bridged call. -- 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 -- -- www.ringfree.biz 828-575-0030 -- -- www.ringfree.biz 828-575-0030 -- _ -- 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 stop ringing when incoming PSTN call is answered externally?
On Tue, May 22, 2012 at 11:32:19PM -0400, ft...@mindspring.com wrote: The calls are routed just fine, but when a call is answered at one of the extensions or externally (by a home telephone) the asterisk extensions continue to ring one more time. Is there a way to have Asterisk drop an incoming PSTN call as soon as it's answered? I have the same problem, and earlier discussion here suggests it's insoluble. -- _ -- 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] twenty thousands (20, 000) users, which asterisk and how many servers?
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? Regards Bilal -- _ -- 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?
20.000 users is really a big number, as big as 2000 concurrent calls. As previously stated on this list, it depends... it depends by the type of calls for example. If all media is offloaded from the server letting the phones to reinvite each other, than your server CAN support the call volume. If instead even a tiny portion of the call volume uses service on the pbx, like IVR, music on hold, conferences, queues or even worst, transcoding, then the server is obviously underpowered. From my point of view, servicing 20.000 users with a single piece of hardware is highly risky. It can broke in the middle of the day, leaving all your users without service. I think a better approach will be to have more less powered servers working all together to serving your users. If a day one or two of them broke, you have not to worry because the other will continue to serve your users and nobody notice the little decrease in power. There are a lots of way to achieve the high availability, load sharing, each with its pros and cons. Right now I am building a pbx with high availability and load sharing in mind, for a client who wants to achieve numbers you have just said. Let's see how it works in few months. Leandro 2012/5/23 bilal ghayyad bilmar...@yahoo.com 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? Regards Bilal -- _ -- 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 Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- 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
Alejandro's setup looks correct; you can also get the correct config using Vitelity's wizard tool for setting up the trunks. The only thing I would add is that if your account is setup with a session border controller you will need to use the SBC's IP address instead of the IP the wizard gives you. If you have an SBC, the fact will be noted in your account including the IP address. I've found Vitelity's tech support to be pretty helpful too, should you need to contact them. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Wed, May 23, 2012 at 9:14 AM, Alejandro Imass a...@p2ee.org wrote: On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- 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
Word of warning - I have had a lot of issues with Vitelity's routing. Lots of troubles to the Caribbean, lots of troubles with ordinary US 800 numbers (major corporations like Nicor, American Airlines). Cheers, Jeff LaCoursiere SunFone On Wed, 2012-05-23 at 09:22 -0500, Stephen J Alexander wrote: Alejandro's setup looks correct; you can also get the correct config using Vitelity's wizard tool for setting up the trunks. The only thing I would add is that if your account is setup with a session border controller you will need to use the SBC's IP address instead of the IP the wizard gives you. If you have an SBC, the fact will be noted in your account including the IP address. I've found Vitelity's tech support to be pretty helpful too, should you need to contact them. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Wed, May 23, 2012 at 9:14 AM, Alejandro Imass a...@p2ee.org wrote: On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- 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 -- _ -- 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] Planned service outage for community services
On May 31, 2012 from approximately 9:00AM to 12:00PM (Central Daylight Time, GMT-5), the servers that Digium uses to provide many services to the Asterisk community will be relocated. This will mean that these services will be unavailable during most, if not all, of this time window. Once the move is complete, the services will be available again, with no user-visible changes. The services affected include: bamboo.asterisk.org code.asterisk.org downloads.digium.com downloads.asterisk.org git.asterisk.org issues.asterisk.org packages.asterisk.org reviewboard.asterisk.org svn.asterisk.org svnview.digium.com wiki.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] Vitelity Setup
On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote: Word of warning - I have had a lot of issues with Vitelity's routing. Lots of troubles to the Caribbean, lots of troubles with ordinary US 800 numbers (major corporations like Nicor, American Airlines). We had lot's of trouble with the 800 numbers as well but after help from Vitelity's support we were able to determine that the problem was that toll free require _exactly_ 10 digits to accept the toll free call. Regarding call to the Caribean we had a lot trouble with cell phones in Venezuela and it seems they were using pre-paid lines that ran out money but they eventually got around and solved it. So I think that if you insist with their support they usually resolve the issue. Best, -- Alejandro Imass Cheers, Jeff LaCoursiere SunFone On Wed, 2012-05-23 at 09:22 -0500, Stephen J Alexander wrote: Alejandro's setup looks correct; you can also get the correct config using Vitelity's wizard tool for setting up the trunks. The only thing I would add is that if your account is setup with a session border controller you will need to use the SBC's IP address instead of the IP the wizard gives you. If you have an SBC, the fact will be noted in your account including the IP address. I've found Vitelity's tech support to be pretty helpful too, should you need to contact them. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Wed, May 23, 2012 at 9:14 AM, Alejandro Imass a...@p2ee.org wrote: On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am unable to register vitelity SIP trunk, where its keep on sending registration request, and I am using Asterisk 1.4.39.2, my registration procedure as follows, sip.conf register = username:sec...@sip41.vitelity.net:5060 We use viteity w/o registration like so: [vitel-inbound] type=friend dtmfmode=auto host=inbound24.vitelity.net context=vitelity-inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=vitelity-outbound allow=all insecure=very -- _ -- 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 -- _ -- 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] How to stop ringing when incoming PSTN call is answered externally?
Roger Burton West wrote: On Tue, May 22, 2012 at 11:32:19PM -0400, ft...@mindspring.com wrote: The calls are routed just fine, but when a call is answered at one of the extensions or externally (by a home telephone) the asterisk extensions continue to ring one more time. Is there a way to have Asterisk drop an incoming PSTN call as soon as it's answered? I have the same problem, and earlier discussion here suggests it's insoluble. Nonsense It is easily solved. Simply answer through a supervised extension ( asterisk ) Don't expect analog lines to work properly with Asterisk and a POTS phone bridged, with the call answered on a POTS phone Technically that is a poor method of connection. Simply have the PSTN line go into Asterisk, and ALL analog phones in the home or business become extensions off of Asterisk There ARE other ways, but it requires some coding and testing to get things to work, with no real gain I am doing this to capture and screen caller ID numbers, but it is not the best solution John Novack -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] No caller id when using cadence with DAHDI
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? Cheers, Panos -- _ -- 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] Disable All Asterisk Features (blind xfer, disconnect, etc)
Hi Guys, is there any way to disable all Asterisk Features? We are having false dtmf detections and randon calls being put on-hold and suspect that dtmf features is the cause. Changing features.conf aparently keeps the default options. Since we dont use it, is there any way to disable it? Thanks, Eduardo -- _ -- 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 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: Word of warning - I have had a lot of issues with Vitelity's routing. Lots of troubles to the Caribbean, lots of troubles with ordinary US 800 numbers (major corporations like Nicor, American Airlines). We had lot's of trouble with the 800 numbers as well but after help from Vitelity's support we were able to determine that the problem was that toll free require _exactly_ 10 digits to accept the toll free call. That's not the trouble we were having - if you call these large companies and sit on hold waiting for an agent, then finally get transferred, you get a message Cannot complete this call from your location. Oddly, it is the same message no matter what large company you are calling. It was reproducible every time, not random. Through other carriers we work with the same numbers were no problem. I finally concluded that these large companies are probably offloading their support overseas, and that they were doing some kind of PRI transfer, offloading the new leg upstream in some manner that eventually resulted in the call being rejected. Why this seems to happen exclusively through Vitelity I can't say. Support emails went unanswered. With so many other termination providers it was easier to simply switch our 800 carrier than chase this down with a support infrastructure that won't answer emails. Regarding call to the Caribean we had a lot trouble with cell phones in Venezuela and it seems they were using pre-paid lines that ran out money but they eventually got around and solved it. So I think that if you insist with their support they usually resolve the issue. Our troubles were first with ALL calls to Jamaica. It took all day to get someone to look into it, and two days later we still couldn't complete calls there. Again, switched to a different carrier and the problem went away. Next was Trinidad. Same story. Haven't gone back to see if they were eventually resolved. I don't know who they are trying to use in the Caribbean to save cost on their routes, but I would rather work with someone that is using white routes and pay a bit more than spend all my time resolving call routing for my customers. 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. Cheers, 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
Re: [asterisk-users] No caller id when using cadence with DAHDI
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
Re: [asterisk-users] Vitelity Setup
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? Thanks, -- Alejandro Imass Cheers, 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
Re: [asterisk-users] Vitelity Setup
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
Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers
Dear; So it is a hardware issue and not software? I am afraid that asterisk software it self is not able to support 20 000 users and 2000 concurrent calls. About the high availability: is there a method that if the first asterisk server down, then the call will stay connected and failover to second asterisk server? Regards Bilal -- 20.000 users is really a big number, as big as 2000 concurrent calls. As previously stated on this list, it depends... it depends by the type of calls for example. If all media is offloaded from the server letting the phones to reinvite each other, than your server CAN support the call volume. If instead even a tiny portion of the call volume uses service on the pbx, like IVR, music on hold, conferences, queues or even worst, transcoding, then the server is obviously underpowered. From my point of view, servicing 20.000 users with a single piece of hardware is highly risky. It can broke in the middle of the day, leaving all your users without service. I think a better approach will be to have more less powered servers working all together to serving your users. If a day one or two of them broke, you have not to worry because the other will continue to serve your users and nobody notice the little decrease in power. There are a lots of way to achieve the high availability, load sharing, each with its pros and cons. Right now I am building a pbx with high availability and load sharing in mind, for a client who wants to achieve numbers you have just said. Let's see how it works in few months. Leandro 2012/5/23 bilal ghayyad bilmar...@yahoo.com 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? Regards Bilal -- _ -- 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
you can find more details @AsteriskSCF project. On Wed, May 23, 2012 at 11:16 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; So it is a hardware issue and not software? I am afraid that asterisk software it self is not able to support 20 000 users and 2000 concurrent calls. About the high availability: is there a method that if the first asterisk server down, then the call will stay connected and failover to second asterisk server? Regards Bilal -- 20.000 users is really a big number, as big as 2000 concurrent calls. As previously stated on this list, it depends... it depends by the type of calls for example. If all media is offloaded from the server letting the phones to reinvite each other, than your server CAN support the call volume. If instead even a tiny portion of the call volume uses service on the pbx, like IVR, music on hold, conferences, queues or even worst, transcoding, then the server is obviously underpowered. From my point of view, servicing 20.000 users with a single piece of hardware is highly risky. It can broke in the middle of the day, leaving all your users without service. I think a better approach will be to have more less powered servers working all together to serving your users. If a day one or two of them broke, you have not to worry because the other will continue to serve your users and nobody notice the little decrease in power. There are a lots of way to achieve the high availability, load sharing, each with its pros and cons. Right now I am building a pbx with high availability and load sharing in mind, for a client who wants to achieve numbers you have just said. Let's see how it works in few months. Leandro 2012/5/23 bilal ghayyad bilmar...@yahoo.com 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? Regards Bilal -- _ -- 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, Ali R. Taleghani -- _ -- 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] SIP endpoints CANCEL when PRI receives Cause Code 31
We have an Asterisk server which connects to another Asterisk server acting as a PSTN gateway. This gateway machine has Digium TE210P card connected to a pair of PRIs. For the most part, all is working well, however there are some specific telephone numbers that my users have attempted to call, but we unable to. I set debugging on and determined that when the the gateway machine dials one of the numbers in question, we receive from the PSTN an ISDN cause code 31, which in my understanding is not an error. This is then passed back to the originating Asterisk server via IAX as progress. It is then sent to the originating endpoint as a sip message 183 'Session Progress'. 2 seconds after this 183 progress message is sent, the endpoint sends a SIP CANCEL message and the channel is torn down. I have the prematuremedia=yes and progressinband=never in the sip.conf file which looks like it could be a solution, however I believe that because we are getting ISDN Call Proceeding and a corresponding SIP 100 Trying message that this setting has no effect. I have tried from several different endpoint types with the same results. I have verified that the numbers in question are in fact operational. Any suggestions? Asterisk version is 1.8.7 on both hosts Dahdi version 2.5.0 libpri version 1.4.12 Thanks, Dale -- _ -- 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] SIP endpoints CANCEL when PRI receives Cause Code 31
We have an Asterisk server which connects to another Asterisk server acting as a PSTN gateway. This gateway machine has Digium TE210P card connected to a pair of PRIs. For the most part, all is working well, however there are some specific telephone numbers that my users have attempted to call, but we unable to. I set debugging on and determined that when the the gateway machine dials one of the numbers in question, we receive from the PSTN an ISDN cause code 31, which in my understanding is not an error. This is then passed back to the originating Asterisk server via IAX as progress. It is then sent to the originating endpoint as a sip message 183 'Session Progress'. 2 seconds after this 183 progress message is sent, the endpoint sends a SIP CANCEL message and the channel is torn down. I have the prematuremedia=yes and progressinband=never in the sip.conf file which looks like it could be a solution, however I believe that because we are getting ISDN Call Proceeding and a corresponding SIP 100 Trying message that this setting has no effect. I have tried from several different endpoint types with the same results. I have verified that the numbers in question are in fact operational. Any suggestions? Asterisk version is 1.8.7 on both hosts Dahdi version 2.5.0 libpri version 1.4.12 You should upgrade Asterisk to at least v1.8.8. A regression in the Asterisk v1.8.7 ./configure script does not setup Asterisk to use libpri correctly. Most supplementary service features and a hangup fix supported by that version of libpri do not get enabled. Richard -- _ -- 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
the solution lies in kamailio/opensips's despatcher module. Sent from my iPhone On 23 maj 2012, at 20:46, bilal ghayyad bilmar...@yahoo.com wrote: Dear; So it is a hardware issue and not software? I am afraid that asterisk software it self is not able to support 20 000 users and 2000 concurrent calls. About the high availability: is there a method that if the first asterisk server down, then the call will stay connected and failover to second asterisk server? Regards Bilal -- 20.000 users is really a big number, as big as 2000 concurrent calls. As previously stated on this list, it depends... it depends by the type of calls for example. If all media is offloaded from the server letting the phones to reinvite each other, than your server CAN support the call volume. If instead even a tiny portion of the call volume uses service on the pbx, like IVR, music on hold, conferences, queues or even worst, transcoding, then the server is obviously underpowered. From my point of view, servicing 20.000 users with a single piece of hardware is highly risky. It can broke in the middle of the day, leaving all your users without service. I think a better approach will be to have more less powered servers working all together to serving your users. If a day one or two of them broke, you have not to worry because the other will continue to serve your users and nobody notice the little decrease in power. There are a lots of way to achieve the high availability, load sharing, each with its pros and cons. Right now I am building a pbx with high availability and load sharing in mind, for a client who wants to achieve numbers you have just said. Let's see how it works in few months. Leandro 2012/5/23 bilal ghayyad bilmar...@yahoo.com 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? Regards Bilal -- _ -- 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] twenty thousands (20, 000) users, which asterisk and how many servers?
On 05/23/2012 07:16 AM, bilal ghayyad wrote: 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] Unable to execute 'dahdi_scan /etc/asterisk/dahdi_scan.conf'
On Wed, May 23, 2012 at 02:02:51PM +0500, p070075 Muhammad Atif Ramzan wrote: Hi Can anyone help me with this error Unable to execute 'dahdi_scan /etc/asterisk/dahdi_scan.conf' i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call reached the destination but no voice is coming from destination my voice reflects back Have you verified the user asterisk is running as can execute dahdi_scan? This was asked not too long ago on the forums as well: http://forums.asterisk.org/viewtopic.php?f=1t=82659 -- 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
Re: [asterisk-users] SIP endpoints CANCEL when PRI receives Cause Code 31
On 05/23/2012 02:59 PM, Richard Mudgett wrote: We have an Asterisk server which connects to another Asterisk server acting as a PSTN gateway. This gateway machine has Digium TE210P card connected to a pair of PRIs. For the most part, all is working well, however there are some specific telephone numbers that my users have attempted to call, but we unable to. I set debugging on and determined that when the the gateway machine dials one of the numbers in question, we receive from the PSTN an ISDN cause code 31, which in my understanding is not an error. This is then passed back to the originating Asterisk server via IAX as progress. It is then sent to the originating endpoint as a sip message 183 'Session Progress'. 2 seconds after this 183 progress message is sent, the endpoint sends a SIP CANCEL message and the channel is torn down. I have the prematuremedia=yes and progressinband=never in the sip.conf file which looks like it could be a solution, however I believe that because we are getting ISDN Call Proceeding and a corresponding SIP 100 Trying message that this setting has no effect. I have tried from several different endpoint types with the same results. I have verified that the numbers in question are in fact operational. Any suggestions? Asterisk version is 1.8.7 on both hosts Dahdi version 2.5.0 libpri version 1.4.12 You should upgrade Asterisk to at least v1.8.8. A regression in the Asterisk v1.8.7 ./configure script does not setup Asterisk to use libpri correctly. Most supplementary service features and a hangup fix supported by that version of libpri do not get enabled. Richard Richard, Thanks. That appears to have done the trick. Very much appreciated. Dale -- The truth speaks for itself. I'm just the messenger. Lyta Alexander - Babylon 5 -- _ -- 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] Detecting Fax Tones over IAX2
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? Thanks, Cody -- _ -- 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
