<html><head><meta content="text/html; charset=us-ascii" http-equiv="Content-Type"></head><body><div><div style="font-family: Calibri,sans-serif; font-size: 11pt;">Unsubscribe<br><br>Elvin G. Nodalo<br></div></div><hr><span style="font-family: Tahoma,sans-serif; font-size: 10pt; font-weight: bold;">From: </span><span style="font-family: Tahoma,sans-serif; font-size: 10pt;">asterisk-users-request@lists.digium.com</span><br><span style="font-family: Tahoma,sans-serif; font-size: 10pt; font-weight: bold;">Sent: </span><span style="font-family: Tahoma,sans-serif; font-size: 10pt;">7/10/2013 1:00 AM</span><br><span style="font-family: Tahoma,sans-serif; font-size: 10pt; font-weight: bold;">To: </span><span style="font-family: Tahoma,sans-serif; font-size: 10pt;">asterisk-users@lists.digium.com</span><br><span style="font-family: Tahoma,sans-serif; font-size: 10pt; font-weight: bold;">Subject: </span><span style="font-family: Tahoma,sans-serif; font-size: 10pt;">asterisk-users Digest, Vol 108, Issue 14</span><br><br>Send asterisk-users mailing list submissions to<br> asterisk-users@lists.digium.com<br><br>To subscribe or unsubscribe via the World Wide Web, visit<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br>or, via email, send a message with subject or body 'help' to<br> asterisk-users-request@lists.digium.com<br><br>You can reach the person managing the list at<br> asterisk-users-owner@lists.digium.com<br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of asterisk-users digest..."<br><br><br>Today's Topics:<br><br> 1. analog phone digit delay (Justin Killen)<br> 2. Re: analog phone digit delay (jg)<br> 3. Re: analog phone digit delay (Justin Killen)<br> 4. Re: analog phone digit delay (jg)<br> 5. Re: analog phone digit delay (Steve Edwards)<br> 6. Re: PCI Passthrough of T1 cards (Mauricio Tavares)<br> 7. Re: PCI Passthrough of T1 cards (Nick Khamis)<br> 8. Fwd: AQuA Meter ? waveform analysis to get continous MOS<br> scores for your network (Sevana Oy)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Mon, 8 Jul 2013 10:14:31 -0700<br>From: Justin Killen <jkillen@allamericanasphalt.com><br>Subject: [asterisk-users] analog phone digit delay<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <asterisk-users@lists.digium.com><br>Message-ID:<br> <55B5D66C43B57F44BC89CB4650FD32F80118FFC2B99F@MAL.sg1.allamericanasphalt.com><br> <br>Content-Type: text/plain; charset="us-ascii"<br><br>I have an installation that has analog phones connected via T1 channel banks. I'm getting complaints from users that they will enter a partial number (eg 91213), then turn away to get the next few digits, and the system will start dialing before they have a chance to put in the rest of the dialing string. Is there a way to increase this delay? The only use these 4 dialing patterns:<br><br>Internal 3 digit numbers<br>91 XXX XXX XXXX (for backwards compatibility)<br>9 XXX XXXX (also for compatibility)<br>XXX XXXX<br><br><br>I'm using the freepbx distro if that helps. Asterisk 11.2.<br><br>Thanks,<br><br>-Justin<br><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130708/a09901d8/attachment-0001.htm><br><br>------------------------------<br><br>Message: 2<br>Date: Mon, 08 Jul 2013 19:21:10 +0200<br>From: jg <webaccounts@jgoettgens.de><br>Subject: Re: [asterisk-users] analog phone digit delay<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <asterisk-users@lists.digium.com><br>Message-ID: <51DAF506.5070900@jgoettgens.de><br>Content-Type: text/plain; charset=UTF-8; format=flowed<br><br>Have a look at the documentation of the channel bank. I guess some kind of overlap dialing is <br>enabled, which is typically associated with a timeout value. chan_dahdi.conf also has entries <br>like this.<br><br><br><br>------------------------------<br><br>Message: 3<br>Date: Mon, 8 Jul 2013 10:45:52 -0700<br>From: Justin Killen <jkillen@allamericanasphalt.com><br>Subject: Re: [asterisk-users] analog phone digit delay<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <asterisk-users@lists.digium.com><br>Message-ID:<br> <55B5D66C43B57F44BC89CB4650FD32F80118FFC2B9BB@MAL.sg1.allamericanasphalt.com><br> <br>Content-Type: text/plain; charset="us-ascii"<br><br>The channel banks are Adtran TA-624's using ESF/B8ZS. When a handset is picked up, I can see the offhook in the asterisk console, so it looks that the channel is immediately connected through the channel bank (not delayed until after digits are dialed), so it looks that overlap dialing isn't a factor and that asterisk has complete control.<br><br>As for options in chan_dahdi.conf, I simply can't find any that relate to this problem. I have looked at the page here: http://www.voip-info.org/wiki/view/chan_dahdi.conf and the closest thing I can find is 'ringtimeout' which is obviously not what I want. I would expect to see something like 'dialtimeout' or 'interdigittimeout'.<br><br>-Justin<br><br>-----Original Message-----<br>From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of jg<br>Sent: Monday, July 08, 2013 10:21 AM<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>Subject: Re: [asterisk-users] analog phone digit delay<br><br>Have a look at the documentation of the channel bank. I guess some kind of overlap dialing is <br>enabled, which is typically associated with a timeout value. chan_dahdi.conf also has entries <br>like this.<br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> http://www.asterisk.org/hello<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br><br><br><br>------------------------------<br><br>Message: 4<br>Date: Mon, 08 Jul 2013 20:38:20 +0200<br>From: jg <webaccounts@jgoettgens.de><br>Subject: Re: [asterisk-users] analog phone digit delay<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <asterisk-users@lists.digium.com><br>Message-ID: <51DB071C.2030405@jgoettgens.de><br>Content-Type: text/plain; charset=UTF-8; format=flowed<br><br><br>> The channel banks are Adtran TA-624's using ESF/B8ZS. When a handset is picked up, I can see the offhook in the asterisk console, so it looks that the channel is immediately connected through the channel bank (not delayed until after digits are dialed), so it looks that overlap dialing isn't a factor and that asterisk has complete control.<br>This also means that you should see the digits as they are dialed. When something times out you <br>should also see a message why there was a timeout.<br>I am using ISDN for PSTN connections and where I live there must be some kind of overlap dialing <br>enabled, otherwise P2P configurations don't work. With current DAHDI drivers I no longer need <br>special settings to make things work (Wanpipe/Woomera was different), so I guess overlap dialing <br>is enabled. Some SIP phones distinguish between "Overlap Dialing" and "Automatic Dialing", so <br>your channel bank might also have something like an Automatic Dialing option with some timing <br>value.<br>> As for options in chan_dahdi.conf, I simply can't find any that relate to this problem. I have looked at the page here: http://www.voip-info.org/wiki/view/chan_dahdi.conf and the closest thing I can find is 'ringtimeout' which is obviously not what I want. I would expect to see something like 'dialtimeout' or 'interdigittimeout'.<br>There is an "overlap" option in configs/chan_dahdi.conf.sample.<br><br>I am currently assembling an Asterisk box that has 48+2 analog channels (+ SIP + ISDN). If your <br>problem doesn't go away I could tell next week what my system is doing.<br><br>jg<br><br><br><br><br>------------------------------<br><br>Message: 5<br>Date: Mon, 8 Jul 2013 11:55:21 -0700 (PDT)<br>From: Steve Edwards <asterisk.org@sedwards.com><br>Subject: Re: [asterisk-users] analog phone digit delay<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <asterisk-users@lists.digium.com><br>Message-ID: <alpine.DEB.2.02.1307081154360.13329@ws><br>Content-Type: text/plain; charset="iso-8859-7"; Format="flowed"<br><br>On Mon, 8 Jul 2013, Justin Killen wrote:<br><br>> I have an installation that has analog phones connected via T1 channel <br>> banks. ?I?m getting complaints from users that they will enter a partial <br>> number (eg 91213), then turn away to get the next few digits, and the <br>> system will start dialing before they have a chance to put in the rest <br>> of the dialing string. ?Is there a way to increase this delay?? The only <br>> use these 4 dialing patterns:<br><br>Will 'show function TIMEOUT' help?<br><br>-- <br>Thanks in advance,<br>-------------------------------------------------------------------------<br>Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST<br>Newline Fax: +1-760-731-3000<br><br>------------------------------<br><br>Message: 6<br>Date: Mon, 8 Jul 2013 17:07:53 -0400<br>From: Mauricio Tavares <raubvogel@gmail.com><br>Subject: Re: [asterisk-users] PCI Passthrough of T1 cards<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <asterisk-users@lists.digium.com><br>Message-ID:<br> <CAHEKYV60YsVa76GJ+TX2ToVA1w=AV2gi+=F4GHdz8cAnmd25XA@mail.gmail.com><br>Content-Type: text/plain; charset=ISO-8859-1<br><br>On Wed, Jun 19, 2013 at 11:52 AM, Nick Khamis <symack@gmail.com> wrote:<br>> Hello James,<br>><br>> Thank you so much for your response. I should have chose my words<br>> carefully. PCI pass-through in terms of virtualization of devices and<br>> it's draw back are well know. I was leaning more towards near host<br>> performance virtualization using SR-IOV.<br>><br> I know I am late in the show, but what are the drawbacks as far<br>as using Asterisk is concerned?<br><br>> This moves emphasis back to the production drivers of the interface<br>> card using virtual functions etc., and can provide near host<br>> performance. Rephrasing my question, are any of the T1 pci<br>> manufactures providing support for virtualization using SR-IOV and<br>> virutal functions?<br>><br>> Kind Regards,<br>><br>> Nick<br>><br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br><br><br><br>------------------------------<br><br>Message: 7<br>Date: Mon, 8 Jul 2013 19:11:36 -0400<br>From: Nick Khamis <symack@gmail.com><br>Subject: Re: [asterisk-users] PCI Passthrough of T1 cards<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <asterisk-users@lists.digium.com><br>Message-ID:<br> <CAGWRaZY0Q_jAGjLPGzqF6X=utMcTKRrwBBH5ZRLFyJo=UZ_sCw@mail.gmail.com><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Asterisk does fine in a virtual instance. The key is finding hardware that<br>would<br>support more than just virtualization (i.e., SR-IOV).... Not sure if such a<br>card<br>exist.<br><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130708/4b8e1919/attachment-0001.htm><br><br>------------------------------<br><br>Message: 8<br>Date: Tue, 9 Jul 2013 19:34:34 +0400<br>From: Sevana Oy <sales@sevana.fi><br>Subject: [asterisk-users] Fwd: AQuA Meter ? waveform analysis to get<br> continous MOS scores for your network<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <asterisk-users@lists.digium.com><br>Message-ID:<br> <CAMyj0v=ez4dkj9hPi205QQ9ySQs-wbV_pHq8b4uJRVeDhJfbMA@mail.gmail.com><br>Content-Type: text/plain; charset="windows-1252"<br><br>Hi,<br><br>Although this is a repost from Asterisk biz, we would like to ask if<br>somebody may help us to develop a native Asterisk module using AQuA<br>technology for voice quality monitoring using the same web service AQuA<br>Meter is using.<br><br>Thanks,<br>Sevana Finland/Estonia<br><br>---------- Forwarded message ----------<br>From: Sevana Oy <sales@sevana.fi><br>Date: Mon, Jun 17, 2013 at 7:30 PM<br>Subject: AQuA Meter ? waveform analysis to get continous MOS scores for<br>your network<br>To: asterisk-biz@lists.digium.com<br><br><br>[image: AQuA Meter]<http://blog.sevana.fi/wp-content/uploads/2013/03/screenshot.png><br><br>Hi,<br><br>We would like to offer you to learn about our new application that performs<br>scheduled voice test calls to a predefined<br>echo server and then uses our AQuA web service to evaluate the call quality.<br><br>We developed it because several VoIP service providers have inquired us for<br>a possibility to make test calls from local machines within<br>their customers? network.<br><br>A typical example is when you provide VoIP communications to a company that<br>rents its premises (including an Internet connection) in a<br>business center. In this case it is quite important to monitor voice call<br>quality from different computers in the office space to the<br>service provider?s server.<br><br>This is a cross platform (Windows, Linux, MAC) Java application and uses<br>our latest developments in waveform analysis to evaluate voice call<br>quality: http://www.sevana.fi/aquameter.zip<br><br>The setup is simple: our application calls the echo server (apparently<br>provided by the VoIP service provider), plays a reference audio and records<br>the playback from the echo server and can thus provide overall (both ways)<br>call quality analysis.<br><br>We are very interested to receive your feedback and feature wishlist. The<br>application is free.<br><br>Best Regards,<br><br>Sevana Oy/O?<br>Finland/Estonia<br><br>http://blog.sevana.fi/aqua-meter-waveform-analysis-to-get-continous-mos-scores-for-your-network/<br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130709/18261bfe/attachment-0001.htm><br><br>------------------------------<br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by http://www.api-digital.com--<br><br>AstriCon 2010 - October 26-28 Washington, DC<br>Register Now: http://www.astricon.net/<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br><br>End of asterisk-users Digest, Vol 108, Issue 14<br>***********************************************<br></body></html>