Re: [asterisk-users] asterisk.ctl file
Hi Shaun, thanks for the reply. This file is needed for debugging? -- _ -- 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] RFC List
Hi There, Where can I get a complete set of RFCs and other specifications supported by Asterisk? 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] RFC List
On 08/08/2012 06:30 AM, Kannan wrote: Where can I get a complete set of RFCs and other specifications supported by Asterisk? To my knowledge there is no such list. In addition, Asterisk (like many other pieces of software) does not claim 100% compliance with every RFC that is relevant, so usually it's better to ask about the specific features you are interested in. -- 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] Suggestion of Server Specifications for Asterisk
Let us know how does it performs... Leandro 2012/8/6 Shahid H shah...@gmail.com I have bought a new server today: i7-2600 CPU, 8GB and 2 x 256GB SSDs. 100Mbit Connection. I hope CPU is powerful enough for 200 concurrent calls. On Sun, Aug 5, 2012 at 1:57 AM, Michelle Dupuis mdup...@ocg.ca wrote: That's how we do it - write to a memory based (ramdisk) disk then write to HDD upon call completion. We haven't tried a SSD but that may be necessary depending on your call volumes. -- *From:* asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner [ rswago...@gmail.com] *Sent:* Saturday, August 04, 2012 7:34 PM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Suggestion of Server Specifications for Asterisk On Sat, Aug 4, 2012 at 1:22 PM, Shahid H shah...@gmail.com wrote: Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to do 200 calls recordings. Once the call hangup/completed it will then move recording file to SATA HDD. What do you think of this? You want some form of raid for redundancy. I usually go with two 15K SAS drives in raid 1 or four 7.2k SATA drives in raid 10. Performance between the two should be similar. With drives being as cheap as they are skip raid 5. Ryan -- _ -- 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] OT - Patton - FXO - Reduce incoming call delay
Hi, I'm benchmarking the performance of a Patton Smartnode 411X gateway. My setup is : GSM phone --PSTN-- SN411X --SIP-- Asterisk --SIP-- SIP phone My reference setup is: GSM phone --PSTN-- analog phone In the first case, it takes roughly 10s from the moment GSM user hits Send button to the moment Asterisk gets the incoming call (or the SIP phone rings). In the 2nd case with my reference setup, this delay drops to 4s. Can I (or should I try to) reduce this delay without loosing Caller ID ? In my country, Caller ID format is ETSI. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] qualifysmoothing
Greetings list, I have a scenario where half a dozen phones at a site appear to be dropping offline for a few seconds every few hours, but the connection between them and the asterisk server remains up. It's been suggested to me that the problem might be to do with qualify - which is enabled in this case. However, I don't really want to disable it if at all possible - it's a very good early warning indicator of network problems, and has often proved useful in diagnosing network faults - especially with end users' *DSL connections not provided by us. AIUI from the documentation, qualifysmoothing effectively averages the last two qualify results. Is there any way to increase this, so a device won't be considered unavailable until, for example, 3 consecutive qualify packets have been missed? Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] alwaysauthreject=yes not working as expected
Asterisk 1.4.42 Set alwaysauthreject=yes in [general] section of sip.conf. Restarted asterisk However when I attempt to register I still get: [2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from 'sip:000333082261...@domain.com' failed for '121.98.1.1' - Wrong password [2012-08-08 21:12:42] NOTICE[15689] chan_sip.c: Registration from 'sip:00033308226...@domain.com' failed for '121.98.1.1' - No matching peer found Based on the Asterisk security advisory (http://downloads.asterisk.org/pub/security/AST-2011-011.html) I would have expected 1.4.42 to respond the same in both cases (since the issue was fixed in 1.4.41.2). Am I missing something obvious? -- _ -- 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] Block outbound calls based on IP address
Thanks for the reply however it is not possible to get the public IP address using the SIP_HEADER function (see my original post). We have many devices connecting from hundreds of dynamic external IPs. -- _ -- 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] alwaysauthreject=yes not working as expected
Asterisk 1.4.42 Set alwaysauthreject=yes in [general] section of sip.conf. Restarted asterisk However when I attempt to register I still get: [2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from 'sip:000333082261...@domain.com' failed for '121.98.1.1' - Wrong password [2012-08-08 21:12:42] NOTICE[15689] chan_sip.c: Registration from 'sip:00033308226...@domain.com' failed for '121.98.1.1' - No matching peer found Based on the Asterisk security advisory (http://downloads.asterisk.org/pub/security/AST-2011-011.html) I would have expected 1.4.42 to respond the same in both cases (since the issue was fixed in 1.4.41.2). Am I missing something obvious? Yes. Those are log messages for the administrator's benefit. They are not SIP messages sent in response to the REGISTER request. The SIP messages sent are supposed to be the same not the logging messages. 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
[asterisk-users] tls is up but no audio
Hi All, I'm headbanging on this from a couple of days, begging here for some help :) I'm configuring tls on asterisk for the first time to experiment with an open (public) service idea about having asterisk accepting any sip user (with the sip.conf option 'autocreatepeer=yes') and call each other on the same server and perhaps to other asterisk servers with the same configuration. Something like 'skype for poors' for the 'average joe'. I'm using asterisk 10.7.0 on a debian squeeze dedicated server (with public ip). I've followed this tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial and got no errors but when dialing a test context: exten = _X.,1,Answer exten = _X.,n,playback(tt-weasels) exten = _X.,n,echo exten = _X.,n,Hangup() i get no audio. On the client side, I've tried with many softphones (bink, jitsi, microsip, phonerlite) on both windows and linux, on two different computers but same result. I've also enabled srtp, checked the sip debug trace, recompiled libsrtp from sources, tried different combination of parameters in sip.conf, enabled and disabled some port forwardings on the client's router but same result: all looks ok, but i get no audio. If not using tls (but the usual udp and rtp), audio works full-duplex :) Anyone had a similar problem ? Any hints ? Let me know if i can provide more info. Thanks for supporting, regards and have a nice day, Mike -- _ -- 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] IAX with two asterisk boxes
Hello, I have two asterisk boxes running and both are using DAHDI PRI Card. I wish to know if IAX is the best method to connect both the boxes? Also, need some help with the following? 1. For incoming call on server2 I wish to run an IVR to the user for which all my prompt sound files resides on server1. Is there a way I can achieve this? 2. I am also using .call file at times to make outgoing call to the user where IVR will be played but I will initiated the .call file from server1 spool but the call should use server2 dahdi lines and also stream the file from server1? Please suggest -- Regards, Ashish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
