Re: [asterisk-users] Asterisk as useragent registered using 2 accounts
Rizwan Hisham wrote: > I am having a strange problem. I am using my asterisk server AST1 to > register with another asterisk server AST2 using 2 accounts (2 register > commands in sip.conf). I have made 2 local users in AST1, and configured my > dialplan in such a way that both local accounts on AST1 use different trunks > to send the call to AST2 server. These 2 different trunks are for 2 accounts > i have registered on AST1. > (skiped) > > How can i make asterisk realize it? > You must enable authentication of INVITE that AST1 send to AST2. Now you have no authentication of incoming INVITE and AST2 make decision about used account based only on IP address of caller peer. Changing insecure=port,invite to insecure=port should help. -- Best regards, Igor A. Goncharovsky ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6
Hi! Paul Hales wrote: > > I just installed Asterisk 1.6 beta5 and moh is not working - is there a > trick? Or is something wrong with my system? > This bug already fixed, you can check latest 1.6 branch or try to use 1.6 beta4. This version must not have this issue. -- Best regards, Igor A. Goncharovsky ___ ICQ: 648337 blog: http://igorg.ru mailto: [EMAIL PROTECTED] ___ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf Problem with DTMF_sequence
Tilghman Lesher wrote: > On Tuesday 22 April 2008 05:22, Sergey Shumeyko wrote: > >> I have following problem with my Asterisk installation (version 1.6.0. beta >> 7.1). I want to assign start record conversation to #7 and stop record >> conversation to #8, but it isn't working (on previous Asterisk 1.2.17 it >> was working fine). When I assign those functions to 7/8 (without #) >> correspondingly it also works fine, but it works only from caller side. I >> would appreciate very much if somebody can take a look at my configuration >> below and give me comments what I am doing wrong. >> > http://bugs.digium.com/view.php?id=12299 > No, I think this issue have other root. The issue I've reported have deal only with pickup number. Sergey, it is interesting for me, I'll try to look at you configuration till weekend. -- Best regards, Igor A. Goncharovsky ___ ICQ: 648337 blog: http://igorg.ru mailto: [EMAIL PROTECTED] ___ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk server and DSCP QOS
Hi! Steve Johnson wrote: > The network we're setting up has data on the default VLAN, Asterisk > server and phones on VLAN 4, and we're using Polycom phones with a PC > hooked up to the phone's pass-thru port. > If you are using VLAN, than you also look at new options in trunk cos_sip and cos_audio to set 802.1p. (If you run Linux). It will help with QoS too. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where does the call go in the dialplan after a call disconnects
Arpit Mehta wrote: > I would like to do some cleaning up after my call disconnects. For > this I need to know where in the dialplan a call goes after it > disconnects ? Is there any special place in the dialplan a call goes > to when it disconnects ? > After call disconnected, dialplan execution moves to h extension, if it exists. For example: [default] exten => 10,1,Dial(SIP/10) exten => h,1,NoOp(Call End. Cleanup.) -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura provisioning
Hi! Philip Prindeville wrote: > Yes, you can "tickle" an SPA94x or 962 and have it fetch a config from a > TFTP server... But is there no way to simply "push" a couple of lines > of XML config to it directly via an HTTP POST (sans TFTP server)? > If you have HTTP access to this phone, why not? Just look how it interface done. I think you can post value of any parameter, but I have not tried this. Also you can at any moment notify you phone, that there is new configuration. this can be done via resync URL or by SIP Notify. You can look how work Linksys firmware update utility ;) -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] resync linksys SPA9XX config file from Asterisk
Andres wrote: >> Anyone know the sip header to send to a Linksys to resync it's config file? >> > You will have to set the parameter Auth Resync-Reboot: to NO on the > phone so it will not ask for credentials. > Or you can use patch for asterisk that enable authorization of outgoing sip notify: http://bugs.digium.com/view.php?id=9896 This is more secure way to notify devices. There are more event in linksys devices for cold and war reboot. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
Hi! d tbsky wrote: > ok. i will add linksys to our testing list. but cisco tend to lock things. > can we get firmware for linksys easily ? or we must pay like cisco > routers and switches? > You can download latest firmware from linksys.com, also here is firmware release notes with full changes list. There is some support issues: support of VoIP devices only for itsp, but community can give answer on very-very advanced questions. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between TE121 and TE122
Hello! Guilherme Loch Waltrick Góes wrote: > What's the difference between the TE121 and TE122. I read the description on > Digium's site and it isn't clear to me. > Best regards, > The only one difference is interface: one of them have PCI and other have PCI-Express. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.17 and RXFAX via T38
Hello! Robert Moskowitz wrote: > I was pointed to the following: > > http://asteriskforum.ru/viewtopic.php?t=1761 > > It is in Russian, which I don't speak, but it references an Asterisk patch. > > Is this patch in 1.4.17? > Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?) > > Anyone work with this? > No new features wold be in 1.4 branch. But there is much work to bring full T.38 support in 1.6. You can use this patch for you 1.4.17, I think there is no problem for Cache to update this patch. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and 802.1Q
Hi! Coco Richard wrote: > How can i use different VLANs for signaling and audio, e.g vlan 100 for sip > and vlan 200 for rtp? Where can i find documentations for this? > > Comments and suggestions are welcomed (a sample config too :-))) > If you need to make RTP traffic priority higher then signaling or other data, you can use different values of "CoS" field. This setting available in 1.6 version for Linux. Also if you RTP going directly to host you may no need extra settings. -- Best regards, Igor A. Goncharovsky ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Perl AGI defunct process
Hi! Ruddy Gbaguidi wrote: > I'm using DeadAgi and has set AGISIGHUP to no because I don't want my > script to stop if the user hangs up. > But when it reach the end of the script, the child process should die. > And I don't see why I only have this trouble with perl agis. > Can you check if your script realy don't get SIGHUP? Some time ago I have problem with that setting AGISIGHUP to 'no' have no effect. -- Best regards, Igor A. Goncharovsky ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
