similar to: sip.conf "limitonpeers=yes" in asterisk 1.4

Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the

Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This

Earlier I posted the message below to R-announce. Please, let me appologize for this mistake, it was not intended. The solution was provided by Duncan Murdoch who passed on a message he recieved from Brian Ripley which informed me that settimg R_Interactive to 1 changes the default behavor of aborting on an error. Thankyou, Don Wingate. -----Original Message----- From: Don Wingate

2008/11/4 Igor Zamocky <asterix at ponozky.sk> > > Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers > ? > I didn't. Now I did and it's working the way I wanted. Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and SIPPEER but limitonpeers is much more concise. Thanks a lot. > > > Hi, > > > In this

Hi, I'm currently using Asterisk 1.4.23.1, and I have a problem (also on previous version). Sometimes, when I try to do an attended transfer to another internal with default feature *2, Asterisk doesn't make it (it doesn't play 'pbx-transfer'). Sometimes on second time, Asterisk make transfer correctly. I have this problem on variuos type of SIP phones (GrandStream, Aastra,

Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch

Hi guys i'm using asterisk 11.18.0. I need to send the pound # sign to my SIP provider, but each time it's reencoded in %23. I try to put pedantic=yes in the sip.conf as advised here: http://www.voip-info.org/wiki/view/Asterisk+SIP+pedantic but nothing's changed. Have someone already met this issue please ? thanks a lot, regards, Alan

I am trying to get the presence/hints/BLF working along with Realtime SIP but I never get any "busy" notification. core show hints always shows the realtime sip user as idle. I have tried setting call-limit to various values, including 1 but nothing seems to help. I have tried limitonpeers both yes and no. Anybody got any other ideas? I do know the hinting is working as I can

Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI> show hints -= Registered Asterisk Dial Plan Hints =- 30@default : State:Unavailable Watchers 3 29@default :

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

Hi to all. I'm using Asterisk 1.4 with Sjphone as softphone. My problem is that when a SIP user is busy, he still receive calls from asterisk. I've tried to setup the call-limit preference to 1, but with this kind of configuration the user can't transfer calls, as the system block the 2nd call generated to do the transfer. I've also tried to set the user as friend, limitonpeers

It's my understanding that the backport is available now in 1.4. However, seem to be having some issues with it. Just wondering if I have everything setup right. I'm running 1.4.26.2 realtime. queue_members: `uniqueid` int(10) unsigned NOT NULL auto_increment, `membername` varchar(40) default NULL, `queue_name` varchar(128) default NULL, `interface` varchar(128) default NULL,

Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic

Hello guys, I've one SIP trunk that support multiple DID. Only the trunk is documented in sip.conf (called DID is taken from the sip-header in real time). I would like to limit the number of simultaneous calls on each DID. Is there a way to achieve this ? My understanding is that the SIP configuration parameter "limitonpeers" will limit at the trunk level, right ? Thanks in advance

Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648 Peer User/ANR Call ID Seq (Tx/Rx) Format Hold

" Hello, I have been trying to get my Grandstream GXP2000 phones to stop showing ringing state on monitored extensions. But no matter where I put notifyringing=no asterisk still sends the ringing state to the phones. Is this a bug I should report or is there another way around it. Here is how i have my subscriptions setup: extensions.conf [demo] exten => 6100,hint,SIP/100 exten =>

Hello all I'm suggesting to add line 'writable = yes' for Profiles share in file examples/smb.conf.default. Then the complete share def looked like ;[Profiles] ; path = /usr/local/samba/profiles ; browseable = no ; guest ok = yes ; writable = yes The reason is simply I spent two days figuring out why my Samba PDC did not work. That line was missing. Not very clever, I

Ok it's now Solved ! In spite of trying all day long to figure it out what's wrong in my sintax i could find, ...using a pencile and a pice of paper, combining all commands parameters invoked so far and, came to the conclusion that -L does not stand with -l , think that i couldn't find in any docs or faq so far. Now, symlinks are being copied without a trouble. Tnks Paul for

Hello Andrew I'm trying nothing ! But when I do /etc/init.d/samba restart i've the problem that you can see on the screenshot. nbmd is not necessary on a AD DC ? Regard Le 29/06/2017 à 10:18, Andrew Bartlett a écrit : > On Thu, 2017-06-29 at 09:22 +0200, Hénoch Hervé via samba wrote: >> Hello, >> >> We have followed the migration from samba4.2 to samba4.6 from

On Thu, 29 Jun 2017 12:55:46 +0200 Hénoch Hervé <h.henoch at isc84.org> wrote: > Yes i'm on a AD DC. When I have installed AD DC two years ago and > "apt-get install" has installed on the system nmbd (jessie). > Ok, put your smb.conf back to this: [global] workgroup = SC1 realm = sc1.local netbios name = VSPDC1 server role =