similar to: Queue: SIP status not set to busy

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

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

Anyone have any info on the date (or bug tracker number) of the change from incominglimit to call-limit, and is it call_limit or call-limit? Does it work with SIP friends? Running CVS head 8/24 (right before 1.2 release). -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to all the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? This is a piece of my sip.conf: [202] type=friend secret=202 host=dynamic ; This device registers with us username=202 ; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored.... I've also tried

Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. I configure sip.conf like following: [10] type=friend qualify=yes

Hi, We've been using asterisk as our main telephone-communications platform for years now, and we wrote several extra scripts and features for it. Now we 're looking for a solution to limit the number of channels going to an external SIP provider. We recently upgraded our system from asterisk 1.0 to 1.2(.9 now) to be able to use such features, but nothing helped... When we configure a new

For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available

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

Hi all, I'm using Dovecot 2.1.16 and I'd like to know if the following configuration is supported: * An SQL dictionary quota * A static userdb to get uid/gid/home information * An SQL userdb to get per-user quota limit Currenlty, current quota usage are properly stored into the database but per-user limits are not found. If I replace the SQL dictionary by the maildir backend (my

Hi, An observation on this feature, which I may have completely misunderstood, so flame away if I am being dumb :) Looking at the code, setting "limitonpeers=yes" causes all user and peer calls to be ref-counted as if they are peer calls (assuming a user and peer of the same name exist). A side-effect of this is that an incoming call seems to have its call-limit evaluated based on the

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

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, Summary of my problem: Remote X forwarding is apperently randomly impossible for different display numbers. At the end of this mail you will find a recipe for how to reproduce this behaviour easily. I use SuSE 10.2 with the following openssh version: OpenSSH_4.4p1, OpenSSL 0.9.8d 28 Sep 2006 Clients (Linux and Windows (Cygwin)) connect to the server with X-Forwarding enabled

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

Hello, I have a problem with the meetme application, but I'm not sure if it's a bug or just a misuse. I'm trying to get a 3 way call system working as follow : A calls C B calls C C who's speaking with A or B, presses one keypad (only one) and the 2 incoming SIP (A, B) and C are redirected into a conference room. Therefore, I created an entry in the applicationmap

Hello, I'm sorry if this is not the correct place to ask this question but what I'm looking for is any data on the reality of Speex working on an ARM7 core? I'm looking at doing a few diffrent projects where I will be using a proc such as the LPC2378 with the ARM7TDMI-S core running at 72Mhz (although errata show only 60Mhz is currenlty possible). Do you know if there is a port

Hi all, I'm setting up a CentOS 4 in AMD64 box. The goal is to mount a terminal server with LTSP software. Currenlty I'm in troubles with the multimedia support. I've added Dag's, Karanbir's and Dries' repositories according that. ?Is there some intention to make a multimedia group as others groups (FTP Server, Web Server... etc)? It would be great. ?What packages do you

Hi, I'm running a highschool network with Debian etch and the etch-cran "backports" on top of it. Being bio-informatics, the newest R is essential to us. I did some upgrading experiments; switching to ubuntu is not an option, neither is fetching the sid debs for r. When trying to upgrade from etch to lenny with etch-cran packages installed things block on etch-cran dependencies.

Hi! Let's say I have vector x with positive integer values ranging from 1 to N. I need to obtain another vector y of size N where y[i] contains the number of times value i occurs in x. It is in a sense similar to hist() (with appropriate number of breaks) or table() with numeric "factors". Currenlty I use a custom function for that, but thought maybe there is a more