similar to: queue_variables() function

2009/12/4 Olivier <oza-4h07 at myamail.com> > Hello, > > Has someone successfully used this QUEUE_VARIABLES() function (in > 1.6.2-rc7) ? > I tried to use it as I'm using SIPPEER() but without success. > > A previous question about it remainded unanswered ( > http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). > > Regards > How can

hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the

Hi, Can somebody please shed some light on how to use the QUEUE_VARIABLES() function? The built-in help says ---cut--- Return Queue information in variables [Description] Makes the following queue variables available. QUEUEMAX maxmimum number of calls allowed QUEUESTRATEGY the strategy of the queue QUEUECALLS number of calls currently in the queue QUEUEHOLDTIME current average hold time

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, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1

Hello, In sip.conf, each peer/friend/user entry gathers several parameters such as type, canreinvite or mailbox. How can you specifically access to mailbox value from dialplan ? I know how to access custom parameters (ie setvar=FOO=value) but I don't know to access standard parameters. I'm specifically concerned to access to mailbox's value (from a given entry) but would be

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Here is what I have in my extensions.conf file now. Trustrcid and sendrcid are turned to "yes" in the conf file. I'm not fully sure how the SIPCalledRPID works though. The example I found seems to try and provide the stuff automatically (id and name), but does the SIPPEER stuff even exist? I think this is probably the right track though. Any insight would be much appreciated.

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

Hi, Is anyone using Fring as a SIP client to an Asterisk server ? A prospective customer of mine is asking to integrate its iphones with an Asterisk server and after googling, I still have some unanswered questions : 1. Which codecs are available when calling from fring ? 2. Is it easy and natural to change your presence status (available, busy, ...) with Fring or will users prefer to use

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

Hi All, I'm using Asterisk 1.4 branch and checking the status of some SIP Peers with the functions ${SIPPEER(101:status)} and the result is "OK (48 ms)". Seems to work fine. Now I would like to use the function CUT to set a variable with the 'OK' portion of the status "OK (48 ms)" and then do some follow on stuff if the status is OK. I'm running into syntax

try, database get SIP/Registry/<peername> it gives you a string which contains the info, then pass it to CUT to extract ip-adr and port Freddi > To do that you need to get the remote ip address and port of the sip peer! > > I found the function: > > ${SIPPEER(exten:ip) > > But how can I get the port??? > >

On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup, such as: == Registered custom function 'SIP_HEADER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPPEER' in tree 'description' == Registered custom function 'SIPPEER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find

Howdy, I'm trying to re-write my voicemail check extension. I formerly used the SIPPEER function to get the mailbox for a peer with ${SIPPEER(${peer},mailbox)} Is there a way to do this with PJSIP now that I've converted over? I see a function PJSIP_ENDPOINT and it has a mailboxes subset but I'm not retrieving any data from it when I query it. -- A human being should be

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");

I'm still struggling to pickup calls. I now have a single context (entryocginternal) where I have "include => parkedcalls". The log below shows me calling from one internal extension to another, then picking up, then parking the call. -- SIP/239-0915d5c8 is ringing -- SIP/239-0915d5c8 answered SIP/233-0915bf40 -- Packet2Packet bridging SIP/233-0915bf40 and