similar to: Intercom with Dial() works, but not with Page()

I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. I am using chan_sip2z.c and

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

Hi, I'm testing paging using snom 360. Can someone correct my dialplan? Regards, Jason. ================================================== ;exten => _99XXXX,1,SIPAddHeader(Call-Info: Answer-After=0) ;exten => _99XXXX,n,SIPAddHeader(Call-Info: <sip:192.168.1.113>\;answer-after=0) ;exten => _99XXXX,n,Dial(SIP/${EXTEN:2}) exten => _99XXXX,1,Set(__SIPADDHEADER=Call-Info:

Hello, I am struggling to have a SPA504G to auto answer (for intercom/paging). I have tried the following SIP headers (not all together), but without luck: SIPAddHeader(Call-Info:\;answer-after=0); SIPAddHeader(Call-Info: answer-after=0); SIPAddHeader(Alert-Info: info=intercom); SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(P-Auto-Answer: normal); Any other ideas? Leandro PS I have set

Hi list, I'm stuck for the moment @ the following : My Query (in a macro) : exten => s,n,MYSQL(Query resultid ${connid} SELECT\ vakantie_set\ vakantie_data1\ vakantie_data2\ FROM\ AstDB\ where\ SIPACCOUNT="${ARG1}") Asterisk CLI : [Aug 30 14:07:42] -- Executing [s at macro-vakantie:2] MYSQL("IAX2/zoiper-9238", "Query resultid 1 SELECT vakantie_set

I'm having hard times with paging intercom Heres my dialplan exten => 777,1,Goto(intercom,777,1) [intercom] exten => 777,1,SIPAddHeader(Call-Info: <sip:192.168.16.105>\;answer-after=0) exten => 777,2,Page(Local/308 at page& Local/309 at page& Local/310 at page) [page] ; Paging context exten => _X.,1,Macro(page,SIP/${EXTEN}) [macro-page] ;

I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call "22" and the phone rang it did not auto answer. Did I miss something? exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten => 22,n,SipAddHeader(Alert-Info: Ring Answer) exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten =>

Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's for our general phones. I have a global defined which has all the SIP channels concatenated together - this is ${ALL-PAGE-EXTS}. The problem comes when a user is on the line, and someone else uses the intercom function to page

Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer.

I want to implement a one way announcement and paging facility using Asterisk and Sipura phones. The wiki says Sipura phones only support Auto Answer using the Call-Info header which is no lone shipped with asterisk stable since 1.0.4. I would like to ask if anyone has implemented a similiar facility using Sipura SPA-841 or any other SIP phones. If I could take a look at how

Hey, Hi, All, We have a few dozen of the Kirk (ie: Polycom bought this European brand) out there & most all work very well & work very well with most all versions of Asterisk. But we have been tripped-up by one combination of firmware & version & configuration variables. We are running Asterisk 1.4.23.1 (TrixBox CE). We are running latest stable firmware on the handsets. Most

I am trying to page my Grandstream GXP-2000 phones and keep getting the error message: Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete destination '' supplied. How can I fix this error? The two contexts below do either one-way paging or two-way paging to all Grandstream phones in a list. [One_Way_Page_GROUP] ; one to many page exten =>

Have someone running fine Alert-Info with a Snom 370 ( System Information: Phone Type: snom370-SIP MAC-Address: 0004132661BD IP-Address: 192.168.10.170 Firmware-Version: snom370-SIP 7.3.14 14961) i've tried exten => 200,1,SIPAddHeader(Alert-Info:<http://www.notused.com>\;info=alert-external) exten => 200,n,Dial(SIP/${EXTEN},30) Can see into the phone SIP trace is

Hey Gang, Trying to figure out how I can do the following (have each part working individually but drawing a blank on combining) 1) click on-screen which sends an AMI originate (works fine) 2) the originated call is to an internal extension that looks up the number to be dialed (works) 3) then via Perl, adding in a SIPAddHeader for answer-after=0.. (works separate from the above) What I

Hello list, this is the dialplan : <snip> exten => s,n,Dial(SIP/test1&SIP/test2,,t) <snip> exten => 10,1,Dial(SIP/test1) exten => 20,1,Dial(SIP/test2) So there is an incoming call that rings SIPaccounts test1 and test2. Account test1 answers and wants to transfer the call to test2. Transfer is : #20 This is what the CLI shows : [Jul 2 10:55:30] -- Executing [20

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk

Hi guys I'm using asterisk 1.4 and going on to Snom phones. I'm trying to add a sip header to make the Snom phone use a different ring tone on one particular incoming number. I have added the following to the dial plan of the incoming context +------+------------------+-------+----------+--------------+-------------------------+ | id | context | exten | priority | app

This possibly isn't 100% asterisk related, but I'd like some opinions/feedback... A customer wanted different ring-tones to differentiate external and internal calls. No biggie once I'd worked out that details - they have 100% GXP2000 phones, so adding in the relevant SIP header and altering the phones to suit seems like it's going to be a solution... But I started to look at

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)");

Hi, I'm trying to use the SIPAddHeader application to add a header containing to semicolon separated strings like this: exten => 12, 1, SIPAddHeader(X-TestHeader:a=test1;b=test2) But in the resulting INVITE message only the first part (X-TestHeader:a=test1) is added. Setting into quotation mark doesn't change anything. exten => 12, 1,