similar to: App Page() in 1.2.5

Hi. I am thinking about building an asterisk system for a small business and want to be able to page through the phones. It seems like to do this asterisk needs auto-answer support in the phone. I know the SPA-841's support this, as do Cisco phones, but I have been unable to determine if the new Cisco/Linksys SPA-941's do. Does anyone here have experience with trying to use auto-answer

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include

Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root at ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk

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, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

I have gotten intercom working on my office phones (Linksys SPA-942s), but I have noticed that if someone is in a call, it places the call on hold and sends the intercom audio to the person holding the phone that is being paged. I'd like to add logic to my dialplan that doesn't send a page to a phone that is currently in a call. But to do this I need a function that will tell me if a

All other phones we work with will auto-answer when we do this: [macro-paging1way] exten => s,1,SIPAddHeader(Call-Info: answer-after=0) exten => s,n,Page(${PAGINGLIST}) exten => s,n, Hangup The SPA phones simply ring. I have verified that Auto Answer Page is set to yes (the default). We've tried a variety of firmware versions and phone ages, going back to an old 942 and new 504s.

Sipura has implemented auto-answer in version 0.9.5 of the SPA-841 firmware. However, it is implemented via the Call-Info header, which Asterisk stable doesn't currently support. The attached patch implments a quick hack to support the Call-Info header from the Dial() application by way of setting the CALL_INFO variable. For example, the following macro can be used to dial up a single

Hi, In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind Transfer when transferer hangs up before callee answers : - in Blind Transfer, caller (ie transferee) is hearing Ringing tone when callee's phone is ringing - in Attended Transfer, caller (ie transferee) is hearing Music On Hold when callee's phone is ringing - in Attended Transfer, if callee don't answer

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Does anyone have an (overhead) paging system that they like that works with SIP? We've got a client with an old paging system that (supposedly) just takes an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn't auto-answer the call, so paging never happens. [cid:image001.png at 01D4DFF6.9C1F1AA0] Michael J. Munger, dCAP, MCPS, MCNPS, MBSS Microsoft Certified

Hello, I am using AuthLogic as my authentication gem. Everything is great on my development server. When I upload to my passenger driven production server at Dreamhost, the sessions are persisting relentlessly. In other words, a user cannot log out. I have tried implementing the Active Record session store and have changed the :secret key in initializers/session_store.rb, but for some reason,

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

How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader("pchargingvector","val") in outgoing Invite. How can I achieve this? regards, Asif

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 all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP

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 All I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with with one of my ATAs not ringing. Basically, when I execute the Dial command, an error occurs: "Got SIP response 400 "In alert-info header: Empty value expected" Now in 1.2, I just issued the following command to overcome this problem: Set(_ALERT_INFO=). Now in 1.4, _ALERT_INFO is deprecated, so I

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

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