similar to: a=recvonly

I'm trying to use a simple page function. It starts a MeetMe conference with the devices I've listed, but the devices hang up after 3-5 seconds. After doing some research I found this was a problem, and I needed to remove a (5) from app_page.c Well, my app_page.c didn't have the (5). I did make clean; make install again just in case I had some weird compiled version installed that

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

I've been working on writing a subroutine to page groups of phones at once and I'm having some difficulty. My goal is to have a user call an extension, I record the page they wish to play, I then page out that recorded file to the phones in groups. [sub-masspage] exten => s,1,NoOP same => n,Answer same => n,Set(filename=$PAGE) same => n,Wait(1) same =>

Hey Guys! I have perl script for allpage which is working fine with asterisk 1.8.2.3 version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is there anything changes ? If i run this script from command like it works but not from asterisk dialplan. This script nothing but just connecting AMI interface and using Variable: SIPADDHEADER=Alert-Info: Ring Answer variable to

Ladies, Gentlemen We are experiencing an unusual problem in our asterisk 1.4.34.. We are attempting to determine if channels are in use before paging to them. This works correctly, as in it pages the phone.. however, we see the error message below on the console... after googling, we discovered limited information regarding the issue... -- Executing [NPANXX7298 at from-pstn:1]

I am having a weird issue with setting the recording file for the Page app. Here is some quick background info I have a macro that pages all my phones: [macro-pageall] ; Context for paging all devices. ; This will search the sip table in the realtime database ; for all phones that start with a number. That number is ; passed to this macro as ${ARG1}. ; ; ARG1 = The

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

I use SIPAddHeader today to put some proprietary info into the SIP header of an outbound call. Now I'd like to add some proprietary info to the SDP portion of an outbound call. Can this be done with SIPAddHeader? Thanks in advance, Tom -------------- next part -------------- An HTML attachment was scrubbed... URL:

Group I'm having some trouble with asterisk and the page cmd. Any help would be great! This is what's in my extensions.conf exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0) exten => _**2,2,Page(SIP/36651)|d exten => _**2,3,Hangup CLI output ******************** Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid =

hello list, i have asterisk 1.4.43 installed and i want to configure the auto-answer exten => 506,1,SIPAddHeader("Call-Info:\; answer-after=0") exten => 506,n,Dial(SIP/105) when i call the 506 the SIP/105 still ringing, i have snom 320 and i have set the Auto Answer Indication: on i test with Dial and page() but the issue still the same any help please -------------- next

Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly -------------- next part -------------- An HTML attachment was scrubbed... URL:

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>

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Hi everyone I’m building an Asterisk 16/PJSIP server and my dialplan uses SIPAddHeader & SIPRemoveHeader but the apps don’t appear to be installed in v16. Can anyone tell me where they went and how to get them installed please? Thanks Mark. Mark Farmer Senior UC Systems Architect Intercity Technology Limited HQ 101-114 Holloway Head, Birmingham, B1 1QP Tel: 0330 332 7933 / 07872542107 /

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 list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

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

We just finished an upgrade of our Asterisk system to an HA environment on a pair of servers using Linux-HA. As part of the upgrade, we also moved to Asterisk version 1.8.4.3 Most things are working quite nicely on the new system. However, I?m having trouble getting a paging feature to work. In Asterisk 1.4, we simply used the Page() application like this:

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 =>