similar to: SIPAddHeader not working

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 Everyone, Our switch is sending P-Asserted info to asterisk however the information is getting removed when asterisk forks the call. Is it possible to have asterisk retain the P-Asserted on the leg. This is quite important for valid functionality of our network. Tried setting `sendrpid = yes` and still same problem. We really don't want to have to `SipAddHeader` as it is already being

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u

Hello all, I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233445 at 192.168.1.100 SIP/2.0 From: "123456789"<sip:*123456789*@192.168.1.100>;tag=333333333 To: <sip:*6700112233445 at

Hello. Is it possible to configure Asterisk to preserve specific SIP INVITE headers when setting up a call? Specifically, I have a custom SIP client that sends an additional header in the INVITE request when originating a call. This is to request that the call is auto-answered by the destination phone. i.e. Call-Info: <sip:192.168.100.50>;answer-after=0 If I use wireshark to sniff

Good evening, I have a problem with SAMBA domain, I have many pc's with S.O Windows when I try to put then on a samba domain, then don't locate my domain. The OpenSuse Linux 10 show me any lines on the logfile, i'm put these lines below. I'm have OpenSuse Linux 10 + SAMBA 3 with LDAP authentication. Below the SMB.CONF + Slices of a LOG file. Thanks to all !

Hi all, We currently have 2 Asterisk boxes and we pass calls to a fro. All works great except we now need to pass variables between them. For example now on box 1 we have: exten => _23XX,1,SetVar(Foo=1234) exten => _23XX,2,Dial(SIP/${EXTEN:0}@Box2) When the call dials into Box 2 the variable Foo does not get passed... Does anyone have any clever ideas? -------------- next part

Hi, all I wanted to fetch data from Bloomberg for govt bonds, and analyse it further. I am having trouble in getting data as when I use field=PX_LAST, it is giving the prices but when I use field=CPN, or ISSUE_DT, it is not giving the results and just bouncing back <NA> for that. This is the piece of code: > library(rJava) Warning message: package 'rJava' was built

Dan Gohman wrote: > Hello, > > On Nov 4, 2009, at 1:51 AM, Hans Wennborg wrote: >> >> / Hans >> Index: lib/Analysis/BasicAliasAnalysis.cpp >> =================================================================== >> --- lib/Analysis/BasicAliasAnalysis.cpp (revision 86023) >> +++ lib/Analysis/BasicAliasAnalysis.cpp (working copy) >> @@ -633,6 +633,15

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

The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk "..the most powerful tool for manipulating and accessing CPN data.." > http://www.theregister.co.uk/2004/07/07/hackers_gut_voip/ I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time.

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Hello, I'm trying to use SIP_HEADER function on my dialplan but I receive this message (on the console): pbx.c:3367 ast_func_read: Function SIP_Header not registered Why? Thank's - Bakko

We have 2 T1's coming from our phone switch to a digium TE220B. We have managed to get CPN and the extension outpulsed from the switch, but call setups are really slow. Our T1's are set up as E&M Wink, and they send us the last 5 digits dialed followed by the 10 digit calling party number (we couldn't get the switch to be happy with *CPN*+5* to use featd). We are using asterisk

On Nov 6, 2009, at 7:49 AM, Hans Wennborg wrote: > > I'm not sure what you mean by generalizing. > Do you mean I should do the check on O1 and O2, which are the results of calls to getUnderlyingObject? > > Something like: > > if (const ConstantPointerNull *CPN = dyn_cast<ConstantPointerNull>(O1)) > if (CPN->getType()->getAddressSpace() == 0) >

Dan Gohman wrote: > On Nov 6, 2009, at 7:49 AM, Hans Wennborg wrote: >> I'm not sure what you mean by generalizing. >> Do you mean I should do the check on O1 and O2, which are the results of calls to getUnderlyingObject? >> >> Something like: >> >> if (const ConstantPointerNull *CPN = dyn_cast<ConstantPointerNull>(O1)) >> if

I remember seeing something somewhere that described how I could get SIP header information with Asterisk. It was a command or a variable. Anyone know what it is? Thanks. Doug.

Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184 at CallFromQueue-000007f4;2 to 'Local/23 at from-internal' (thanks to SIP/myaccount184-00003729)

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, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part