similar to: How to add SipAddHeader in outgoing call file.

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

In tracking down a bug in AuthLogic I realized that I had made the same error in one of my libraries. Knowing what the error was made creating a test to expose it (using cucumber) rather trivial. However, it has occurred to me that this sort of issue is far more subtle than I first appreciated. So, I am looking for some opinions on how to test for this sort of thing in a more general sense. At

I''m a big fan of stubbing Time.now so it returns a known value. I used to be able to use stubba and say: @time_now = Time.parse("Jan 1 2001") Time.stubs(:now).returns(lambda{@time_now}) However, something in trunk broke that. Fine, rspec''s got its own stubbing lib now, so I tried switching to that: @time_now = Time.parse("Jan 1 2001")

Hello! I got an interesting task to make with asterisk: pstn--- * ---sip--- * pstn This sounds common till now. What I have to make is: 1.the call is routed through PSTN to asterisk1 (#1) which has ISDN PRI interface(s) - leg1 2.#1 doesn't pick up the call, neither rejects, it just place into state CALL PROGRESS (?) [maybe gives back alerting tone? probably not] 3.#1 notifies that there is

Signed-off-by: krishnaiah bommu <krishnaiah.bommu at intel.com> --- moduli.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/moduli.c b/moduli.c index 578fc48..c1fa484 100644 --- a/moduli.c +++ b/moduli.c @@ -153,12 +153,14 @@ static int qfileout(FILE * ofile, u_int32_t otype, u_int32_t otests, u_int32_t otries, u_int32_t osize, u_int32_t ogenerator, BIGNUM *

Following up on this, I've figured out what's going on. What I'm not sure of is how best to address the situation. I'd like to avoid going down to the MIB level because I just don't understand it. "Split phase" output devices seem to fall into an undefined area. They're not three phase, but require additional MIB values so APC uses their 3-phase MIB

Loan, Thanks for your help in this matter. Having never used astdb before, can you point me to an example on this?? Thanks hugely, Clara >> Hi Clara, You could put some data into astdb and query for the outgoing line and callerid based on internal callerid (extension). something like user/201/outline 89859715 user/201/outcallerid 89859715 and so on... By the way:

Hi everyone, How can I add the equivalent of: exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) in a .call file? This is to support paging to Polycom phones... Thanks for all info! Steve

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.

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

Does SipAddHeader only allow headers to be added to INVITEs, or should it also allow headers to be added BYEs or SIP responses as well?

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,

Hi All, I'm doing some simple paging functions and using the SIPAddHeader cmd. * 1.2 branch. Using it in the extensions.conf file, it works fine: exten => _*2XX,1,SIPAddHeader(Call-Info: sip:\;answer-after=0) in * console: lab2*CLI> -- Executing SIPAddHeader("SIP/204-0818dcd0", "Call-Info: sip:;answer-after=0") in new stack When i put the same cmd in Realtime

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:

Hi, I am using the XML-browser and Call-Info header features for some SIP phones. SIPAddHeader(Call-Info: ...) seems to work only in the outgoing direction. Does somebody know a way to send a Call-Info header to the originating SIP device by using only the dial plan? Currently, I am using the XML-browsers to update callee info, but I'd like to use the icon purpose to do that. It's

Hi list, is there any existing IAX command to add information to a call like SipAddHeader? Another solution is sending text frame (0x07) frame type, but I don know how do it in a dialplan. Thanks for any hint. -- Daniel

Hi Guys I'm using asterisk 1.8.23.1 When I add a SIP Header from inside the extensions.conf (SIPAddHeader(Alert-Info:<http://www.notused.com>\;info=alert-internal\;x-line-id=0) ) it works fine. When I try to do the same thing from within a database table, all of the string apart from x-line-id=0 gets ignored. I've tried escaping the semicolon and not escaping it and the result is

Greetings: I'm in the situation where I'm trying to splash information picked off by an asterisk IVR into a Cisco call center environment. I'm under the impression that the ONLY way to do this is to setup socket connections with the Cisco "voice processor", or CVP, and send packets corresponding to GED-125. Cisco has a detailed 100+-page document detailing the internals of

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 /

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>