search for: sip_header

...I-pseudo-header")==0) > { > ast_copy_string(buf, p->initreq.rlPart2, len); > -----Original Message----- > From: Steve Langstaff > Sent: 23 October 2006 09:58 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [asterisk-users] SIP_HEADER function; what names > are available? > > Looking at the source code for Asterisk 1.2.7.1 (just what > I've got handy), it appears that the SIP_HEADER() function > just parses the SIP INVITE for whatever SIP *header* you specify - so: > a) there's no list of headers yo...

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

Hi * users, I would like to extract the information present in the SIP REFER message that comes to asterisk. Would SIP_HEADER() allow me to do that ? I have used SIP_HEADER() for extracting the to and from SIP headers previously. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

Hi all, is there anyway i can set SIP_HEADER(To) to the value i like? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070409/528077f9/attachment.htm

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>

...rned: SIP/2.0 400 Bad Request request uri is neighter my realm or a valid dns and Asterisk prints smth similar on the CLI. However it appears that I cannot get access to "400 Bad Request" from the dialplan because this error is not part of any SIP header, and therefore the function SIP_HEADER won't do the trick. Right or wrong? ;-) Cheers, Philipp

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.

...voip/${EXTEN:1}) exten => _955X.,1,Dial(SIP/acevoip/${EXTEN:1}) [from-acevoip] include => dialstring exten => 073.......,1,Answer exten => 073.......,2,Dial(Zap/g1/100,60,tn) exten => _073.....XX,1,Answer exten => _073.....XX,2,System(mkdir /mnt/data/Recording/${SIP_HEADER(TO):12:3}) exten => _073.....XX,3,Set(CALLFILENAME=/mnt/data/Recording/${SIP_HEADER(TO):12:3 }/${SIP_HEADER(TO):12:3}-Received-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-$ {CALLERID(num)}) exten => _073.....XX,4,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0)) exten => _073.....XX,5,Dial(SIP/${SIP_...

Hi, I am trying to capture "Server" header in a 200 OK reply message. My idea was to use Dail(SIP/user at domain,30,M(GetOtherPartyInfo)), and inside of GetOtherPartyInfo macro use SIP_HEADER function. For example: [default] exten => _X.,1,Dial(SIP/user at domain,30,M(GetOtherPartyInfo)) exten => _X.,n,Hangup() [macro-GetOtherPartyInfo] exten => s,1,NoOp(SIP Server: ${SIP_HEADER(Server,1)}) unfortunately the above doesn't seem to work: -- Executing [s at macro-GetOther...

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)

...too exten => _1234567800,1,Dial(SIP/int) Except from SIP invite with tcpdump: INVITE sip:123456780000 at provider:5060 SIP/2.0 From: <sip:013579246800 at provider>;tag=as6bc7cbbc To: <sip:1234567800099 at other:5060> I wonder, if I really need to grab the extension with Set(DN=${SIP_HEADER(TO):5}) or something similar? Another issue is, that I don't like asterisk to decline foreign INVITE requests. Any best practices from within asterisk on how to ignore SIP invitations, that don't match certain criteria (neither local nor from sip provider)? System: openSUSE 42.2, Aster...

...voip/${EXTEN:1}) exten => _955X.,1,Dial(SIP/acevoip/${EXTEN:1}) [from-acevoip] include => dialstring exten => 073.......,1,Answer exten => 073.......,2,Dial(Zap/g1/100,60,tn) exten => _073.....XX,1,Answer exten => _073.....XX,2,System(mkdir /mnt/data/Recording/${SIP_HEADER(TO):12:3}) exten => _073.....XX,3,Set(CALLFILENAME=/mnt/data/Recording/${SIP_HEADER(TO):12:3 }/${SIP_HEADER(TO):12:3}-Received-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-$ {CALLERID(num)}) exten => _073.....XX,4,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0)) exten => _073.....XX,5,Dial(SIP/${SIP_...

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

Hello, I would like to verify if a specific SIP header exists, and if yes, extract the partial content from another header. 1. Is there a way to verify if a specific header exists? 2. How do I extract data that is between the first : and the following @? Specifically, The data looks like <sip:1234567890 at 10.0.0.1:5060> and I would like to get only the 1234567890 I tried to use the CUT()

Asterisk 1.4.12 I wish to extract some custom headers from a SIP REFER message but am unable to do so. However I can extract them from an INVITE. The code is: exten => _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ; exten => _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ; Examples of the INVITE (works) and REFER (doesn't) messages are below. U 147.202.001.001:5060 -> 127.0.0.1:5065 INVITE sip:0116499123123 at 127.0.0.1:5065 SIP/2.0 Via: SIP/2.0/UDP 147.202.001.001;bran...

I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To) same=>n,.... But when a call comes in to the gv-voice context, "s" doesn't match the extension: res_pjsip_session.c:2991 new_invite: Call from 'gv-voice' (UDP:10.10.10.80:5062) to extension '<xxxxxxxxxx>' rejected because...

...from-acevoip] > > > > include => dialstring > > > > exten => 073??.,1,Answer > > exten => 073??.,2,Dial(Zap/g1/100,60,tn) > > > > exten => _073?..XX,1,Answer > > exten => > _073?..XX,2,System(mkdir /mnt/data/Recording/${SIP_HEADER(TO):12:3}) > > exten => > _073?..XX,3,Set(CALLFILENAME=/mnt/data/Recording/${SIP_HEADER(TO):12:3}/${SIP_HEADER(TO):12:3}-Received-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${CALLERID(num)}) > > exten => _073?..XX,4,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0)) > > exten =>...

I am trying to set up my Asterisk server so that it will recognize an incoming call to the Asterisk's own Location Routing Number (LRN), validating the "rn" in the INVITE and then using the Called Number from the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200:5060;user=phone;transport=udp SIP/2.0

...but CUT(). It's so easy to make false assumtions about angle brackets (< >), whitespace (LWS), quotes (") around the display-name, character escaping etc. All of the applications of CUT() I have seen are way too simplistic. Example of how it could work: Set(addr=${SIP_PARSE_HEADER(${SIP_HEADER(P-Asserted-Identity)},addr-spec)}); Interesting parts include: name-addr, display-name, addr-spec, scheme, userinfo, user, telephone-subscriber, host, hostname, port, ... Actually headers like P-Asserted-Identity can even have more then one value. ---cut--- PAssertedID = "P-Asserted-Id...

...going to require that I do some matching and > > substring-type variable replacement to hit a context with just the > > Called Number part of the request, but I wondered if anyone had a > > working example of this before I started putting too much effort into it. > > Use the SIP_HEADER function > > http://www.voip-info.org/wiki/view/Asterisk+func+sip_header I am not sure that this is needed here. The Request URI has all of the values that I need. I agree that I might need to CUT part of the R-URI, but I don't need access to any other header to find the info I need. W...