similar to: cut ip adress from caller id number display (ci$co 7941)

Minor update - use the following: > if (strcasecmp(data, > "x-Asterisk-Request-URI-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]

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Extensions No Problems Panasonic Ext -> SIP Extensions No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1]

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

I did a NoOp and see what the callerid was and when coming from the SIP Ext->Voip it is set to the Extension Number of the SIP Extension (as you would expect). When coming from the Panasonic the CallerID is blank, I tried setting it to nothing again, and I tried setting it to the callerid of the voip provider, a sip extension id, the extension number on the Panasonic side, the zap channel

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Ext No Problems Panasonic Ext -> SIP Ext No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",

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, 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:

I have logs full with this messages... I must have qualify turned on, because phone is behind firewall, main problem si, that phone is each hour about one hour unavailable! :'( I tried to modify minexpiry/maxexpiry sip.conf timeouts, but nothing help me. I'm using latest firmware 8.4 in phone, will be better to downgrade? to what version? (latest asterisk 1.4branch) [Dec 7 00:36:56]

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 ->

Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but it does work. For prosperity, the SIP service is through Internode. Here is my "extensions.conf" file: exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one exten => s,2,Set(pseudodid=${SIP_HEADER(To)}) exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)}) exten =>

Hi, Just to be sure: Is there a dialplan function in Asterisk that parses custom "name-addr"-style SIP headers for me? If I wanted to do it right the syntax name-addr *(SEMI generic-param) is quite complex to parse in the dialplan using nothing but CUT(). It's so easy to make false assumtions about angle brackets (< >), whitespace (LWS), quotes (") around the

Hi, I want my Samba 2.2.4 Machine to join my nt-domain. When I try this I recognize that samba seems to use a wrong IP-Adress! I have no Idea where this wrong adress comes from!? Any Ideas? --- cut --- # /usr/local/samba/bin/smbpasswd -D 10 -j CAD -r bsadpp01 -Uadministrator ... a lot of text ... Adding chars 0xe7 0xe8 (l->u = True) (u->l = True) Adding chars 0x9c 0x0 (l->u = False)

----- Asterisk 1.8 will allow to read SIP response codes in the dialplan via ${HASH(SIP_CAUSE,<channel-name>)} Asterisk 1.8 also comes with a 'use_q850_reason' configuration option = for generating and parsing, if available:=20 ----- That will give you what you want if you consider upgrading to v1.8. =09 -----Original Message----- From: asterisk-users-bounces at

is there anyway to change Sip headers in local channels? if a user sets forward on their handset, calls coming in to the handset get diversion header added: Diversion: "202" <sip:202 at 192.168.1.46>;reason=deflection Then asterisk sends the call to local channel: - Now forwarding SIP/201-00000483 to 'Local/3333333333 at test' (thanks to SIP/202-00000484) and not all

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

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? Please help me, where can I add SipAddHeader() in

Hi, when attempting to dial an invalid number with Nikotel this is returned: 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

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > 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 >

To any that may be interested in the implementation of the conditional expression in the expression parser (ast_expr2*) in asterisk, I've filed the patch at: http://bugs.digium.com/view.php?id=4459 Right now, a comment has been added noting that the IF func provides this capability, and asks if both would really be necessary. It's a good question. I haven't been following the

Hi, I just started with setting up a new asterisk system, that will operate on a sip trunk, but I wonder, how to transfer the calls to different extensions, because all calls appear as being send to the base number of the trunk. E.g. given the trunk range of 1234567800-12345678099, a call to 1234567800 is matched by the same pattern as a call to 12345678099. ; matches 12345678099, too exten