similar to: Configure Asterisk to preserve SIP header?

Hi, I notice that if two or more existing connections match an ingress policing filter, the input bandwidth does not get evenly divided up between the n connections. Kinda like litters of baby animals, where the stronger babies get more access to the mothers teats and grow up bigger and faster than their siblings. The only workaround that''s working for me is to set explicit ingress

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.

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:

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 ->

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>

Quick question for those who know the innards of chan_sip: Does chan_sip use the "To:" header of an incoming INVITE request, for anything other than setting SIP_HEADER(TO) ? As far as I can tell so far, the target extension is taken from the request URI, i.e. sip:extension at domain, and the target context is taken from the section in sip.conf that matches the request's source IP

Hi, I''ve released an alpha of ''pyshaper'' - a python prog which simplifies traffic-shaping. Requires python2.2 or later, a 2.4 or later Linux kernel with QoS options compiled in, the iproute2 suite and optionally GeoIP as well. pyshaper periodically netstats the current TCP connections, matches them against your rules, and dynamically calculates/generates/executes

Hi, I have the following situation. I have one account created in my VoIP provider. Asterisk registers this account with the usage of 'register = ' command in the sip.conf file. I have a number of aliases assigned to my user which correspond to different public/PSTN numbers through which I am accessible. When there is an incoming call from my sip provider 'some_extension' which

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

I'm working for a higher education institution, and we have Novell Netware for our file sharing services. We are looking at what migration paths are available. I know samba works, we use it on a number of Solaris and Linux boxes and have it authenticate against our Windows ADS. Manually editing samba confiiguration files for up to a hundred users is OK. The challenge is how do you manage a

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()

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

Hello. I have discovered that mongrel does not correctly take on all the groups of the requested user/group combination. It seems that while the specified user and group is correctly activated, all the other groups that are associated with this user are not enabled and the group permissions remain the same as the caller (i.e. root). This problem (and solution) is discussed in the Ruby Forum:

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

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 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",

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 Dmitriy and others and thanks for your help so far. The option "match_auth_username=yes" seems to have had no effect. From my reading, this option will try to match the username of the incoming SIP account to a section heading. If that is how it must work then i can see a big problem. I'm trying to present the receptionist with a nice display of which line the call came in on.

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 >

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