similar to: call transfer to asterik.. asterisk as an end point

Hi, I have a dlink dvg-1120M (mgcp) box that i will like to use with asterisk. Is it possible? has anyone done that? Here's a link to the product page at dlink. http://support.dlink.com/products/view.asp?productid=DVG%2D1120M Also, does anyone has or know where to get the firmware for Dlink DVG-1120S (sip model)? thanks. -- Zahid

I'm a bit confused about how * constructs the RURI when it generates a BYE. For the situation where * send the initial INVITE it constructs the RURI for the BYE from the contact header of the 200 OK response which is well and good. However when * receives the initial INVITE it does not use the contact header contained within to construct the BYE's RURI but constructs it from scratch. This

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it --------------------------------- Get your own web address. Have a HUGE year through Yahoo! Small Business. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I was having question about mysql driver support ( not odbc). Do we still need the asterisk-add-on to be installed for mysql support.? If yes, Which version should be used and from where I should get it? Thanks in adavance. ---- Thanks & Regards, PrashantAbhang -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -------------- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad

Dear Alex; Thanks for your kindly reply. Please explain for me what do u mean exactly in "a la" in the following sentence u wrote it below? " in SIP, this can be done via "re-INVITEs" a la the canreinvite= option for SIP peers in sip.conf" Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full

Hi, I'm trying to port our configuration form sip to pjsip channel and have following issue. Sip.conf has a host parameter that sets the RURI to a given value. This functionality is needed in some of our scenarios where we need to send requests to specific IP address with specific domain in RURI. I did not found an equivalent to the host parameter in pjsip configuration. Did I

I want to calculate the commulative sum of any numeric vector with the following command but this following command does not work "comsum" My question is , how we can calculate the commulative sum of any numeric vector with above command Thanks Zahid Khan Lecturer in Statistics Department of Mathematics Hazara University Mansehra. ---------------------------------

Schlatter Christian wrote: >Dear list members > > > >I'm very new to R but a little informed about geostatistics. > >As I was looking for possibilities of geostatistical analysis in R I encountered at least two very interesting packages: > > > >Rgeo and gstat > > > >And of course I'm wondering now about the one which fits better my needs

Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give " cat /proc/zaptel/* " it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1

Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> To: <sip:8009499014 at X.YYY.32.10 :5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898

Hi, When I have the following extension matching defined: exten => _an_.,1,NoOp(-- Context routing-sip-announcement for ${EXTEN} --) Asterisk doesn't find it when it receives such SIP request: <--- SIP read from 192.168.129.38:7160 ---> INVITE sip:an_hellboy at ms.sip.rd.touk.pl SIP/2.0 Record-Route: <sip:192.168.129.38:7160;lr=on> ... for instance when I use such extension:

Hello, Is there any way to identify an incoming session based on the Diversion header? In my scenario, I have some unregistered endpoints (mobile phones) that make calls through our Asterisk, which controls the external call rights based on the endpoint's context. In a normal call, their number will be in the From header and the destination in the To an RURI, but when they make a call

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]

Hey list, Sorry for all the questions today. But I am trying to wrap up this ftp business and still having some issues. I appreciate your input. SELinux is temporarily disabled (until I can work this all out) and I am now able to log into the FTP server. [root at LCENT05:~] #/usr/bin/ftp localhost Connected to localhost (127.0.0.1). 220 FTP Server ready. Name (localhost:root): bluethundr

Gentlemen , I am getting this error when listing groupmaps [root@lndc1 lib]# net groupmap list [2004/12/26 15:21:38, 0] lib/smbldap.c:smbldap_search_suffix(1155) smbldap_search_suffix: Problem during the LDAP search: (No such object) [2004/12/26 15:21:38, 0] passdb/pdb_ldap.c:ldapsam_setsamgrent(2458) ldapsam_setsamgrent: LDAP search failed: No such object [2004/12/26 15:21:38, 0]

https://bugs.freedesktop.org/show_bug.cgi?id=95188 Bug ID: 95188 Summary: GeForce 840M (GM108) troubles Product: xorg Version: git Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau Assignee: nouveau at lists.freedesktop.org

hi, is there a way of calculating of measuring dependence between two categorical variables. i tried using the chi square test to test for independence but i got error saying that the lengths of the two vectors don't match. Suppose X and Y are two factors. X has 5 levels and Y has 7 levels. This is what i tried doing >temp<-chisq.test(x,y) but got error "the lengths of the two

using an example from r online help > state <- c("tas", "sa", "qld", "nsw", "nsw", "nt", "wa", "wa", "qld", "vic", "nsw", "vic", "qld", "qld", "sa", "tas", "sa", "nt", "wa", "vic",