similar to: Can asterisk PRI/BRI support redirect calls

Hi I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI Already installed). according to https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Asterisk 1.8 include this application but I cannot see it with "core show applications" Do I need to install mISDN or other modules for using that ? Regards M.Shirazi

The Asterisk Development Team has announced the release of Asterisk 1.8.13.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.13.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hi The only way to have CD service is using: DAHDISendCallreroutingFacility(<destination-5551212>, <original-my-number>, cfu|cfb|cfnr|unknown) or it is possible also with Dial() command before answering the call also ? Regards Babak -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. Calling from a Tenovis phone to a SIP phone (i.e. traditional phone -> Tenovis PBX -> QSIG -> Asterisk -> SIP phone) works with the following messages: --- Don't know what to do if second ROSE component is of

Dear Lists, Here are my latest posts on retrenching and locating facilities better. Do you have anything in academic papers that use data mining- analytics for 1) Better Employee ROI quantitative measurement or cutting costs rationally based on data rather across the board. 2) Locating facilities based on pure quant based approaches than local supply demand conditions. Specific application for

can some one please how to transform this into TCNG code? tc filter add dev ${DEVICE} parent 1: protocol all u32 \ match u16 0x8864 0xFFFF at -2 flowid 1:${ID} 8864 is the PPP session ethernet protocol the above code is used to shape all PPPoE traffic on the same device so this way i can make use of the borrowing facilty which is not given by PPPoE. PPPoE gives me each time a device =>

On Tue, Jan 15, 2002 at 06:00:50PM -0000, John Bowman wrote: > > Date: Tue, 15 Jan 2002 17:29:44 +0100 > > From: Markus Friedl <markus at openbsd.org> > > Cc: openssh at openbsd.org > > Content-Type: text/plain; charset=us-ascii > > Content-Disposition: inline > > User-Agent: Mutt/1.3.25i > > > > On Tue, Jan 15, 2002 at 03:46:15PM -0000, John

First off, I'd like to give all the people involved with the development of Samba a big thanks. I'd have to say that Samba is probably the single most important piece of software that we run in our research lab. I'd also like to say that I have had so few problems that I probably haven't written about one in 3 years. With that being said, I'm having problems with my Samba

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

That was my impression as well, I tried adding switchtype=qsig in zapata.conf, but all I see is the capi information... No connection... Janne > -----Ursprungligt meddelande----- > Fr?n: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] F?r Peter Svensson > Skickat: den 23 februari 2005 21:34 > Till: Asterisk Users Mailing List -

I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make calls from the Toshiba to Asterisk and internal calls from Asterisk to the Toshiba. What I can't do is make an call with an outside destination from Asterisk to the Toshiba. The Toshiba is looking for 9 to grab an outside line then it expects to see the 10 digits. In the FreePBX dial plan I use 9|. which sends 9 plus the 10

I am looking to get the info about QSIG support in Asterisk. Does Asterisk have QSIG support? Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking? If so, How to configure that? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060503/3cb7f966/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Please cc in replies as I am not subscrubed to the list - However I have just searched backwards over the list archive looking for an answer and I can't find any posts relating to this. I am currently backing up my family's windows machine over the network to my linux box by mounting the windows disk as a file share an using cp to copy

[Reposting, as was bounced for non-member, sorry if this is a dupe] Arrangement: { PSTN }--E1--[PBX]--E1--[*]--LAN--[SIP phones] \__[PBX system phones] Normal calls between PBX system phones and SIP phones work, in both directions. The call logs look like (ignore the no answer, it did ring):

Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk---> Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send

Hello, If I connect an Asterisk 1.6 to a PBX via Q.SIG and A (on the PBX) calls B (a SIP phone on Asterisk). B answers and puts A on hold. Then B calls C (on the PABX) and does an attended transfer. Is it possible to configure Asterisk so that it does the Q.SIG ?Path Replacement Feature? ? The Q.SIG "Path Replacement Feature" requires the following: After both legs of the

I installed ntop on a centos5.2 system (which got all yum updates a few days ago)... # rpm -q ntop rrdtool ntop-3.3.8-1.el5.rf rrdtool-1.2.28-1.el5.rf went through the password configuration when I start it, its erroring. # service ntop start Starting ntop: Processing file /etc/ntop.conf for parameters... Mon Nov 24 23:56:36 2008 NOTE: Interface merge enabled by default Mon

Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined

I have read most of Eicons information on Q.SIG, and I am able to load the Q.SIG protocol (instead of ETSI for example). No strange logging in divacrtl mlog. But how do I tell Asterisk to understand Q.SIG? My PBX is configured for QSIG, but I cannot see anything on my trace when trying to make a call via the S0(Q.SIG) Janne > -----Ursprungligt meddelande----- > Fr?n:

Hi, Is is possible to implement this: Hicom150 --- BRI (QSIG) ---- Asterisk I've been reading Siemens documentation and they say: "Digital nailed connections Corporate communication networks can be implemented over digital S0 or S2M nailed connections between several Hicom systems using the CorNet N protocol and between Hicom and non-Siemens systems using the QSig protocol. The