similar to: asterisk-users Digest, Vol 105, Issue 39

Hello, I'm facing a rights issue on with Asterisk 11.3.0 running on CentOS release 5.8. Asterisk process is running with asterisk since it is define in asterisk.conf as following: runuser = asterisk rungroup = asterisk You can see asterisk proccess here: ps aux |egrep 'python|asterisk' root 11581 0.0 0.1 65940 600 ? S Apr17 0:00 /bin/sh /usr/sbin/safe_asterisk

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, 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

We are connecting an asterisk box to a Nortel Option 61 via a T1 with PRI. We have hit a problem we cannot overcome; specifically, the Nortel is asking for the ROSE information element (IE) over the PRI connection. This causes libpri to drop the connection, with cause 100. The Nortel cannot "turn off" asking for ROSE. Since our Nortel box does not support QSIG (too old), we have to

Hi everyone, I was referred to this mailing list by Richard Mudgett regarding the following thread on Issue Tracker as it's a feature request and not a bug:https://issues.asterisk.org/jira/browse/PRI-170 We've got a Nortel BCM (actually 2 BCM50s and 1 BCM400)... all showing the same problem. When a call comes in the number shows up OK but the name does not. In looking at the ISDN messages

[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):

Guys, I need to make asterisk connect to another device over a pri using QSIG signalling (master/slave). Is this feasible and has it been done before? Thank you! al. _________________________________________________________________ Get ready for school! Find articles, homework help and more in the Back to School Guide! http://special.msn.com/network/04backtoschool.armx

Hi, 1. From chan_dahdi.conf.sample (asterisk 1.6.1.18) : ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: AT&T 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN (common in Europe) ; ni1: Old National ISDN 1 ; qsig: Q.SIG ; ;switchtype=euroisdn At the same time, dahdi_genconf

Hi, Thanks for your reply. I have gone through the tutorial how to write llvm pass .. and but not able to figure out how should i proceed for this pass.. Can u please tell me some starting point for this.. Let say i have some add function.. define <4xfloat> add(<4xfloat>% in1, <4xfloat>% in2) { entry: %0 = fadd <4xfloat> %in1, %in2 ret <4xfloat> %0 }

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

Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27

Woops, forgot to reply to the list too. > Hi, > > Have you read this page? http://llvm.org/docs/DebuggingJITedCode.html > > It contains pretty much what you're asking - it uses a command-line > switch to lli as opposed to the C++ interface, but if you're using > llvm::cl for options processing you get that flag automatically (try > ./your_tool --help | grep

Hi all, We we send a server out to a customer with a PRI card I basically tell the PBX person that we want national, NI2, b8zs, D channel is on 24, and we are the CPE and your the NET and us a cross over cable. Allow extensions, local calls and long distance. on the surface I would have thought this would be enough for them to connect the digium card to the PBX and get calls working. The PBX

Hi, Can anyone please tell me how can I scalarize or de-vectorize the llvm vector ir. In this (http://old.nabble.com/Re%3A-Thoughts-about-the-llvm-architecture---p2961720 3.html) thread I found LegalizeTypes will do this while generating machine code from llvm ir.. How do I convert llvm ir to selection Dag. And scalarize the vector ir and again get back llvm ir. Thanks &

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

Hi All, Our experience with * to date has been a bit limited. It's a 4xCisco 7960 network, linking our head office with a faraday caged datacenter. As a way of putting voicecomms into a sealed room, it was quick and easy to deploy, and works very well. As typically happens, we've now thought about extending the use of asterisk - and a new opportunity has cropped up. In three months

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

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

Hi, I know using llc i can print machine code (-print-machineinstrs). How can i print machine code using llvm c++ api.. i looked into llc code but not able to find it.. Thanks & Regards, Pachauri