similar to: TM Malaysia E1 PRI signaling

Greetings from Hack in The Box -- We are pleased to announce that the Call for Paper (CfP) for HITBSecConf2006 - Malaysia is now open! Set to take place from September 18th - 21st 2006 at The Westin Kuala Lumpur, this years conference promises to once again deliver an International deep-knowledge security conference. HITBSecConf has been described as "the most intimate of hacker

Hi all, I face a strange problem. I'm in France using Completel as operator for the E1 line. I move a client from ccm to Asterisk keeping the 2811 gateway. Set up is complete, outgoing and incoming calls are sended to the 2811. The problem is that in 90% of the time I get a 500server error back from 2811 when placing a call. Incoming calls are OK. The 2811 was configured in primary-4ess

We have 3 existing switches interconnected via dpnss, we need to integrate asterisk with these switches via a dpnss link. Any suggestions? also does anyone have a link to the differences between isdn30 and dpnss. Thanks in advance Bails

Greetings from sunny Malaysia! This is a reminder that the Call for Papers for the upcoming HITBSecConf2007 - Malaysia is closing on the 1st of May. HITBSecConf2007 - Malaysia is set to take place from the 3rd till the 6th of September in Kuala Lumpur. Our event last year attracted over 600 attendees from all corners of the globe and this year we are expecting this number to grow to well over

Hello, I have an E100P, and in the zaptel.conf I have: span=1,1,0,ccs,hdb4,crc4,yellow fxsks=1-10 the light on the card is green( BTW what do all those states of the card that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or for the card?) in the asterisks` zapata.conf I have: [channels] context=default switchtype=euroisdn signalling=fxs_ks usecallerid=yes hidecallerid=no

hallo everyone, fyi ... working SIP Trunk configuration between Asterisk and Mitel 3300 ICP attached. let's refine further, please test and share your feedback, regards, Joseph Okoegwale Abuja, Nigeria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070709/4f7f15e1/attachment.htm --------------

Hello I am creating a linear model with the command net5 = lm( X[,1] ~ PrinComp[,1:5]) where my vector PrinComp looks like this > head(PrinComp[,1:5]) PC1 PC2 PC3 PC4 PC5 [1,] 1.8626055 -3.34190998 -0.5448889 2.8296751 0.3994096 [2,] 3.1124144 -1.68113572 1.7187314 -2.0162583 -0.2935675 [3,] 3.3538049 -0.05471002 -2.9385065 0.6921495 -2.2743761 [4,]

A not so technical friend in India is shopping for a laptop. He often travels and stays months in Malaysia and so needs to be able to use the laptop there as well. He typically connects to the internet via wifi, but sometimes must use a telephone line (yes, with a modem). And of course there will be times when he has to plug into mains power to recharge the battery. So to be able to fully use

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

hi All I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI. 1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the Jumper. 2. I Want To seeting E1 in ASterisk/PC Back To Back To Cisco E1 AS5300 Use ISDN Signaling, my configutration :

Hi all, Our company has deployed a Mitel 3300 system (only about 2.5 years ago) and we are experimenting with setting up Asterisk in our head office (for business continuity, ie we have a bird flu epidemic and no-one can come in, therefore use SIP softphones at home to co-ordinate activity) and at a remote site in the Isle of Man (connected via 2Mbps SDSL) Ideally we'd like anyone on either

We are setting up an office in Malaysia. We contacted Telekom Malaysia and are surprised to be told that ISDN-30 is no longer available. They are yet to give us information of the replacement technology. Does anyone have any experience and information with this? Thanks in advance.

Hey, We are deploying an Asterisk machine in Malaysia for a client. The machine has a TDM400P with 2 x FXO & 2 x FXS. I'm wondering if there are any Malaysian users out there who might be able to help me out with a running zap config for their phone system. Cheers, Darryl.

Merci pour la transmission de cet appel ?? tous ceux et celles qui pourraient y ??tre int??ress??s Conf??rence Internationale HMI 2013 Nous avons le plaisir de vous annoncer que l'unit?? de recherche SETIT et son partenaire l'ASDF viennent de conclure un accord. Cet accord pr??voit l'octroi d'une bourse d'un montant de 150$ et l'h??bergement gratuit (5 nuit??es) ?? tous

Hi I am trying to connect asterisk with a Cisco GW 2600 with E1 pri using a Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02), Errors: [Jan 28 17:32:33] VERBOSE[6182] logger.c: == Primary D-Channel on span 1 up [Jan 28 17:32:33] WARNING[6182] chan_dahdi.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jan 28 17:32:34] NOTICE[6182]

Hi, Asterisk 1.4.17 Sangoma a102DE I'm having some issues sending CallerID Name to a Dialogic based phone app. According to the pri debug (asterisk2a-pri-debug.txt in [3]) you can see that it is sending the CallerID Name "Mike - Budgetone - reachme.com" to the Dialogic card, but it isn't regestering on the Dialogic based system. I can receive CallerID Names from our Paetec,

Greetings, I read from archive that OCFS is for kernel 2.6. which means RHAS4.0 is supported. However, I can't find the download for this version. Is the support official and where can I download it. Thanks / regards, Ivan Wong Database Administrator e2Open Inc. (www.e2open.com) Suite 34.03, Level 34, Menara Citibank, 156, Jalan Ampang, 50450 Kuala Lumpur, Malaysia DID: +603 2776 6397

As requested by Ralph in my introduction e-mail to this list [1], I've gone ahead and done some sample test timings for encoding between libtheora-1.0 and libtheora-1.1beta3, as available from the Theora website [2]. So far, I've tested with 4 different files that I generated with content I already have. The files, in brief, are the following: 1. planet-earth-360x240.yuv4mpeg - A

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently