similar to: Asterisk 1.6, B410P and TE/PtmP mode. Could you get it running ?

Hi, I've read that the new dahdi channel supporting B410P doesn't support PtmP when in NT mode. Does it support PtmP when in TE mode ? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081209/f422b5d0/attachment.htm

Hi all, So, we reverted the LibPRI version and tested it, and then tried with the latest version of everything. Still no changes. The BRI line is in PTMP. If we set the configs to PTMP in the genconf_parameters and try it, we get the following: [Jan 21 17:32:20] ERROR[20341]: chan_dahdi.c:12645 dahdi_pri_error: Unable to receive TEI from network! If we set it to PTP (which it is not) we

Hi thanks to everybody who has been assisting me in solving the various problems I had to dial out from Asterisk to a PSTN number with SIP using Nikotel's VoIP service. I have drafted a mini-how-to which is available at http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf This is a first draft, I will amend this further, in particular the "verify and debug" section

Has anyone had any experience using a VOIP provider in Japan? No matter what I try, my REGISTER string kicks back one of 2 errors: Got SIP response 481 "Call/Transaction Does Not Exist" back from x.x.x.x or Got SIP response 400 "Bad Request" back from x.x.x.x My register string is as follows: 05075034132@ipphone2.freebit.ne.jp I have tried the following also:

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk

Hello, Is it possible, for testing, to connect an cat5 straight patch cord between 2 ports of a Digium B410P card and use these 2 ports as a normal dahdi trunk ? I've tried this: One port is set as NT, the other as TE. I would expect timing to come for system hardware so I choose in /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami Results: - both ports lights are green -

asterisk-users at lists.digium.com has now been added to the filters white list! Anyway, 100ohm termination isn't required for ptp - but is required for ptmp. I know the DAHDI package(s) no longer include make b410p - hence the reason it is included in the docs.

Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC

Apart from you were dialling out on your inbound context and vice-versa. The best advice I can give now is to change to mISDN - as this is proven to work with v1.4 and v1.6. Actually - have you tried putting the 100ohm termination on for your NT port? I need to do that with mISDN as it only allows ptmp for ISDN extensions. Cheers Andy -------------- next part

Hi, Until now I've been using my Digium B410P cards with misdn 1.0.x. I would like to upgrade my systems and am now wondering which is the "best" route to take: - use the latest release of misdn v1 - upgrade to the latest "stable" kernel and use the built-in misdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn

The DAHDI docs actually include the 'steps' that used to be in Zaptel's 'make b410p'. These steps involve downloading and compiling mISDN. Why re-invent the wheel? Just a thought :-). As for the 100ohm termination bit - it's simply changing a couple of dip switches on the b410p card (as described in the manual). Andy -----Original Message----- From:

Hi, As you may know, these ISDN BRI features are very important here in Europe as ISDN Basic Rate Access is very popular among Small & Medium Entreprises. I don't really know why but it seems that in many countries, default is to install small PBX using Point-to-Multipoint (PtMP) mode as opposed to Point-to-Point (PtP) which is the norm for PRI. So basically, in several countries, SME

Hi, Reading http://www.asteriskguru.com/tutorials/bri.html , it seems PtP is the way to connect businesses but if you read http://public.swbell.net/ISDN/connect.html you would think the opposite. Can anyone elaborate a bit PtP or PtmP respective advantages ? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. When I call my RNIS numbers (with a mobile phone for example), I can see 2 incoming calls on the IPBX, which should not happend. I'm not sure if it's a problem with the telco France Telecom and their ISDN setup, or if it's a problem with the MISDN driver on the IPBX itself. I'm stuck ... Any advices for troubleshooting that? Someone provide working configuration files

Hi Olivier It seems wrong configuration, because according to your mail Asterisk it will be acting as terminal mode (ie Patton gateway acting as network and asterisk as terminal). But Asterisk message is indicated Asterisk a s a NT mode [ 212.226050] wctdm24xxp 0000:0a:01.0: xhfc: Configuring port 0 span 1 in NT mode with termination resistance ENABLED. It could help by checking parameters on

Hi! > How to do this ?? > To proceed with your answer on PICKUPMARK, where do I put this ??? Look at the example for Asterisk 1.4 on this page: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup Philipp

Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK,

Hi, I need to rebuild a system which has 4 BRI ports and is connected to Point-to-multiPoint lines, in a country where telco often "drop lines for energy savings". I'm planning to use latest 11.4.0 asterisk version along with dahdi and libpri (no misdn). Which version are recommended for Dahdi and Libpri ? My main requirements are, beside having calls coming in and out: - keep a

Hi, After an upgrade, I discovered yesterday strange things I would like to share here. Basically, I'me comparing platforms: The first one is a 2.6.26 (Debian Lenny) platform, with Asterisk 1.6.1.18, Libpri 1.4.10.2, Dahdi revision 8853 (must be between 2.3 and 2.5, I think). The second one is a 2.6.32 (Debian Squeeze) platform, with Asterisk 10.5.1, Libpri 1.4.12, Dahdi 2.6.1. Both are

Hi, a partner, who exchanges voip traffic with my asterisk box, complains, that asterisk ignores hints about ports to use. Hints about ports to use, seem to be a feature of H323. (I'm not firm enough with H323 to verify this.) The remote party opens the media-in channel: remote-ip:port-A -> local-ip:port-B My local Asterisk-box uses the same channel for media-out: local-ip:port-B ->