similar to: How does TDM work?

Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig

This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because you current number it is not being answered, and you don't want to hangup before dialling again. /Obelix

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because the current number it is not being answered, and you don't want to hangup before dialling another number. /Obelix

Hi All, Please bear my ignorance but what is TDMoE used for? Illustrations with practical applications, scenarios or set ups will be most appreciated. Thanks Luan

If I have a system with 1 machine to handle incoming H.323 calls and then multiple machines to distribute them to T1 ports over TDMoE, where does the codec translation take place? Does it take place in the master system or does it take place in each of the slave TDMoE systems? Also, any idea how many concurrent G.729 calls a system like this would handle?

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

Is there a key sequence to stop a call as its ringing, before the call is answered? The * key stops a call after it is answered, but I'd like a way to cancel the call during the ringing phase. /Obelix ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

Which Actions and events to the read/write options in manager.conf give access to, ie the options below. read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Are they documented somewhere? /Obelix

I'm just experimenting with dynamic spans using TDMoE, which looks ideal for clustering Asterisk boxes with low latency (i.e. 1ms for TDMoE compared with 20ms for IAX). When using normal E1s and T1s, I've always used ISDN signalling: either EuroISDN or NI2. So naturally I tried that over the TDMoE. I've found that I get a lot of HDLC errors and the D-channel is up and down like a

Hi all, Information on this topic seems a little scarce, so I thought I'd try the list.... Apart from the the coolness factor can anyone explain to me in what situation one would use TDMoE rather than IAX for communication betwwen 2 Asterisk servers?

Hello all, recently i stumbled upon the I-TDM standard, e.g. see here http://www.picmg.org/v2internal/news2005.htm "SFP.1, also known as I-TDM (Internal TDM), is a companion protocol specification to SFP.0 that is optimized for TDM traffic over high-speed fabrics such as 1 and 10 Gigabit Ethernet (PICMG 3.1), Advanced Switching (PICMG 3.4), Infiniband (PICMG 3.2), etc. SFP.1 and SFP.0

Just a Question. I would like to know if TDMoE follows specifiaciones of TDMoIP RAD protocol that says that there is a compression of 16/1 when you do TDMoIP. Manuel Marin Garcia TRANSTELCO S.A. DE C.V. Campos Eliseos 9050 B4 – Cd. Juárez, Chih. 32452 - México Oficina: +52 656 692 11 09 – Fax: +52 656 692 1112 - Celular: 915 727 6141 http://www.transtelco.com.mx

> Hi folks, > My SMB server is a Helios Ethershare 3.1 on a Tru64 Alpha machine. > We are now integrating more and more gentoo-Linux Servers. These are the > SMB Clients. > > On Obelix (one of our gentoo-Servers) I can mount the SMB Volume from the > Ethershare Server. > obelix nc4smb # whoami > root > obelix # mount -t smbfs -o username=m.heckmann

I'm scoping out HA for a relatively simple Office/Call Center PBX. Current setup uses a TE412P with 4 PRI our telco with SIP hard/soft phones for users. Some outbound also goes to a SIP provider. Active/Active looks to be too much hassle for an installation this size, so we're looking at adding an extra * in an active/passive configuration with Linux-HA in between them. Does anyone

I have been trying to connect via sip and things don't seem to work. What do messages like this mean? Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834 (len 361) to 216.127.66.119 returned -1: Invalid argument Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 000638cf3adb579455c0d20b2051ba1d@127.0.0.1 for seqno 102

Hi! I have some problems with the download of Asterisk-addons. I try to follow instructions that I found in http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql , but nothing to do. This is my shell: [root@obelix root]# cd /usr/src [root@obelix src]# export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot [root@obelix src]# cvs login Logging in to

Hello all, I have no problems setting up trunk groups in general, but is there a way to set up a trunk group for outbound calls that includes channels on multiple servers? I might have missed something somewhere, but I couldn't find any reading about this topic. Thanks! Sean

Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load

We have been pretty much able to solve our echo problems, except for the primary mode in which we desire to operate our system. See system diagram at bottom. Prior to making adjustments to cancel echos (all echocancel=no): Call Type Result (Before) --------- --------------- CP <- LEC PRI * TDMoE * FXO -> AP