similar to: FXO modules for TDM400P?

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

I got the devkit tdm400P(tdm10b) + x100p bundle. Where does one buy more ports for the tdm400p? I couldnt find anywhere on digium site or resellers to buy them? -Dan

Semi-offtopic, Anyone know of voip providers who can provide tollfree number service? E.g. route 800-xxx numbers to our * ? Even better if they are familiar with * or can speak IAX ... -Dan

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

Just a warning to dovecot listmembers. The list is being harvested. test3943395 is a unique address I created only for communication to the dovecot list. The following spam came from: Received: from dial-dynamic-62-69-52-187.surfdial.murphx.net (dial-dynamic-62-69-52-187.surfdial.murphx.net [62.69.52.187]) by sasami.anime.net (8.11.6/8.11.6) with SMTP id j7FLxtv03775 for

Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

I am sure that quite a lot of people would like to have Caller ID working with their X100P and TDM400P cards outside of USA. Judging from previous threads this is just a matter of implementing this support in the software driver! So, I was thinking, if we get together and put few $(USA DOLLARS) into a basket, we could then ask Digium to actually properly implement Caller ID for non USA

asterisk-dev-bounces@lists.digium.com wrote: > I have a DSP based system that is working on a four port FXS system > using a 200MHz arm processor. Well.. since we are talking about this topic I owe you guys notes of my experience with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch etc.). We made a Linux distro and compacted it into 32MB flash. Installed asterisk and

Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with "1" in order To successfully make a call to other USA destinations? ---- I have not been to USA (yet) :) Ta SJ

Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

Hi I have been playing with trying to get voicemail.conf from DB, I am using cvs-head, but when I start asterisk, it dies a horrible death, because it cant load any voicemil setting. I looked at my mysql logs to see what query was being sent, and I get SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY

Hi , I have an asterix (1.2.7.1) box with 3 TDM400P and 10 FXO modules, and zaptel (1.2.5). The problem is that 6 modules can't be initaized and i get the folowing error: May 7 09:17:37 WARNING[7739] chan_zap.c: Unable to specify channel 9: No such device May 7 09:17:37 ERROR[7739] chan_zap.c: Unable to open channel 9: No such device here = 0, tmp->channel = 9, channel = 9 ztcfg output

Hi all, Let me be the first to wish everyone, especially the Digium team, an awesome year in 2004.. Later..

Anyone have one of these yet? http://www.voxilla.com/shop/index.php?action=item&id=38 -Dan

hi all anyone that've heard of any working IAX hardphones yet? roy

I have some code working that lists only items from a particular user. The code in my list action finds the user and then conditionally lists only his/her items: def list user = User.find(session[:user]) user_id = user.id @product_pages, @products = paginate :products, :per_page => 10, :conditions =>[''user_id = ?'', user.id]

Hi All, This is the first time I'm submitting a patch, and I hope it fixes more than it breaks. I'm putting it here, since John Todd mentioned a while ago about the heavy load Mark and crew have at Digium (doing such good work), so I thought all of us could test this first, and if ok submit for inclusion in CVS later if appropriate. This is an extension to work done earlier (sorry I