similar to: Digium TDM400p troubles

Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

hallo all could anyone tell me how to get the * to send keepalive packets over a registration "trunk" or how to increase the amount I'm having natting issues, (the machine is siting behind 2 nat firewalls) thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML

I've setup * with TDM400P w/1 FXS, 3 FXO modules. I've one analog phone connected to TDM400P FXS module, 1 PSTN line to one of the FXO module(ZAP) , and 2 analog phones connected to Sipura 2000 (SIP). The calls between SIPs and zap phone (fxs) are OK. But 2 issues cannot be solved: 1. To dial to PSTN via zap phone, the setup in extensions.conf with the following exten =>

Hi, I have successfully configured an IAX trunk between 2 asterisks, calls can go through both ways without any problems, NAT in the middle of course (iptables) Now , leave them for a while , and make a call from the external server , it doesn't go through, Dial from the internal one, everything works fine again.. Now , it is clearly a problem in the NAT engine,

I have a question regarding to OS platform. As I see on Wiki -s homepage there are many type of linux version. And in some of them there are reported errors (regarding to asterisk ) for exemole in rad hat . Can you tell me what is the best linux paltform ,( version ), which is supported by digiroom card (T1 and TMD )and asterisk run on it stable ?. Which linux is prefereable ? for asterisk ?

I'm looking to add a single FXO port to my Asterisk box. It looks like my options are a Digium Wildcard X100P off eBay for $6.99, or a Wildcard TDM400P with an FXO Module from Digium for $125. Can anyone explain the tradeoffs (other than the ability to put 4 FXO/FSO modules on the TDM400P). What about RTC for the system - I know the TDM400P provides it. Does the X100P? Thanks!

I need to setup quota on //server/home for user a to 1GB and on //server/client (for all users on this share) to 10GB May you please let me know the exact command to be used for this? Also, do I need to add any other parameter in smb.conf in order to make quotas work? I have compiled samba with quotas option. Thank you. --------------------------------- Do you Yahoo!? Yahoo! Search presents

Hello, In an attempt to randomise the basic blocks in a function, is it possible that I can randomise the entry block as well? And maybe insert some instructions in the pass to call entry block while running the program ? Is it feasible? What does entry block consist of ? Thanks Tanu --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second

Hello, Is it possible to display the data structure graphs created by the Data Structure Analysis using a visualization tool? Thanks --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have setup a SAMBA + OPENLDAP server following Samba-3 By Example, but I have run into several problems. All the tests described in the Chapter 6 such as pdbedit -Lv, slapcat, and ldapsearch -x -b give the desired results. But, getent can't read passwd or group information from ldap backend. I have done my best to solve the problem, but it just doesn't work. I have compiled the nss_ldap

Dear All, . After installing, testing and like many other I found that Asterisk is reliable and a great Open source telephony solutions for a professional use. I would like to create a business with *, offering it as a IP PBX solution to customer, as a server, whatever with some Digium HW and SIP phone, .. etc. Asking for something that might have been asked. What are the implications for

I have been at this problem for almost a week on my own... I am running Samba 3.0.4 on a linux 2.4.26 kernel. I have a share that I want to be read only to the world and writable by members of the write list: [Content] path = /mnt/win/Content read only = yes quest ok = yes write list = sam I set a password for sam using smbpasswd. I believe all of the UNIX permissions are set correctly. I can

Tanu Sharma wrote: > Hello, > > In an attempt to randomise the basic blocks in a function, is it > possible that I can randomise the entry block as well? And maybe insert > some instructions in the pass to call entry block while running the > program ? > > Is it feasible? > > What does entry block consist of ? The entry block, by definition, is the first basic

Thanks a lot for replying. I have another query. If branching to the entry block is not legal in llvm how should I be able to create a new entry block for any existing list? I created a new block and inserted it into the present basic block list , but I get this error: opt: BasicBlock.cpp:83: virtual llvm::BasicBlock::~BasicBlock(): Assertion `getParent() == 0 && "BasicBlock

Hello All, I am looking for a stable calling card billing software for Asterisk & find that there are a lot of softwares freely available at http://www.voip-info.org/tiki-index.php?page=Asterisk%20Prepaid%20Applications Does anyone know which is the best of them all or has most features ? Any suggestions ? Suggestions for paid softwares are also welcomed preferably not

Bon soir, * People, You guys are fantastic. This list is hyper-helpful; thanks for that. I have an organization that needs to roll-out * for each of its 30+ locations across the country & we want all features in a DB where they can be easily toggled on & off, as needed. I have done some research in the usual places & got a bit of it, but not all of it, & am hoping someone else

Hi I was trying to use ChanSpy command, but it seems like it is not implemented, or is not included in the standard asterisk distribution. Can someone tell me how to obtain this. I am trying to monitor IAX channels, and I know now that ZapMonitor cmd doesnt work on this,so can anyone tell me how to monitor IAX channels. Thanks in advance --------------------------------- Do you

Hi all How do one "record" the tone that a pbx gives for hangup, so that one can use that in indications.conf? thanks --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am trying to configure * to work with SER (Sip Express Router), the configuration that I am trying is as follows. I have 2 windows machines running X-Lite soft phones The * registers with the SER, I want a call from one X-Lite to asterisk(after registration) which is to be forwarded to other X-Lite. For now, I am running both SER and * on the same machine (IP:10.232.2.249) with * on

I've a problem about the SIP phone registration. 1. with Linux 8.0, I can setup 2 or more Sipura-2000 boxes with no registration issue. 2. with Fedora Core 1 or Fedora Core 3, I've found that the extensions on 2nd Sipura-2000 cannot register well. This cause no dial tone from the phone connected to the 2nd Sipura-2000. Does anyone encountered the same problem? Please help on this