similar to: Settings for SIP to dial PSTN with TDM400P w/FXO module

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

Hi all I installed a tdm400p into a old p2 machine. I'm not able to see it under /proc/interupts or using lspci.. we removed all other cards. changed slots, forced irq to that slot.. etc etc. what is the min specs needed to get one of these cards running? thanks L --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'

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:

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

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

I had a similar issue with sending a flash to the PSTN for call waiting. I found that my dial plan in the extensions.conf file was not allowing me to dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78, *79, etc. Training the wife on how to actually use it was an entirely different issue. I have not tried to enable 3-way calling via the PSTN. "Jeff Otterson"

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

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

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

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 ?

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

"My Documents" folder doesn't sync with the server unless users log off from the machine and log back in. Is there a way to sync the "My Documents" folder during the session? Thank you. --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'