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Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Colleagues- I am working on the design of a fairly large samba deployment, and I am looking for feedback on some of my design ideas. I have 10 buildings spread out in and around a city, all interconnected via 1.5Mb leased lines. There are samba servers in each building. I have some users that move from building to building. We are using primarily windows 98 desktops, with a few 2K and XPP

I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what

Hi All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian

Hi all I have asked this question before but have not got any helping input. I'm really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand Brands is not used, Cards just makes the cards. Routed in the dialplan and pricelist, Trunks is for ASTCC to

Hi all, I want to make a Samba Sever as amember of a Domain, using a Windows NT 4.0 as PDC. I have my smb.conf configured to do this, but users ca't connect on Samba useing the Windows NT account. Is it possible or I have to create them on a smbpasswd to make them login on it? I have a Samba Server 3.0.7 running under a Slackware Linux 10.0, with Kernel 2.6.8.1. I'm sending my smb.conf

Hi All I seem to have a small problem with the music on hold button on SJPhone. I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS. On the rapid dist when I press the music on hold button on my SJPhone I get music on hold. When I do the same I get no music on hold just silence. I create extension like this exten => 1111,1,MusicOnHold(Default),

I need some help with getting the following to work SipPhone <------> Asterisk <------> H323 GK (quintum) And H323Phone <------> Asterisk <------> H323 GK (quintum) I have tried to run the Asterisk from the newest ports and could after some digging around in the configs register the SipPone to Asterisk and Asterisk to the H323 GK. But when I try to make a call from

Hi All, I'm new to * I wonder if anyone have an idea how to make the following with ASTCC. I would like to have all SIP phones to work on prepaid basis and without need to dial any access number, instead I would like to use the phone as normal dialing only the destination number, for example 00464090510. Then I would like to authenticate the call using the SIP Username and Secret check

Hello all, I'm a little stuck in getting astcc working. I went to configure, set up the DB, Created Tables, but what now? What does the Users_Configure, Brands, Cards,Trunks, Routes, Sip Friends etc... do? How do I configure Users (In the Users_configure I get Not Configured) How do I configure Cards and assign a card to a user? Is there a tutorial on this on the web? I have searched

After upgrading to Samba 3.0.7 via apt-get I get an error when I try to copy files from the share from a windows server The error is Can not copy xxxxxx.pdf, error performing inpage operation If I try to open the file then adobe lounches but there is nothing inside. Can anyone tell me what could be wrong? ---

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing