Thank you for your help, when i finish the work on my first asterisk-costumer i like to donate to you. EL Asterisk version sail I can get the card visible in the PCI Probe under both, but no trunk lines are created.. Suggest you go up to As an aside, the tdm is quite different to the tdm and uses different drivers as you'd guessed.
The TDM is it's 4 port equivalent. What I have found out so far. The problem however is that, that is where it stops. Trying to narrow down the problem - it seems that modprobe is not loading the driver at startup If I run modprobe wctdm24xxp - it see the card If I do not run this - the card is not recognised.
The problem is that when you commit your changes - the wctdm24xxp is unloaded. As such the is no info in the card info screen I have run the latest yum updates and sail is still only on How would I go about taking it to like you suggest. Much appreciated. Here is the situation currently update to Sail to 2. Loaded wctdm24xxp Does anyone out there have any help??
Janeth iyv4nt4h at outlook dot com 09 October Placa eu ne3o achei, realmente essas sf3 etexsim em pree7os acima de 1 mil. Bem, obrigada pelo elogio, o trabalho ne3o foi sf3 meu, uma pessoa e9 a renie3o de muitas coisas, de sua propria personalidade e iclusive de amigos como vocea.
Um grande abrae7o. Ah vou clicar ali para vc. Thank youGilles. Can you teach me how to solve the "2 channels to configure" issue? Ive done all the steps but still I did not resolve it. I am using TDMP just like above. And why is it that theres no documentation with regards to Zap configuration? I always find how to configure Iax and Sip in the internet..
I did. One sign of that is that "core show channeltypes" will not list a Zap type. I am using TDM11B card. Thanks in advance souvik. Ashwini ashwini dot pict at gmail dot com 30 January Connecting two asterisk servers in same network We have two asterisk servers in a same LAN , we can able to make calls using both the asterisk servers individually.
But we need to communicate between two asterisk servers and to make call to the number exist in the another server. I have searched online for several times and tried the options given , but nothing seems to be working. Please guide me in this issue. Marcelo marcelodanza at gmail dot com 04 January I would like to recive tutorials. Thanks for share!! Please Help me!!!!!!!!!!!!!!!!!!!!!! Or you can also re-read them one by one, by running sip reload to restart SIP settings and dialplan reload to restart the dialing scheme.
If you execute sip show peers , you will see the list of available users on the server, which of them are registered now and what their IPs are. At this stage, your users in Moscow are already able to call each other, but you also have users on the Saint-Petersburg server they should be able to join as well.
To allow this, you have to set up a connection between your two Asterisk servers. Accordingly, this server will register on the Moscow server, Now the two servers are connected; the dialing scheme must be defined to allow Moscow users to make calls to Saint-Petersburg and vice versa. Restart the SIP settings and the dialing scheme. Now all our users in Moscow as well as in Saint-Petersburg both servers running Calculate Directory Server with Asterisk can join each other.
OK, now your users in Moscow can call each other, can call their colleagues in Saint-Petersburg, but they still cannot call standard numbers. You must configure your interface card to allow them to. Now configure channels on your card. Your card has a module trunk including 4 FXO ports, with channels numbered from 5 to 8 if you look on your card, these will be 4 left ports.
Channels are numbered from right to left; consequently, the eighth channel is at the extreme left and the first channel is at the extreme right. An FXO Foreign eXchange Office port receives the line from the exchange; that is, it acts as a terminal device a phone set and uses fxsks signaling. An FXS Foreign eXchange Station port generates the ready signal and the line voltage; that is, it acts as an exchange and uses fxoks signaling.
Thus the ports have names which correspond to the nature of their connections, and use signaling that is appropriate in their case. Now you have to configure channels right on the Asterisk server. The first context defines how calls are handled for channel 7; the second context does the same for channel 8. The main line is on channel 7, while channel 8 is used for faxing. This will permit you to make calls through the first available line.
You can now configure the dialing scheme for receiving calls on your lines. Here you are answering the call and sending it to the relevant queue. Queue welcome,n,,,12 send the call to the queue welcome for 12 seconds; the n parameter allows to dial while the call is in the queue and the Music On Hold MOH is playing. In the welcome queue, on the other had, the greeting message will be played, as we will describe later. All phone sets will thus ring when the majority of staff are not in office, and in working hours the normal order is respected.
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