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...all - if I specify GSM as the format, the resulting files are of size 0. I did not request mixing of the files or anything else. Any ideas why the monitoring fails? Cheers Robert Spielmann ----------------------------------------------------- TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399

...te_addr: No such host: phone1 What the..? I use Asterisk CVS-v1-0 at the moment. Oh by the way, some other AGI commands refuse to work: ANSWER HANGUP GET VARIABLE Cheers Robert Spielmann ----------------------------------------------------- TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399

...ke[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 I downloaded zaptel, libpri and asterisk. Any hints? Robert Spielmann ----------------------------------------------------- TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399

...t return value -1, if I put a channel name behind "HANGUP", I also get return value -1. Maybe I'm missing something, or is the AGI command parser broken? :-) Robert Spielmann ----------------------------------------------------- TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399

Hey, whenever I try to load, I get these errors Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to bind to 0.0.0.0 port 4569: Address already in use Jan 15 16:37:24 WARNING[7573]: loader.c:345 ast_load_resource: chan_iax2.so: load_module failed, returning -1 == Manager unregistered action IAXpeers == Unregistered channel type 'IAX2' Jan 15 16:37:24 WARNING[7573]:

...lways got a hangup and no further processing of the dialplan. Any hints? ( the call mustn't go into Nirvana if the AGI server isn't available!) Thanks for any help Robert Spielmann ----------------------------------------------------- TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399

Hello. I have just arrived to Asterisk. I would like to know if Asterisk can perform some functionalities I am looking for. I want to allow voip over sip to some users. All of them must have their own user name and password to login to Asterisk so only allowed users can login. All calls started by users have to be redirected to one account at our voip provider. I think those functionalities can

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

...m unable to know which of the john.s is being called, hence I cannot route the call correctly. Hope the question is clear enough ;) TIA, Robert -- Mit freundlichen Gr??en Robert Spielmann ----------------------------------------------------- TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399