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Hi my Digium FXO card isn't picking up the callerid I get from the PSTN. I have verified with a deskphone that can display the callerid that the facility works. So, it's definitely the FXO card not picking it up. As I am in Japan, I guess that NTT uses a different method to provide the callerid and so I guess that it is just a matter of configuring the FXO card so that it uses the

Hi A few days ago, Kelly remarked that he had previously observed that Nikotel charged him for calls he did not actually complete. I have made a number of test calls to my landline without picking up the calls. I just let it ring once and hung up on the calling phone. A look at the call records on MyNikotel reveals that I was charged six seconds for every of these calls. I have raised a

Hi since "callerid=" in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? thanks regards bk

Hi I thought I share this one, just in case this is an indication of some bug ... When I was trying to use music on hold at first, I didn't bother to copy any music into /var/lib/asterisk/mohmp3 since there was a sample- hold.mp3 in there which played just fine in a standalone MP3 player. But after uncommenting one of the lines in musiconhold.conf and doing reload on the console, there

Hi on the download site for asterisk, there is one directory called webmin and I presume that this contains a webmin module for Asterisk, yet this doesn't seem to be mentioned anywhere. Is this unfinished work in progress? typically webmin modules are in form of .wbm packages that can be installed using webmin itself, but I couldn't find any such module in the webmin directory nor

Hi thanks to everybody who has been assisting me in solving the various problems I had to dial out from Asterisk to a PSTN number with SIP using Nikotel's VoIP service. I have drafted a mini-how-to which is available at http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf This is a first draft, I will amend this further, in particular the "verify and debug" section

Hi I have just been struggling for four days to get SIP working and now as I created a voicemail box, Asterisk has become very unstable and it can't bridge SIP phone to SIP provider calls anymore. Calling internally from one SIP phone to another works fine. Calling internally from a SIP phone to an analog phone on a Zap channel and vice versa works fine. Incoming PSTN calls delivered to

Hi I have defined my SIP phones like this ... [Sip1] username=gs1 callerid= "Full name" <1001> etc etc Now, when I do this in a given extension exten => nnnn,1,NoOp(${CALLERIDNUM}) then I get "<gs1>" as callerid and not "<1001>" as defined with callerid= Sure, I could set the usernames to their respective extensions, but I don't want

Hi please excuse if this seems obvious, but I am new to this and the SIP section in the Asterisk handbook do not give any clues nor do the SIP examples in there seem to represent real-world situations. I am using Nikotel as a VoIP provider (for now) and I would like to configure Asterisk to sign on with Nikotel so that I can use the telephones connected to Asterisk to make calls using the

Hi in order to keep the dial tone after pressing 9 for 'outside line' I have this in my extensions.conf [localpstn] ignorepat => 9 exten => _9[123456789]XXXXXXX,1,Dial,${PSTN}/${EXTEN:1} exten => _9[123456789]XXXXXXX,2,Congestion this is properly included in the handsets' context but the dial tone disappears after pressing 9. am I missing something? thanks in advance

Hi thanks to everybody who responded to my earlier post. I have looked at all the material and links provided and tried everything in there, but it simply won't work for me. My SIP phones register with Asterisk, but they cannot be called (everybody is busy at this time) nor can they call anything (error code 4, whatever that means) not even internal (yes I did give them appropriate

Hi is there any ritual sacrifice a newbie has to perform to be welcome on this list? I am new to this whole PBX thing in general and Asterisk in particular. I had hoped that the community on this list would welcome a newbie like myself and help me with some answers to my stupid questions, but somehow it seems to me that nobody likes to respond to somebody who appears to be a complete

Hi I followed the instructions on the Asterisk website for download and building Asterisk. I checked out a fresh copy from the CVS tree as described and that went smooth, but when I try to build as described, I get a truckload of errors and I have absolutely no clue what this all means. Can anybody please give me some hints or perhaps provide a link to a pre-compiled version? thanks in