search for: militz

...to remove an option to strip the first digit of incoming calls and found nothing. The wiki and the mailinglist archives can't enlight me either, why asterisk behaves like this, or how I can turn it off. So if someone could give me a hint, I would be very delighted! Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller L?tticher Stra?e 10 Tel 0241/701333-11 km@westend.com D-52064 Aachen Fax 0241/911879

...I ran asterisk 1.2.10 on the machines and then updated to 1.2.11. I looked through the Changelog but coulnd't find anything that seems related, but I guess it's a bug that was introduced somewhere between 1.2.10 and 1.2.11 ... Does anyone else have similar problems? Regards, Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller L?tticher Stra?e 10 Tel 0241/701333-14 km@westend.com D-52064 Aachen Fax 0241/911879

...hannel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I will be glad if someone can throw more light on this for me. Goksie -----Original Message----- From: asterisk-ss7-bounces@lists.digium.com [mailto:asterisk-ss7-bounces@lists.digium.com] On Behalf Of Kai Militzer Sent: Monday, March 27, 2006 3:19 PM To: asterisk-ss7@lists.digium.com Subject: Re: [asterisk-ss7] g723 or g729 on ss7 link Hi! > Is it possible to set codecs on ss7 link? No. E1s channels (which chan_ss7 uses as voice channels) can only use G711 alaw. > Or receiving call with g723 or...

...l(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I will be glad if someone can throw more light on this for me. Goksie -----Original Message----- From: asterisk-ss7-bounces@lists.digium.com [mailto:asterisk-ss7-bounces@lists.digium.com] On Behalf Of Kai Militzer Sent: Monday, March 27, 2006 3:19 PM To: asterisk-ss7@lists.digium.com Subject: Re: [asterisk-ss7] g723 or g729 on ss7 link Hi! > Is it possible to set codecs on ss7 link? No. E1s channels (which chan_ss7 uses as voice channels) can only use G711 alaw. > Or receiving call with g723...

...transmitted correctly, as the communication is SIP only, but this seems a bit strange to me. I can't even tell, if my asterisk produces these messages, or the other side (aka PSTN-Gateway). If anyone can bring some light into this behavior I would be very greatful. Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller L?tticher Stra?e 10 Tel 0241/701333-11 km@westend.com D-52064 Aachen Fax 0241/911879

...connected. Forwarding an incoming ISDN-call to a SIP-Phone works fine. How can this behavior be explained. Should I use an ISDN-CAPI driver to realise the call transfers, or do I need a newer Card? I would appreciate any information that will help me solve this problem. Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller L?tticher Stra?e 10 Tel 0241/701333-11 km@westend.com D-52064 Aachen Fax 0241/911879

> Is it possible to tell asterisk not to strip the leading 0 > of *incoming* MSNs? I use asterisk with i4l and whenever > I get a call from an long-distance party, the leading 0, which > should be there according the german numbering, is not. Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest

...ady), resulting in zombies. Only a killing of the zombies and a restart, makes asterisk usable again. I'm running a RC1 (I know it's not the newest, but beside the hanging-Channel Problem it runs OK). Does anynone have a idea how this problem could be solved? Best regards, Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller L?tticher Stra?e 10 Tel 0241/701333-11 km@westend.com D-52064 Aachen Fax 0241/911879

Hi list, I've noticed that instance-backed AMIs are not available in AWS MP (only the EBS ones are there). Still though, also the instance-backed AMIs are listed as published and available on your wiki: http://wiki.centos.org/Cloud/AWS I've seen that the same question was asked already before but I'm not sure the answer given actually covered the whole question:

This is super easy to do. All you need to do is to put that announcement in a MP3 and set a musiconhold class for that incoming Zap channel. Then basically when ever that PSTN number rings, Asterisk will play the MP3 stream "Your call may be monitored or recorded, please hangup if you do not agree...etc" in a loop until the line is answered. Caller doesn't pay a single dime to