similar to: SIP/2.0 487 Request Cancelled

I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways. I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites. Got SIP response 500 "Internal Server Error" back from 10.1.3.28 SIP/alma-1b77 is circuit-busy Everyone is

Hello, I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in my ata the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP

I've seen this question floating around, but yet to see any answers. PLEASE let me know if anyone has figured this out. I've got a SIP trunk between Cisco Call Manager 4.x (10.200.204.10) and Asterisk 1.4 (10.200.204.40). I'm trying to send calls from CM to Asterisk. It appears Asterisk is sending info back that CM doesn't like. I keep getting a SIP/2.0 400 Bad Request -

Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI> -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack -- Called 5104112978 --

I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN

I installed * to primarily test its voicemail feature. I installed it on a server WITHOUT any telco board (i.e., digium). Installation looks ok, however I am having problems. MY SETUP: 2xATAs are configured to use * as GkorProxy Asterisk is registered to my SER SIP/RTP Proxy 1.) First test - ATA1 calls to ATA2. When voicemail starts playing, it just cuts-off after 5-10seconds (consistently).

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

Does anybody have a good source for hold music? I can see a number of companies on the web that sell royalty-free MOH, but they don't all provide samples. The customer service desk has requested "calming, not sleeping, but calming" and "this is a high-tech company, so make it 'techie' [sic]". Thanks, --Ernest

The following series flushes my documentation queue and replaces previous postings of those patches. The main difference is that the xl cfg file is now formatted using POD instead of markdown and presented as a manpage. I have setup a cron job to build docs/html and publish it at http://xenbits.xen.org/docs/unstable/ (it''s a bit bare right now). The motivation for some of these patches