search for: scheepmakersstraat

Hi, all!!! What will Asterisk do in the following case: For example, we have 4 licenses, and have 4 simultaneous calls, using G729. Will asterisk allow incoming calls from peer, that can talk G729 and ulaw, and will it force it somehow to use ulaw in this case? All phones there in LAN behind Asterisk prefer GSM codec, so it does transcoding. So, what I mean is will Asterisk fall back to use

I've successfully got inbound/outbound calling working with our Asterisk using the Asterisk-OH323 channel driver. We are using a parent gatekeeper and the NuFone H323 channel driver would not work with the parent gatekeeper... I'm trying to determine a way to ensure that the line used for outbound calling is always available i.e. like trunking.. >From what I can tell when I place an

...ort Sep 9 09:25:45 WARNING[467828746]: Asked to transmit frame type 256, while native formats is 32 (read/write = 256/256) With lots of the last variety. Does Asterisk support T.38 (froma Google search it seems not), is anyone working on this? -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

...m here with asterisk. Wehn Asterisk sends it reinvite, it uses its own codecs, not those of the other endpoint. So until someone fixes that (when possible), there's no way this will work. We're using a CVS version of approx. a month ago. -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

...understand what I want it to do: Aug 6 15:46:42 NOTICE[524301]: pbx.c:4700 pbx_builtin_gotoif: Not taking any branch I've been looking all over the wiki and google, but can't find any example doing what I want to do. Is it even possible? -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

...${connid}) This somewhat mimics the way the Borland implements this type of queries in their products like Delphi. It is a quick hack, but we've been using it for a couple of days now and have not seen any issues with it. (yet? ;-) ) Enjoy. -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540 -------------- next part -------------- A non-text attachment was scrubbed... Name: app_mysql.c Type: application/octet-stream Size: 13518 bytes Desc: app_mysql.c Url : http://lists.digium.com/pipermail/asterisk-users/attachment...

Hi, need a quick help ... it should be easy but ... exten =>_9898,1,Answer exten =>_9898,2,VoiceMailMain(${CALLERID}@domain) Accepting overlap call from '342' to '9' on channel 0/2, span 3 -- Executing Answer("Zap/8-1", "") in new stack -- Executing VoiceMailMain("Zap/8-1", "@domain") in new stack As you can see there

Hi, Our gateway has asked that we send DTMF as RFC 2833. Although chan_oh323 seems to do this, it doesn't specify the DTMF mode during the H323 setup headers. Is there an easy way around this? Thanks, Andrew

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hi, I run asterisk with oh323 plugins.It runs correctly with sjphone H323 Gatekeeper. But When i run openphone it doesn't recognize my asterisk server like a gatekeeper !! What is the problem ? Thx

I am looking at building an IVR product with a few interesting features and need some more information about how asterisk and VoIP work and what I can get from them. As far as I can tell when I use ISDN/GSM telephone networks the DTMF information travels as data representing 'start tone' and 'stop tone' for each button pressed, it is then generated at the other end if an

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

...e" release, but I'd need to jump > through all sorts of hoops to get Woody working properly). I wouldn't make a fuss about this. sarge is at least as good as woody and much more up to date for the stuff asterisk can do / needs. -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C