search for: kokinos

i've been searching the archives but can't find anything substantive on this. most of the music on hold documentation discusses integrating with zap hardware, but i am trying to send it across a sip channel. I have the following in extensions.conf: exten => 2100,1,Answer exten => 2100,2,MusicOnHold(default) and have uncommented the "default" line in musiconhold.conf:

in my setup i have several users with DID lines coming in from various sip/iax providers. within our old phone system, a user could call their own DID line, then hit the * key when they hear their voicemail greeting and be prompted for their password. is there any way this could be replicated within asterisk? i'm having trouble figuring it out since it steps through things sequentially,

Hello- As part of the asterisk build/installation instructions it mentions that the zaptel drivers should be built and configured first. My question is whether they are required at all, in the case of a system with no hardware cards at all (as is the situation in my case). With them loaded I continually get the following message on my console (server not asterisk): Zapata Telephony Interface

Hello- I'm obviously doing something wrong here in trying to get an inbound DID to work with voicepulse. I have an outbound context set-up for those calls in iax.conf, and the appropriate register in- statement. within extensions.conf I am doing something like this: exten => 212xxxxxxx,1,Dial(SIP/admin,t) (where admin is the phone i am looking to forward to from sip.conf). i'm

I'm working on setting up a macro that will allow users to call their own DID number, and when they hear their voicemail greeting hit the * key and be prompted for their password to check vmail. For some reason though the background command isn't working as I'd expect it to: [macro-vmessage] exten => s,1,Answer exten =>

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

Does anyone know of any provider(s) that can provide DID's for the Czech Republic? Regards, -Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040802/4885836b/attachment.htm

Hello- I have noticed that since i upgraded my kernel, asterisk spawns many copies (usually approximately 18) when starting up. It then runs fine, but there doesn't seem to be any reason for this behavior. I have tried moving between different kernel versions, and all but the stock fedora core 1 kernel exhibits the same behavior. I have verified running this with both safe_asterisk (as I

Hello- I am currently testing with a carrier that seems to be having some trouble around toll-free (800 number) access. While a problem, its the resulting behavior that I'm finding disconcerting. When I dial an 800#, I get the following response: -- Executing Macro("SIP/2700-e10b", "carrier-out|18005558355|70|r") in new stack -- Executing

...Apr 2004 13:05:06 -0400 (EDT) Subject: Re: [Asterisk-Users] Zapata required? From: "Mark Phillips" <kc2eni@nyc-ares.org> To: asterisk-users@lists.digium.com Reply-To: asterisk-users@lists.digium.com You did unhash the ztdummy in the Makefile before compiling it right? > Steven Kokinos wrote: > >>Ho do I go about loading the ztdummy driver after unloading zap? >> >> >> > $ su - > # modprobe ztdummy > > >>Thanks, >> >>-Steve >> >> >> > > _______________________________________________ > Asterisk-Us...

Hello- I am in the process of adding a new provider to my asterisk box (both for outbound termination as well as inbound DID). They are going to be delivering and receiving traffic via SIP only. Now, in IAX via Voicepulse or others I know that I can simply have one registration statement along with an inbound context, then in extension.conf map the outbound context. from iax.conf: register

I've been intermittently seeing the following warning: Apr 14 18:35:04 WARNING[1192437440]: rtp.c:386 ast_rtp_read: RTP Read error: Resource temporarily unavailable which doesn't appear to have any effect on the current call, but isn't anything I've seen before either. Any thoughts on whether this is something to be concerned with? Regards, -Steve

thanks to those who replied. I have managed to get the functionality I was looking for working with a series of Macros. However, it doesn't work as simply as I would like. There are two issues I've run into: (1)Goto provides no way to pass variables between one context and another. (2)I can't find any way to Goto a specific point within a Macro when calling it. Mostly this is a

I currently have several asterisk servers geographically distributed (for automatic fail-over in the event of either a network or server problem). My carrier delivers to each server based on the same priorities that I have set in the DNS SRV records which the clients point to. Users always have dialtone regardless of a single server failure. In addition, once they have re-registered with the

hello all, I'm having problems getting my polycom soundpoint ip 500 working, and was wondering if anyone would be willing to share their config files with me (the polycom configs). I have managed to get my boot server up and running, and the phone successfully updated its ROM, and downloaded the config files i have put together for it (the display shows correctly but the line won't