similar to: Inband announcement of parking slot from app _parkandannounce?

I'm trying to use Asterisk app_parkandannouce to build a global parking pool from within a couple of Norstar PBXes. Right now I can blind transfer calls into the parking lot, but the slot announcement relies on calling back the 'transferee' after the call is parked and I can't pass enough callerid data out from within the PBX to be able to route the call back in (ie. no PRI

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Gary Reuter > Sent: Thursday, August 11, 2005 12:59 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped > betweenpstn & norstar > > > I poured over my logs most of

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

I am trying to build asterisk-0.9.0 on 533MHz 160MB Redhat Linux 9.0 machine. I have followed the instructions to build asterisk. Building zaptel and libpri seemed to go well (lots of messages but nothing that indicated an error) However, when I do the make clean ; make install for asterisk-0.9.0 after running for sometime I get the following: gcc -shared -Xlinker -x -o app_senddtmf.so

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

I've got a need to do something like the following: [foo-context] exten => _.,1,SetCIDNum(123) exten => _.,2,SetCIDName(XYZ) include => local include => tollfree But of course, this example won't work. The goal here is this: if a call ends up being handled by the "local" or "tollfree" contexts, I want those SetCID*** commands executed. Otherwise, I

Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars =

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

To answer my own question for the record: The relevant timing parameters in zaptel.h are #define ZT_MINPULSETIME (15 * 8) /* 15 ms minimum */ #define ZT_MAXPULSETIME (100 * 8) /* 150 ms maximum default, lowered to 100ms */ #define ZT_PULSETIMEOUT ((ZT_MAXPULSETIME / 8) + 50) And the pulse detecion loop that consumes these parameters begins at line 4866 of zaptel.c The

Is it possible to make cmd_dial() bridge the audio going out to the network back to the calling party as soon as dial() starts? Put another way, is it possible to have the caller hear the outside dialtone and subsequent DTMF digits? I notice that there is an option 'r' to dial(), thus: r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one

I'm using a T100p to interface to a channel bank and from there to analog PSTN lines. Because of my particular setup I have to do post-connect inband DTMF dialing - which takes up to 5 seconds for a 10 digit number, assuming 0.5/sec per digit (ie. using "zap/g1/31|5|D(6045551212)". Even with an 'outside transfer' voice prompt before commencing dialing my users are getting

Dear all, I have 3 models (from simple to complex) and I want to compare them in order to see if they fit equally well or not. From the R prompt I am not able to see where I can get this information. Let´s do an example: fit1<- lm(response ~ stimulus + condition + stimulus:condition, data=scrd) #EQUIVALE A lm(response ~ stimulus*condition, data=scrd) fit2<- lm(response ~ stimulus +

using a zap fxo and zap fxs card how can I set up caller announce? like this. 1 call comes in and a prompt asks the called to identify themselves. 2 the system would then put the caller on hold and pick up the FXS and play the message for the users prompting them to hit 1 to accept the call and have it connected or hit 2 to dump the live caller to voicemail. Can this be done with * Dave