search for: aisling

...ame errors were display in the log file on startup and it didn't allow me to interrupt the menu. [incomingpstn] exten => s,1,Wait(1) exten => s,2,Background(MainMenu) ;exten => s,3,WaitExten(10) exten => 1,1,Goto(internalExt,s,1) exten => 2,1,Goto(mainconfmenu,s,1) Many Thanks, Aisling. -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 11 January 2006 10:14 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu Hi Kokmeng, Unfortunately that's wa...

...y hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an "Unable to read password" message on the asterisk console. Has anyone experienced issues with XLite dtmf? Many thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in...

Just got a call from a company in Warren, MI . They recently had an Asterisk system put in by a vendor, and are having issues which need analysis and correction. They have a tremendous sense of urgency. They have about (40) users, and need DID's assigned to extensions and are having some echo issues at the site. If anyone is in the Warren, MI area, and is interested in some cavalry work,

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW: SER Asterisk Voicemail Date: Thu, 10 Feb 2005 16:45:53 -0000 Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services....

...aying ?MainMenu? (language ?en?) -- other messages (not relevant I think) == Spawn extension (outgoing, 021123456, 1) exited non-zero on ?SIP/2092-5837? == Spawn extension (default, 2093, 2) exited non zero etc etc I?m very stuck on this and can?t figure it out. Any help appreciated. Many thanks, Aisling. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Giovanni Miano Sent: 05 January 2006 21:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Incoming PSTN Calls Is Exist...

...n when I create a new context, a directory is not created in /var/spool/asterisk/voicemail. The default context resides there but new ones are not created. Has anyone ever experienced this or does anyone have any idea as to how I would solve this? Hope someone can shed light on this, Many thanks, Aisling. -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 07 September 2005 13:54 To: 'asterisk-users@lists.digium.com' Subject: Eeven Stranger - Asterisk BT100 Password Issue Following on from my below email, things have taken another bizarre twist.. I have c...

Dear R communities May I seek your advices on how to change color the default in levelplot(), e.g. from the default of pink and light blue, to e.g. red and green ? The levelplot function has 1 of the arguments being panel (which is actually panel.levelplot), but I am not sure where the commands to alter the color. For example, I type: p1<-levelplot(my.mat,colorkey=FALSE), how could I

...oicemail (u2092) exten => 2092, 102, Voicemail (b2092) exten => 2092, 103, Hangup exten => 9999, 1, VoicemailMain(${CALLERIDNUM}) ;voicemail.conf [general] format=wav [from-sip] 2092 => 2092, 2092, emailaddress Has anyone any inkling as to what the cause could be? Many thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in erro...

...d I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java Rockx using sipsak for sending mwi sip notify messages to the phone but is there a simpler way which I am blindly ignoring?? Thank you in advance, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in erro...

...en I do alter table voicemail_users add column delete varchar(3) NOT NULL default 'no'; I get a message telling me that I have an error in my MySQL syntax...Is this because the 'delete' word I s a reserved word and if so is this something others have experienced? Many thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in erro...

...ay be to connect to a third party voice/pstn gateway?? Is that simply a matter of forwarding all sip traffic destined for the pstn to another provider with a gateway and then they have to worry about the number of lines etc??And if that is the case, I presume no extra hardware is required? Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in erro...

...ame old error as before. I tried plugging out the phone again but it did not make a difference. Does anyone know what those extra messages on the console mean or how I can solve this? I am obviously missing something important - How do I get it? Many Thanks. -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 06 September 2005 18:09 To: 'asterisk-users@lists.digium.com' Subject: Asterisk BT100 Password Issue Hi, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain...

..."sip reload" in the asterisk console, it says parsing /etc/asterisk/sip.conf, so it's definitely the correct file. Do I need to change the asterisk port somewhere other that sip.conf? Does anyone have other suggestions for what could be making Asterisk behave so oddly? Many thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in er...

...enu, when I press '1' to interrupt the menu and move to menu option 1 (another sound file) it won't let me interrupt and I eventually get the error "Timeout but no rule 't' in context 'default". Does anyone have any ides where the problem might be? Many thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in erro...

...everal calls are possible then do I need a pbx with a PRI interface?? Also where does all the digium cards come in all this??Where do they fit in?? I would be extremely grateful if somebody could shed some light on my currently very hazy understanding of voip telephony with asterisk Thanks again, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in erro...

...d "make upgrade". However I get an error: Makefile:16: *** missing separator. Stop. Make[2]L Leaving directory '/usr/src/asterisk' Make: *** [depend] Error 1 Has anyone come across this or does anyone know a way of solving this? Many thanks -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 26 August 2005 15:44 To: 'asterisk-users@lists.digium.com' Subject: cvs update error? Hi, I'm experiencing a problem with playing back my voicemail. (Failed to write frame). It has been indicated in the archives that this is problem can be s...

...rhaps a voip number) and this number would enable them to be reached via the pstn, pots, gsm etc.... Does anyone have ideas where I could start looking at sites to research this or how asterisk might fit into this?. It would be great if someone could maybe point me in the right direction. Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in erro...

...n the ser.cfg file, my clients can't register. The only thing I can think of is that SER is behind NAT and my clients may/may not be behind NAT....I have included my ser.cfg file below...I have spent along time trying to understand why this is happening so any help will be appreciated! Thanks, Aisling. # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to ente...

...n_sip.c: 2520 sip_write: Asked to transmit frame type 4 while native formats is 256 (read/write=64/64). When I comment this line out again I am back to my original situation where outgoing calls work and incoming don't. Has anyone any idea how I can work around this? Many thanks in advance, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in er...

Hi guys im trying to get NAT working on my system. im using 3 phones, 2000 = xlite, 2001 = xlite, and 2010 = some piece of crap voip phone. when i ring from anywhere to anywhere u can either never hear anything on both ends, or just 1 end can hear stuff. below is the output of sip show peers 2010/2010 203.1.68.90 (D) 255.255.255.255 49534 UNREACHABLE 2001/2001 203.1.68.90