similar to: How to distinguish between UNEXISTENT channels v/s UNAVAILABLE channels

I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway user at a rate

I've defined my dialplan as showed below. My internal lines are numbered as 12345XX, and internal users can call another by the entire 7-digits extension, or by just last 2 digits. [invalid] exten => _X.,1,Playback(pbx-invalid) exten => _X.,2,Hangup() [internal] include => invalid exten => _XX,1,Dial(SIP/12345${EXTEN}) ; Short alias for internal lines exten =>

Is there a way to access only certain positions in the $EXTEN variable? I'd like to filter my international calls based on the destination country: My dialplan looks like this (1XX0. is the international calling convention for Chile) exten => _1XX0.,1,Dial(SIP/${EXTEN:4}@external_provider) But, I'd like to, depending on the destination country (digits 5 and eventually 6 of EXTEN),

I'm developing an application that monitors the state of the incoming calls using Manager events. So, as a part of it, I need to "override" the control of the extensions by the dialplan itself. The problem is that, if I don't declare the incoming extension, Asterisk hangs up the call by default. So I want to know if there's some kind of "ManagerControl() application

I've set voicemail almost successfully, only a minor detail remains :-) I can't get the dates in my local language (spanish). In sip.conf, zapata.conf and voicemail.conf, I've set: language=es and my locale is "es" also. However, the days and months names still appear in english in the emails!!! Thursday 11 de May de 2006, 18:49:34. instead of Martes 11 de mayo de

I'm trying to set the Pickup feature. I'm setting my extensions.conf as: exten => _*.,1,Pickup(SIP/${EXTEN:1}) but if, for example, extension 03 is ringing by a call made from extension 01, and I try to pick it up from extension 02 (by dialing *03 from extension 02), I can see in the Asterisk console (Verbosity set to 10): -- Executing Dial("SIP/01-512c",

Hello! I''ve noticed a strange thing: when a client system generates an arp query for an unexistent host, the routing cache entry is being made. My system is Fedora 2 with vanilla 2.6.11. the client is 10.1.1.2 with mask 255.255.0.0 the router/firewall is 10.1.1.1 with mask 255.255.255.0 Yes, the masks are different and this cannot be fixed easily. So, when the client generates ARP

The problem description below is relevant for those who use samba + LDAP. We installed four Intel Xeon servers with standard SuSE 8.2, samba + ldap. The W2K client complained about very, very, very slow reponse from the server. Below is we descripe the reasons and the solution. We have big problems with openldap version 2.1.12 (standard suse 8.2 rpm) and the name server cache daemon versiom

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

I've the following dialplan. exten => _XX,hint,SIP/${EXTEN} exten => _XX,1,Dial(SIP/${EXTEN},10,j) exten => _XX,2,VoiceMail(${EXTEN}@default,u|j) exten => _XX,3,Hangup() exten => _XX,102,Goto(110) exten => _XX,103,Playback(pbx-invalid) exten => _XX,104,Hangup() exten => _XX,110,VoiceMail(${EXTEN}@default,b|j) exten => _XX,111,Hangup() exten =>

Hello all I need your support in a big decision in front of two alternatives related with *. I must buy an E1, in order to manage 30 channels, given this big price; or I could opt for a 15 BRIs without cost to replace the same number of channels, and the question that inmediatly emerge is : ? can asterisk manage 15 BRIs ? If yes to the latter, could posible somo guide, for instance, wich Digium

hello list, I subscribe to Sixtel.net for a DID just to see how it worked. they say DID active inmediatly , but after 4 days , I have no DID , I tried to call tech support office and it seems they all ways are in hollidays, a support ticket no good luck, I call back service also not good luck. so anyone here has experience with them? are they a SCAM? fortunally only put there $10 bucks but

Hello, I'm running Asterisk 1.6.1.0 compiled from scratch under debian Lenny and I can't delete message from VoiceMailMain using option 7 Default folder is /var/spool/asterisk/voicemail and it's owned by asterisk:asterisk with 777 permissions Apparently VoicemailMain delete the message and inmediatly undelete it ! This the same issue as in this post :

This is the first time i wrote you, i would like to know if you have any idea about my problem: ? I install and conigure the last version of openssh and it is working like a root user ? $ ssh root at server ? and it?s just fine, even i can use CVS software, but if a try to use it with other user it doesn`t work ? $ ssh any at server ? ask me for a password and it wrote me, ?have a lot of fun? and

Hello I used to have an extension like this which worked fine with asterisk 1.0.7 I first dial to see if an IAX phone is present, if not I would try on SIP instead exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout exten=s,2,Goto(s-${DIALSTATUS},1) ; Default action exten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto 301

Hello everyone. Since I''ve found very poor documentation about the rails FCKEditor plug-in and Easy-FCKEditor (which is a fork of the very same plug-in), I decided to bother you with my noobish questions. I have this weird sensation that the image-uploading feature of FCKEditor isn''t supposed to work magically, without some sort of server side preparation, but I have no idea

Hi I recently updated to R 2.11 and see a strange problem. When run into the console, R does not warn when calling an unexisting object, see below. I don't know if this is related, but I am not able to run the R CMD BATCH properly... Did someone see a similar problem? Which information can I provide more on this problem? Thanks for help! Matthieu $R R version 2.11.0 (2010-04-22)

I am running Asterisk 1.0.9 and the latest bristuff in combination with a HFC isdn card, connected to a BRI interface. For some reason, I am not able to have it dial out (see below). It exits with DIALSTATUS=CHANUNAVAIL. One thing that may be misconfigured is that it says: Signalling Type: PRI Signalling. How do I change this? Or is this the correct value? I posted my configs below. ------

Well, I've installed Wine Doors, and I have problem. First of all, I'm trying to emule GUITAR PRO 5. The instalation runs right, and all seems to work, but once opening GP5, it pups a window saying that: Violation d'?cces ? l'adresse 39840950 dans le module "gdiplus.dll". Lecture de l'?dresse 00000048. Well, after that I've installed timidity and executed it, but

Hi, I want to use G.723.1 on *, I read it is supported in Pass Through mode, but I don't understand whats the meaning of that. I have a GW 5300 and an ATA 186 and I want to place calls to PSTN. I setup this config: [general] port = 5060 bindaddr = xx.xx.xx.xx context = sip tos=throughput maxexpirey=360 defaultexpirey=120 [gw5300] type=friend insecure=yes host=xx.xx.xx.xx