similar to: Don't receive the prefix

Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture. My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn?t work! Is there any possibilities to do caller based routing in

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

Hello, Does someone know how to setup snmp monitoring on asterisk. I?ve plan to deploy 50 asterisk, so I need some monitoring tools. I try with nagios as I read in the wiki, there is some project on it, but I can?t reach the end. Can someone help me? Thanks. GIBERT Fr?d?ric Ste VigiNetworks Mobile: +33 6 72 08 35 16 -------------- next part -------------- An HTML attachment was scrubbed...

Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI> -- Executing Dial("SIP/16008-3d17", "SIP/16007&SIP/16006|20|tr") in new stack -- Called 16007 -- Called 16006

Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Thanks by advance.

Hi, sipp (http://sipp.sourceforge.net/) seems to be a good app. Take a look at http://www.voip-info.org/wiki-SIPP on the wiki to have more info about it... Basically, there is scenario which are describe there and I personnally generated about 3,000,000 calls before having to restart asterisk and i placed about 90 concurrent calls. Good luck! -----asterisk-users-admin@lists.digium.com a ?crit :

It's because you're misunderstanding how the lookup() function works. It gets ALL the search parameters, including the "mailbox inbox". This is intentional, and not a bug. Two reasons being: 1) The FTS plugin in theory could support indexing/searching any kinds of searches, not just regular word searches. So I didn't want to limit it unnecessarily. 2) Especially with

Thanks for this informations. Do you know where I can find the icd-snmp package for a redhat 9 distri? I can't find it. Thanks. Message: 6 Date: Fri, 09 Jul 2004 15:45:57 +0200 From: Andrea Fino <af@faino.org> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SNMP Monitoring Reply-To: asterisk-users@lists.digium.com Holger Schurig wrote: >>You could try

Inbox appears in the list of arguments, because fts_backend_xapian_lookup() is parsing the search args wrong. Not sure about the other issue. > On 21 Apr 2019, at 19.31, Joan Moreau <jom at grosjo.net> wrote: > > For this first point, the problem is that dovecot core sends TWICE the request and "Inbox" appears in the list of arguments ! (inbox shall serve to select teh

hello, i was thinking of using postfix for a mail server, the question is can i use this just for in bound email only and have my clients access this server locally. the clients would be outlook. I want this to check certain email accounts i have and download them to the postfix mail server. The reason i would think about doing this is my wife and i need to access both of these accounts and i do

Hi all, I know Hylafax is an application and not Asterisk but I'd like to post a problem found in configuring such application and Asterisk. I am able to reveive fax,but , I can't receive it in e-mail. Although I put my e-mail in /etc/hylifax/Dispatch I can't receive. Anybody know where I must to add something else in order to make it works! Thanks in advanced!! Att,

Hi I'm from spain, so forgive my English. I'm somewhat new to asterisk, and i'm having trouble in getting my line to work (i'm in a little hurry!). I have an E100P, and it seems everything is configured ok, but when receiving a call, I get the following message "D-Channel on span 1 up" (four times), then the call ends, asterisk says "D-Channel on span 1

Hi, I search How To transfer call between my SIP phone. I have an PSTN line (X100P) and 10 grandstream budge tone phone. For example I want : - Reveive an external call and send it to SIP/phone1. At this point no problem. - After my receptionnist want transfert extern call at SIP/phone2... I don't known how to properly transfert call.... Thanks

Hi, i got a little question about the NA's in vectors and matrices. When I want to do some operations on a matrix with some NA's it is possible to stripp them before the computation proceed (by na.rm=TRUE). But how can I stripp NA's when I want to proceed a 'if', 'for' or 'while' string? Because when I don't stripp them I always reveive the message

No, the parsing is made by dovecot core, that is nothing the backend can do about it. The backend shall *never* reveive this. (would it be buggy or no) PLease, have a look deeper And the loop is a very big problem as it times out all the time (and once again, this is not in any of the backend functions) On 2019-04-21 10:42, Timo Sirainen via dovecot wrote: > Inbox appears in the list of

hi at all, i'm trying to migrate a vm between two host but fails, this is what I did: virsh # start win2008 Domain win2008 started virsh # list Id Name State ---------------------------------- 1 win2008 running virsh # migrate --live win2008 qemu+ssh://host2/system error: internal error Attempt to migrate guest to the same host

Hi everyone, I use Dovecot in lmtp mode to receive mails. I would like an option to tell Dovecot to not add a "Reveived" header on each server (I use a director, so Director also adds this header). Is it possible to do this ? Or could it be a future feature ? Thank you. Florent

Hello, Groups in asterisk are summarized here ( http://www.voip-info.org/wiki/view/Channels+and+Groups). Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789 (as I've been advised in another thread, to switch from one notation to the other and I can't see the reason behind that) ? Regards -------------- next part -------------- An HTML attachment was scrubbed...

Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number.

Timo, A little of logic here : 1 - the mailbox is passed by dovecot to the backend as a mailbox * pointer , NOT as a search parameter. -> It works properly when entering a search from roundcube or evolution for instance. -> therefore this is a clear bug of the command line 2 - the loop : Actually, the timeout occurs because the dovecot core is DISCARDING the results of the backend