similar to: How to get Asterisk acting like a Multi-thread application?

Hi list, I'm trying to get some ideas on this subject. Normally astersik sends emails with voicemail attached trough local MTA. As far as i know there is no way for asterisk to authenticate to an external mailserver to relay these emails. Well, these days every provider has some sort of spam blocking, to add to that usually users of asterisk are behid a dynamic IP with no PTR and list grows

HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

> On 12. Jul 2020, at 23.38, Sami Ketola <sami.ketola at dovecot.fi> wrote: > > > >> On 12. Jul 2020, at 12.54, Vitalii <vnagara at yandex.com> wrote: >> >> Panic: file imapc-sync.c: line 328 (imapc_initial_sync_check): >> assertion failed: (mail_index_is_expunged(view, lseq) || >>

On Sat, 11 Jul 2020 20:36:44 +0300 Sami Ketola <sami.ketola at dovecot.fi> wrote: > > On 11. Jul 2020, at 14.22, Vitalii <vnagara at yandex.com> wrote: > > > > Greeting > > > > I've managed to backup my emails from external email server via > > dsync and imapc: protocol like this: > > > > doveadm -Dv -o imapc_user='user at

Hello, Long time subscriber/reader of this list - thank you for all the great ideas. Scenario: We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality: - Agents having the ability to

Dear all using top -H i can see that some asterisk thread are consuming many CPU (sometimes more than 50%) Is there a way to understand what is doing the process with pid 9429 ? i've tried the core show thread command, but it doesn't seem to print any PID information. Thanks to all in advance PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 9429 root 20 0

Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register:

Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli

Hi Guys, do any body know how to receive debug info on RxFAX application? i am experiencing a lot of fax failures and can't guess the reason behind. Thank you very much for any help! -- Abdullah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100402/cf26573a/attachment.htm

Hello list i have configured astersik 1.4 with sip i have a question when i put in dial plan.conf exten => _0678922645.,1,Set(CALLERID(number)=520460587) exten => _0678922645 .,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => _0678922645 .,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)) exten => _067892264*5*,2,Hangup() i can not call my

Hi all, I've uploaded a free app for the iPhone called AsteriskRef to the Apple AppStore. This allows you to lookup applications and functions using your iPhone or iPod touch so you don't have to jump out of extensions.conf or open another terminal tab. It currently supports applications and functions from Asterisk 1.4, but I'm adding 1.6 and trunk at the moment. It currently

Hi all, As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00 pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. Any idea or solution. Regards Farooq --

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031207/e5e9b8eb/attachment.htm -------------- next part -------------- Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message "Phone

Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : alice at pbx.com should be able to call bob at

HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

re-posting the question. ----------- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload.

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm hosting a few customers on a dovecot Server. Most users speak german and have german as their main language. I configured the IMAP Foldernames in a Dovecot configuration file like this: ##### root at srv04:~# cat /etc/dovecot/conf.d/105-mailboxes.conf imap_capability = +XLIST namespace inbox { inbox = yes location = separator = / mailbox "Entw?rfe" { auto

Hi all, I am very new with Astersik. Could some body tell me if it is possible to run Asterisk without any .conf file in /etc/asterisk ? I just want to test if my Asterisk has been installed correctly and as I am waiting for digium cards ... I have already tried but nothing happened after some verbose it stop... Thanks Angel __________________________________ Do you Yahoo!? Yahoo! Tax

Good day all I want to know what the best codec is to use for asteris for VOIP We have two towns connected with a 64k line that's going to do VOIP with astersik.At the moment with the default installation the quality is bad and the bandwith is high. Is this even a codec problem Pleas help ALtus