similar to: voicemail use external smtp server for sendingmail

Hi, I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0. Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail. There's been no changes to our configuration I bet I'm expereiencing a Linux problem rather than an Asterisk problem, but because I know only as much Linux as required to get

I know Vonage doesn't officially have a "bring your own device" type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/5bc9f644/attachment.htm

Hi All, I have set up three servers in a development environment. Via CR they're updated to Centos 6.2 It appears that these servers have postfix installed on them by default, which unfortunately I'm not very well acquainted with. All I want is a quick and dirty way to enable these hosts to send email through my own SMTP host. My (sendmail) SMTP host uses SMTP AUTH on a non-standard

I've googled around and although I get a lot of hits about postfix smarthost authentication with ssl, I can not find out how to actually accomplish the task. I've read through smatterings of postings from Neophasis and the like searching for just the syntax and what file (I assume it's main.cf) I should be using; however, any smtpd_ lines I have tried result in postfix hanging and

Hi I need to set some of our boxes to send mail through our smarthost. In the sendmail.mc file i have set the smarthost and rebuilt the sendmail.cf file. When i send a mail locally on the box though i get this error Diagnostic-Code: SMTP; 553 5.1.8 <root at localhost.localdomain>... Domain of sender address root at localhost.localdomain does not exist What else do i need to edit as

I've been using my Asterisk (At my house - 2 modem-type fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and I've been really happy with it, but I'm still having an echo problem that I've exhausted google with, and can't get straight... I think I've determined that because I'm using $7 voice modem clones for my FXOs that bad echo is going to

Hi., do you think that sbr policy in queue strategy will be useful? Bye ---------- Forwarded message ---------- From: nik600 <nik600 at gmail.com> Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com> Hi to all isn't there any plan to add the Skills Based Routing strategy in

Hi, I''m writing puppet modules since a couple of weeks now, so I''m still considering myself as a new comer in this field. It seems that modules I can find on the web or the recipe page on the wiki almost all use a design pattern where the module is shipped in one or several class(es) whose configuration are determined by "global" or nodes variables ala: module: class

Hi to all can i play a sound or a dtmf tone on a specific channel using AMI? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

thanks John yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new

Hi to all the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? This is a piece of my sip.conf: [202] type=friend secret=202 host=dynamic ; This device registers with us username=202 ; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored.... I've also tried

this is the point, and the strange thing: DTMF is set to rfc2833, but is working both on incoming and outgoing calls, it is not working only on calls generated with the Originate AMI command, or with the queue member that point to Local dialplan, as you suggested 2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>: > Looking at your logs it looks like you may need to

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

Hi i have a PRI connected to a TE205P. Actually, can i send and receive FAX through Asterisk using stable solutions? Or shall i connect an ATA to Asterisk and then a modem with Hylafax? -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser

i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/

Hi is it possible to spy (not record, spy) partially on a channel? for exaple, i'd like to listen only the input or the output voice. is it possible? thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser

Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMF&Channel=SIP/200-sdadsadkioah&Digit=1 the result is: <ajax-response> <response type='object' id='unknown'><generic response='Success' message='DTMF successfully queued' /></response>

Hi to all i'm planning the migration of a company on Asterisk, i have planned this scenario: 2 server with * 4 GB RAM * 2 CPU 64 bit dual core * RAID 1 * 2 network interfaces 1000 Mbit/s Each server will have a virtual interface that will be switched from one to the other in case of hardware problem. The question is: can one server with those settings manage up to 200 simultaneous call?

hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the