similar to: IAX hardphones? anyone?

Hi all! I'm using Asterisk as a SIP server with 2 ip phones and it works great, the only think that I can?t make it wotk is: 1. I leave the messages but when I receive them in my mailbox , and open them I hear only noises -------voicemail.conf------- [general] format=wav49 servermail=asterisk attach=yes maxmessage=180 pbxskip=yes fromstring=The Asterisk PBX [default] 123 =>

Hello all, I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to send the voicemails as attachment to e-mails and delete the voicemails from my PBX once it has been sent. But, I don't have a running MTA here even on the PBX itself. I just want to send the e-mails to my GMail account from my PBX. Can I just use the mail or mailx command to send the e-mail and attach the

I'm having tons of trouble getting the fromstring to work in voicemail.conf. I've tried both voicemail and voicemail2 but the emails still seem to be coming from asterisk pbx. Has anyone had any luck with this? ================= Here's my voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav49|gsm|wav ;

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Mon, 2 Mar 2015, Oliver Welter wrote: > Am 02.03.2015 um 06:03 schrieb Dan LaSota: >> I have dovecot version 2.2.10 >> dovecot -n output below >> >> I am seeing connection errors being written to my dovecot error log: >> Mar 1 19:51:15 mail dovecot: auth-worker(2224): Error: mysql(localhost): >> Connect

On Tue, May 28, 2002 at 12:59:33PM +0200, Ralf G. R. Bergs wrote: > Hi there, hey Ralf. > sorry for being semi-off-topic, but since this has at least *soemthing* to do > with XFS I hope you forgive me. :-) > > I'm trying to compile Samba 2.2.4 under Debian 2.2 (stable). I'm running a > kernel compiled from yesterday's XFS 2.4.18 kernel source (I'm mentioning

Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal ____________________________________________________________________________________ Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545469

I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:1234@213.137.73.178/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -----Original Message-----

Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ

Asterisk Users: I've been working hard on app_voicemail2 which is an enhanced scalability version of app_voicemail. Specifically, its features are: * Highly improved internal architecture (maybe someone else can actually code on it) * Foot print for getting mailboxes from DB (for Vonage) * Segmentable mailboxes, allowing you to truly multihost voicemail for multiple companies

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. What I want to do is execute a remote command with the voicemail as an argument. The remote machine command would email the message. I'm thinking of: same =>n,VoiceMail(vm,u) same =>n,System(ssh myserver "emailVM

Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

Hi, I have enabled ztdummy in order to have * compile it. I can modprobe ztdummy with no problems. The sole reason, i need ztdummy is to heve musiconhold and meetme working. However when I start *, it says this and does not start. ---------------------------------------------------------------------------- ---------------------- == Parsing '/etc/asterisk/zapata.conf': Found

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq --

Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with "1" in order To successfully make a call to other USA destinations? ---- I have not been to USA (yet) :) Ta SJ

Hello all, I am sure that this is possible - for helpdesk envioroments i.e. when you hear an announcment "Your call may be recorded for quality purposes" can asterisk record all calls onto disk or similar - hopefully as MP3? Thanks Nick