search for: mielnik

Hi, I can not send voicemails as an attachement. When setting the "attach=yes" option in voicemail.conf the mails get rejected from the mail server: ----- Transcript of session follows ----- 451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection timed out with higgs.elka.pw.edu.pl. 451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection timed out

Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? Regards, Dave

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea?

...VIC 3820 AU Ph: (+61) 1300 665 575 Fx: (+61) 1300 556 595 -----BEGIN GEEK CODE BLOCK----- Version: 3.12 G! d- s: a-- C+ UL++ P+ L++ E---- W+ N+ o- K- w+$ O--- M-- V? PS !PE Y-- PGP- t 5 X+ R- tv b- DI-- D--- G-- e* h* r- z++ ------END GEEK CODE BLOCK------ -----Original Message----- From: Dawid Mielnik [mailto:D.Mielnik@elka.pw.edu.pl] Posted At: Friday, 2 January 2004 8:24 PM Posted To: Asterisk Conversation: [Asterisk-Users] one way choppy sound problem ! Subject: [Asterisk-Users] one way choppy sound problem ! Hi all, I have my asterisk setup as following: IP 2 x E1 x-l...

Hello, I can not seem to be able to get StripMSD and Prefix to work for me in extensions.conf. This is an example of what I have: exten => _050.,1,StripMSD,1 exten => _50.,Prefix,01051 exten => _001051.,1,Dial(${TRUNK2}/${EXTEN}) exten => _001051.,2,Busy exten => _001051.,102,Busy What I want to achieve is to call 001051501657887 on TRUNK2 when dialing 0501657887. dialing

Hi all, How scalable is asterisk ? I am considering using asterisk as a VoIP platform/gateway between Internet and PSTN (switches) to offer services to home customers. What goes along with it is eventually a lot of users - upto thousands probably. Is load balancing possible with multiple asterisk boxes ? Does anyone have any sort of info/experience with such projects ? How would asterisk cope

Hi all, I am planning to use VoIP gateways to connect remote offices to Asterisk. Not having much experience with these and Asterisk I would appreciate any info/experience that you might share with me as to their interoperability with Asterisk. I am interested with in rather bigger gateways (order of E1's) from: AudioCodes - Mediant Mediatrix - 1531 Quintum tenor Multupath D3000 Has anyone

Hi, I have problems with Asterisk recognizing dtmf tones sent by Mediatrix 1104 FXS. I can not enter mailbox number (voicemail) or pin code (meet-me). Asterisk shows 'username not entered' when dialing in voicemail. Both asterisk and Mediatrix have set inband dtmf. Can anyone help me out ? Best regards, Dave

Hi ! I am trying to configure two E100P cards, but I am a bit confused with zapta.conf in what I am trying to achieve. The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines will be used for incoming calls as well as outgoing calls. My problem now is what to put in zapta.conf, I would like to group all channels from both cards together (if that's possible). Does this

...n managing the list at asterisk-users-admin@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: Has Nufone gone belly-up (Steve Underwood) 2. SIP - fax / voicemail (Dawid Mielnik) 3. Re: Has Nufone gone belly-up (Girish Gopinath) 4. app_queue and dialplan (Anton Yurchenko) 5. Know if a call is answered (Asterisk List) 6. Re: rc.local dont works (Jeroen) 7. Re[2]: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller (Frankie Gravato)...

Has anyone installed openh323 on redhat 9 ? I have problems compiling pwlib: In file included from /usr/include/openssl/ssl.h:179, from ../../ptclib/pssl.cxx:195: /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory

Hi, I am trying to test a Mediatrix 1104 FXS SIP gateway with Asterisk, but I have some problems. When registering the Mediatrix gw doesnt respond to Asterisk's 'proxy authrisation required' messages as if it didnt understand them. Strnage thing, when I have type=friend, asterisk says that the Mediatrix is unauthorised - get fast busy in handset. When I put type=peer in sip.conf, I

Hi, Just to clear things out.. Can asterisk transmit faxes over IP ? If not, are there any works being done towards implementing t.38 on asteisk ? Also dialing in from a mediatrix fxs sip gateway to voicemail, asterisk does not see the digits entered after mailbox prompt. I have dtmftone settings correct - inband (also tried others to make sure), however asterisk shows 'username not

Hi ! I have a small problem. When switching a call (pstn -> sip user), I get the sip phone ringing - ie. everything is OK, but I do not get a ringtone in the handset on the pstn side. Can anyone help me out in how to make * play tones ? My setup: E1 IP pstn ------ Asterisk ------ sip phone Regards, Dave