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..., Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816 Some tip? I have a ide>flash adaptor to make the install... I need recomendations in Linux distro... asterisk min. install ...etc..any info i can get. Thanks for any help Miklos Atenciosamente Cl?udio Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP ( BR - 55 11 3801-3702 ( USA - 1 360-968-1591 ( FWD - 64662 ( sip:ipfone@sipserver.com.br www.ipfone.com.br info@ipfone.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk...

Hi all! I get this error when trying to start asterisk: ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk What can be the problem? Thank you! Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 UK +44 870 - 3403539 FWD 64662 sip:ipfone@sipserver.com.br www.ipfone.com.br info@ipfone.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/att...

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

> -----Original Message----- > From: listas iPfone [mailto:listas@ipfone.com.br] > > Somebody knows why asterisk gives me that error wile playing .mp3 files? > > The files play well but the message aperas any way: > WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of > 4 > bytes) (No such file or direct...

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

--- listas iPfone <listas@ipfone.com.br> wrote: > Hi all > > I need to know if the video support for h.263 is > active in version stable > 1.0.7 to use with eyeBeam in asterisk it works for me... [2222] type=friend secret=xxxx auth=md5 callerid="myCallerId" <2222> canreinvi...

Hi all! Every time i receive a sip call MOH begin to play and i can?t talk to the caller. My setup is the default. Someone knows what is the problem? thanks Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 FWD 64662 ICH 31451543 www.ipfone.com.br info@ipfone.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031031/dc0da4f1/attachmen...

Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don?t in the initialization... I have in my file that comands: touch /var/lock/subsys/local modprobe zaptel modprobe wcfxo safe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening Somebody

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

After apt-get update && apt-get upgrade -y wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz tar xfvz chan_capi.02.5c.tar.gz cd chan_capi-0.2.5c make && make install shutdown -r now asterisk seg faults upon calling in via ISDN. Any ideas are greatly appreciated. rgds pos

...the criteria for such evaluation? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP 192.168.129.74:5070;branch=z9hG4bK5b33ae78;rport=5070 From: "IPFon" <sip:0225761853 at 192.168.129.74:5070>;tag=as7217acbc To: <sip:tzl at voip.touk.pl>;tag=as7217acbc Call-ID: 307fda656066a7e264e85cea0742e601 at 192.168.129.74 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: r...

...n advance bests -tomasz <--- SIP read from 192.168.0.165:7060 ---> SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 192.168.0.165:7070;branch=z9hG4bK274b7d50;rport=7070 Contact: <sip:poczta at voip.rd.touk.pl:7170> To: <sip:hellboy at voip.rd.touk.pl:7060>;tag=12711d6c From: "IPFon"<sip:0225761853 at 192.168.0.165:7070>;tag=as66773d49 Call-ID: 08027b6d45abc7a00c2c6a8630d2de47 at 192.168.0.165 CSeq: 102 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Jan 4 10:43:27] --- (9 headers 0 lines) --- [Jan 4 10:43:27] --...

Hi everyone, I know someone makes a product that's a POTS phone to SIP converter, where you just plug your POTS phone in one side and the network cable in the other. Has anyone successfully used any of these with Asterisk, and if so how expensive were they? I ask partly out of frustration with the FXS cards but mostly because it would make installation MUCH easier for what we're

...s Topics: > > 1. Re: Incoming SIP matching (James H. Thompson) > 2. Re: Has Nufone gone belly-up (John Baker) > 3. GSM phone to *? (Max Tulyev) > 4. Re: Has Nufone gone belly-up (David Liu) > 5. GSM modems (Steve Underwood) > 6. Re: rc.local dont works (listas iPfone) > 7. Need Europian vendor for Digium hardware. (Anton Tinchev) > 8. Wildcard X100P usable in Germany? (Roger Schreiter) > 9. RE: Need Europian vendor for Digium hardware. (Low, Adam) > 10. RE: Asterisk Indications (Philipp von Klitzing) > 11. He really doesn't care...

I've just down loaded Xten Lite and it is now build 1079. It now finds the NAT firewall type and has loads more to configure. But it doesn't work on my poor W95 tablet PC. -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene 04 90 23 30 81 "Internet Sheriff Technology" revendeur en France <http://www.linuxautrement.com> IAX 17004902330

Hi List ! I received an unit of the Symbol NetVision Phone and i will test it with asterisk using H.323 or Skinny , somebody tested this phone with asterisk and can share experience? Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040113/2ea43296/attachment.htm

I'm wanting to have a phone at home next to the garage door that when my bride comes home, she can see that there is a new message, push a button and have the messages played to her. Otherwise, she will not let me install asterisk on my home line. Can someone suggest relatively inexpensive hardware that will do this for me (us)? Thanks, -Peter

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

....... > >Bug generraly author of that article is an idiot. He does not >understand the difference beteween VOIP and ISDN PRI. > > >-----Original Message----- >From: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of listas >iPfone >Sent: Wednesday, July 07, 2004 6:26 PM >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID > >This is very interesting... > >Regulations..USA... > >But... what can i do faking a caller id? stolen what? what is the point? >...

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on