search for: ipfone

Displaying 20 results from an estimated 21 matches for "ipfone".

2004 Jun 16
2
embedded Asterisk
..., 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-...
2004 Jan 16
1
ERROR[8192]
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/atta...
2003 Aug 04
14
Mysql CDR
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
2003 Sep 23
2
error message playing .mp3
> -----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 directo...
2003 Dec 08
2
snom X MOH
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:
2005 May 26
1
VIDEO ON 1.0.7 stable
--- 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> canreinvit...
2003 Oct 31
2
MOH problem
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/attachment...
2004 Jan 23
6
rc.local dont works
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
2003 Nov 27
13
Asterisk behind NAT << How to do it.
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
2003 Oct 08
7
chan_capi and latest Debian package
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
2007 Dec 21
1
Asterisk SIP handling - why 491 Request Pending response
Hi, I have the following situation I use asterisk as o gateway between networks. What is the reason for such response? What are 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
2008 Jan 04
1
Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)
2003 Oct 06
6
Alternatives to FXS cards?
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
2004 Jan 26
0
Digium FXO Card
...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...
2003 Oct 02
3
Xten Lite Build 1079
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
2004 Jan 13
1
Symbol NetVision Phone
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
2005 May 26
5
Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages
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
2003 Sep 24
10
SIP / GrandStream Configuration
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
2004 Jul 08
2
Shady dial anyone??
....... > >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? > &...
2004 Jun 17
4
Problems with PRI with T410 messages
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