Re: [Asterisk-Users] Voicemail for Toshiba dk280
On Wednesday 28 April 2004 03:37 pm, Barton Hodges wrote: > I would like to use Asterisk for voicemail, connected to a Toshiba > dk280. Has anyone done this with this model or similar system? Are > there any documents available that could give me some insight into how > I can do this? You may want to see: http://www.voip-info.org/tiki-index.php?page=Asterisk%20legacy%20integration ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P FXO, 2 slots?
On Sunday 02 May 2004 08:56 pm, Jamin W. Collins wrote: > On Sun, May 02, 2004 at 09:07:37PM +0100, Kevin Walsh wrote: > > The same Digium shop page suggests that two PCI slots would be required. > > I'll assume the card is too fat, with the daughter board(s) fitted, to > > fit into a single slot. > > This is something I would like to see confirmed, does this card really > take 2 pci slots? No, the TDM400P does not take up 2 slots. In fact, in my box I an IDE card in slot 3, next to it I have the TDM400P in slot 4, and a X100P right next to it in slot 5. No space problem at all. Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Sunday 02 May 2004 08:07 pm, Kevin Walsh wrote: > Someone on IRC once pointed out the conflict between "suggested" and > "must" on a similar page and said that their TDM400P (FXS-only at the > time) was working on a PCI 2.1 system. Can anyone confirm whether a > PCI 2.2 bus is mandatory? Yes, PCI 2.2 _is mandatory_. I know because I just upgraded my TDM400P card on an RMA, and the gentlemen tech emphasized that the motherboard for my new TDM400P card MUST have PCI 2.2. Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Security Issue in Asterisk with sip.conf configuration.
uhm, strange but does this work on your setup? even with permit and
deny, if a user is not matched in the conf, it is allowed access to the
default context stated in the conf.
On Wed, 2004-04-28 at 16:12, James H. Thompson wrote:
> I think the problem is that using permit= alone does nothing.
> You need to combine it with a deny= as in:
>
> deny=0.0.0.0/0.0.0.0 ; deny all
> permit=123.123.123.123 ; allow only this address - netmask defaults to:
> /255.255.255.255
>
> order matters, the deny needs to come first.
>
> for reference here is the code from acl.c that checks the rules:
>
> int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin)
> {
> /* Start optimistic */
> int res = AST_SENSE_ALLOW;
> while(ha) {
> /* For each rule, if this address and the netmask = the net address
>apply the current rule */
> if ((sin->sin_addr.s_addr & ha->netmask.s_addr) == (ha->netaddr.s_addr)
> res = ha->sense;
> ha = ha->next;
> }
> return res;
> }
>
>
> Jim
>
> James H. Thompson
> [EMAIL PROTECTED]
>
> - Original Message -
> From: "William Zhang" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, April 27, 2004 2:43 PM
> Subject: [Asterisk-Users] Security Issue in Asterisk with sip.conf configuration.
>
>
> > I had tried many ways with some advanced user help, but without
> > success(at one point I thought I had it worked).
> >
> > Here Asterisk is working as a SIP PSTN Gateway, and in the sip.conf
> > file, there are a lot of entries with just "host=a.b.c.d", thinking
> > that * will only accept calls from host "a.b.c.d", but in my test, no
> > mater how you set up the sip.conf entries, either * will NOT accept
> > calls for that user account at all, or it will accept calls from any
> > where without VERIFYING the source IP(whether it is "a.b.c.d" or not),
> > so long the sip userid is the username in sip.conf. This post a very
> > serious security problem.
> >
> > Of course we can put "secret=" for each entries, but giving Asterisk GW
> > and SIP proxy are in 2 TRUSTED IPs, no Authentication is neccessary,
> > otherwise it increase the SIP traffic quite a bit.
> >
> > Following are the 4 different entries that I had tried:
> > #Notice that in the "general" section, context is pointed to a none
> > existant context "INVALID".
> >
> > ;
> > ; SIP Configuration for Asterisk
> > ;
> > [general]
> > port = 5060 ; Port to bind to
> > bindaddr = 212.213.66.68
> > context = INVALID ;
> > ;srvlookup = yes; Enable SRV lookups on outbound calls
> > ;pedantic = yes ; Enable slow, pedantic checking for
> > Pingtel
> > ;tos=lowdelay
> > ;tos=184
> > ;maxexpirey=3600; Max length of incoming registration
> > we allow
> > ;defaultexpirey=120 ; Default length of incoming/outoing
> > registration
> > ;notifymimetype=text/plain ; Allow overriding of mime type in
> > NOTIFY
> > ;videosupport=yes ; Turn on support for SIP video
> > disallow=all; Disallow all codecs
> > allow=ulaw ; Allow codecs in order of preference
> > allow=g729
> > allow=ilbc
> > ;
> > ;dtmfmode=info
> > ;dtmfmode=inband
> > dtmfmode=rfc2833
> >
> >
> >
> > [20034]
> > type=friend
> > callerid=TEST <61331045>
> > host=212.213.65.66
> > nat=yes; This phone may be natted
> > canreinvite=no
> >
> > [20035]
> > type=peers
> > callerid=TEST <61331045>
> > host=212.213.65.66
> > nat=yes; This phone may be natted
> > canreinvite=no
> >
> > [20036]
> > type=friend
> > context=default
> > callerid=TEST <61331045>
> > host=212.213.65.66
> > permit=212.213.65.66
> > nat=yes; This phone may be natted
> > canreinvite=no
> >
> > [20037]
> > type=peers
> > context=default
> > callerid=TEST <61331045>
> > permit=212.213.65.66
> > nat=yes; This phone may be natted
> > canreinvite=no
> >
> > Thank you in advance.
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP: Current CVS works for you?
Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542 http://bugs.digium.com/bug_view_page.php?bug_id=881 and other MGCP related bugs/fixed. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New ENUM service, what do you think?
John Todd wrote: TRIP (RFC 3219) is the answer, but I'm the only one pounding that drum, it seems. If anyone here on the list has $100,000 to put together a real programming effort towards getting that implemented, y'all let me know. The longer this waits, the more lame and broken become the solutions offered. One small oversight in your thinking, something like TRIP will only benefit large telcos and VOIP providers with interconnects, I don't see this flowing down to a tangible benefit to the average person, where as something like enum.164 is. TRIP is based on BGP and BGP already does most of the IP routing smarts TRIP is supposed to be beneficial for, however that $100k would be better spent on improving the smarts in the call routing software rather then turning things back into a hub and spoke model, p2p is way more efficient if it can be utilised to it's full potential. At this stage the only potential method to prevent VOIP spam is something like SPF records, which would only end up duplicate enum. It's a lot harder to get phone numbers then IP addresses, so this would overcome people's concerns about dynamically allocated IPs, phone numbers aren't. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk remains in the media path
Le lun 03/05/2004 à 18:48, Jeremy McNamara a écrit : > Actually its cuz chan_h323 sucks like that. Correct me if I'm wrong, but I browsed the archives and I got the feeling that you (Jeremy) were one of the main developers of the chan_h323... aren't you a little harsh about your own work? :-) Anyway, is there any plan in the chan_h323 roadmap to support direct RTP between endpoints? Thanks, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Siemens cordless phone
The SDK for the Siemens USB cordless phone was just released a few days ago. I understand from a few people I spoke with when this was first released that this could be ported to work for Asterisk. Does anybody have any thoughts now they have seen the sdk information? Cheers, Dean Gigaset M34 USB PC Adapter offers an open interface to enable third party solution providers to integrate cordless phones into their applications for VoIP, messaging and home control. To encourage developers Siemens provides a free Software Development Kit (SDK), Internet-based support, as well as 24 hours hotline via the Siemens mobile developer portal: http://www.siemens-mobile.com/developer. The portal contains detailed information about the interface, the SDK and the hardware and tool environment. <>
Re: [Asterisk-Users] grandstream transfer, park and conference
Hi! > I have 2 grandstream budgetone 100 series. I can call allright, but I > cant do call transfer, park and call conference. (all features works > with tdm devices) the 1. Check if Asterisk is always in the media path, i.e. you need the t or T option (or something similar) in your Dial statement. Alternatively you could introduce a canreinvite=no in sip.conf for the GS phones. 2. Check your context setup in extensions.conf and make sure that in call cases your GS phone has the "parkedcalls" context available Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - no outband DTMF with Mediatrix
Dear List members, I have this problem with Mediatrix 24-FXS-line gateway and out-of-band DTMF. It seems not working - the RTP mode is not working and when I select INFO mode, the Mediatrix is behaving just the same as with RTP mode. Here is a bunch of logs to explain this: 1. The RTP out-of-band mode (dtmfmode=rfc2833): This is OK reply from Asterisk to Mediatrix when RTP mode selected. Seems OK ;-): [...] SIP/2.0 200 OK CSeq: 1091919829 INVITE v=0 o=root 35059 35059 IN IP4 xxx s=session c=IN IP4 xxx t=0 0 m=audio 12210 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 [...] And then, during connection with asterisk, when we use DTMF, this shows on debug: [...] May 3 17:49:42 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec 96 received May 3 17:49:42 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec 96 received [...] 2. INFO mode (dtmfmode=info): Proper INVITE from Mediatrix: [...] INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 CSeq: 1657017135 INVITE Content-Type: application/sdp Contact: Port 3 Supported: replaces User-Agent: MxSipApp/4.4.11.68 MxSF/v3.2.7.30 v=0 o=MxSIP 4563726510189014186 6429835688411497953 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 a=sendrecv m=audio 5004 RTP/AVP 8 18 4 0 13 111 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:111 X-nt-inforeq/8000 [...] And then nothing happens, Asterisk shows no DTMF events. Thanks for any help, Arek Bekiersz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New ENUM service, what do you think?
You may be quite right, I have read parts of the rfc at least, I remember, but the lure of using cheap existing infrastructure is probably to great. KHB - Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, May 04, 2004 03:20 Subject: RE: [Asterisk-Users] New ENUM service, what do you think? > While I wish the guys at Stealth the best of luck, I'll say again > that ENUM is _NOT_ the solution for VoIP routing in the current real > world. See the mailing list archives for more of my rants on why DNS > is not the answer for cost-based routing (where "cost" is monetary, > distance, qos, or any other comparative metric.) > > TRIP (RFC 3219) is the answer, but I'm the only one pounding that > drum, it seems. If anyone here on the list has $100,000 to put > together a real programming effort towards getting that implemented, > y'all let me know. The longer this waits, the more lame and broken > become the solutions offered. > > JT > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Resolved: sipgate.de
I know it's exciting to get things working, however,
there are some things wrong with your configuration, despite it perhaps
"working" ok.
Is it really? You can make outbound calls this way?
In your friends definition (friend-sipgate) you don't have a host
specified.
host=sipgate.de
Without that I doubt you can make any calls, since asterisk won't know where
to
send the call to.
Further, since you're using fromdomain, it should be the authentication
realm,
which is sipgate.de, not sipgate.net. But this won't hurt your call
completion
Fromdomain will get placed into the From headers instead of your ipaddress.
some domains are picky about it when you're using special services and they
want to make sure you're actually a domain member.
Also, your localnet= parameter should be the network address, not the host
address, but you're probably ok, since the mask cuts it off.
Since you don't have a valid friends definition, your incoming calls come
into
the default context, and you need to be carefull what you make available
there.
It's never a good idea to have calls coming in this way, without restriction
or authentication.
Enjoy
- Original Message -
From: "Jay Milk" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, May 03, 2004 15:23
Subject: [Asterisk-Users] Resolved: sipgate.de
> (History: Getting my home asterisk system set up; one land-line,
> multiple SIP trunks; goal is to have a wife-proof transparent phone
> system)
>
> Just wanted to let everyone know that I got sipgate.de working with my
> asterisk system; relevant settings below:
>
> --account--
> Sipgate number 8001234 (change to suit yours)
> Password "password"
>
>
> --network--
> External static IP routes to internal 192.168.254.204 (static)
>
>
> --sip.conf--
>
> [general]
> port=5060
> bindaddr=192.168.254.204
> externip=x.x.x.x ; insert your external IP here
> localnet=192.168.254.204
> localmask=255.255.255.0
> nat=yes
>
> register => 8001234:[EMAIL PROTECTED]/99049 ; 99049 =
> incoming/Germany
>
> [friend-sipgate]
> username=8001234
> secret=password
> fromuser=8001234
> fromdomain=sipgate.net
> type=friend
> nat=yes
> dtmfmode=rfc2833
> canreinvite=no
>
>
> --extensions.conf--
>
> exten => _01149.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30)
> exten => _01149.,2,Congestion
>
> exten => 99049,1,Wait,1
> exten => 99049,2,Answer
> exten => 99049,3,Dial(SIP/sipura2b,30)
>
>
> Hope this helps someone else. The "register" allowed me to receive
> incoming calls, but outgoing calls failed until I set the externip and
> nat settings.
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Probs with oh323 driver: audio only in 1 direction
Hi,
try to setup asterisk as an ISDN2H323-Gateway. The only problem
i have after establishing a call is, that Audio works only from IP to
ISDN-Phone but not from ISDN to IP-Phone.
A known problem ???
Thanks in advance
Michael
i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus)
and oh323-0.6.0
Here are my config's
##
# modem.conf #
##
[interfaces]
context=isdn
driver=i4l
language=en
stripmsd=0
dialtype=tone
mode=immediate
msn=8540340
context=8540340
device => /dev/ttyI0
device => /dev/ttyI1
##
#extensions.conf #
##
[general]
static=yes
writeprotect=no
[8540340]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,Dial(OH323/192.168.70.227)
[voip-h323]
exten => s,1,Answer
exten => s,2,Dial,Modem/ttyI1:${OH323_DSTE164}
##
# oh323.conf #
##
[general]
listenAddress=192.168.70.1
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=no
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
gatekeeper=192.168.70.1
gatekeeperTTL=600
userInputMode=TONE
amaFlags=default
accountCode=H323
context=voip-h323
[register]
alias=isdn
gwprefix=8
[codecs]
codec=G711A
frames=20
--
Michael Niehren __ _ powered by
/ / (_)__ __ __
/ /__/ / _ \/ // /\ \/ /
//_/_//_/\_,_/ /_/\_\
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New ENUM service, what do you think?
While I wish the guys at Stealth the best of luck, I'll say again that ENUM is _NOT_ the solution for VoIP routing in the current real world. See the mailing list archives for more of my rants on why DNS is not the answer for cost-based routing (where "cost" is monetary, distance, qos, or any other comparative metric.) TRIP (RFC 3219) is the answer, but I'm the only one pounding that drum, it seems. If anyone here on the list has $100,000 to put together a real programming effort towards getting that implemented, y'all let me know. The longer this waits, the more lame and broken become the solutions offered. JT At 1:28 PM -0400 5/2/04, Joe Baptista wrote: On Sat, 1 May 2004, Dean Collins wrote: Yes but no information about how this will operate, what regulation or restrictions on joining, what connection protocols will be used etc etc agreed - you see alot of business fluff - but the technicals are very important and on many of these ventures they fail to include them. regards joe www.baptista.god Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Reid A. Forrest Sent: Saturday, 1 May 2004 8:21 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New ENUM service, what do you think? >From http://www.thevpf.com/ To join, please e-mail [EMAIL PROTECTED] or telephone 1-212-232-2020 (Mon-Fri 9AM-5PM EST). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jimfl Sent: Saturday, May 01, 2004 5:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New ENUM service, what do you think? >Jim/frank, >Can you give us more information about how to access this enum? I've >been to the stealth web site and there is no information about access. > >I look forward with interest to what you have up and running today for >asterisk users to benefit from. > >Cheers, >Dean Sorry, I am not associated with Stealth in any way. Just saw the news story and thought it would be of interest to Asterisk users. It sounds like you don't have to be a VOIP provider to get access to their service. They talk about businesses using the service. If anyone finds out how to get access to their service, please post. > Jim > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Czech sound files
Hi, if there is somebody working on Czech support please contact me off list, so we can work together. Petr Mosnicka -- YieldTech - linuxova divize ATAX Group, spol. s r.o. V zavetri 6 tel: +420-777-2LINUX 170 00 Praha 7 mailto: [EMAIL PROTECTED] Ceska republika http://www.YieldTech.cz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN WAN ISDN bridge possible?
Patrick, doe a google search for "ISDN over IP", maybe that's your solution. jo Patrick Stuckenberger wrote: Hi list, is it possible to create something like a ISDN-WAN-WAN-ISDN bridge? We have to change our location, but our number and the telephone system should shoulb stay the same. kind regards, Patrick Stuckenberger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
