Re: [asterisk-users] Check for the voicemail
Ok: this is the complete recipe. first, make a sendmail.cf configuration file so that you are sure that sendmail tries to deliver directly to your server, and not to spool the file locally in order to send it later. cp /etc/mail/sendmail.cf /etc/mail/direct-delivery-sendmail.cf vi +/^DS /etc/mail/direct-delivery-sendmail.cf end modify the line that begins with DS (for smart relay host) into: DSyour.exchange.server.name save and then write a script like this: #!/bin/bash # this is the location of the temporary file for redirecting stdout- anything unique will go TEMPFILE=/dev/shm/checkeddelivery.$$ /usr/lib/sendmail -v -g -C /etc/mail/direct-delivery-sendmail.cf $* 21 /dev/shm/checkeddelivery.$$ if [ $? != 0 ]; then do domething: sendmail returned an immediate local error elif [ $( grep -c ^250\\.2\\.0\\.0 $TEMPFILE) == 0 ]; then something: sendmail did not get a 250 2.0.0 OK message from the remote server just check that your.exchange.server.name answers wth 250 2.0.0 OK, or adjust the script accordingly ... here you can also check for some different errors and behave differently for over quota, fi #on the first runs, you may leave the file in order to inspect it rm $TEMPFILE and invoke the script as: myscript recipientaddress piping over standard input the complete mail you want to send (not only the .wav attachment) On Wed, Aug 22, 2012 at 12:09 PM, Danilo Dionisi dionisi.dan...@gmail.com wrote: How can I, with a bash script to take the standard output? When I take the standard output, I'll do the grep to see if there is a code 450. Right? Il 22/08/12 11:56, Roberto Piola ha scritto: no. when you issue sendmail -v , the output is sent on the standard output -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roberto Piola, Ph.D. Senior Network Engineer Outsourcing Infrastructure VISIANT OUTSOURCING strada del Drosso 128/6 - 10135 Torino T +39 011 3473520 - T +39 02 45413318 - F +39 011 3473522 M +39 3356961505 roberto.pi...@visiant.it www.visiantoutsourcing.it Questo messaggio è destinato alle sole persone indicate e può contenere informazioni riservate. Se avete ricevuto questo e-mail per errore siete pregati di comunicarlo immediatamente al mittente o di inviare un e-mail a: info.outsourc...@visiant.it. Ogni altro uso del messaggio è vietato. This e-mail may contain confidential and/or privileged information. It is intended solely for the addressee. If you are not the intended recipient please notify immediately the sender or email: info.outsourc...@visiant.it. All other use is prohibited. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Check for the voicemail
I would simply send the message with sendmail -v and then grep the output for the error message Il giorno 22/ago/2012 04:19, Raj Mathur (राज माथुर) r...@linux-delhi.org ha scritto: On Tuesday 21 Aug 2012, Ruben Rögels wrote: Hello, no problem at all, I think this is the tricky part. A smtp dialogue between your email client and a smtp server normally looks like this: user@box:~? netcat mx1.example.com 220 postfix ESMTP mx1.example.com helo me.local 250 mx1.example.com mail from: ruben.roeg...@wiseape.de 250 2.1.0 Ok rcpt to: ruben.roeg...@example.com 450 5.7.1 ruben.roeg...@example.com: Mailbox Full The tricky part is writing or finding a console smtp client that gives you feedback about the 450 error that just happened. Right now I cannot give you a precise way to do that, but I have basic understanding of the technology, so I know that it is possible to do so ;-) I'm looking around in the net, because I think I'll soon have to handle your problem aswell in my company ;-) If I can find solution, I'll post it. Something like this ought to do it: (sleep 5; echo HELO foo; sleep 1; \ echo mail from: f...@example.com; sleep 1; \ echo rcpt to: userid.t...@youwant.to.check; sleep 1; \ echo data; echo test; echo .; sleep 1; echo quit) | \ telnet mail.ho.st 25 21 | fgrep -q '450 5.7.1' notify-user.sh Of course, it's probably better to wrap this into a Perl or equivalent script, but it should work on the shell too. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about EuroBRI final 2 digits
In Italy, you must enable overlapdial=yes On Thu, Feb 3, 2011 at 7:45 PM, Cassius Smith cass...@cassius.org wrote: Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to switch on electric heaters remotely?
we're using a Damocles Mini (http://www.hw-group.com/products/damocles/damocles_mini_en.html). of course, the damocles will have to drive a high-power relay. the damocles can be driven via snmp, so you'll have to simply call the snmpset unix standard utility On Mon, Oct 18, 2010 at 1:24 PM, Gareth Blades list-aster...@skycomuk.com wrote: Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only control 750W so you will probably need to get it to control a more powerfull relay as a heater is going to take a lot of current. It can be controlled by a virtual serial port so you just program the extension to make a system() call to a simple script which sends a string of characters to the serial port. That device is quite expensive. You could probably find something much cheaper on ebay. Gilles wrote: Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roberto Piola, Ph.D. Senior Network Engineer Outsourcing Infrastructure VISIANT OUTSOURCING strada del Drosso 128/6 - 10135 Torino T +39 011 3473520 - F +39 011 3473522 M +39 3356961505 roberto.pi...@visiant.it www.visiantoutsourcing.it Questo messaggio è destinato alle sole persone indicate e può contenere informazioni riservate. Se avete ricevuto questo e-mail per errore siete pregati di comunicarlo immediatamente al mittente o di inviare un e-mail a: info.outsourc...@visiant.it. Ogni altro uso del messaggio è vietato. This e-mail may contain confidential and/or privileged information. It is intended solely for the addressee. If you are not the intended recipient please notify immediately the sender or email: info.outsourc...@visiant.it. All other use is prohibited. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User-invoked call restrictions
Hint: an AGI application that looks into a database for passwords, and the decides, according to the prefix, if the call is allowed or not On Wed, Aug 26, 2009 at 4:39 AM, David A. Bandeldavid.ban...@gmail.com wrote: Folks, Had a request from a customer: is it possible for a customer, using a password to restrict others from making long distance/cell calls? That is, the user set a level of service? Something like this: Customer dials a number -- operator asks for password, then service level (another number). Service level would be something like: 1 - allow inbound calls only 2 - allow 1 + local/toll free calls 3 - allow 2 + long distance national 4 - allow 3 + cell calls 0 - allow 4 + international calls (basically cancel all call restrictions) Dialing to restricted zones would evoke a message from the operator that the phone is blocked by owner. Code examples? Hints? RTFM URL? TIA, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto Visit my blog at: http://www.pananix.com/cgi-bin/blosxom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roberto Piola, Ph.D. Senior Network Engineer Outsourcing Infrastructure VISIANT OUTSOURCING strada del Drosso 128/6 - 10135 Torino T +39 011 3473520 - F +39 011 3473522 M +39 3356961505 roberto.pi...@visiant.it www.visiantoutsourcing.it Questo messaggio è destinato alle sole persone indicate e può contenere informazioni riservate. Se avete ricevuto questo e-mail per errore siete pregati di comunicarlo immediatamente al mittente o di inviare un e-mail a: info.outsourc...@visiant.it. Ogni altro uso del messaggio è vietato. This e-mail may contain confidential and/or privileged information. It is intended solely for the addressee. If you are not the intended recipient please notify immediately the sender or email: info.outsourc...@visiant.it. All other use is prohibited. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QoS VPN
I do not have examples, but if you are using the 1700 series router in order to originate the ipsec vpn, you may use command qos pre-classify (please search for it on cco.cisco.com) On Thu, May 7, 2009 at 9:54 PM, Brent Davidson br...@texascountrytitle.comwrote: I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that I need to VNC in to machines at the various offices for tech support while the user is also on the phone. Unfortunately the VNC connection apparently takes priority and makes it impossible for me to understand anything the person on the phone is saying, although they can still hear me fine. Our Main office uses a Cisco PIX 506 for the main firewall and VPN concentrator. Each branch office used a Cisco 1700 series router with IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the main router or the branch routers to make sure the voice quality takes precedence over the VNC? (Any example configs would be greatly appreciated) Would I be better off routing the voice packets over the internet rather than the VPN, and could I safely do that without exposing the asterisk boxes to unnecessary security risks? (At present all of our asterisk boxes are behind the firewalls and only talk to each other over the VPN. All PSTN connection is done through TDM boards so they have no direct exposure to the internet.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 7, Issue 323
I fear that list digest did not forward to me all the messages... buying cell phone adapters is quite unfeasible at this point, since the installation at hand uses 8 BRI for outgoing calls, and the customer negotiated very special rates for handling all the traffic through his voice carrier. Moreover, in italy you have 4 cell phone operators, and you should add a bunch of call phone adapters for each of them (call.operator A to cell.operator B costs much more than land operator X to any cell. operator) in another installation, with a single 4BRI card conntected to point-to-multipoint ISDN lines, everything works fine (even calls to cell phones), and land calls are fine. this is the problem that puzzles me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, February 28, 2005 6:14 PM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 7, Issue 323 Date: Mon, 28 Feb 2005 18:07:42 +0200 From: Mark Elkins [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain On Mon, 2005-02-28 at 17:50 +0200, Herman Cremer wrote: http://www.psitek.co.za/gsm.html These guys are also in RSA, and Australia. This unit does exactly the same as the DigiCell, which mark is talking about, but is a much better product (and more expensive) maybe they export ? Except that when I've been to the USA - I've needed a 1900Mhz phone - this is only 900 and 1800... *GSM INTERFACE * GSM output 900MHz: Class 4/5, 2W EGSM * GSM output 1800MHz: Class 1, 1W DCS * SIM interface: 3V mini SIM but look at the website (Hey, it looks like my box!) as the features are what you are looking for... I believe Motorola was one of the earlier producers of this type of device - but would think that most of the manufacturers would have a similar type of unit. Push the Telephone access in disaster areas, where wire-network infrastructure is damaged point... :-) -Herman On Mon, 2005-02-28 at 17:21, Mark Elkins wrote: On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote: Hello Mark , C. All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL No idea. The Unit I have is a locally manufactured device called Digi-Cell - frmaritz (at) global.co.za is the email address on the box it came in Its probably 900Mhz GSM only - in the US - You'll need a 1900Mhz unit??? On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phone problem
We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10) and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are configured in TE mode and connected to the PSTN; the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear, while the speaker at the internal side hears perfectly. CPU usage is quite low, and asterisk -rvvv does not show anything particular Any suggestion ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Problem with a new italian service provider...
We got the same problems in connecting to uni.it . the problems were solved (under indications of Mr. Cardone, from Unidata) by using the ip address of the machine as the SIP account, and by using two different IP addresses, on the asterisk server: one for outgoing calls and one for incoming ones. -Original Message- Message: 1 Date: Tue, 30 Nov 2004 18:32:55 +0100 From: Massimo De Nadal [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem with a new italian service provider... To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I've a problem connecting uniVoice (http://voice.uni.it) from asterisk. Using my account data I can place a call smoothly using xlite or my budgetone phone directly, but I'm not able to use uniVoice as a peer from asterisk. Registration seems to work correctly, but when I try do dial, the sip authentication fails every time. Their tech people told me that they are unable to make asterisk working properly due to some asterisk authentication features lacks. The problem persists using chan_sip2 too. So, the question is: how can I help investigating the problem to obtain some good bugs report for digium ?? maxx ___ 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] Problems in autnenticating with SER / PortaSIP
We have a problem in authenticating with a SIP server running PortaSIP.
first, my exten.conf says:
exten = _396262X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _39064040.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
and sip.conf:
register=390645416983:[EMAIL PROTECTED]/390645416983
[to-uni]
type=peer
secret=XX ; i tried also using md5secret= instead of secret=... but
it's the same
username=390645416983
fromuser=390645416983
host=sip.uni.it
nat=yes
our asterisk pbx correctly registers on sip.uni.it (it is displayed as
registered in sip show registry, and if I issue a sip debug I see the
answer to the registration, correctly reporting the name of the remote
server and our balance:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK1bd57ff9
From: sip:[EMAIL PROTECTED];tag=as4fb9a73e
To: sip:[EMAIL PROTECTED];tag=a4a48d8b20978897d8e0f5c399e6cc29.fbc4
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
PortaBilling: available-funds:5.00 currency:EUR
Contact: sip:[EMAIL PROTECTED];q=0.00;expires=115
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0
The problem is when I try to call a number on the othere side
(39064040): the call is correctly routed, the remote server asks us for
the proper credentials, and it seems to me that asterisk answers their
challenge:
Authorization: Digest username=390645416983, realm=sip.uni.it,
algorithm=MD5, uri=sip:[EMAIL PROTECTED],
nonce=419358e858969bef4a5c77326f2b205b97c672bf,
response=188824ee848f9ed095990999fb2e3893, opaque=
but for some reason it seems that the remote server does not like the
answer. the helpdesk of uni.it says that this is an old bug of asterisk
(actually, the account works with an X-Lite softphone ).
I'm using CVS-v1-0-11/08/04-10:57:05. I hoped that the latest version
corrected this problem as well, but it appears that it is not the case
I enclose the sip debug trace of the call
-- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/[EMAIL PROTECTED]) in
new
stack
We're at 217.18.104.75 port 10880
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 11 Nov 2004 12:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 12361 12361 IN IP4 217.18.104.75
s=session
c=IN IP4 217.18.104.75
t=0 0
m=audio 10880 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 217.72.100.4:5060
-- Called [EMAIL PROTECTED]
janis*CLI
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport=5060
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED];tag=a4a48d8b20978897d8e0f5c399e6cc29.b98a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: Digest realm=sip.uni.it,
nonce=419358e858969bef4a5c77326f2b205b97c672bf
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0
9 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED];tag=a4a48d8b20978897d8e0f5c399e6cc29.b98a
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 217.72.100.4:5060
We're at 217.18.104.75 port 10880
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK588b0624;rport
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Authorization: Digest username=390645416983, realm=sip.uni.it,
algorithm=MD5, uri=sip:[EMAIL PROTECTED],
nonce=419358e858969bef4a5c77326f2b205b97c672bf,
response=188824ee848f9ed095990999fb2e3893, opaque=
Date: Thu, 11 Nov 2004 12:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 12361 12362 IN IP4 217.18.104.75
s=session
c=IN IP4 217.18.104.75
t=0 0
m=audio 10880 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 217.72.100.4:5060
janis*CLI
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK588b0624;rport=5060
From: Roberto Piola sip:[EMAIL PROTECTED];tag=as2fb0ecc6
To: sip:[EMAIL PROTECTED];tag=a4a48d8b20978897d8e0f5c399e6cc29.5919
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
WWW
[Asterisk-Users] Re: Asterisk WITH Swyx... Any Idea?
We use asterisk coupled with swyxIt via a gnugk (version 2.0.8) and oh323 module: both swyx server and asterisk register on the gnugk. asterisk receives sip calls from the exterior and routes them to the gk. I've set up a prefix on swyx so that if I prepend +996 to my phone numerb, the call gests routed to asterisk (which, in turn, strips the prefix and sends the call via iptel or iaxtel. H.323 phones register on swyx. SIP phones register directly to asterisk) -- Message: 11 Date: Wed, 25 Aug 2004 19:35:16 +0200 (CEST) From: Zineddin Karzazi [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk WITH Swyx... Any Idea? To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 --- Loek Gijben [EMAIL PROTECTED] schrieb: Hello Zineddin, Hi, I'm a student and my thesis work consist in testing Asterisk with Swyx(SwyxWare). Until 2 years I was a student too so I think I can still relate to your state of mind :^) My approach is to declare asterisk as h323 gateway for the Swyxserver using oh323 Plugin. Is there any possibility to connect Asterisk with Swyx? how? Commercially we took a look at Swyx, it has great Windows (Active Directory) integration. But despite Swyx told us for nearly a year now that SIP support was coming we still haven't seen anything yet. So we left Swyxware where it belongs: on the shelf ;-)) 95% of innovations in VoIP are based on SIP I wonder why uou want to set up a system like this. Merely for testing purposes? Or does it have real life implications? Yes!! It s just for testing Purpose. If calling with swyxit is imminent then you can bypass the Swyxserver alltogether. And the Swyxserver can be hung on PSTN also, so if you need the AD integration than you can bypass the * server. I?m Tryin to connect the PBX without using ISDN or any Hardware. i already have a Nikotel Account to Be reachable under a regular phone number. but this is not possible to achieve with Swyx because it is based on H.323 and not SIP. I know this does not answer you question, and I'm not into H323. But like all other * users I've plowed myself through the Wiki and Googled a lot for answers. So if even I stumble on H323 on Asterisk info then it must be possible for you too. IMHO it must be possible to set up a system you describe, my hunch would be installing * on a testPC with H323 support, then first try to attach some softphones (like Swyxit) and the route through Swyxserver. Already Done for testing purpose. * with OH323 Plugin, works with 12 PC Clients (using Xlite(SIP),OpenPhone(H.323),IAXCOMM)and can also make calls to PSTN and be reachable from outside the LAN. The Problem is,that i cant use the Swyxit/Handsets or the Swyx IP-Phones. Succes with graduation! Loek Gijben Remotica Thank you. ___ 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] Re: telnet and Root
you can login as root only on the console or on the lines listed in /etc/securetty if you want to log in remotely as root, you can either: - log in as a regular user and then issue the su - command in order to become root - use a ssh client (secure shell) instead of telnet (well, you can disable root access in ssh as well) -- Original message: From: neil [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Fri, 20 Aug 2004 10:39:03 +0100 Subject: [Asterisk-Users] telnet and Root Sorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!! I am running Fedora 1 and I can telnet in to my asterisk box as any user except root and am using the same credentials as logging in locally. I am new to Linux and any help would be gratefully appreciated. Thanks ___ 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] H323 under asterisk RC1 ?
This evening I tried to install asterisk RC1 .rpm on a Fedora box... but how can I re-add openh323 support? or does it contain an alternate h323 support? thanks 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] Strange message, and one-way audio between sip and H.323
we are trying to use asterisk for converting SIP to H.323 calls. asterisk (0.9.1) runs on the same linux (Redhat 8) box of our gatekeeper (gnugk version 2.0.8). the calls are going out through a cisco gateway. when I make a call from a SIP phone to a PSTN number reachable through the cisco gateway: asterisk diaplays Aug 5 23:24:26 WARNING[1255648560]: chan_oh323.c:2898 alerted_h323_connection: Call ip$localhost/22666 in unexpected state (PLAYONLY). I hear (on the SIP phone) clearly what the other person is saying, but the other person (on the PSTN side) hears nothing from me. gatekeeper and the cisco gateway work fine, when using H.323 terminals. The result does not change if I use a IAX phone instead of a SIP one. The gatekeeper configuration contains [RoutedMode] GKRouted=1 H245Routed=1 and has no Proxy Section I tried also [RoutedMode] GKRouted=0 H245Routed=0 and I also tried to enable the [Proxy] function on the gatekeeper, but the result is the same I tried to search the internet for the message, but I got no results Roberto Piola, Ph.D. Senior Network Engineer Divisione VAIPS - SOFTPEOPLE - IHNET .: Strada del Drosso 128/6 - 10135 Torino .: tel. +39 011 3473520 - mob. +39 335 6961505 - fax. +39 011 3473522 .: mail:[EMAIL PROTECTED] .: http://www.softpeople.it .: http://www.ihnet.it Business Unit di SOFTPEOPLE - Questo messaggio è destinato alle sole persone indicate e può contenere informazioni riservate. Se ricevuto per errore, si prega di avvisare immediatamente il mittente e cancellare l'originale. Ogni altro uso del messaggio è vietato. ___ 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
