RE: [asterisk-users] dundi problem * 1.4.2
Hi,
I'm not working with Asterisk 1.4.X, but i think your Dundi.conf is not
correct.
If you want i can send you my complete working exemple with Asterisk 1.2.x
(I think the config is the same)
Fred
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Remco Post
Envoyé : mardi, 24. avril 2007 23:15
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] dundi problem * 1.4.2
Hi All,
I've been banging my head on a small dundi problem...
I have two * servers setup, both have almost identical dundi.conf files:
[EMAIL PROTECTED]:/opt/asterisk/etc# cat dundi.conf
[general]
department=thuis
organization=pipsworld
locality=Amsterdam
stateprov=NH
country=NL
[EMAIL PROTECTED]
phone=+31207508308
;bindaddr=0.0.0.0
;port=4520
entity=00:02:b3:49:69:5e
ttl=16
autokill=yes
;secretpath=dundi
[mappings]
;pipsworld =
pipsworld,1,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
;pipsworld =
external,1000,IAX2,[EMAIL PROTECTED]/31207508308,nounsolicited,nocomun
solicit,nopartial
[02:60:8c:f2:3e:aa]
model = symmetric
host = pipc.pipsworld.nl
inkey = pipsworld
outkey = pipsworld
include = pipsworld
permit = pipsworld
qualify = yes
and:
[general]
department=thuis
organization=pipsworld
locality=Amsterdam
stateprov=NH
country=NL
[EMAIL PROTECTED]
phone=+31207508308
;bindaddr=0.0.0.0
;port=4520
entity=02:60:8c:f2:3e:aa
ttl=16
autokill=yes
;secretpath=dundi
[mappings]
pipsworld = pipsworld,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
; pipsworld =
external,0,IAX2,[EMAIL PROTECTED]/${NUMBER},nounsolicited,nocomunsolic
it,nopartial
[00:02:b3:49:69:5e]
model = symmetric
host = tsjonge.pipsworld.nl
inkey = pipsworld
outkey = pipsworld
include = pipsworld
permit = pipsworld
qualify = yes
But for some reason dundi-lookups fail.
tsjonge*CLI dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 3 ms
ETx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: DPDISCOVER
(Command)
Flags: 00 STrans: 23682 DTrans: 0 [145.100.55.14:4520]
VERSION : 1
DIRECT EID : 00:50:da:73:18:c6
CALLED NUMBER : 29
CALLED CONTEXT : pipsworld
TTL : 16
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 23682 DTrans: 0 [145.100.55.14:4520]
ENTITY IDENT: 00:50:da:73:18:c6
KEYCRC32: 1754443205
ENCDATA : [IV a7a14a4b63ccc33286b2b23108b05f49] 4 encrypted
blocks
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 21677 DTrans: 23682 [145.100.55.14:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 23682 DTrans: 21677 [145.100.55.14:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 15333 DTrans: 0 [145.100.55.14:4520]
ENTITY IDENT: 00:50:da:73:18:c6
SHAREDKEY : [ 5b c1 3c b5 41 6d a9 11 62 40 16 0a a4 b9 11 1f
54 ae b1 7f bd af de f7 aa 5a 72 13 2e d8 b1 e7 56 17 4a 48 6a 82 3b 66
ef c4 07 b7 ce 3e ab 39 d0 75 b4 b4 0f 08 af 21 9f d6 a9 45 34 be bd 59
bc e2 a2 5b a3 d8 60 7d 8d d2 31 01 24 73 ba 27 e0 3d ce ca 22 50 c6 ef
83 ba b6 24 b3 7d 34 5b c2 c0 31 36 b5 1d bf 62 73 56 77 61 b5 5f 9e cf
d3 d2 8b 98 25 e6 47 54 7f a6 0f 97 42 ab 96 74 ]
SIGNATURE : [ d3 d9 4f d2 05 9d 71 b3 4f 76 32 29 74 02 51 2f
90 40 10 c8 6c 49 3d 67 e4 8b e4 bd 2b ca 32 ed 65 d3 b0 bc 87 ff 30 60
05 e6 f2 e2 52 2f 04 6a a4 6a fe 6e ca 9c d0 e5 24 fa e6 35 9d 38 0a 93
61 46 84 04 03 c2 f8 9d eb b5 06 60 5b 23 f3 33 69 82 3c ba 2c 57 f9 af
1a be a9 b5 23 0d 53 58 f0 fa 07 13 c1 79 b8 37 5e 7c 87 dc 14 1b a3 ec
78 6e 91 8d 1d fa 52 db 54 ce 03 3e d8 ac 96 86 ]
ENCDATA : [IV a7a14a4b63ccc33286b2b23108b05f49] 4 encrypted
blocks
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 15402 DTrans: 15333 [145.100.55.14:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
as you can see from the dialplan the extension is available:
pipc*CLI dialplan show pipsworld
[ Context 'pipsworld' created by 'IAX2' ]
'20' = 1. Noop(remco)[IAX2]
'22' = 1. Noop(tsja) [IAX2]
'23' = 1. Noop(sipura1_tst) [SIP]
'24' = 1. Noop(sipura2_tst) [SIP]
'28' = 1. Noop(s450_1) [SIP]
'29' = 1. Noop(s450_2) [SIP]
'sipura1_lijn' = 1. Noop(sipura1_lijn) [SIP]
'sipura2_lijn' = 1. Noop(sipura2_lijn) [SIP]
also, tcpdump shows that both dundi-peers are communicating (as does the
dundi debug output).
Any hints?
--
Remco Post
I didn't write all this code,
RE: [asterisk-users] dundi problem * 1.4.2
Hi,
My configuration:
SERVER 1: 192.168.1.1 = submusic
SERVER 2: 192.168.1.2 = vns
SERVER 1: Extension 32XX
SERVER 2: Extension 31XX
If you want, I can explain off list for more informations or Dundi concept
Tell me if you understand my configuration.
Fred
; DUNDI.conf SERVER 1 (Submusic)
[general]
bindaddr=0.0.0.0
port=4520
entityid=00:04:76:DB:54:7F
cachetime=1200
ttl=32
autokill=yes
storehistory=yes
[mappings]
asterisk-france =
dundi-priv-canonical,0,IAX,asterisk-france:[EMAIL PROTECTED]/${NUMBER},n
opartial
; VNS
[00:00:F8:04:C4:51]
model = symmetric
host = 192.168.1.2
inkey = vns
include = all
outkey = submusic
permit = asterisk-france
qualify = 3000
order= primary
; DUNDI.conf SERVER 2 (VNS)
[general]
bindaddr=0.0.0.0
port=4520
entityid=00:00:F8:04:C4:51
cachetime=1200
ttl=32
autokill=yes
storehistory=yes
[mappings]
asterisk-france =
dundi-priv-canonical,0,IAX,asterisk-france:[EMAIL PROTECTED]/${NUMBER},n
opartial
; SUBMUSIC
[00:04:76:DB:54:7F]
model = symmetric
host = 192.168.1.1
inkey = submusic
include = all
outkey = vns
permit = asterisk-france
qualify = yes
order= primary
; IAX.conf (Same for both)
[asterisk-france]
type=user
dbsecret=dundi/secret
context=dundi-priv-local
=
; Extension.conf Server 1 (Submusic)
=
; This macro is used to do the lookup and the match to the other host over
the Dundi Network
[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
switch = DUNDi/asterisk-France
; This Context is where the Lookup function is looking for extension
matching, just put the priority 1 and a NoOP
This server is just responding for 3 Extension over the Dundi Network
[dundi-priv-canonical]
exten = 3202,1,NooP(DUNDI LOOKUP 3202)
exten = 3216,1,NooP(DUNDI LOOKUP 3216)
exten = 3220,1,NooP(DUNDI LOOKUP 3220)
; This context is used to receipt the IAX Call, it must match with the
iax.conf.
[dundi-priv-local]
exten = 3202,1,Dial(SIP/3202)
exten = 3216,1,Dial(SIP/3216)
exten = 3220,1,Dial(SIP/3220)
; This Extension is used for the lookup and the dial over the Dundi Network.
; You must put it in the context that allow tu dial over the Dundi Network
exten = _31XX,1,Macro(dundi-priv,${EXTEN}) ; VNS
=
; Extension.conf Server 2 (VNS)
=
; This macro is used to do the lookup and the match to the other host over
the Dundi Network
[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
switch = DUNDi/asterisk-France
; This Context is where the Lookup function is looking for extension
matching, just put the priority 1 and a NoOP
This server is just responding for 3 Extension over the Dundi Network
[dundi-priv-canonical]
exten = 3101,1,NOOP(DUNDI)
exten = 3102,1,NOOP(DUNDI)
exten = 3103,1,NOOP(DUNDI)
; This context is used to receipt the IAX Call, it must match with the
iax.conf.
[dundi-priv-local]
; Direct numbers (dundi priority 0)
include = VNS
exten = 3101,1,Dial(SIP/3101)
exten = 3102,1,Dial(SIP/3102)
exten = 3103,1,Dial(SIP/3103)
===
End
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Remco Post
Envoyé : mercredi, 25. avril 2007 00:26
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] dundi problem * 1.4.2
Asterisk [Submusic] wrote:
Hi,
I'm not working with Asterisk 1.4.X, but i think your Dundi.conf is not
correct.
well, things haven't changed in the dundi.conf going from 1.2 to 1.4, so
that should be ok.
If you want i can send you my complete working exemple with Asterisk 1.2.x
(I think the config is the same)
Please do. I've had a friend look at my dundi.conf, he couldn't find
anything wrong with it, but it is quite likely that there is.
Fred
--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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RE: [asterisk-users] Realtime multiple registration for a Hard PhoneSnom 360
Olivier, If I configure the first account in Realtime and the second in the sip.conf or both in sip.conf the phone can register with multiple SIP accounts. I've tested with X-Lite and Snom 3XX Regards Fred _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Olivier Envoyé : mardi, 2. janvier 2007 11:09 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Realtime multiple registration for a Hard PhoneSnom 360 2006/12/29, Frédéric Marti [EMAIL PROTECTED]: Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can someone help for this problem ? Regards Fred ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users Fred, Are you sure Asterisk handles multiline registrations ? Could it be a Snom feature needing another call manager to happen ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Realtime multiple registration for a HardPhone Snom 360 (solved)
Hi all, My problem seems to be solved, When we have multiple SIP accounts on the same phone with RealTIme configuration, Asterisk can't authenticate correctly the second account, I think it's because of the same IP and port number. My solution is to use insecure=invite on the second SIP account in the database. Thanks for your answer Bryan, but I dont like FreePBX, I prefer VI :-) Fred -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Bryan M. Johns Envoyé : vendredi, 29. décembre 2006 15:58 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [asterisk-users] Realtime multiple registration for a HardPhone Snom 360 The device config for the Snom 360 needs to be set to adhoc mode. If you are not comfortable with hand-configuration of the extensions file, take a look at freepbx as a tool to assist you. Thanks, Bryan M. Johns Partner Shelton Johns Technology Group Office: (678) 248-2637 X: 1500 Direct: (678) 229-1809 http://www.sheltonjohns.com **Sent from my mobile phone** -Original Message- From: Frédéric Marti [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 12/29/2006 9:25 AM Subject: [asterisk-users] Realtime multiple registration for a Hard Phone Snom 360 Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can someone help for this problem ? Regards Fred ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Hi, This config is working for me: _ musiconhold.conf [shoutcast] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://stream128.submusic.ch:8004/ ; The '/' in the stream URL is important ! _ extensions.conf exten = 17,1,Answer exten = 17,2,MusicOnHold(shoutcast) _ Regards Frederic De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Robert Chadwell Envoy: mardi, 19. septembre 2006 14:47 : asterisk-users@lists.digium.com Objet: [asterisk-users] Format_MP3, Streaming, File Formats, MOH Format_MP3 appears to play MOH files starting at the beginning of each file, using the .wav file format, making for some repetitive hold music unless you alter the file itself to begin somewhere in the middle. Solution: One stream that all users connect to giving dynamic hold music (tried and tested in A1.0x using mpg123 with some success, and Icecast or Slimserver or Shoutcast) Format_MP3 doesnt seem to stream, and the wiki is wrong about streamplayer being used to play streams, as it is only used to play raw TCP streams. There are many questions in forums on the web with no answers about how to solve this dilemma, How do you get users connected to a constantly-changing stream of music instead of streams starting from the beginning (regardless of whether Linux counts them as one stream or not where the processor is concerned)? Hopefully, at the end of this thread, I will have enough information to go back to these web-forums and post the answer. To get it started here is what I have tried that hasnt worked. In most all cases the response is Music on hold started, immediately followed by Music on hold stopped with no sound in any case. ;[classes] ;mode=custom ;application=/usr/bin/streamplayer 194.158.114.67 8000 ;format=ulaw --- Straight From The Music On Hold Wiki ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy -@,http://www.shoutcast.com/sbin/tunein-station.pls?id=7733filename=playlist.pls --- From the Nerd Vittles Tutorial with the -@ added because mpg123 seemed to ask for it since the file was a .pls ;default = mp3:http://127.0.0.1:9000/stream.mp3 -- From a forum of someone using mpg123 to stream SlimServer (installed mpg123 v0.60 with no success here) [default] mode=files directory= /var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3 -- Tried a 1.2 format ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy,http://193.251.154.243:8000/ -- Thought maybe it was SlimServer so tried to stream the top Shoutcast station ;default = quietmp3:/var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3 -- Tried to stream Slimserver using the old format Thank you in advance I have been at this for a week now. How did you make it work in Asterisk 1.2x? Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released!
Hi Erick, The instructions for the SVN installation are on the Asterisk's website: http://www.asterisk.org/download Instruction from Asterisk's web: === SVN repository Subversion is the best way to keep on the bleeding edge of source releases. If you are wanting to help develop for the Asterisk project, you will want to use SVN to get the most up-to-date source code. Commands to check out code from our SVN repository: # cd /usr/src # svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk # svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel # svn checkout http://svn.digium.com/svn/libpri/trunk libpri Commands to get the current snapshot from the release branch of SVN: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 # svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2 # svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2 An Important Note: You can check out the source at any level of the filesystem. This includes something like svn checkout http://svn.digium.com/svn/asterisk. However, it would be a bad idea to do so, because you will end up checking out the code for every branch and tag that exists in the asterisk repository. Make sure you are careful when checking out the code! After you receive the latest code from SVN, issue the following commands as root to install Asterisk on your system: # cd zaptel # make clean; make install # cd ../libpri # make clean; make install # cd ../asterisk # make clean; make install Alternatively, if you checked out the 1.2 branch, you would use the following commands: # cd zaptel-1.2 # make clean; make install # cd ../libpri-1.2 # make clean; make install # cd ../asterisk-1.2 # make clean; make install Regards Fred -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Erick Perez Envoyé : samedi, 15. juillet 2006 16:46 À : Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Objet : Re: [asterisk-users] Asterisk 1.2.10 and Zaptel 1.2.7 released! Matt, What do you mean the 1.2 svn branch? Where are the download instructions and installation procedure? I always download tar.gz (that means the official release) but i always question what do I do to keep my installation with the latest bug fixes. Thanks, On 7/15/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Julian Varanini wrote: What is the best way to update from 1.2.9 to 1.2.10? If it was downloaded from SVN then you can just type make update in the directory. If it was a .tar.gz download then you will need to reinstall. I would recommend using the 1.2 branch of SVN as it means you don't have to wait for the releases to get the bugfixes. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuLm8S6d5vy0jeVcRAk9RAJ478UyMx8g7WLzkhAp+9VT9eZfXewCggHXo 9bn2Ob7u9jlDsqrKLZVrv/4= =y79J -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] agi variables list
Title: Message Hi, try: AGI debug , in you Asterisk Console, You'll see some variables. an exemple: AGI Tx agi_request: get_dnd.agiAGI Tx agi_channel: SIP/3220-bc90AGI Tx agi_language: frAGI Tx agi_type: SIPAGI Tx agi_uniqueid: 1134238803.113AGI Tx agi_callerid: 3220AGI Tx agi_calleridname: Fred LaptopAGI Tx agi_callingpres: 0AGI Tx agi_callingani2: 0AGI Tx agi_callington: 0AGI Tx agi_callingtns: 0AGI Tx agi_dnid: 3202AGI Tx agi_rdnis: unknownAGI Tx agi_context: macro-stdextenAGI Tx agi_extension: sAGI Tx agi_priority: 3AGI Tx agi_enhanced: 0.0AGI Tx agi_accountcode: Fred - Original Message - From: Olivier Taylor To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Saturday, December 10, 2005 6:44 PM Subject: [Asterisk-Users] agi variables list hello all, where can I find a list of agi variables that can be read by a external script? Thanks, Olivier ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and sendmail
Hi, I think you must spécify a full qualified domain name. The destination mail server try to resolve your Linux Box domain name , and he can't because of you domain name localhost.localdomain If you haven't a domain name you can create a dyndns.org domain linked with you linux public IP. Specify you domain name in you Linux box in the file /etc/host (must by root) like this: # Do not remove the following line, or various program # that require network functionality will fail. 127.0.0.1 localhost.localdomain localhost 0.0.0.0 test.dyndns.org AsteriskBox == change this line test.dyndns.org is the domain name that point to your public IP AsteriskBox is the name of your linux box If you have a static IP you can replace 0.0.0.0. you can try to send mail via linux console: echo test mail | mail -s MAIL TEST 1 [EMAIL PROTECTED] Sorry for my bad english,,, but i think you can decrypt Fred - Original Message - From: Michaël Gaudette [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, November 29, 2005 9:25 PM Subject: [Asterisk-Users] Voicemail and sendmail Hi, I`m a beginning Asterisk and Sendmail user. I am trying to setup my voicemail to send emails to a certain email address. It doesn't work, and I think I've figured out what it is. There is probably a spam-feature at my provider (that I am using as smart host in sendmail) to not accept emails coming from [EMAIL PROTECTED] If I start a telnet session on port 25 locally and go at it manually, an email with MAIL FROM: [EMAIL PROTECTED] never makes it, while the exact same email with MAIL FROM: [EMAIL PROTECTED] actually gwets to my inbox. How do I make it so that asterisk emails as send using [EMAIL PROTECTED] instead of [EMAIL PROTECTED] Is it an asterisk thing or a Sendmail problem? Because my logs show that the email is send from [EMAIL PROTECTED] Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
