Re: [asterisk-users] live audio stream in asterisk
Hi Daniel/asterisk users, You're correct, a typo. If got now to stream configured in musiconhold.conf [Hitz] mode=custom application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-dtc-aa02.stream.aol.com:80/stream/1074 [sbs] mode=custom application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx If I try to play the Hitz stream, it works correctly and if I try to play the sbs stream I hear nothing? exten = s,n,MusicOnHold(Hitz) or exten = s,n,MusicOnHold(sbs) The sbs stream is a mp3 stream with a bitrate of 64/128 kpbs The Hitz stream I don't know what kind of stream this is? Maybe someone knows this? Does anybody have an idea how the sbs stream must be streamend? Regards, Arjan Kroon Mobillion BV -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Daniel Tryba Verzonden: 24-12-2010 16:12 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] live audio stream in asterisk On Fri, Dec 24, 2010 at 02:36:40PM +0100, Arjan Kroon | Mobillion wrote: Is it possible to use a live audio stream in asterisk Yes, there are examples on: http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf#Exampleusingasxmmswmvstreamsoranythingth BTW You have a typo in your config (custum should be custom). -- Daniel Tryba -- _ -- 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] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone
On Mon, 27 Dec 2010, Asim Amin wrote: Also since some of these manufacture only analog cards, does anyone have any experience using these in a single system with digital cards from other manufacturers like Openvox? I've used OpenVox analogue cards. They seem to just work without having to do anything special. (And with OSLEC they just work even better in the UK!) Gordon -- _ -- 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] load balance with 2 wan connections
On Mon, 27 Dec 2010, Olivier wrote: 2010/12/25 dave george dgeo...@teletoneinc.com Need some advise or paid help on running asterisk on two WAN connection. I need load balancing and failover support. WAN: 1 DSL + 1 Cable ISP. I seem to have missed the start of this - however I'd suggest getting hardware to do it for you - e.g. Draytek 2820 - will work out of the box for you as it has both an ADSL port and an Ethernet port for the cable presenstation. Is this WAN connection used to reach an ITSP or to reach phones ? A good point - if to reach the ITSP then it'll be fine if to reach phones then your effective external IP address will alternate between the 2 - so the phones won't always be able to connect - but you could start to fiddle with dynamic DNS solutions, etc. Not perfect though. Gordon -- _ -- 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] load balance with 2 wan connections
The biggest issue with any solution to use two different providers for your IP service that will be used by your VOIP provider to deliver calls to your Asterisk server, is that each internet service will have a separate address. Therefore, for INBOUND calls, your VOIP provider will have to do the load balancing. For outbound calls, it won't be that hard as long as your provider allows you to send calls from both IP addresses. -- _ -- 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] sip attack.. fail2ban not stopping attack
On Sat, Dec 25, 2010 at 04:04:59PM -0700, Dave George wrote: My server is being attached all day and fail2ban is not stopping the attack. I updated stamstamp to match fail2ban requirements. How about posting your fail2ban config? -- Daniel Tryba -- _ -- 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] live audio stream in asterisk
On Mon, Dec 27, 2010 at 09:56:56AM +0100, Arjan Kroon | Mobillion wrote: [sbs] mode=custom application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx The sbs stream is a mp3 stream with a bitrate of 64/128 kpbs The Hitz stream I don't know what kind of stream this is? Maybe someone knows this? Does anybody have an idea how the sbs stream must be streamend? mgp123 doens't understand .asx, you could/should use mplayer (with -playlist) or any other player that understands asx like in the URL I posted. But if you take a look at the content of the .asx you'll see that it contains links to mp3 streams. You could pick one of them manually, but expect the URLs to change in the future and break your MOH. -- Daniel Tryba -- _ -- 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] sip.conf, realtime, and LDAP
On Mon, Dec 27, 2010 at 3:38 AM, Olivier oza_4...@yahoo.fr wrote: 2010/12/26 Richard Kenner ken...@gnat.com I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions? I'm also a bit confused about what's possible and not possible. Anyway, my understanding is : - you can directly query an LDAP directory from your dialplan (LDAPget), - you can also use Asterisk Realtime Architecture and use LDAP as a backend. It can be used with any Asterisk version (at least 1.4 and later). Cheers http://www.zentyal.org/ does a really good job with the LDAP integration. Try it out and see what they did. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- 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
[asterisk-users] Queue Member relationship and AstDB
I need clarification on couple of issues of Realtime Queue. It seems that when Agents(Memebers) are added using AddQueueMember, Asterisk puts this Queue-Member relationship information into AstDB, So that on asterisk restart this can be preserved. My question is, why does asterisk not store call information for Queue (holdtime, talktime, W, C, A, SL%) in AstDB, So that it can also be retained on restart? Though Queue-Member relationship information is stored in AstDB, it still forgets number of calls taken by member on instance of asterisk restart. Thanking you, -AsteriskMan -- _ -- 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] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone
On Mon, 27 Dec 2010 09:14:22 + (GMT), Gordon Henderson gordon+aster...@drogon.net wrote: I've used OpenVox analogue cards. They seem to just work without having to do anything special. +1. I have an OpenVox with a single FXO module, and it's been working for 4 years now. I don't know the other manufacturers listed by the OP. -- _ -- 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] load balance with 2 wan connections
some providers do serve inbound by sending the traffic to exact IP, some do accept the registers from any IP. in second case for Inbound failover, you might just to register = using another interface/IP address. here a new question arose: how to sip-ping some phone number to see if it's alive? On Mon, Dec 27, 2010 at 11:51 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: The biggest issue with any solution to use two different providers for your IP service that will be used by your VOIP provider to deliver calls to your Asterisk server, is that each internet service will have a separate address. Therefore, for INBOUND calls, your VOIP provider will have to do the load balancing. For outbound calls, it won't be that hard as long as your provider allows you to send calls from both IP addresses. -- _ -- 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 -- Mvh, Aurimas Skirgaila -- _ -- 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
[asterisk-users] G729a and G729 interoperability
Hello! I am wondering how the differences between G729, G729a, and G729b effect call bridging and server interoperability. For example, can one server use the G729 code with another server that uses the G729A codec? Also, which version is Asterisk set up to use? Thanks! Elliot -- _ -- 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] callerid and user on voicemail
On Wednesday, December 22, 2010 04:59:42 pm Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan
Kayhan Sent: Wednesday, December 22, 2010 4:11 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] callerid and user on voicemail
Hello,
There is a problem that i can not figure out how to solve.
I got users with 5 digit usernames for sip.
Some users has a callerid for outside calls.
I have such problems
When a user activates (for ex) call forwarding, System creates that entry
on
database as CFIM/callerid not the username,
So this rule works only if a call is made from outside to the callerid. Not
the local calls made to username.
Or, if that user dials *97 and tries to enter voicemail, voicemail
application looks for callerid instead of username , so it can not find it.
And got similar problems in some other applications too.
So, how can i make to use callerid only for outbound calls, or to forward
incoming calls to local extensions.
This won't completely solve your questions, but here are some tips. #1.
You can define a different callerid than the user-id in sip.conf. For
example, your user 12345 may look like this
[12345]
Type=peer
Context=default
Add this line
Callerid=Joe Cool 5551212
Hi and thanks for the reply,
Userid and caller ids are different. Thats making the problem bigger.
If it was like 55512345/12345 i can crop the number in dialingplans and doenst
have any headaches.
#2. *97 is just a dialplan line like this:
Exten = *97,1,voicemailmain(${CALLERID(n...@default)
You can either do some error trapping or use ex-girlfriend logic like this
Exten = *97,1,noop(new *97 logic)
Exten = *97/12345,n,voicemailmain(1...@default)
Exten = *97,n,voicemailmain(${CALLERID(num)@default)
Making such static definitions can work with small number of users but, it is
planned to have about 5000 users which makes impossible to make such
definitions.
Maybe i should match username and callerid together.
mmm that might work :)
Hope this is useful.
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[asterisk-users] malformed SIP / routing issue
Hi, I wonder what conditions might lead, that SIP packets from provider P destined to my external SIP server A, are reaching my internal SIP server B? the fun factor is that internal B server is used for outbound calls via the same provider P. I found no routing issues. Is it possible to build SIP header specifying the final destination - the internal IP address? -- Mvh, Aurimas Skirgaila -- _ -- 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] load balance with 2 wan connections
We have another gateway in the USA that will send traffic to both IPs. The US gateway will load balance the traffic to both IPs. This is not used for phones. It is used mainly for wholesale traffic. Asterisk is being used as an SS7 gateway. Each DSL limits us to about 16 calls. We are thinking of combining DSL + DSL + Cable ISP on the same box and have our USA box send traffic to all 3 IPS. Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Monday, December 27, 2010 4:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] load balance with 2 wan connections The biggest issue with any solution to use two different providers for your IP service that will be used by your VOIP provider to deliver calls to your Asterisk server, is that each internet service will have a separate address. Therefore, for INBOUND calls, your VOIP provider will have to do the load balancing. For outbound calls, it won't be that hard as long as your provider allows you to send calls from both IP addresses. -- _ -- 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] malformed SIP / routing issue
Hi, Have you checked SIP messages on B server? Maybe your provider P sends traffic to incorrect IP. -- Best Regards, Giedrius -- _ -- 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] malformed SIP / routing issue
Surely. B responds 404 Not Found., as it's not configured to receive these SIP packets. provider P sends to correct IP, and moreover B has no external IP. On Mon, Dec 27, 2010 at 3:54 PM, voipas voi...@gmail.com wrote: Hi, Have you checked SIP messages on B server? Maybe your provider P sends traffic to incorrect IP. -- Best Regards, Giedrius -- _ -- 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 -- Mvh, Aurimas Skirgaila -- _ -- 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] sip attack.. fail2ban not stopping attack
jail.conf
[asterisk-iptables]
enabled = true
filter = asterisk
action = iptables-allports[name=ASTERISK, protocol=all]
sendmail-whois[name=ASTERISK, dest=root,
sender=fail2...@example.org]
logpath = /var/log/asterisk/messages
maxretry = 5
bantime = 259200
filter asterisk.conf
[INCLUDES]
# Read common prefixes. If any customizations available -- read them from
# common.local
#before = common.conf
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile. The
# host must be matched by a group named host. The tag HOST
can
# be used for standard IP/hostname matching and is only an alias
for
# (?:::f{4,6}:)?(?Phost\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for 'HOST' - Wrong
password
NOTICE.* .*: Registration from '.*' failed for 'HOST' - No
matching peer found
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for 'HOST' - Device
does not match ACL
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' \(from HOST\)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*' (.*)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
ignoreregex =
logger.conf
[general]
;
; Customize the display of debug message time stamps
; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)
;
; see strftime(3) Linux manual for format specifiers. Note that there is
also
; a fractional second parameter which may be used in this field. Use %1q
; for tenths, %2q for hundredths, etc.
;
dateformat=%F %T ; ISO 8601 date format
;dateformat=%F %T.%3q ; with milliseconds
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Monday, December 27, 2010 5:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip attack.. fail2ban not stopping attack
On Sat, Dec 25, 2010 at 04:04:59PM -0700, Dave George wrote:
My server is being attached all day and fail2ban is not stopping the
attack. I updated stamstamp to match fail2ban requirements.
How about posting your fail2ban config?
--
Daniel Tryba
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Re: [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone
- Original Message - Anyone who has experience using Digium analog card clones from any of the following: 1. Zycoo 2. CTVON 3. Chinaroby 4. Etross 5. Immediate IT (IIT) 6. Realtone and can give review which one is good quality with easy configuration and error free running. Also since some of these manufacture only analog cards, does anyone have any experience using these in a single system with digital cards from other manufacturers like Openvox? I've used Openvox cards in production heavily for FXO/FXS applications and a few installations using their single port T1/E1 card. Other than a bad run of analog cards we experienced last year, they've been solid. Also, I've used Zycoo analog boards which work very well, not a problem anywhere. One other manufacturer you might want to look at is 'Phonic EQ'. [1] I just received a T1/E1 board (wanted something very inexpensive as this is for my test rig in the lab) that seems to work well but I have not used it long enough to know if there are any problems. --Tim [1] http://www.phoniceq.com/ -- _ -- 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] G729a and G729 interoperability
On 12/27/2010 08:05 PM, Elliot Murdock wrote: Hello! I am wondering how the differences between G729, G729a, and G729b effect call bridging and server interoperability. For example, can one server use the G729 code with another server that uses the G729A codec? Also, which version is Asterisk set up to use? Thanks! Elliot There is no compatibility issue between basic G.729 and G.729A. That is why they use the same SDP code. In practice it is rare to see a G.729 codec in real world use. They are almost all G.729A. G.729 sounds better, but G.729A uses half the CPU power. Cheaper normally wins over better in the real world. G.729 annex B is an add on, providing VAD features. It may be used with G.729 or G.729A. A separate entry in the SDP says whether this option is supported. Steve -- _ -- 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] sip attack.. fail2ban not stopping attack
Le 27/12/2010 16:20, dave george a écrit :
[...]
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile. The
# host must be matched by a group named host. The tag HOST
can
# be used for standard IP/hostname matching and is only an alias
for
# (?:::f{4,6}:)?(?Phost\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for 'HOST' - Wrong
password
NOTICE.* .*: Registration from '.*' failed for 'HOST' - No
matching peer found
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for 'HOST' - Device
does not match ACL
NOTICE.*HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' \(fromHOST\)
NOTICE.* .*: HostHOST failed MD5 authentication for '.*' (.*)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
ignoreregex =
[...]
How looks your asterisk notice file?
---
Daniel
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Re: [asterisk-users] sip attack.. fail2ban not stopping attack
Simply to reduce the attack, and then improve the defense: If you don't need traffic from some area that is attacking you, just put the whole area in IPTables. A list is available on VOIP-INFO.org. Cull out what you want to allow. Then tune Fail2Ban at your leisure. Cary Fitch -- _ -- 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] sip attack.. fail2ban not stopping attack
With asterisk 1.8+ it should be:
failregex = NOTICE.* .*: Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - Wrong password
NOTICE.* .*: Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - No matching peer found
NOTICE.* .*: Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - Device does not match ACL
NOTICE.* .*: Registration from '.*' failed for
'HOST(:[0-9]{1,5})?' - Peer is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' \(from HOST\)
NOTICE.* .*: Host HOST failed MD5 authentication for
'.*' (.*)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
since format of notice has changed (asterisk now adds port after HOST)
Nick
On Mon, Dec 27, 2010 at 6:03 PM, Administrator TOOTAI ad...@tootai.net wrote:
Le 27/12/2010 16:20, dave george a écrit :
[...]
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile.
The
# host must be matched by a group named host. The tag HOST
can
# be used for standard IP/hostname matching and is only an alias
for
# (?:::f{4,6}:)?(?Phost\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for 'HOST' -
Wrong
password
NOTICE.* .*: Registration from '.*' failed for 'HOST' - No
matching peer found
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
Device
does not match ACL
NOTICE.*HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' \(fromHOST\)
NOTICE.* .*: HostHOST failed MD5 authentication for '.*'
(.*)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
ignoreregex =
[...]
How looks your asterisk notice file?
---
Daniel
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Re: [asterisk-users] live audio stream in asterisk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon | Mobillion Sent: Monday, December 27, 2010 2:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] live audio stream in asterisk Hi Daniel/asterisk users, You're correct, a typo. If got now to stream configured in musiconhold.conf [Hitz] mode=custom application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-dtc-aa02.stream.aol.com:80/stream/1074 [sbs] mode=custom application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx If I try to play the Hitz stream, it works correctly and if I try to play the sbs stream I hear nothing? exten = s,n,MusicOnHold(Hitz) or exten = s,n,MusicOnHold(sbs) The sbs stream is a mp3 stream with a bitrate of 64/128 kpbs The Hitz stream I don't know what kind of stream this is? Maybe someone knows this? Does anybody have an idea how the sbs stream must be streamend? Regards, Arjan Kroon Mobillion BV ASX is ASF (Microsoft Advanced Systems Format). This format is not presently supported by mpg123 (as of version 1.13.0). If you want to use the ASF stream, you're going to have to either use something besides mpg123 or repackage it into a regular mpeg format. -- _ -- 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] sip attack.. fail2ban not stopping attack
On Mon, Dec 27, 2010 at 10:20:13AM -0500, dave george wrote: [snip fail2ban config] Well, all looks fine. Your filter is correct. Your message log is also in the correct format. You can test this with: fail2ban-regex /var/log/asterisk/messages /etc/fail2ban/filter.d/asterisk.conf So is fail2ban actually running (like someone already suggested)? $ ps auxwww | grep fail Other things it could be: -a broken backend in jail.conf (try polling). -running as an unprivileged user (can't read asterisk/messages). -- When you do things right, people won't be sure you've done anything at all. Daniel Tryba -- _ -- 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] Agents login
Dnia Sat, 25 Dec 2010 15:31:57 +0200 Michael voip.quest...@gmail.com napisał(a): Is that possible?? From what we saw, the agents login works on a constantly open line. Which version of Asterisk you're using? -- Damian Ryszka aka Rychu rychu(at)sileman.net.pl -- _ -- 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
[asterisk-users] CEL and custom values.
I am setting up CEL with asterisk 1.8 and so far so good. The issue I was hoping to address here was also being able to get storage of other values such as HANGUPCAUSE and other variables that are used for billing and quality of service. The CEL documentation starts out by saying that we can not store any other variables but then at the top of that section it says this is incorrect and that section of the documentation needs to be changed. So how can I set a variable for storage when a CEL log event is fired. I want to be able to add some additional fields to my database so when a CEL storage event is fired that the values of variables are stored to my database or CSV if the variable is set. Is there something like the CDR(field)=value but for CEL(field)=value. Any help is appreciated. Bryant -- _ -- 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
[asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?
Hi Everyone, I use Asterisk for regularPBX use it's made for. But I want to take it a bit further and use it at cmmand level to be able to send SIP notifies to restart a phone or take advantage of a phone's UPnP capabilities. Is Asterisk capable of that? If so, what is a simple SIP reboot message like and how can I invoke it from a Asterisk CLI? If Asterisk is not the best tool for this purpose what is a very simple to implement SIP stack out there that can do this? Thanks, -- _ -- 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] Using SIP stack within Asterisk to reboot phones -Possible?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Monday, December 27, 2010 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Using SIP stack within Asterisk to reboot phones -Possible? Hi Everyone, I use Asterisk for regularPBX use it's made for. But I want to take it a bit further and use it at cmmand level to be able to send SIP notifies to restart a phone or take advantage of a phone's UPnP capabilities. Is Asterisk capable of that? If so, what is a simple SIP reboot message like and how can I invoke it from a Asterisk CLI? If Asterisk is not the best tool for this purpose what is a very simple to implement SIP stack out there that can do this? Thanks, On Polycom phones, you can do sip notify foo xxx and it will reboot that phone, when foo is set up per the following link http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip_notify.conf I assume it will reset any phone that responds to the check-sync command, but since I only have Polycom's, this is just a guess. -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
On 12/27/2010 12:08 PM, Bruce B wrote: Hi Everyone, I use Asterisk for regularPBX use it's made for. But I want to take it a bit further and use it at cmmand level to be able to send SIP notifies to restart a phone or take advantage of a phone's UPnP capabilities. Is Asterisk capable of that? If so, what is a simple SIP reboot message like and how can I invoke it from a Asterisk CLI? If Asterisk is not the best tool for this purpose what is a very simple to implement SIP stack out there that can do this? The very first hit when I did a Google search for Asterisk SIP notify was the voip-info page that documents exactly what you are looking for. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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
[asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe
Hi, we upgraded an Asterisk 1.4 with mISDN to 1.6 with chan_dahdi. Due to problems with iax channel posted earlier, we wanted to switch back to 1.4 version. Server has 2 HBA cards, everything is running fine with 1.6, bri_cpe is recognized and the 7 euroISDN channels are running well, ingoing and outgoing. Now we installed 1.4.38 version and no more ISDN. In logs we found this: [2010-12-24 14:50:38] VERBOSE[1773] logger.c: == Parsing '/etc/asterisk/chan_dahdi.conf': [2010-12-24 14:50:38] VERBOSE[1773] logger.c: Found [2010-12-24 14:50:38] ERROR[1773] chan_dahdi.c: Unknown signalling method 'bri_cpe' [2010-12-24 14:50:38] ERROR[1773] chan_dahdi.c: Signalling must be specified before any channels are. We think about a bug in libpri 1.4.11.4 so installed 1.4.11.5, same result. Dahdi linux and tools are 2.4.0 And yes, Asterisk is build with libpri ;-) d...@myphoneserver:/usr/src$ strings /usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony w/PRI DAHDI Telephony Driver w/PRI Thanks for your help -- Daniel -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
On Mon, Dec 27, 2010 at 3:08 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I use Asterisk for regularPBX use it's made for. But I want to take it a bit further and use it at cmmand level to be able to send SIP notifies to restart a phone or take advantage of a phone's UPnP capabilities. Is Asterisk capable of that? If so, what is a simple SIP reboot message like and how can I invoke it from a Asterisk CLI? If Asterisk is not the best tool for this purpose what is a very simple to implement SIP stack out there that can do this? Thanks, To restart phone at extension 1234 cli sip notify polycom-check-cfg 1234 ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
On Mon, Dec 27, 2010 at 3:33 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/27/2010 12:08 PM, Bruce B wrote: Hi Everyone, I use Asterisk for regularPBX use it's made for. But I want to take it a bit further and use it at cmmand level to be able to send SIP notifies to restart a phone or take advantage of a phone's UPnP capabilities. Is Asterisk capable of that? If so, what is a simple SIP reboot message like and how can I invoke it from a Asterisk CLI? If Asterisk is not the best tool for this purpose what is a very simple to implement SIP stack out there that can do this? The very first hit when I did a Google search for Asterisk SIP notify was the voip-info page that documents exactly what you are looking for. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org or https://wiki.asterisk.org/wiki/display/AST/ManagerAction_SIPnotify also has accurate and up to date info. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- 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 1.4.38 - unknown signalling bri_cpe
On 12/27/2010 12:37 PM, Administrator TOOTAI wrote: Hi, we upgraded an Asterisk 1.4 with mISDN to 1.6 with chan_dahdi. Due to problems with iax channel posted earlier, we wanted to switch back to 1.4 version. Server has 2 HBA cards, everything is running fine with 1.6, bri_cpe is recognized and the 7 euroISDN channels are running well, ingoing and outgoing. Now we installed 1.4.38 version and no more ISDN. In logs we found this: [2010-12-24 14:50:38] VERBOSE[1773] logger.c: == Parsing '/etc/asterisk/chan_dahdi.conf': [2010-12-24 14:50:38] VERBOSE[1773] logger.c: Found [2010-12-24 14:50:38] ERROR[1773] chan_dahdi.c: Unknown signalling method 'bri_cpe' [2010-12-24 14:50:38] ERROR[1773] chan_dahdi.c: Signalling must be specified before any channels are. We think about a bug in libpri 1.4.11.4 so installed 1.4.11.5, same result. Dahdi linux and tools are 2.4.0 And yes, Asterisk is build with libpri ;-) d...@myphoneserver:/usr/src$ strings /usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony w/PRI DAHDI Telephony Driver w/PRI Asterisk 1.4 has never had BRI support in chan_dahdi. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
Lots of good info and pointers so far. But do keep in mind that not all phones will automatically reboot just because you sent it a check-sync or resync event with the sip notify command. I vaguely remember that for e.g. the Polycoms some other condition had to be true: either the phone's config file on the ftp/tftp server had to have a newer time-stamp than the one that was downloaded during the phone's last boot, or a config option had to be set to a non-default value to make the phone reboot unconditionally upon receiving the SIP notify, regardless of the config file's modification date. -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
Thanks Kai-Uwe and everyone else. I have seen all those examples and I am exploring the sip_notify.conf file now which makes things more clear to me. However, when sending a SIP notify to a phone that is not registered to Asterisk via it's IP address should I expect to receive a success of fail packet back or that is not how SIP Notify works? *sip notify aastra-check-cfg 192.168.0.5* *Sending NOTIFY of type 'aastra-check-cfg' to '192.168.0.5'* * * That is all I see and the phone is not restarted. There might be a few things different about Aastra phones to get them accept SIP Notifies and I would like to hear your experience about it and what features and notifies are available to me as it pretains particulary to Aastra phones. P.S. Are these SIP notifies anything different than simple HTTP get or XML push and receive or do they require a sip stack or a program like Asterisk and it's much more complicated than I think? I want to get a simple page where some phone controls can be done without relying on a heavy program like Asterisk but again if it get's too complicated I won't mind using Asterisk for this purpose. Just want to know my options. Thanks again, On Mon, Dec 27, 2010 at 1:59 PM, Kai-Uwe Jensen kujen...@gmail.com wrote: Lots of good info and pointers so far. But do keep in mind that not all phones will automatically reboot just because you sent it a check-sync or resync event with the sip notify command. I vaguely remember that for e.g. the Polycoms some other condition had to be true: either the phone's config file on the ftp/tftp server had to have a newer time-stamp than the one that was downloaded during the phone's last boot, or a config option had to be set to a non-default value to make the phone reboot unconditionally upon receiving the SIP notify, regardless of the config file's modification date. -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Monday, December 27, 2010 1:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible? Thanks Kai-Uwe and everyone else. I have seen all those examples and I am exploring the sip_notify.conf file now which makes things more clear to me. However, when sending a SIP notify to a phone that is not registered to Asterisk via it's IP address should I expect to receive a success of fail packet back or that is not how SIP Notify works? sip notify aastra-check-cfg 192.168.0.5 Sending NOTIFY of type 'aastra-check-cfg' to '192.168.0.5' That is all I see and the phone is not restarted. There might be a few things different about Aastra phones to get them accept SIP Notifies and I would like to hear your experience about it and what features and notifies are available to me as it pretains particulary to Aastra phones. P.S. Are these SIP notifies anything different than simple HTTP get or XML push and receive or do they require a sip stack or a program like Asterisk and it's much more complicated than I think? I want to get a simple page where some phone controls can be done without relying on a heavy program like Asterisk but again if it get's too complicated I won't mind using Asterisk for this purpose. Just want to know my options. Thanks again, On Mon, Dec 27, 2010 at 1:59 PM, Kai-Uwe Jensen kujen...@gmail.com wrote: Lots of good info and pointers so far. But do keep in mind that not all phones will automatically reboot just because you sent it a check-sync or resync event with the sip notify command. I vaguely remember that for e.g. the Polycoms some other condition had to be true: either the phone's config file on the ftp/tftp server had to have a newer time-stamp than the one that was downloaded during the phone's last boot, or a config option had to be set to a non-default value to make the phone reboot unconditionally upon receiving the SIP notify, regardless of the config file's modification date. This is just my opinion: #1 - if you are using sip notify to send a command to an unregistered peer, you won't get a result back unless you are in a very verbose/debug mode. #2 - they are probably xml push/receives and there are lighter clients than Asterisk out there to accomplish what you want. Google for lightweight SIP server -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
On Mon, 27 Dec 2010, Bruce B wrote: Thanks Kai-Uwe and everyone else. I have seen all those examples and I am exploring the sip_notify.conf file now which makes things more clear to me. However, when sending a SIP notify to a phone that is not registered to Asterisk via it's IP address should I expect to receive a success of fail packet back or that is not how SIP Notify works? Maybe it's just me being a 1.2 Luddite, but I'd rather not have to explain to my boss that I crashed the production server because of a bug or because I was 'trying something out' and forgot which server I was on. Any chance sipsak or sipp can handle this task? I reboot my aging SPA3K using an http request via wget. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
[asterisk-users] Panasonic KX-TGP500 w/Asterisk
Has anybody have any experience using the KX-TGP500 Cordless DEC 6.0 System with asterisk ? I run a small asterisk server at home using two SPA3102s, and thinking of upgrading my cordless analog phones to something a little newer. -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
I've never worked with Aastras, so don't have any additional data over what's been said by others. Also, I've never sent the SIP check-sync notify to a phone that wasn't already registered with the asterisk server the SIP notify was sent from. My best *guess* would be that actual behavior of the phones in that case might vary (based on manufacturer and/or firmware version). Some might ignore the notify, others *may* accept it *if* the asterisk server was their provisioning server previously, some might even blindly accept it. The actual payload/content of the Notify is configured in the sip_notify.conf file. My version (running 1.8.1.1) does not have anything specific to Aastra in it, so I would try all the other ones to see if one of them works. If none work, I'd go search for some Aastra administrator documentation (or provisioning guide, or some such). As you can see from the file, the payload itself is not complex at all. For Polycoms as an example, the Event field in the SIP NOTIFY header is set to check-sync. That's all there is to it. The SIP Notify we're talking about here is a simple SIP event being sent to the phone by asterisk. Best to use a packet sniffer, or you could turn on SIP debug in asterisk for a single peer and send it the notify. The behavior I see with my Polycoms makes it appear as if there is no handshake between asterisk and the phone at all for these. The phone reboots as soon as I send the notify. Even more, asterisk does a few retransmissions of the notify packet, so it might've expected a SIP response from the phone (which the phone did not send). Lastly, to state the obvious, the SIP Notify itself does not convey any configuration data to the phone at all. It only tells the phone to go check your config. So to help with your provisioning scenario, you'd have to update/modify the original config data/files for the phone on the provisioning server, then trigger the phone to reboot (or reload). From all I've seen with Polycoms and Ciscos, you'll set up a phone's provisioning server through your DHCP server (or manually on the phone). The SIP Notify will cause the phone to go back to that provisioning server and re-load the latest config data. -- _ -- 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] Panasonic KX-TGP500 w/Asterisk
I have no direct experience. But I know that E4 Technologies has been using this phone with Asterisk Switchvox. Panasonic made an effort earlier this year to have it certified with Asterisk. It's also Broadvoice certified. Michael --Original Message Text--- From: William Stillwell Date: Mon, 27 Dec 2010 16:40:57 -0500 Has anybody have any experience using the KX-TGP500 Cordless DEC 6.0 System with asterisk ? I run a small asterisk server at home using two SPA3102s, and thinking of upgrading my cordless analog phones to something a little newer. -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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
[asterisk-users] How to use google voice for voicemail transcription
No, I don't know how to do this. Does anybody? I'd like to take a voicemail file from asterisk (*.wav, *.gsm, *.mp3 ?) and send it to googlevoice as a voicemail, then get the transcription over gmail. I know about pygooglevoice (is it still maintained?). But I can't figure out how to dial gv and leave a message (robo dial??) from a system command. This would allow me to get transcriptions of voicemails left on asterisk. Anybody know how to do this? Or doing it? sean -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
SIPp is a good option. Thanks Nikhil On 12/27/2010 11:38 PM, Bruce B wrote: Hi Everyone, I use Asterisk for regularPBX use it's made for. But I want to take it a bit further and use it at cmmand level to be able to send SIP notifies to restart a phone or take advantage of a phone's UPnP capabilities. Is Asterisk capable of that? If so, what is a simple SIP reboot message like and how can I invoke it from a Asterisk CLI? If Asterisk is not the best tool for this purpose what is a very simple to implement SIP stack out there that can do this? Thanks, -- _ -- 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] Using SIP stack within Asterisk to rebootphones - Possible?
What type of phones? Easy to do with Polycom and several others from Asterisk CLI. Sent from my BlackBerry® smartphone -Original Message- From: Nikhil d.nik...@cem-solutions.net Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 28 Dec 2010 08:42:22 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible? -- _ -- 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] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone
Hi, I have used 4-PRI card from atcom.cn and it works perfectly for me. Regards, Faisal +923214059996 On 12/27/2010 12:25 PM, Asim Amin wrote: Hello All, Anyone who has experience using Digium analog card clones from any of the following: 1. Zycoo 2. CTVON 3. Chinaroby 4. Etross 5. Immediate IT (IIT) 6. Realtone and can give review which one is good quality with easy configuration and error free running. Also since some of these manufacture only analog cards, does anyone have any experience using these in a single system with digital cards from other manufacturers like Openvox? -- Asim Amin Partner Technical Manager, Telco Division Horizon Technologies Cell: +92-323-3314151 E-mail: a...@horizontech.biz mailto:a...@horizontech.biz Web: http://horizontech.biz http://horizontech.biz/ http://hostht.com http://hostht.com/ -- _ -- 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] Agents login
Hi, We're using version 1.6.2.X. I think that the command we need is AgentCallbackLogin. We're building a script to study the entire functionality of queues, agents and everything around it. Happy New Year to all, Michael 2010/12/27 Damian Ryszka ry...@sileman.net.pl Dnia Sat, 25 Dec 2010 15:31:57 +0200 Michael voip.quest...@gmail.com napisał(a): Is that possible?? From what we saw, the agents login works on a constantly open line. Which version of Asterisk you're using? -- Damian Ryszka aka Rychu rychu(at)sileman.net.pl -- _ -- 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
[asterisk-users] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact
Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a number and save the contact or save it without a number. Seems to me that Outcall is not harvesting the CLID number as it should or maybe it's not passing it to outlook so that the old contact which already exists for that number to be pulled. I am wondering if anyone else has experienced this or if you guys think OutCall is really not reliable and I should look for an alternative. Please let me know if there is a solid alternative out there. Thanks -- _ -- 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] Agents login
Dnia Tue, 28 Dec 2010 08:02:51 +0200 Michael voip.quest...@gmail.com napisał(a): I think that the command we need is AgentCallbackLogin. We're building a script to study the entire functionality of queues, agents and everything around it. Perhaps you noticed, that AgentCallbackLogin() has been removed in 1.6 series. To log in Agents into queues I'm using AddQueueMember() and to remove RemoveQueueMember(). To make authorization I've tried to use Read() application to read PIN and username from caller and compare his input with mysql (ODBC driver) with success but I didn't have time to finish it. I hope that above will be useful. Greets, -- Damian Ryszka aka Rychu rychu(at)sileman.net.pl -- _ -- 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
