Re: [asterisk-users] RTP Packets Going To Wrong IP Address
What does the call setup look like on this? You can either debug sip in the console or 'ngrep -s 1500 -T -W byline host 75.36.34.98' From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicholas Blasgen Sent: Monday, July 21, 2008 16:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTP Packets Going To Wrong IP Address I have a user behind a firewall who's had no issues in the past connecting though his firewall. He's registered just fine. But when he places a call, a large number of them have no audio on either side of the connection. No one can hear him, he can't hear anyone as well. After a lot of poking around (and changing many settings) I noticed that Asterisk is communicating the RTP packets to an internal IP address. My server has no internal IP address, only an external address, so it's not like we're trying to route this anywhere else. As can be seen below, I've already identified the host as being behind a firewall and therefor to not trust packets from it. Anyone have a suggestion? Name/username HostDyn Nat ACL Port Status Realtime jfabriquer/jfabriquer 75.36.34.98 D N 55266OK (145 ms) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Asterisk SVN-branch-1.4-r118365 -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Problems w/Asterisk Realtime + MySQL + SIP
Try to delete whole column 'md5secret' from DB peers table. Leave only 'secret'. And try then. Same result. Regards, Walter Stanish Owner / Director Occident Systems (+86 15808 700 801) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Problems w/Asterisk Realtime + MySQL + SIP
On Mon, Jul 21, 2008 at 6:40 PM, Walter Stanish
[EMAIL PROTECTED] wrote:
[Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) -
Command in SIP REGISTER
[Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be
handled, bad request: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg.
It looks like Asterisk is unhappy with the SIP REGISTER request coming
from your softphone for some reason. It's very strange that it's
occurring for two different softphones though.
Trun on SIP debugging by typing sip debug on your Asterisk console
and then post up the 4 SIP messages invloved in the register
transaction so we can take a look and spot why it could be getting
rejected.
Sure.
Here's what happens when kphone starts up:
==
--- SIP read from 192.168.0.25:5060 ---
REGISTER sip:192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C
CSeq: 35 REGISTER
To: Walter sip:[EMAIL PROTECTED]
Expires: 900
From: Walter sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
Content-Length: 0
User-Agent: kphone/4.2
Event: registration
Allow-Events: presence
Contact: Walter
sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE,
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER
black*CLI
-
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.25 : 5060 (no NAT)
--- Transmitting (no NAT) to 192.168.0.25:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25
From: Walter sip:[EMAIL PROTECTED]
To: Walter sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 35 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
--- Transmitting (no NAT) to 192.168.0.25:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25
From: Walter sip:[EMAIL PROTECTED]
To: Walter sip:[EMAIL PROTECTED];tag=as59de1023
Call-ID: [EMAIL PROTECTED]
CSeq: 35 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7864265a
Content-Length: 0
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in
32000 ms (Method: REGISTER)
==
Kphone prompts for a password, then the following occurs.
==
--- SIP read from 192.168.0.25:5060 ---
REGISTER sip:192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C
CSeq: 36 REGISTER
To: Walter sip:[EMAIL PROTECTED]
Authorization: Digest username=walter, realm=asterisk,
nonce=7864265a, uri=sip:192.168.0.2, cnonce=abcdefghi,
nc=0001, response=10a7024959390c04b4d09c708fac6130, opaque=,
algorithm=MD5
Expires: 900
From: Walter sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
Content-Length: 0
User-Agent: kphone/4.2
Event: registration
Allow-Events: presence
Contact: Walter
sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE,
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER
-
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.25 : 5060 (no NAT)
--- Transmitting (no NAT) to 192.168.0.25:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25
From: Walter sip:[EMAIL PROTECTED]
To: Walter sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 36 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
--- Transmitting (no NAT) to 192.168.0.25:5060 ---
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25
From: Walter sip:[EMAIL PROTECTED]
To: Walter sip:[EMAIL PROTECTED];tag=as59de1023
Call-ID: [EMAIL PROTECTED]
CSeq: 36 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[Jul 22 00:59:38] NOTICE[2414]: chan_sip.c:15049
handle_request_register: Registration from 'Walter
sip:[EMAIL PROTECTED]' failed for '192.168.0.25' - Wrong password
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in
32000 ms (Method: REGISTER)
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER
==
Just to confirm, the password supplied was 'aaa'.
In MySQL md5secret = md5('aaa') and secret = 'aaa'.
Here's what happens with zoiper (one registration click only)...
==
--- SIP read from
Re: [asterisk-users] Problems w/Asterisk Realtime + MySQL + SIP
It looks like Asterisk is unhappy with the SIP REGISTER request coming from your softphone for some reason. It's very strange that it's occurring for two different softphones though. I couldn't see any sign of the console message from your first trace [Jul 21 15:40:47] DEBUG[2105]: chan_sip.c:15372 sipsock_read: SIP message could not be handled, bad request: [EMAIL PROTECTED] which is a bit strange. The 403 Forbidden message you are getting with the KPHone is definitely realted to credentials, either the username or password is configured incorrectly somewhere. You can test by adding the SIP account into sip.conf to get ti working and then after that move onto your relatime config. It definitely seems to be an asterisk realtime config issue. Just to confirm, the client and server _are_ on the same network with no intermediate firewalls or NAT. I just switched back to static config with the same username/password and authentication worked first time in kphone, as it was working before. Zoiper, however, still has issues - unsure why, probably something related to the fact it's pulling that external IP from somewhere. (Some kind of automated NAT detection/STUN initialisation?) At this point I'm unable to get any results from realtime except a different response when I enter a username that doesn't exist at all in the realtime tables. Any further ideas? Other than trying a different asterisk version, I'm feeling pretty stuck. Regards, Walter Stanish Owner / Director Occident Systems (+86 15808 700 801) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] increase ring time out
Fidel Garcia wrote: I need to increase the ringing timeout on the AA50 appliance. How do I accomplish this? I need the phones to ring a bit more before the caller gets to the voicemail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 Could you show your extensions.conf? Normally you'd do that in the Dial command: exten = _XX,1,Answer exten = _XX,n,Dial(SIP/1,20) ... Where 20 is the time you're letting the phone ring... :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Incompatible voice frame panic! [SOLVED]
Vazquez David wrote: Hi all, Panic! Panic! When I get a call over mISDN to my IAX extension and try to transfer it to another IAX/SIP, I get this message: Dropping incompatible voice frame on ... of format ulaw since our native format has changed to alaw Immediately followed by one almost the same: Dropping incompatible voice frame on ... of format alaw since our native format has changed to ulaw and so on, and so forth... Any ideas??? Thanks, David Vazquez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 Phew! Solved... The only thing I changed was, in my iaxprov.conf changed codecpriority=host to codecpriority=reqonly. Now everything works smoothly... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Problems w/Asterisk Realtime + MySQL + SIP
On Tue, Jul 22, 2008 at 10:32 AM, Walter Stanish [EMAIL PROTECTED] wrote: Any further ideas? Other than trying a different asterisk version, I'm feeling pretty stuck. Is there an equivalent of the Asterisk console odbc show command for MySQL? That would show you whether Asterisk has a connection yo your db. Can you post up the contents of your extconfig.conf for us to have a look at? Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Problems w/Asterisk Realtime + MySQL + SIP
extconfig.conf contents (as per first message) were: sipusers = mysql,asterisk_config,sip_users sippeers = mysql,asterisk_config,sip_peers However, I've now commented them all out to static config for testing purposes. I was unable to locate a long-term changelog, so I'm currently compiling 1.4.21.1 to see if that makes any difference. Is there an equivalent of the Asterisk console odbc show command for MySQL? That would show you whether Asterisk has a connection yo your db. I tried this before posting here (ie: with the two extconfig.conf lines enabled), and it reported the status as fine. MySQL is version 14.12 Distrib 5.0.54 Regards, Walter Stanish Owner / Director Occident Systems (+86 15808 700 801) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Overlap dialing via SIP
i can see from your dialplan that all the extensions except international
extension are of 12 digits. International extensions are of 13 or more
digits. here is what you can do with the international extensions, all other
extensions remain the same:
[084x]
exten = _9084,1,Macro(dialout-pstn)
[outbound-national]
exten = _90[1-2]X,1,Macro(dialout-pstn)
[087x]
exten = _9087,1,Macro(dialout-pstn)
[0906]
exten = _90906XXX,1,Macro(dialout-pstn)
[outbound-international]
exten = _900XX.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _900XX.,2,Congestion
If you see closely i have put a dot at the end of each international
extension, this will allow you to dial atleast 13 digits. so no need to
crate extension of every length.
On Mon, Jul 21, 2008 at 9:10 PM, Ben Thompson [EMAIL PROTECTED] wrote:
Hi
I have set up an asterisk system which allows the use of Overlap Dialing
from
SIP handsets. In order to do this I had to list the various patterns of
numbers
which can be dialed in the UK. We also dial with a prefix of '9' for and
outside
line so much of my dialplan looks like this :-
[084x]
exten = _9084,1,Macro(dialout-pstn)
[outbound-national]
exten = _90[1-2]X,1,Macro(dialout-pstn)
[087x]
exten = _9087,1,Macro(dialout-pstn)
[0906]
exten = _90906XXX,1,Macro(dialout-pstn)
...
I was able to download the mappings for 0800 numbers and other special
ranges
from the ofcom website and I have incorporated these. For international
dialing
I have not been able to find an easy way of doing this so I created the
folling
contexts whcih make use of the WaitExten feature :-
[outbound-international]
exten = _900XX,1,Set(oldexten=${EXTEN})
exten = _900XX,2,Goto(international-number-length-check,s,1)
[international-number-length-check]
exten = s,1,Answer
exten = s,2,WaitExten(8)
exten = _X,1,Set(enddigits=${EXTEN})
exten = _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial
${oldexten}${enddigits})
exten = _X,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _X,4,Congestion()
exten = _X,104,Busy()
exten = _XX,1,Set(enddigits=${EXTEN})
exten = _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial
${oldexten}${enddigits})
exten = _XX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XX,4,Congestion()
exten = _XX,104,Busy()
exten = _XXX,1,Set(enddigits=${EXTEN})
exten = _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial
${oldexten}${enddigits})
exten = _XXX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XXX,4,Congestion()
exten = _XXX,104,Busy()
exten = t,1,Dial(${OUTBOUNDTRUNK}/${oldexten})
exten = t,2,Congestion()
exten = t,102,Busy()
This works fairly well but I have noticed that occasionally the WaitExten
feature does
not seem to catch the first digits if they are dialed too quickly. It is
almost as if
there is a some sort of delay and the thirteenth digit is sometimes missed.
Can anyone suggest why WaitExten might be ocasionally missing a digit or
can anyone think
of a better way of doing this?
Thanks
Ben Thompson
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Best Regards
Rizwan Hisham
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Re: [asterisk-users] Help With dial plan
maybe the user is dialing something other than 3000 and that extension is
not registered on your asterisk. just a wild guess.
On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED] wrote:
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo calls
to trixbox. I am using sip to send the calls from the gateway to trixbox. I
have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I be doing
wrong?
James
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Rizwan Hisham
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Re: [asterisk-users] Overlap dialing via SIP
On Mon, 21 Jul 2008, Ben Thompson wrote:
Hi
I have set up an asterisk system which allows the use of Overlap Dialing from
SIP handsets. In order to do this I had to list the various patterns of
numbers
which can be dialed in the UK. We also dial with a prefix of '9' for and
outside
line so much of my dialplan looks like this :-
[084x]
exten = _9084,1,Macro(dialout-pstn)
[outbound-national]
exten = _90[1-2]X,1,Macro(dialout-pstn)
You'd better learn more about the UK before going further...
Don't forget that we now have 03 numbers too.
And UK geographic numbers can be 10 or 11 digits long. Mine is 11 digits,
but the town down the road from me is 10 digits. (So locally, I can dial a
5 or 6 digit number!)
[087x]
exten = _9087,1,Macro(dialout-pstn)
[0906]
exten = _90906XXX,1,Macro(dialout-pstn)
york.ac.uk and you're allowing 0906 numbers? Where do I sign up ;-)
I was able to download the mappings for 0800 numbers and other special ranges
from the ofcom website and I have incorporated these. For international
dialing
I have not been able to find an easy way of doing this so I created the
folling
contexts whcih make use of the WaitExten feature :-
[outbound-international]
exten = _900XX,1,Set(oldexten=${EXTEN})
exten = _900XX,2,Goto(international-number-length-check,s,1)
[international-number-length-check]
exten = s,1,Answer
exten = s,2,WaitExten(8)
exten = _X,1,Set(enddigits=${EXTEN})
exten = _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial
${oldexten}${enddigits})
exten = _X,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _X,4,Congestion()
exten = _X,104,Busy()
exten = _XX,1,Set(enddigits=${EXTEN})
exten = _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial
${oldexten}${enddigits})
exten = _XX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XX,4,Congestion()
exten = _XX,104,Busy()
exten = _XXX,1,Set(enddigits=${EXTEN})
exten = _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial
${oldexten}${enddigits})
exten = _XXX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XXX,4,Congestion()
exten = _XXX,104,Busy()
exten = t,1,Dial(${OUTBOUNDTRUNK}/${oldexten})
exten = t,2,Congestion()
exten = t,102,Busy()
This works fairly well but I have noticed that occasionally the WaitExten
feature does
not seem to catch the first digits if they are dialed too quickly. It is
almost as if
there is a some sort of delay and the thirteenth digit is sometimes missed.
Can anyone suggest why WaitExten might be ocasionally missing a digit or can
anyone think
of a better way of doing this?
I'm sure there are some codes in Germany that are only about 7 digits
long... (My brothers is 9 digits though + 49 for the country takes it to
11 + 00 is 13) Where do you draw the line? I think it's always going to be
hard to guess every country (and our own!) dialling lengths...
Personally I think you're making life hard for yourself, although
potentially nice for the users, I guess.
Or maybe you want to look at the ! match pattern, or just give-up on
overlap dialling. I like to be able to 'edit' numbers on my phone before
hitting the 'send' button. Too used to doing in on mobiles and DECT
handsets or years now I guess, and this is what I teach my customers -
pretend the handset is a mobile, dial the number, push 'send' (or the
green button, or the 'tick' key, or whatever the phone uses to transmit
the number)
So I have:
exten = _0.,1,Noop(Outside line request: Dialled 0... for ${EXTEN})
exten = _0.,n,Macro(dialOut,${EXTEN})
exten = _0.,n,Hangup()
; Dial 9 for an outside line:
exten = _9.,1,Noop(Outside line request: Dialled 9... for ${EXTEN:1})
exten = _9.,n,Macro(dialOut,${EXTEN:1})
exten = _9.,n,Hangup()
I make sure the SIP phones have early/overlap dial turned off, and it
just works for me...
(That Macro does other stuff for me, it might as well be a
Dial(${OUTBOUNDTRUNK}/${ARG1}) as far as this is concerned)
Gordon
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Re: [asterisk-users] Overlap dialing via SIP
On Tue, Jul 22, 2008 at 11:41:45AM +0100, Gordon Henderson wrote: [0906] exten = _90906XXX,1,Macro(dialout-pstn) york.ac.uk and you're allowing 0906 numbers? Where do I sign up ;-) Err no, this is not my actual dialplan - just an example. Personally I think you're making life hard for yourself, although potentially nice for the users, I guess. Or maybe you want to look at the ! match pattern, or just give-up on overlap dialling. I like to be able to 'edit' numbers on my phone before OK, the ! match pattern sounds interesting. Does this allow overlap dialing though? Thanks Ben Thompson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Overlap dialing via SIP
On Tue, 22 Jul 2008, Ben Thompson wrote: On Tue, Jul 22, 2008 at 11:41:45AM +0100, Gordon Henderson wrote: [0906] exten = _90906XXX,1,Macro(dialout-pstn) york.ac.uk and you're allowing 0906 numbers? Where do I sign up ;-) Err no, this is not my actual dialplan - just an example. Personally I think you're making life hard for yourself, although potentially nice for the users, I guess. Or maybe you want to look at the ! match pattern, or just give-up on overlap dialling. I like to be able to 'edit' numbers on my phone before OK, the ! match pattern sounds interesting. Does this allow overlap dialing though? TBH, I've only just noticed it when writing that email - my bible is a beaten-up 1st edition of copy of the starfish book which doesn't mention it, but I checked the WiKi (which seems to have had a bit of a facelift since I looked last!) and it makes mention of the ! match character.. According to the WiKi: http://www.voip-info.org/wiki-Asterisk+Dialplan+Patterns ! wildcard, matches zero or more characters immediately (only Asterisk 1.2 and later, see note) Note: The exclamation mark wildcard, which is available only in Asterisk 1.2 and later, behaves specially it will match as soon as can without waiting for the dialling to complete, but it will not match until it is unambiguous, and the number being dialled cannot match any other extension in the context. It was designed for use as follows, so that as soon as the digits dialled don't match '001800...' the outgoing telephone line will be picked up and overlap dialling will be used (with full audio feedback from 'earlyb3' etc.) Context outgoing: Extension Description _001800NXXCalls to USA toll-free numbers made by VoIP _X! Other calls via normal telco, with overlap dial. It looks hopeful... (If you believe in overlap dialling that is ;-) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] OT Astricon/Digium Beach Ball Mailing
Alex Balashov wrote: randulo wrote: On Sat, Jul 19, 2008 at 7:09 PM, Alex Balashov [EMAIL PROTECTED] wrote: I've asked a number of others I know in real life who got the beach balls and all are reported as being fully functional. So this is not a case for the bug tracker? Perhaps a bounty... I've already submitted plastic patches to Beach Ball-rc5-pl5-beta trunk. Hopefully there's no white space involved in that patch. Sorry. Couldn't resist. :-) BJ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Cisco vs Asterisk
Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
On Tue, Jul 22, 2008 at 8:52 AM, voip crazy [EMAIL PROTECTED] wrote: Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? Thanks. VoipCrazy. You said migrate to a Cisco, what do they have now? Sell them all Cisco. You will make more money and great residual income for MACs ;-) Anyways, you could ditch the Cisco entirely and use Asterisk. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
voip crazy a écrit : Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? To answer your questions, one would need to know what exactly are all the functionalities of a Cisco Unity server, and more specificaly, what are the needs of your client. But i'm pretty sure the voip-info wiki can answer the asterisk part... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
Hi, I don't use asterisk since 1.2.x version and never deployed an big project with Asterisk, so I don't know if currently Asterisk can replace to Cisco Unity as Voice Mail, but Cisco Unity is not only for voice mail the main objective is to be part of all Unified Communications infrastructure. Then Integration with Active Directory / Exchange (Lotus Notes) and other features is only possible with Cisco Unity. Maybe I'm wrong and Asterisk can do it.. so I would like to read about that... Rgds. On 7/22/08, voip crazy [EMAIL PROTECTED] wrote: Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 -- Omar E.P.T - Certified Networking Professionals make better Connections! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Echo Issue
The Aastra 480i is also known for having pretty high sidetone volumes which some may interpret as an echo. These are the gain/sidetone settings I use on all my Aastra's (9112i, 9113i, 480i, and 57i) headset tx gain: -3 headset sidetone gain: -2 handset tx gain: -6 handset sidetone gain: -5 handsfree tx gain: 0 You can probably safely ignore the headset settings unless you are using headsets obviously... I've found those settings work fairly well with Plantronics headsets (have a few different models in use using those same settings). Don;t know anything about the SNOM's so I can help ya there. -- Matt http://www.mattgwatson.ca On 7/21/08, Noah Miller [EMAIL PROTECTED] wrote: Hi Joseph - I have Astra 480i's and Snom M3's. I am using a SIP provider so I do not have any peripheral cards. I am on voip-wiki now reading about the echo canceller tuning, thanks! For your particular case, you're probably not going to find much useful info on the wiki about echo cancellation. The info there is about reducing echo when there is an analog-to-digital conversion (in other words, if you're connecting to PSTN lines somewhere). If you have echo on calls that go through your SIP provider, it is possible that they are not doing a very good job with echo cancellation. If the echo is exclusively on these calls, you'll probably want to call them to discuss this. If you have echo on calls between your Astra and/or Snom handsets, you may want check the gain settings on these devices. Reducing the gain would probably lessen the effect of the echo. You may also want to check if either of these phones is doing any AEC (acoustic echo cancellation), and if there are any AEC parameters that are adjustable. I don't have experience with either of these phones, so I can't give you direct info on how to do this, but I'm sure that at least Snom support can help you. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
Call me crazy, but why are you so keen on selling them an Asterisk box when you don't even know if its capable of doing what you want to sell it for? thats kinda scray actually. -- Matt http://www.mattgwatson.ca On 7/22/08, voip crazy [EMAIL PROTECTED] wrote: Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] issue with high latency
Hi, Is there a specific latency that asterisk accepts? I encountered a problem wherein when the latency was unusually high,my xlite's (i have 2 xlite) cannot register. but when the link suddenly went stable, the x-lite just registered. what i forgot to look at is if the registration packet is reaching my asterisks. -- when xlite cannot register --- Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data: Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=651ms TTL=56 --- when xlite can register Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data: Reply from 202.203.204.205: bytes=32 time=43ms TTL=56 Reply from 202.203.204.205: bytes=32 time=12ms TTL=56 Reply from 202.203.204.205: bytes=32 time=13ms TTL=56 even if the latency is high i still have internet access as i can still browse and using yahoo messenger. anyone encountered something similar? regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
[EMAIL PROTECTED] a écrit : Call me crazy, but why are you so keen on selling them an Asterisk box when you don't even know if its capable of doing what you want to sell it for? I won't, i had the same felling ... thats kinda scray actually. Yep ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] issue with high latency
On 22 Jul 2008, at 14:36, Nhadie wrote: Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data: Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=651ms TTL=56 Never going to work with that latency. I would say anything over 150 is probably pushing it. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
At this stage connectivity is great between asterisk -- sipX(3.8) -- Exchange UM And still i dont see the features needed. 2008/7/22 Benoit Plessis [EMAIL PROTECTED]: [EMAIL PROTECTED] a écrit : Call me crazy, but why are you so keen on selling them an Asterisk box when you don't even know if its capable of doing what you want to sell it for? I won't, i had the same felling ... thats kinda scray actually. Yep ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Help With dial plan
James Mutuku wrote:
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I be
doing wrong?
James
As a sanity check, you may want to place a NoOp(${EXTEN}) prior to the dial. If
you set the verbosity high on the Asterisk console, then you can see what the
value of EXTEN is when the NoOp occurs.
Mark Michelson
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Re: [asterisk-users] increase ring time out
Where exactly do I have to change it?
This is the extensions.conf file:
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Tue Jul 22 15:14:28 2008
;!
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[globals]
trunk_1 = Zap/g1
trunk_1_cid = asreceived
[dundi-e164-canonical]
[dundi-e164-customers]
[dundi-e164-via-pstn]
[dundi-e164-local]
include = dundi-e164-canonical
include = dundi-e164-customers
include = dundi-e164-via-pstn
[dundi-e164-switch]
switch = DUNDi/e164
[dundi-e164-lookup]
include = dundi-e164-local
include = dundi-e164-switch
[macro-dundi-e164]
exten = s,1,Goto(${ARG1},1)
include = dundi-e164-lookup
[macro-trunkdial]
exten = s,1,set(CALLERID(all)=${IF(${LEN(${CALLERID(num)})} 6 ?
${CALLERID(al
l)} : ${ARG2})})
exten = s,n,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp
[iaxtel700]
exten = _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider]
[trunkint]
exten = _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten = _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
exten = _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
ignorepat = 9
include = local
include = trunkld
[local]
ignorepat = 9
include = default
include = parkedcalls
include = trunklocal
include = iaxtel700
include = trunktollfree
include = iaxprovider
[macro-stdexten]
exten = s,1,Dial(${ARG2},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-stdPrivacyexten]
exten = s,1,Dial(${ARG2},20|p)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = s-DONTCALL,1,Goto(${ARG3},s,1)
exten = s-TORTURE,1,Goto(${ARG4},s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-page]
exten = s,1,ChanIsAvail(${ARG1}|js)
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
exten = s,n(autoanswer),Set(_ALERT_INFO=RA)
exten = s,n,SIPAddHeader(Call-Info: Answer-After=0)
exten = s,n,NoOp()
exten = s,n,Dial(${ARG1}||)
exten = s,n(fail),Hangup
[demo]
exten = s,1,Wait(1)
exten = s,n,Answer
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n(restart),BackGround(demo-congrats)
exten = s,n(instruct),BackGround(demo-instruct)
exten = s,n,WaitExten
exten = 2,1,BackGround(demo-moreinfo)
exten = 2,n,Goto(s,instruct)
exten = 3,1,Set(LANGUAGE()=fr)
exten = 3,n,Goto(s,restart)
exten = 1000,1,Goto(default,s,1)
exten = 1234,1,Playback(transfer,skip)
exten = 1234,n,Macro(stdexten,1234,${CONSOLE})
exten = 1235,1,Voicemail(u1234)
exten = 1236,1,Dial(Console/dsp)
exten = 1236,n,Voicemail(u1234)
exten = #,1,Playback(demo-thanks)
exten = #,n,Hangup
exten = t,1,Goto(#,1)
exten = i,1,Playback(invalid)
exten = 500,1,Playback(demo-abouttotry)
exten = 500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = 500,n,Playback(demo-nogo)
exten = 500,n,Goto(s,6)
exten = 600,1,Playback(demo-echotest)
exten = 600,n,Echo
exten = 600,n,Playback(demo-echodone)
exten = 600,n,Goto(s,6)
exten = 76245,1,Macro(page,SIP/Grandstream1)
exten = _7XXX,1,Macro(page,SIP/${EXTEN})
exten = 7999,1,Set(TIMEOUT(absolute)=60)
exten =
7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]/n
|d)
exten = 8500,1,VoicemailMain
exten = 8500,n,Goto(s,6)
[page]
exten = _X.,1,Macro(page,SIP/${EXTEN})
[default]
exten = 6050,1,VoiceMailMain
exten = 7000,1,Goto(voicemenu-custom-1|s|1)
exten = 702,1,Goto(voicemenu-custom-2|s|1)
exten = 500,1,Goto(voicemenu-custom-2|s|1)
[voicemenu-custom-1]
include = default
comment = Welcome
alias_exten = 7000
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Background(thank-you-for-calling)
exten = s,4,Background(if-u-know-ext-dial)
exten = s,5,Background(otherwise)
exten = s,6,Background(to-reach-operator)
exten = s,7,Background(pls-hold-while-try)
exten = s,8,WaitExten(6)
[numberplan-custom-1]
plancomment = DialPlan1
include = default
include = parkedcalls
exten = _91700XXX!,1,Macro(trunkdial,${}/${EXTEN:1})
comment = _91700XXX!,1,IAXTEL,standard
exten =
Re: [asterisk-users] issue with high latency
Not true. Voip is done over satellite every day and those ping times are at least 540 and upwards of in the 700's depending on the technology used. The key here is keeping the latency stable. If the packet flow fluctuates too much in latency this is when a problem arises. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Howes Sent: Tuesday, July 22, 2008 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] issue with high latency On 22 Jul 2008, at 14:36, Nhadie wrote: Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data: Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=651ms TTL=56 Never going to work with that latency. I would say anything over 150 is probably pushing it. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] issue with high latency
Jitter is what your describing, it's a bad thing. http://en.wikipedia.org/wiki/Jitter While VoIP may work (third party 128ms echo cancellers, etc) most support organization won't go outside ITU-T G.114 recommendations. I've done Cisco 7940 phones deployed in the Gulf of Mexico on a oil platform using Callmanager based in US in 2003. The company controlled the satellite and prioritized voice, ping was 600ms. Worked well except the local calls were to Venezula which was too expensive per minute from US. Two phones ran up more than $1000US in 30 days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Moore Sent: Tuesday, July 22, 2008 11:40 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] issue with high latency Not true. Voip is done over satellite every day and those ping times are at least 540 and upwards of in the 700's depending on the technology used. The key here is keeping the latency stable. If the packet flow fluctuates too much in latency this is when a problem arises. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Howes Sent: Tuesday, July 22, 2008 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] issue with high latency On 22 Jul 2008, at 14:36, Nhadie wrote: Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data: Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=651ms TTL=56 Never going to work with that latency. I would say anything over 150 is probably pushing it. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Vitelity dtmfmode=rfc2833 started working!
Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked sip show peer and saw that Vitelity for inbound was now reporting DTMFmode : rfc2833 (it didn't used to), so switched my ountbound dtmfmode to rfc2833 and my problems went away! Yay! It looks like Vitelity now supports rfc2833 on SIP channels. I thought others might be interested in knowing this, as at least in my case it broke things until I changed my settings, and I see this has been a prior source of frustration for many others. Mark -- Mark G. Thomas ([EMAIL PROTECTED]) voice: 215-591-3695 http://mail-cleaner.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Problems w/Asterisk Realtime + MySQL + SIP [SOLVED!]
After upgrading to 1.4.21.1 I received some more verbose debug messages. Apparently the cause of the problem was that the following table columns MUST exist for a SIP peer to register. ipaddr (i used varchar(32)) port (i used int unsigned) regseconds (i used bigint unsigned) This is apparently because realtime stores the IP address, port and registration time for a dynamic IP, realtime peer in the database along with the static accounting information. After adding those columns I was able to register fine with asterisk from kphone. Unfortunately the console commands 'sip show peers', 'sip show registry', etc. don't seem to work (is this normal?) ... but otherwise all is fine. Thanks to all who offered help while debugging this issue. Regards, Walter Stanish Owner / Director Occident Systems (+86 15808 700 801) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Voicemail email to alternative ports...
The main DSL provider in Mexico is no blocking access to port 25 so the email notification for voicemail is stuck in the server. I suppose that I have to change the sendmail configuration so it can send email to an alternative port but I wanted to ckeck first if there is an option to do it from the voicemail.conf file. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] what is the magic needed from upgrading from 1.4 to 1.6
On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote: �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: Call from 'devcentos5x64_to_ebox4300' to extension 'mediaport_audio_visual' rejected because extension not found. Jerry-- from the console, type dialplan show smvoice-mediaport, and let's verify for certain that it's in there. I'll try to reproduce your problem in my test system here. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] what is the magic needed from upgrading from 1.4 to 1.6
dialplan show default There is no existence of 'default' context Command 'dialplan show default' failed. I am getting the same thing for default What gives? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] what is the magic needed from upgrading from 1.4 to 1.6
On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote: / // �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: / Call from 'devcentos5x64_to_ebox4300' to extension 'mediaport_audio_visual' rejected because extension not found. Jerry-- from the console, type dialplan show smvoice-mediaport, and let's verify for certain that it's in there. I'll try to reproduce your problem in my test system here. murf Steve, I get this: dialplan show smvoice-mediaport There is no existence of 'smvoice-mediaport' context Command 'dialplan show smvoice-mediaport' failed. my extensions.conf has a context: ; media [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf Then express.dnis.conf has: ; This file is generated from MessageNet EMACS ; Phone Caller ID DNIS Manager screen ; MMAUDIO : EBOX 4300 - exten = mediaport_audio_visual,1,Goto(smvoice-mediaport-audio-visual,s,1) [smvoice-mediaport-audio-visual] exten = s,1,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup Not seeing what the problem is here. especially since 1.2 and 1.4 both work. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] increase ring time out
Nobody will support the gui generated conf files here, you got the answer,
YOU should try to understand how dialplan works, if not, find the way to
change it in gui.
2008/7/22 Fidel Garcia [EMAIL PROTECTED]:
Where exactly do I have to change it?
This is the extensions.conf file:
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Tue Jul 22 15:14:28 2008
;!
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[globals]
trunk_1 = Zap/g1
trunk_1_cid = asreceived
[dundi-e164-canonical]
[dundi-e164-customers]
[dundi-e164-via-pstn]
[dundi-e164-local]
include = dundi-e164-canonical
include = dundi-e164-customers
include = dundi-e164-via-pstn
[dundi-e164-switch]
switch = DUNDi/e164
[dundi-e164-lookup]
include = dundi-e164-local
include = dundi-e164-switch
[macro-dundi-e164]
exten = s,1,Goto(${ARG1},1)
include = dundi-e164-lookup
[macro-trunkdial]
exten = s,1,set(CALLERID(all)=${IF(${LEN(${CALLERID(num)})} 6 ?
${CALLERID(al
l)} : ${ARG2})})
exten = s,n,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp
[iaxtel700]
exten = _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL
PROTECTED]http://[EMAIL PROTECTED]/$%7BEXTEN:[EMAIL PROTECTED]
)
[iaxprovider]
[trunkint]
exten = _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten = _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
exten = _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
ignorepat = 9
include = local
include = trunkld
[local]
ignorepat = 9
include = default
include = parkedcalls
include = trunklocal
include = iaxtel700
include = trunktollfree
include = iaxprovider
[macro-stdexten]
exten = s,1,Dial(${ARG2},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-stdPrivacyexten]
exten = s,1,Dial(${ARG2},20|p)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = s-DONTCALL,1,Goto(${ARG3},s,1)
exten = s-TORTURE,1,Goto(${ARG4},s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-page]
exten = s,1,ChanIsAvail(${ARG1}|js)
exten = s,n,GoToIf([${AVAILSTATUS} = 1]?autoanswer:fail)
exten = s,n(autoanswer),Set(_ALERT_INFO=RA)
exten = s,n,SIPAddHeader(Call-Info: Answer-After=0)
exten = s,n,NoOp()
exten = s,n,Dial(${ARG1}||)
exten = s,n(fail),Hangup
[demo]
exten = s,1,Wait(1)
exten = s,n,Answer
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n(restart),BackGround(demo-congrats)
exten = s,n(instruct),BackGround(demo-instruct)
exten = s,n,WaitExten
exten = 2,1,BackGround(demo-moreinfo)
exten = 2,n,Goto(s,instruct)
exten = 3,1,Set(LANGUAGE()=fr)
exten = 3,n,Goto(s,restart)
exten = 1000,1,Goto(default,s,1)
exten = 1234,1,Playback(transfer,skip)
exten = 1234,n,Macro(stdexten,1234,${CONSOLE})
exten = 1235,1,Voicemail(u1234)
exten = 1236,1,Dial(Console/dsp)
exten = 1236,n,Voicemail(u1234)
exten = #,1,Playback(demo-thanks)
exten = #,n,Hangup
exten = t,1,Goto(#,1)
exten = i,1,Playback(invalid)
exten = 500,1,Playback(demo-abouttotry)
exten = 500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = 500,n,Playback(demo-nogo)
exten = 500,n,Goto(s,6)
exten = 600,1,Playback(demo-echotest)
exten = 600,n,Echo
exten = 600,n,Playback(demo-echodone)
exten = 600,n,Goto(s,6)
exten = 76245,1,Macro(page,SIP/Grandstream1)
exten = _7XXX,1,Macro(page,SIP/${EXTEN})
exten = 7999,1,Set(TIMEOUT(absolute)=60)
exten =
7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]/n
|d)
exten = 8500,1,VoicemailMain
exten = 8500,n,Goto(s,6)
[page]
exten = _X.,1,Macro(page,SIP/${EXTEN})
[default]
exten = 6050,1,VoiceMailMain
exten = 7000,1,Goto(voicemenu-custom-1|s|1)
exten = 702,1,Goto(voicemenu-custom-2|s|1)
exten = 500,1,Goto(voicemenu-custom-2|s|1)
[voicemenu-custom-1]
include = default
comment = Welcome
alias_exten = 7000
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Background(thank-you-for-calling)
Re: [asterisk-users] Cisco vs Asterisk
voip crazy wrote: Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? I don't know that any OSS piece ever has *all* the features of a proprietary platform, especially since a lot of those features tend to be very esoteric and designed to complement the vendor's other service platform and handset gear. The question is: 1. What are you trying to do? 2. Can Asterisk do it? 3. Can Asterisk do it well? 4. Can Asterisk do it at the scale, volume and scope you're looking for? The question is NOT: 1. Is Asterisk basically like a free version of CallManager? 2. Can Asterisk duplicate CallManager? -- Alex Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: dialplan show default There is no existence of 'default' context Command 'dialplan show default' failed. I am getting the same thing for default Check the console and logs from when you started Asterisk to see if there were any errors reported when loading/parsing the dialplan. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] what is the magic needed from upgrading from 1.4 to 1.6
I dont see any errors in the dialplan while loading. I tried to past the whole log but it was rejected. Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files. I cant even dialplan show default at this time. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
Alex Balashov schrieb: The question is: 1. What are you trying to do? 2. Can Asterisk do it? 3. Can Asterisk do it well? 4. Can Asterisk do it at the scale, volume and scope you're looking for? The question is NOT: 1. Is Asterisk basically like a free version of CallManager? 2. Can Asterisk duplicate CallManager? Come on. People want simple answers. So: Can Asterisk duplicate CallManager? [y/n] *scnr* Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: I dont see any errors in the dialplan while loading. I tried to past the whole log but it was rejected. Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files. I cant even dialplan show default at this time. It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that should have been included in the source code. If you read that you'll realize that dialplan show command was deprecated in 1.4 and be removed in 1.6. Until your read those files you are going to continue to have strange problems. The thing is that they are not strange problems. They are problems you should expect if you don't read the upgrade notes. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files. We aren't disputing that, so you don't need to keep repeating it :-) You'll have to open a bug on bugs.digium.com and attach the log file there; we won't be able to help you any further until we can find out why your dialplan was not loaded when you used 1.6. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Voicemail email to alternative ports...
On Tue, 22 Jul 2008, Carlos Chavez wrote: The main DSL provider in Mexico is no blocking access to port 25 so the email notification for voicemail is stuck in the server. I suppose that I have to change the sendmail configuration so it can send email to an alternative port but I wanted to ckeck first if there is an option to do it from the voicemail.conf file. Asterisk seems to call a local sendmail running on the host that asterisk is running on, so you need to get that incanation of sendmail (or whatever sendmail look-a-like you're using) to relay the email using a different port. However, why don't you get the local sendmail to simply relay outgoing email via the ISPs mailservers than get it to send email directly? That's what you probably ought to be doing anyway. This is an m4 sendmail.mc template you might want to use: divert(-1) divert(0)dnl OSTYPE(linux) FEATURE(nullclient, RELAY_HOST) MASQUERADE_AS(MASQ_HOST) Substitute RELAY_HOST for your ISPs email server (smtp.telecomabmex.com ?) and MASQ_HOST for your local domain (maybe just telecomabmex.com) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
It's amazing... the man starts the thread with a simple question: Can anybody tell him if Asterisk can do the same things that the Cisco Unity Server can do?, if it can do some better, some the same, and/or some worse, can someone indicate which ones? Also, can Asterisk complement the Cisco call manager functionalities?... I wish I knew the answers, and I am myself interested in the educated straight opinions of some of the members of this forum. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Tuesday, July 22, 2008 2:15 PM To: Asterisk Users Subject: Re: [asterisk-users] Cisco vs Asterisk Alex Balashov schrieb: The question is: 1. What are you trying to do? 2. Can Asterisk do it? 3. Can Asterisk do it well? 4. Can Asterisk do it at the scale, volume and scope you're looking for? The question is NOT: 1. Is Asterisk basically like a free version of CallManager? 2. Can Asterisk duplicate CallManager? Come on. People want simple answers. So: Can Asterisk duplicate CallManager? [y/n] *scnr* Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] what is the magic needed from upgrading from 1.4 to 1.6
Eric ManxPower Wieling wrote: It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that should have been included in the source code. If you read that you'll realize that dialplan show command was deprecated in 1.4 and be removed in 1.6. Until your read those files you are going to continue Eric, I think you're mistaken; show dialplan was depreciated, not dialplan show. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] ?? Vitelity dtmfmode=rfc2833 started working!
I appreciate your report (below), but it's a strange and disturbing coincidence for me. DTMF out through Vitelity was not working for me until 1-2 days ago when I changed it from rfc2833 to inband! Maybe I just missed the change date and I should change it back? Date: Tue, 22 Jul 2008 12:23:39 -0400 From: Mark G. Thomas [EMAIL PROTECTED] Subject: [asterisk-users] Vitelity dtmfmode=rfc2833 started working! To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked sip show peer and saw that Vitelity for inbound was now reporting DTMFmode : rfc2833 (it didn't used to), so switched my ountbound dtmfmode to rfc2833 and my problems went away! Yay! It looks like Vitelity now supports rfc2833 on SIP channels. I thought others might be interested in knowing this, as at least in my case it broke things until I changed my settings, and I see this has been a prior source of frustration for many others. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Call Recordings...
Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.4/1566 - Release Date: 7/22/2008 6:00 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] what is the magic needed from upgrading from 1.4 to 1.6
I sit corrected. He should still be reading the upgrade files. Doug Lytle wrote: Eric ManxPower Wieling wrote: It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that should have been included in the source code. If you read that you'll realize that dialplan show command was deprecated in 1.4 and be removed in 1.6. Until your read those files you are going to continue Eric, I think you're mistaken; show dialplan was depreciated, not dialplan show. Doug -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Looking for a more robust Click to Dial/Web Dial solution than AsteriDex (potential for a bounty!)
I realize this may be less of an Asterisk question and more of a... well... everything but asterisk, but still relating to asterisk question. I was looking for a Click to Dial/Web Dial solution and I found AsteriDex. I'm looking for something I can use on the road where I can hit an internal Click to Dial/Web Dial page from my cell, tap on a number and have it bridge a call between my cell and the other number so I can use my office PBX for company LD, clients see my company's CallerID etc. AsteriDex seems to have almost everything that I'm looking for, but I need something with a few more enhancements and I'm wondering if such a thing exists or if I need this to be custom made. - I need something that can import a phone book from vcards and/or pull names and numbers from an LDAP directory, not just MySQL (I don't even really care about keeping my numbers in AsteriDex's MySQL database). - I need something that, when I hit it with a web browser (specifically, Mobile Safari on my iPhone 3G), will also have a field where I can enter a number manually, incase a number I need to dial isn't in the directory. - I need something that has hooks to customize the CallerID fields. It should have configuration hooks somewhere where I can set a couple of different the CallerID Names and Numbers, then have the option to select which CallerID gets set when the outbound call to the client is made. I have control over the CallerID that gets sent to the Telco. Please advise, and if someone is looking for a few extra bucks, let me know how much you will charge to develop something like this. I can provide a deposit if you are credible. Thanks in advance! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Call Recordings...
So basically, He wants all calls recorded, but he wants a sequence that he can push, so that when he rants and raves at a customer, there won't be evidence to say that he did that... :) Just a hunch on that. :) I don't know. Eugen On 7/22/08, Gregory Malsack [EMAIL PROTECTED] wrote: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.4/1566 - Release Date: 7/22/2008 6:00 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
I haven't used Asterisk Voicemail but here are Unity Unified Messaging (for Exchange) 5.x/7.x features, in short I think you need to be a Callmanager/Exchange Server shop with heavy integration with ActiveSync/Direct Push/Outlook 2007/OCS2007. The company that created Unity (Active Voice) was a bunch of ex-microsoft guys. If you are not a enterprise/campus or prefer IMAP/SMTP then I don't think you would see any benefits or ROI. I don't think just hanging Unity Voicemail Only off a Asterisk box would be of much value. I like AVST CallXpress http://www.avst.com/products/callxpressMessaging/ for smaller customers. Unity Unified Messaging (for Exchange) 5.x/7.x Using Exchange Administrator it reads/writes directly to Exchange Message Store (not IMAP or SMTP) Phone View (listen to message as callers leave them, control message on Cisco 79xx phone LCD screen) Windows Mobile/Blackberry intergration (has Blackberry plug-in) Single number for fax/T38 Speech Connect (reply to voicemails via Speech to Text or have them read Text to Speech) Mailbox greetings based on Calendar Integration Unity Digital Networking for multiple sites being able to send each other messages There are flash videos and datasheets here; http://www.cisco.com/en/US/products/sw/voicesw/ps2237/index.html Now I will go read the Wiki and see how Asterik handles Voicemail Note there are several version of Unity (Unity Unified Messaging (with Echange), Unity Unified Messaging (with Domino), Unity Connection 2.x, Unity Unified Express). I choose the version with the most integration with Exchange to discuss here. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Savinovich Sent: Tuesday, July 22, 2008 3:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco vs Asterisk It's amazing... the man starts the thread with a simple question: Can anybody tell him if Asterisk can do the same things that the Cisco Unity Server can do?, if it can do some better, some the same, and/or some worse, can someone indicate which ones? Also, can Asterisk complement the Cisco call manager functionalities?... I wish I knew the answers, and I am myself interested in the educated straight opinions of some of the members of this forum. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Tuesday, July 22, 2008 2:15 PM To: Asterisk Users Subject: Re: [asterisk-users] Cisco vs Asterisk Alex Balashov schrieb: The question is: 1. What are you trying to do? 2. Can Asterisk do it? 3. Can Asterisk do it well? 4. Can Asterisk do it at the scale, volume and scope you're looking for? The question is NOT: 1. Is Asterisk basically like a free version of CallManager? 2. Can Asterisk duplicate CallManager? Come on. People want simple answers. So: Can Asterisk duplicate CallManager? [y/n] *scnr* Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Call Recordings...
I bet the reason is that when his gf calls, he can erase the records so his wife's divorce attorney can not get his hands on them to play in court. Lyle Eugen Soare wrote: So basically, He wants all calls recorded, but he wants a sequence that he can push, so that when he rants and raves at a customer, there won't be evidence to say that he did that... :) Just a hunch on that. :) I don't know. Eugen On 7/22/08, *Gregory Malsack* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Thanks, Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Call Recordings...
Am Dienstag, den 22.07.2008, 14:53 -0500 schrieb Gregory Malsack: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Hi Gregory, I found something about recording at http://www.voip-info.org/wiki/view/Asterisk+config+features.conf (second example). If you combine that with a default_recording_enabled (Monitor() call before Dial(), I would expect), that could be used to turn _off_ recording by pressing a key. I would not know though how to protect against the external call party pressing the same key. Best regards Anselm smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] 3-way calling for IAX channels
How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] features on dial pad
I`m looking for the same for an IAX cliente based softphone I´m trying to build. Did you get some info about it? Thanks in advance, Daniel Arohuanca +51 1 3594122 On Wed, Apr 9, 2008 at 10:29 PM, nhadie ramos [EMAIL PROTECTED] wrote: Hi All, If i were to develop a softphone, how can i add call transfer, call on hold and 3-way conference on it? linksys Ip phone has those built-in button to xfer, conf, on hold. and x-lite also has those, how can i have those if i develop my own? Thank You Regards, Nhadie ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] [Fwd: Re: what is the magic needed from upgrading from 1.4 to 1.6]
On Tue, 2008-07-22 at 13:21 -0400, Jerry Geis wrote: On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote: / // �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: / Call from 'devcentos5x64_to_ebox4300' to extension 'mediaport_audio_visual' rejected because extension not found. Jerry-- from the console, type dialplan show smvoice-mediaport, and let's verify for certain that it's in there. I'll try to reproduce your problem in my test system here. murf Jerry-- I think you've found a bug! I put in an smvoice-mediaport context just like the one you described into my dialplan, and then started asterisk. It looks OK to 'dialplan show', etc, but when I do a 'stop gracefully', I get a core dump-- while deleting *that* context. So, file a bug, assign it to me. I suspect when I find the reason for the core dump, I will also find the reason for your problem. It's too much of a coincidence to believe, that they are separate and independent problems. (but, then, my name is Murphy...!) I'm in the middle of trying to debug this; one context in 22 is corrupted, and at this point I have no idea why. Somthing related to the structure? the name itself? who knows! Hopefully I can find the problem quickly. murf Steve, I get this: dialplan show smvoice-mediaport There is no existence of 'smvoice-mediaport' context Command 'dialplan show smvoice-mediaport' failed. my extensions.conf has a context: ; media [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf Then express.dnis.conf has: ; This file is generated from MessageNet EMACS ; Phone Caller ID DNIS Manager screen ; MMAUDIO : EBOX 4300 - exten = mediaport_audio_visual,1,Goto(smvoice-mediaport-audio-visual,s,1) [smvoice-mediaport-audio-visual] exten = s,1,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup Not seeing what the problem is here. especially since 1.2 and 1.4 both work. Jerry -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] AST-2008-010: Asterisk IAX 'POKE' resource exhaustion
Asterisk Project Security Advisory - AST-2008-010 ++ | Product| Asterisk| |--+-| | Summary| Asterisk IAX 'POKE' resource exhaustion | |--+-| | Nature of Advisory | Denial of service | |--+-| |Susceptibility| Remote Unauthenticated Sessions | |--+-| | Severity | Critical| |--+-| |Exploits Known| Yes | |--+-| | Reported On | July 18, 2008 | |--+-| | Reported By | Jeremy McNamara jj AT nufone DOT net | |--+-| | Posted On | July 22, 2008 | |--+-| | Last Updated On| July 22, 2008 | |--+-| | Advisory Contact | Tilghman Lesher tlesher AT digium DOT com| |--+-| | CVE Name | CVE-2008-3263 | ++ ++ | Description | By flooding an Asterisk server with IAX2 'POKE' | | | requests, an attacker may eat up all call numbers| | | associated with the IAX2 protocol on an Asterisk server | | | and prevent other IAX2 calls from getting through. Due | | | to the nature of the protocol, IAX2 POKE calls will | | | expect an ACK packet in response to the PONG packet sent | | | in response to the POKE. While waiting for this ACK | | | packet, this dialog consumes an IAX2 call number, as the | | | ACK packet must contain the same call number as was | | | allocated and sent in the PONG. | ++ ++ | Resolution | The implementation has been changed to no longer allocate | || an IAX2 call number for POKE requests. Instead, call | || number 1 has been reserved for all responses to POKE | || requests, and ACK packets referencing call number 1 will | || be silently dropped. | ++ +-+ |Commentary|This vulnerability was reported to us without exploit code, less than two days before public release, with exploit| | |code. Additionally, we were not informed of the public release of the exploit code and only learned this fact from a | | |third party. We reiterate that this is irresponsible security disclosure, and we recommend that in the future,| | |adequate time be given to fix any such vulnerability. Recommended reading:| | |http://www.oisafety.org/guidelines/Guidelines%20for%20Security%20Vulnerability%20Reporting%20and%20Response%20V2.0.pdf| +-+ ++ | Affected Versions| || | Product | Release | | | | Series| | |--+-+---| | Asterisk Open Source |1.0.x| All versions |
[asterisk-users] AST-2008-011: Traffic amplification in IAX2 firmware provisioning system
Asterisk Project Security Advisory - AST-2008-011 ++ | Product | Asterisk | |+---| | Summary | Traffic amplification in IAX2 firmware| || provisioning system | |+---| | Nature of Advisory | Traffic amplification attack | |+---| | Susceptibility | Remote unauthenticated sessions | |+---| | Severity | Critical | |+---| | Exploits Known | No| |+---| |Reported On | July 18, 2008 | |+---| |Reported By | Tilghman Lesher tlesher AT digium DOT com | |+---| | Posted On | July 22, 2008 | |+---| | Last Updated On | July 22, 2008 | |+---| | Advisory Contact | Tilghman Lesher tlesher AT digium DOT com | |+---| | CVE Name | CVE-2008-3264 | ++ ++ | Description | An attacker may request an Asterisk server to send part | | | of a firmware image. However, as this firmware download | | | protocol does not initiate a handshake, the source | | | address may be spoofed. Therefore, an IAX2 FWDOWNL | | | request for a firmware file may consume as little as 40 | | | bytes, yet produces a 1040 byte response. Coupled with | | | multiple geographically diverse Asterisk servers, an | | | attacker may flood an victim site with unwanted firmware | | | packets. | ++ ++ | Workaround | The only device which used this firmware upgrade | || procedure was the IAXy ATA device, and the last firmware | || upgrade was more than 18 months ago. It is unlikely that | || any IAXy devices in use today still need the last | || firmware upgrade. Therefore, deleting the firmware image | || from the directory where it is served from and sending a | || reload event to the Asterisk server is sufficient to | || purge the firmware image from the Asterisk server's | || memory. An Asterisk server which is unable to serve out | || the requested firmware image will reply to any such | || request with a much smaller REJECT packet, which is | || smaller than even the FWDOWNL packet. | ++ ++ | Resolution | This firmware download procedure has been disabled by | || default in Asterisk. If you should still need to upgrade | || IAXys in the field, there is an option 'allowfwdownload' | || which can be enabled. However, due to the reasons | || specified on the Workaround section, it is recommended| || that you leave this option disabled and enable it only on | || secure internal networks when an IAXy is initially| || provisioned. | ++ ++ | Affected Versions|
[asterisk-users] Asterisk 1.4.21.2 and 1.2.30 Released
The Asterisk.org development team has released Asterisk versions 1.4.21.2 and 1.2.30. Both of these releases include fixes for two security issues. Both of these issues affect users of the IAX2 channel driver. For more details on these vulnerabilities, see the published security advisories, AST-2008-010 and AST-2008-011. AST-2008-010: Asterisk IAX 'POKE' resource exhaustion - http://downloads.digium.com/pub/security/AST-2008-010.html AST-2008-011: Traffic amplification in IAX2 firmware provisioning system - http://downloads.digium.com/pub/security/AST-2008-011.html Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Suddenly my Asterisk Box Hanged up all calls
Hi everybody, I hope someone could help me. Today one of my PBXs seemed as it were frozen. When I typed *asterisk -rx core show channels; sip show peers *or *queue show *it showed the last activity before the crash. Finally I realise that something was wrong because there were no activity at CLI and incoming calls to IVR didn´t get through, then I decided to restart asterisk and things were good again. What logs say is: /var/log/boot.log: Jul 22 13:51:49 pbx asterisk: Interruption of asterisk succeeded Jul 22 13:51:49 pbx asterisk: Start of safe_asterisk succeeded /var/log/asterisk/messages.1983: [Jul 22 13:40:36] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies [Jul 22 13:40:36] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies [Jul 22 13:40:36] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies [Jul 22 13:40:36] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies [Jul 22 13:40:37] ERROR[2717] channel.c: Translation to slin failed, dropping frame for spies *[Jul 22 13:51:49] NOTICE[3328] loader.c: 1 modules will be loaded.* *[Jul 22 13:51:49] NOTICE[3328] cdr.c: CDR simple logging enabled.* *[Jul 22 13:51:49] NOTICE[3328] loader.c: 157 modules will be loaded.* *[Jul 22 13:51:50] NOTICE[3328] config.c: Registered Config Engine odbc* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Connecting asterisk* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Registered ODBC class 'asterisk' dsn-[asterisk]* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Connecting mysql1* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Registered ODBC class 'mysql1' dsn-[MySQL-asterisk]* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Connecting mysql2* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: Registered ODBC class 'mysql2' dsn-[MySQL-asterisk]* *[Jul 22 13:51:50] NOTICE[3328] res_odbc.c: res_odbc loaded.* *[Jul 22 13:51:50] NOTICE[3328] config.c: Registered Config Engine mysql* *[Jul 22 13:51:50] NOTICE[3328] app_queue.c: Queue members successfully reloaded from database.* *[Jul 22 13:51:50] NOTICE[3328] pbx_ael.c: Starting AEL load process.* *[Jul 22 13:51:50] NOTICE[3328] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'.* *[Jul 22 13:51:50] NOTICE[3328] pbx_ael.c: File /etc/asterisk/extensions.ael not found; AEL declining load* [Jul 22 13:51:50] NOTICE[3437] chan_sip.c: Peer '806' is now Reachable. (61ms / 2000ms) [Jul 22 13:51:50] NOTICE[3437] chan_sip.c: Peer '831' is now Reachable. (62ms / 2000ms) [Jul 22 13:51:50] NOTICE[3437] chan_sip.c: Peer '819' is now Reachable. (30ms / 2000ms) [Jul 22 13:51:50] NOTICE[3437] chan_sip.c: Peer '813' is now Reachable. (30ms / 2000ms) [Jul 22 13:51:51] NOTICE[3437] chan_sip.c: Peer '856' is now Reachable. (26ms / 856ms) First *channel.c *errors **happens when someone use the HOLD and MUTE function in a SIP softphone based on PortSIP Development Kit (we are building an IAX client based softphone to fix these ERRORS and hidden -at logger.conf- WARNINGS). At *Jul 22 13:51:49 *I decided to reboot asterisk *[service asterisk restart] * 4 days ago the system had reboot itself but PBX functions went back automatically, that were no the case today :(. Agents told me they were receiving congestion messages (the most for invalid numbers) in outbound calls just before the crash. My * version: 1.4.17 We have round about 60 users with no more than 30 simultaneous calls at the peak hour (inbound and outbound) My inbound campaing has 10 agents in a simple queue (Local Channels and SIP extensions) My hardware is an IBM iSeries 386 *[Linux pbx_daniel.com 2.6.9-67.0.1.ELsmp #1 SMP i686 i686 i386 GNU/Linux]* I`ll really appreciate every help you can give me. Thanks in advance, Daniel Arohuanca +51 1 3594122 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Suddenly my Asterisk Box Hanged up all calls
On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: My * version: 1.4.17 Please upgrade to 1.4.21.2. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
Philipp Kempgen wrote: Come on. People want simple answers. So: Can Asterisk duplicate CallManager? [y/n] *scnr* I think for questions like this, we should always consider the m (maybe) option. :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Cisco vs Asterisk
Rob Hillis wrote: Philipp Kempgen wrote: Come on. People want simple answers. So: Can Asterisk duplicate CallManager? [y/n] *scnr* I think for questions like this, we should always consider the m (maybe) option. :) Or my preferred approach: Yes means no and no means yes. Can Asterisk duplicate CallManager? [yes/no] -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] 3-way calling for IAX channels
Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] 3-way calling for IAX channels
Thanks for answering Noah, There is no way to enable it at the softphone itself? As is the case for hardphones like my Polycom. Daniel On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED] wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Asterisk dimensioning
Alex Balashov wrote: Conrad Wood wrote: Unless I am mistaken and there *is* some way to run 400 simultaneous calls over 2 PRIs... Traditionally, there hasn't been. But now that they've got that Large Hadron Collider going... :-) Are you thinking that with dark matter we can build dark PRI to go with all the dark fibre? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Help With dial plan
Thanks for the wild guess. But The user(who is myself) is dialing 3000.
It only failes to work when I use patterns. So I thought I am making a
mistake on the syntax, I have checked all the books I have and the
internet and I can't see anything wrong. :-\
Rizwan Hisham wrote:
maybe the user is dialing something other than 3000 and that extension
is not registered on your asterisk. just a wild guess.
On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi list,
Have installed trixbox and I am working with a fxo gateway to get
fxo calls to trixbox. I am using sip to send the calls from the
gateway to trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial
plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I
be doing wrong?
James
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Rizwan Hisham
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begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED]
title:Lead Consultant
tel;work:+254-722-490994
tel;home:+254-722-490994
tel;cell:+254-722-490994
url:www.agile.co.ke
version:2.1
end:vcard
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Re: [asterisk-users] 3-way calling for IAX channels
Hi Daniel - There is no way to enable it at the softphone itself? As is the case for hardphones like my Polycom. A phone can definitely do conference mixing. As you asked about IAX channels on the asterisk-users list, I assumed you were asking about how to do this in asterisk. My experience with IAX softphones is somewhat limited, but maybe if you indicate which phone you're using, somebody could provide you with assistance. - Noah Daniel On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED] wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] sometimes extensions can't be called
Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Help With dial plan
Hi James -
Thanks for the wild guess. But The user(who is myself) is dialing 3000. It
only failes to work when I use patterns. So I thought I am making a mistake
on the syntax, I have checked all the books I have and the internet and I
can't see anything wrong. :-\
Sounds like time for some more in depth troubleshooting. What happens
when you follow Mark's suggestion of adding a NoOp statement? What
happens when you create other pattern-match extensions? Do they work?
What messages are you getting on the console? Is the call being
rejected by the SIP device? What messages do you get when SIP
debugging is turned on? etc, blah, blah, blah...
- Noah
Rizwan Hisham wrote:
maybe the user is dialing something other than 3000 and that extension is
not registered on your asterisk. just a wild guess.
On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED] wrote:
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I be doing
wrong?
James
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Best Regards
Rizwan Hisham
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Re: [asterisk-users] sometimes extensions can't be called
Are the users registered to both active servers? 'sip show peers' in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] sometimes extensions can't be called
Hi, i see my extensions are there: 118103/118103 210.212.213.214 D N 5060 Unmonitored 118101/118101 210.212.213.214 D N 5064 Unmonitored 118102/118102 210.212.213.214 D N 37743 Unmonitored 118102/118102 210.212.213.214 D N 37743 Unmonitored 118101/118101 210.212.213.214 D N 5064 Unmonitored 118103/118103 210.212.213.214 D N 5060 Unmonitored and i have this on both servers: 17 sip peers [Monitored: 0 online, 0 offline Unmonitored: 15 online, 2 offline] regards, nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] How can I Disable call-waiting
Hello I really need to disable the call-waiting on my sip phones I studied most of the posts on internet and did it on my asterisk but not useful. in fact I need a comment that I disable call-waiting but without enable call-limit because I want to keep the waited caller on a queue. I tried many states on sip.conf and also users.conf but I didn't do any changes on my extensions.conf and I don't know am I right? If you can help me to hear the busy tone(!!) on my eyebeam sip phone, in fact you return my work to me. thank you Best regards -- Naraghi e-mail1:[EMAIL PROTECTED] [EMAIL PROTECTED] e-mail2:[EMAIL PROTECTED] [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
