Take out the "" from the SIPmacADDRESS.cnf
# Line 1 Registration Authenticationline1_authname: cisco7960# Line 1 Registration Passwordline1_password: grendel
Also for SIPDefault.cnf
# Proxy Serverproxy1_address: 192.168.144.1 ; Can be dotted IP or FQDN
Reset the phone,"Michael J. Tubby B.Sc."
/04-20:31:38 currently running on server (pid = 2249) -- Executing Macro("SIP/105-8dfd", "iax-out|1700613") in new stack -- Executing SetCIDName("SIP/105-8dfd", "Gonzalo Gasca") in new stack -- Executing SetCallerID("SIP/105-8dfd", "170
=ulawcallerid="Gonzalo Gasca"421058
extensions.conf
;Free World DialupFWDUSERID=421058FWDUSERNAME=Gonzalo GascaFWDGW=IAX2/[EMAIL
, name, email address for attached voicemail msgs;100 = 5045,Gonzalo Gasca,[EMAIL PROTECTED]101 = 1109,Ingrid Gasca,
**
SIP.conf
Check file permissions. create the files in your Linux server do not import them from Windows environment
Restart TFTP server, test from another TFTP client if you are able to download those files,
From which SCCP load you are trying to convert to SIP? There are several known issues. Depending
Hi John/forum
I saw in your config that you are using a Mediatrix box,
I have problems of delay routing for all my calls to the PSTN,
Im also using a Mediatrix box (1204 ) version 2.4.10.68
I was wondering if you are using the same Mediatrix box and if you have the same problem?
Or maybe you can
Hi,
If you still are in the Skinny image Settings --- Network config in that menu press **# and you will get the phone unlock.
Otherwise, if you are in SIP you need to do the following:
Once the telephone has booted -- Settings -- 9 Unlock config --- Enter password
The default password is cisco
I got similar issues, Im running P0S3-07-2-00 loadInyour tftpboot folder in your TFTP server make sure you have these files:
CTLSEP000D651CF3FB.tlvSEP000D651CF3FB.cnf.xml
SIP000D651CF3FB.cnf
[EMAIL PROTECTED] tftpboot]# cat CTLSEP000D651CF3FB.tlvP0S3-07-2-00[EMAIL PROTECTED] tftpboot]# cat
Hi everybody,
I have setup a Mediatrix 1204, the calls worked fine, both incoming and outgoing.
The problem here is the delay.
When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4 seconds after answer or sending the call.
For OUTGOING
My Dialplan for the Mediatrix box
Miguel,
Congrats, i was testing your R2/MFC link, and I was able to made lots of calls, all of them worked fine.Thanks for setting up this link.
When i hang up, there were no dead air, music on hold worked fine, when I called to a conference worked fine also, busy line Telmex recording worked
Hi everybody,
I have setup a Mediatrix 1204, the calls worked fine, both incoming and outgoing.
The problem here is the delay.
When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4 seconds after answer or sending the call.
For OUTGOING
My Dialplan for the Mediatrix box
You need to create a SIP trunk in CCM and in Asterisk a peer in sip.conf with the IP address of the CCM (trunk)
In the trunk configuration change the transport to UDP.
Enter the IP of Asterisk.
And create a route pattern with gateway the SIP trunk
In Asterisk in extensions.conf create the route
Hi, Julio,
thanks for the tip, IAX and the incoming calls confi did the trick! FWD is up and running!
THANKS! and happy holidays!
Do you Yahoo!?
Yahoo! Mail - now with 250MB free storage. Learn more.___
Asterisk-Users mailing list
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP
Hi all,
I have just setup Asterisk, but the problem is that all RTP stream pass through Asterisk, I mean all call setup and voice stream pass trough Asterisk once the call is established.
Is there a way thatcall setup is established, the RTP stream pass just between the SIP endpoints.
Example:
Any example for configuring T1 PRI with Asterisk using a Cisco 2600 series router? MGCP config?
Do you Yahoo!?
All your favorites on one personal page Try My Yahoo!___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Yes seems to be no reason for using Unity instead of *
VM apart that Unity is windows based.
is just to test SIP protocol between them
=
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Hi group
Anyone has perform Unity SIP integration with Asterisk PBX?
Thanks!
Do you Yahoo!?
Check out the new Yahoo! Front Page. www.yahoo.com___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi group!
Anybody has implement Cisco Unity Voice Mail with Asterisk.
I read the Unity can do SIP integrations
Thanks!
Do you Yahoo!?
Check out the new Yahoo! Front Page. www.yahoo.com___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi Michael,
There are not newsthat 7970 support SIP yet, actually the most recent news from 7970´s are that they will have GigaEthernet ports.
I will email the latest SIP image tomorrow.
Thanks!__Do You Yahoo!?Tired of spam? Yahoo! Mail has the
Hey group!
Could someone could help me configure a DIal plan in order that when i dial 9 at the beginning i receive DIAL TONE?
Do you Yahoo!?
Take Yahoo! Mail with you! Get it on your mobile phone.___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Here is my configuration for MEdiatrix 1204, by default the 1204 strips one digit, so it is not necessary to use:
To dial OUTSIDE
EXTENSIONS.CONF
[locales];ignorepat = 9
exten = _9,1,Dial(SIP/[EMAIL PROTECTED])exten = _9,2,Congestionexten = _9,102,Congestion
To receive
Anyone knows some home-use PoE injector that works ok with Cisco 7960s?
Do you Yahoo!?
Yahoo! Mail Address AutoComplete - You start. We finish.
hi man,
if you are trying to upgrade to the latest version, change the permissions of the file, then to the SIPmacaddress.cnf file add a line that says image version = version, copy that line from the Sipdefault.cnf file, .
If the first workaround does not work, try to downgrade to version 2.3
give me a call tomorrow i could help you with your issue
52(55) 150054 54
GonzaloWayde Nie [EMAIL PROTECTED] wrote:
Wayde Nie wrote: I can get a Cisco MC3810 with a mixture of FXO and FXS ports, the MC3810 comes with a built in Ethernet port and I believe it does SIP too... Will this mean that I
Actually im working with Asterisk, a Mediatrix 1204 FXO ports to connect to PSTN SJ
labs softphone, i have the most recent Asterisk version, but when connecting to the
PSTN i have choppy voice problems, not internally just when connecting with my
Mediatrix gateway and ATA, my SJLabs softphone
I have just get an account on Iaxtel.com, and i woud like to know what can i do to
receive my Iaxtel calls in my asterisk server?
Actually i just can make IAX calls.
Thanks
--
___
Get your free email from http://www.hackermail.com
Powered by Outblaze
sorry for posting this basic question is about call forward, i look in the internet
and i get this, in order to make call forward
***
exten = _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2})
exten = _*5X.,2,Hangup
I would like to know if someone could help me when i recieved an incoming call on a
Meditarix 1204 how to redirect the call?
And the configuration i need?
_Thanks
--
___
Get your free email from http://www.hackermail.com
Powered by Outblaze
Create the profile
And a new windows appears:
Profile name
File name
Profile type Calls through SIP proxy
Then in SIP proxy,
click the sip proxy option
enter the Ip address of the proxy domain port
user domain
and proxy for nat and also the port (5060)
be sure u have the sip.conf file correct
I´d like to know if someone could help me with this issue:
Anybody there have the Mibs for .68 ver in 1204 ?
or meavy the .cfg file.
I found mediatrix box so hard to configure.
--
___
Get your free email from http://www.hackermail.com
Powered by
Thank u very much Rich!
I did what u suggested me, but im still having problems with the Mediatrix, actually i
dont have the MIbs for version 2.4.10.68, i tested 1204 with a different SIP server
called 3050 from Mitel www.mkcnetworks.com and it worked ok. Could help me with the
mediatrix
i would like if some could help me with a * and Mediatrix configuration...
i have this in my extensions.conf file
[outbound]
ignorepat = 9
exten = _901,1,Dial(SIP/[EMAIL PROTECTED])
exten = _901,2,Congestion
exten = _9020,1,Dial(SIP/[EMAIL PROTECTED])
exten = _9020,2,Congestion
this host go here
mailbox=100 ; Activate the message waiting light if this
voicemailbox has messages in it
callerid=Gonzalo Gasca 100 ; Caller ID
[line1]
type=friend ; This device takes and makes calls
username=line1 ; Username on device
can however
route properly incoming calls.
I hope this will help you,
Regards,
Wojtek
- Original Message -
From: Gonzalo Gasca [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 07, 2004 9:45 PM
Subject: [Asterisk-Users] Mediatrix 1204
Actually im trying to set up
Someone have the MIB for MEdiatrix 1204 version 2.4.10.68?
thanks
--
Almada Tres SA de CV
Mitel Networks
Eng. Gonzalo Gasca Meza
Service Engineer
52+(55)53730570
Mexico City, Mexico
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
36 matches
Mail list logo