[asterisk-users] Trixbox 1.2.3 - TDM400 FXOs - Outgoing Calls - Transfer # Not Wor king
Trixbox 1.2.3 - TDM400 FXOs - Flash (*) and # Not Working Has anyone run into this problem. I cannot transfer or park a call (#) on outgoing calls. Using Zaptel TDM400 FXO card. This may be normal but I wanted to check. Regards, Juan S. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Need Asterisk VOIP Support for Customer in Louisiana
I have a customer of mine that needs Asterisk VOIP support in Louisiana. Any Asterisk consultants that want to work in LA please let me know. Regards, Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Protocol Invalid when Upgrading to 7.3
Cisco 7960 Upgrading from 6.x to 7.3 get Protocol Invalid. I'm sure this has been discussed but has anyone figured this out. Regards, Juan Staalenburg Teksavers, Inc. (512) 255-8395 x1002 AIM: juanteksavers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 Upgrade Failing
Does anyone know how to get a Cisco 7940 w/FW ver 2.0.3 to v3x and above. Can't get it to upgrade on its own via TFTP. Phones w SCCP image will upgrade fine but I can't get these 2.0.3s to start the firmware upgrade. Thanks. Regards, Juan Staalenburg Teksavers, Inc. (512) 255-8395 x1002 AIM: juanteksavers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940 Upgrade Failing
You da man! Thanks. Regards, Juan Staalenburg Teksavers, Inc. (512) 255-8395 x1002 AIM: juanteksavers -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 08, 2005 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7940 Upgrade Failing Does anyone know how to get a Cisco 7940 w/FW ver 2.0.3 to v3x and above. Can't get it to upgrade on its own via TFTP. Phones w SCCP image will upgrade fine but I can't get these 2.0.3s to start the firmware upgrade. On some older 79x0's, we've had to delete a bunch of the config statements within the *.cnf files in order to upgrade. Deleting comments and the majority of statements won't have any impact in terms of upgrading, just make sure you keep a full copy of the files around so that once the upgrade is complete, you know what parameters were there to start with. The above is based on upgrading several used 79x0's (with experience). Seems not all phones were created equal (eg, maybe differences in buffer sizes, nvram, or something). I never did try to figure out why some worked and some did not without deleting that stuff. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Dial Out Issues - POTS Line
I am having dial out issues and was hoping someone could shed some light.
The problem is Intermittent:
extensions.conf
[globals]
; Trunk Info for outbound calls via PSTN - See the zapata.conf file in
/etc/asterisk
TRUNK=ZAP/G1 ;Trunk Interface
;MSD digits to strip (usually 1 or 0), 1 = remove a leading 9
TRUNKMSD=1
; --
; [trunklocal] - Defines extension for local calls through trunk
; --
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Congestion
exten = _9512NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9512NXX,2,Congestion
; --
; END [trunklocal]
; --
Problem Description:
I have (Qty 2) 4Port FXO Digium Cards with 8 POTS lines.
When dialing local numbers (as in 95551212) I intermittently get a message
from the operator We're sorry your call did not go through, please hangup
and try your call again. Other times the calls go through just fine. It's
almost as if Asterisk is not passing along all numbers dialed (possibly
dropping the first or last digit, 555-121 or 55-1212) causing the dialed
number to not go through.
Any ideas would be appreciated. Could this be a problem with my Telco?
Regards,
Juan Staalenburg
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[Asterisk-Users] Transfer # - Intermittent with Cisco 7905 SIP Phone
We are having some problems with our Cisco 7905 phones (SIP 7.x) and the transfer # functionality not working intermittently. Has anyone else experienced this? Problem Description: User dials # to transfer a call to another SIP extension. Both the user and the party on the other end hear a dial tone when # is dialed but Asterisk does not prompt for extension to transfer to. Intermittent problem, most of the time dialing # to transfer works fine. Regards, J. Staalenburg Teksavers, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
