similar to: gxp2000 and no sound asterisk 1.6

Hi Our sip provider has two servers that sends calls to our asterisk 1.6. When server 1 sends call everything is working, but when server 2 sends call I get [Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303390' rejected because extension not found. And the provider get an "404 not found" error on their side. What

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or

Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description "Asterisk daemon" start on runlevel-2 stop on shutdown respawn exec

Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

Hi I have asterisk 1.6 and running queues with realtime mysql. I am trying to set another musiconhold then "default" but I cant get it to work, I have an musiconhold entry in my queue_table, but don't know what to put in there and where to put the file. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm,

Hi To dial an outside line i have to dial 0. I want to have that when we dial local numbers, that is we are in the 08 area, I don't want to have to dial 08, how to set this up in asterisk 1.6? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460

Hi Is there a user client that a group, like customer service can use? We have today an avaya IP-office with phonemanager pro and I want something equal to phonemanager pro, where you can logon to ques and see how many calls is in that queue and so on. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

Hi I have this in my queue extension and I see this in asterisk when I call to the queue, but no file is created in the directory any ideas? exten => s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID}) -- Executing [s at kundservice:1] Set("SIP/0850001175-b7942770", "MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12") Regards

We have 40 nortel ip2002 phones connected to asterisk 1.6, the problem I have is that the phone looses the connections with the server and then drops calls, we can reconnect but the customers don't like it. Anyone has the same problem? /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir: +46-(0)707548074, vxl:

Hi Im using a sip phone SPA921, and the one that calls me can hear me but I cant hear them, when I make the call I can hear them. Im running asterisk 1.6 behind a firewall, I have port 10000-20000 for rtp and 5060 for sip forward to my asterisk. Any tips? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir:

Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) I get this after I restart my asterisk 1.6, it all worked yesterday. I have the

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to

Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they

I've tried to use a Cisco power injector to supply power over Ethernet to a GXP2000 phone without success. Although when I plugged these phone to a PoE capable Cisco Switch it worked without a problem! Knowing that all these three equipments implement IEEE 802.3af protocol, why doesn't it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho.

1. We just dial the extension directly and have speed dials setup for the first 6 parked positions. We don't use *8 at all. 2. Change the config on the phones under Account to "Send DTMF via RTP (RFC2833)" -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noc Phibee Sent: Thursday, February 08,

Hi i use a lot of Grandstream GXP2000 with BLF How to set up on the same key BLF blinking call interception? So that someone is able to take a call that is destinated to another user phone Thanks bye

Hi i have two problems with my Grandstream GXP2000 : 1- When i wan pickup a call, that's don't work's (*8EXTEN) and when i test whit Softphone, i have a error too, he say me *8201@MyContext not found .. in features.conf, i have: [general] parkext => 700 parkpos => 701-720 context => parkedcalls

Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in "busy" state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = <password> host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502