search for: cisco1

...620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing Dial("SIP/-080e8b58", "SIP/0@cisco1|20|Ttr") in new stack -- Called 0@cisco1 -- SIP/cisco1-f5b3 is ringing == Spawn extension (default, 9342199, 1) exited non-zero on 'SIP/-080e8b58' NOTICE[5126]: File chan_sip.c, Line 1814 (process_sdp): Content is 'multipart/mixed;boundary=uniqueBoundary', not 'ap...

...else has experienced this? i tried to go back down to 1.0.5, but it crashes now when starting so apparently something got overwritten with the install of cvs to really mess stuff up.. I upgraded to get better skinny support, which i kind of regret right now ;-) Executing Goto("Skinny/6609@cisco1-2", "locallines|7501|1") in new stack -- Goto (locallines,7501,1) -- Executing SetVar("Skinny/6609@cisco1-2", "ALERT_INFO=Bellcore-r3") in new stack -- Executing Macro("Skinny/6609@cisco1-2", "oneline|SIP/sipura2") in new stack -- E...

Hello, I tried to configure a cisco 7905 IP phone using the skinny channel but I had not much luck. The relevant portion of skinny.conf is: [cisco1] device=SEP000F3487F8E3 callerid="Alex" <123-456-789> mailbox=500 callwaiting=1 transfer=1 context=default threewaycalling=1 line => 500 ; Dial(Skinny/500@cisco1) I set up the tftp server, and prepared the following XMLDefault.cnf.xml: <Default> <callManage...

...stupid, but I can't seem to see what it is. Anyone have any suggestions? sccp.conf [general] keepalive = 30 context = internal bindaddr = 192.168.1.1 port = 2000 debug = 10 firstdigittimeout = 16 digittimeout = 8 [devices] type = 7960 description = Cisco1 tzoffset = 0 autologin = 104 ; speeddial = 101, 105 device => SEP00036BC3852B [lines] id = Cisco1 pin = 1234 label = 104 description = Cisco1 context = internal ;callwaiting = 1 incominglimit = 2 mailbox = 500 vmnum = 500 cid_name...

Hey all, Anybody been successful running DHCPD on a GRE tunnel? When I tell DHCPD to listen on cisco1 I see this in the log Jul 23 16:21:03 atlantis dhcpd: cisco1: unknown hardware address type 778 Here is the output of ifconfig cisco1 Link encap:UNSPEC HWaddr 8B-8E-28-32-00-00-00-00-00-00-00-00-00-00-00-00 inet addr:10.199.0.2 P-t-P:10.199.0.2 Mask:255.255.255.0...

...4 8:01 PM To: R-Help Subject: [R] Omegahat down? Hello! For 2 days can not connect to www.omegahat.org :( Something happens? traceroute to www.omegahat.org (128.105.174.32), 30 hops max, 38 byte packets .......... 13 144.92.128.196 (144.92.128.196) 161.792 ms 162.260 ms 160.893 ms 14 g1-2.cisco1.cs.wisc.edu (128.105.1.14) 160.996 ms 163.381 ms 161.632 ms 15 * * * ______________________________________________ R-help@stat.math.ethz.ch mailing list https://www.stat.math.ethz.ch/mailman/listinfo/r-help PLEASE do read the posting guide! http://www.R-project.org/posting-guide.html [[alt...

...ll = on autoanswer_tone = 0x32 remotehangup_tone = 0x32 dtmfmode = inband dnd = on [devices] type = 7960 description = Cisco2 tzoffset = 0 trustphomeip = no autologin = 105 device => SEP00036B0123 [lines] id = 104 pin = 1234 label = 104 description = Cisco1 context = internal ;callwaiting = 1 incominglimit = 2 mailbox = 104 vmnum = 500 cid_name = Cisco1 cid_num = 104 line => 104

Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu.... It's just a normal call between to channels that I have to join for few minutes.

...ject: Samba PDC guys, i'm configuring a Samba Primary Domain Controller. i compiled the 2.2.5 version with ./configure --prefix=/usr/samba --with-quotas and followed the Samba-HOWTO-Collection I configured my Win2K workstation with the following parameters: COMPUTER NAME = CISCO1 COMPUTER DOMAIN = COMPLAB USER NAME = root PASSWORD = <the password i set in smbpasswd -a root> i'm having "The speficied network password is not correct" error. many times i changed the smb password of root i tried...

Here's my situation. I have two receptionists that answer incoming lines. Each has a 7960G with 5 incoming lines each. I'm trying to set this up so each line on each phone doesn't utilize call waiting. My problem seems to be that ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always returns cisco1. Here are the sip.conf entries: (mind you, there are entries for frontdesk1-10 and each phone logs in with 1-5 and 6-10 respectively) [frontdesk10] type=friend ;Theresa Sprocket username=frontdesk10 cal...

...right). Better result now: I get as far as "Registered with Asterisk PBX" on the phone, only to have it continue to "no Callmanager found" and retry (ad infinitum). Here's what I get from asterisk debug output: == >> Got message RegisterMessage Auto logging into cisco1 == Sending Packet Type DisplayPromptStatusMessage (48 bytes) == Sending Packet Type RegisterAckMessage (24 bytes) == Sending Packet Type CapabilitiesReqMessage (4 bytes) == >> Got message IpPortMessage == >> Got message HeadsetStatusMessage == >> Got message Version...

...AT session ; qualify=yes uses default value ;callgroup=1,3-4 ; We are in caller groups 1,3,4 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 ;defaultip=192.168.0.60 ; IP address to use if peer has not registred ;[cisco1] ;type=friend ;username=cisco1 ;secret=blah ;qualify=200 ; Qualify peer is no more than 200ms away ;nat=yes ; This phone may be natted ; Send SIP and RTP to IP address that packet is ; receive...

-----Original Message----- From: Adams, Gavin Sent: Monday, August 04, 2003 6:10 PM To: 'asterisk-users@lists.digium.com' Subject: Cisco 7960, SIP, NAT, Voicemal Hey all, I've got a couple 79xx phones working peer-to-peer and am now trying to work on the voice mail. In extensions.conf: [ATL] exten => 4001,1,Dial(SIP/gadams)|10 exten => 4001,2,Voicemail,u4001 exten =>

...; qualify=yes uses default value > ;callgroup=1,3-4 ; We are in caller groups 1,3,4 > ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 > ;defaultip=192.168.0.60 ; IP address to use if peer has not registred > > ;[cisco1] > ;type=friend > ;username=cisco1 > ;secret=blah > ;qualify=200 ; Qualify peer is no more than 200ms away > ;nat=yes ; This phone may be natted > ; Send SIP and RTP to IP address that packet is >...

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,