similar to: Cisco 7960 and Skinny

Hi, Is there a module within the Asterisk standard distribution that provides SNMP features? Is there any third party software for that purpose? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/918b5ebf/attachment.htm

Hi, I use Asterisk 1.0.5 and TDM04B. When an incoming call over ZAP channel hangs-up, it takes 10 seconds until Asterisk realize that. How can I shorten the time of hang-up detection? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050614/de3eec0f/attachment.htm

Good day all How do I get my asterisk and sip to use the password.I'm using x-lite.If I use just the username and no password it still logs on? Here is my sip.conf entry? [101] type=friend callerid="Test User" <101> context = test_1 ; Default context for incoming calls username=101 secret=123456 host=dynamic dtmfmode=inband ; Choices are inband, rfc2833, or info

Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm

Hi all, I have a problem starting the ztdummy. Here is what happens: [root@asterisk /]# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy After this, ztdummy is visible with lsmod, but when I try MeetMe, I get following: == Parsing

Hi, I have stress tested the Asterisk Voicemail. We have encountered problem with simultaneous calls that are sent to the same mailbox. It occurred that several calls were writing to the same file. It seems that there is a synchronization issue in the Voicemail application. Did someone else find this issue? What would be the solution/workaround for it? Regards, Stojan Sljivic

Hi, We have phones registered at another soft switch, and would like to use Asterisk as a Voicemail system. Is it possible and how to configure Asterisk to send NOTIFY messages (for MWI) to the endpoints that are not registered to the Asterisk? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, please tell me, is original skinny support in Asterisk stil under development or is better to try chan_sccp from http://chan-sccp.sourceforge.net ? my first try was unsuccessfull (chan_sccp compile OK, but module loading fail during Asterisk startup) and my phone (C7940) seems to be not supported in original chan_skinny :( PJ

Evnin' (o; As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... The Cisco 7970 registers and is being acknowledged by * but that's it... I see no lines on the 7970 display configured and it is not reachable or it can't make any outboudn calls... The docs are pretty non-existent for skinny and the

Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have

I could use some skinny/Cisco help here. Was finally able to get the phone registered to * but whenever someone tries to call that phone it freezes and reboots itself. Same thing happens when you pick the handset up off the 7960G; it locks and reboots about 5 sec later. Here is what * shows when I plug the phone in: -- Starting Skinny session from 64.72.107.1 Device SEP000F3442E4A7 is

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

Please forgive me my ignorance ... I've spent two days trying to find out something about the format of the default configuration file, which CCM produces. The only example I have so far is the one from the chan_sccp source. There were tons of references on entering the callmanager commands on a cisco command line - which I don't have (don't need thanks to chan_skinny + chan_sccp).

Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After

Hi there We have a Cisco VOIP environment here, with hard and softphones. I have a softphone account/etc, but I'm a Linux user and (as far as I'm aware) there is no Cisco softphone for Linux. However I can run Asterisk. So I was wondering if there is a way to "convert" a SIP phone transaction into a SKINNY transaction so that the Cisco environment thinks it is a Cisco

I am in the process of upgrading a Cisco 7960 to the P0S3-06-3- 00.bin image. It's hanging when it reaches the status message "upgrading". I have two other phones that successfully upgraded. When I researched on the internet it appears that there is a problem with upgrading directly from the image that is currently on this particular phone, PC030300, and that I should go to

Greetings: I have a working configuration for SCCP on our LANS which doesn't route RTP correctly to a skinny phone behind NAT registering from a remote public IP. Configuration: asterisk 1.4.35 servicing only skinny phones trunked to asterisk 1.2.40 which services chan_phone FXS, zap FXO and SIP phones; both instances of asterisk are behind NAT and run on the same host (using different base

Hi All; Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? Regards Bilal

Hello, My Cisco phone has software: Boot Load: PC030300 Ver: 3.2(7.0) And I want to upgrade it to SIP 6.0 Is it possible or I have to upgrade to ealier then 6.0 and then to 6.0 ? bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031121/7331dff5/attachment.htm