similar to: SCCP problem with ATA188, Asterisk@home and chan_sccp

I don't think that will fix my problem. The hints on the individual user extensions (101, 102, 103 and 104 below) are working just fine. sccp.conf example of 1 user: [devices] type = 7970 description = User1 tzoffset = -6 autologin = 101,401 speeddial = 102,User2,102@wct-internal speeddial = 103,User3,103@wct-internal speeddial = 104,User4,104@wct-internal device => SEP000F90CEF9D3

I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

Ok, so multiple people have said that hinting is possible with chan_sccp on the 7940/7960's and such, has anyone got this working? How do you go about getting this to work? I'd use the wiki, but it's link to the mailing list topic on that doesn't work anymore :( -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I have some shares on a media server that are considdered "Local, offline content", namely they should be accessible if the rest of the network is down, and each system has it's own group of users who are allowed to maintain it. the media servers in the livingroom are only for my wife and I, but each person can modify the one in their own bedroom and noone elses bedroom.

I've got a domain member server configured as such: Load smb config files from /etc/samba/smb.conf Processing section "[videos]" Processing section "[music]" Loaded services file OK. Server role: ROLE_DOMAIN_MEMBER [global] workgroup = TEMPEST server string = Media Server (%h) (Livingroom Television) security = DOMAIN map to guest = Bad

Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

It's not just you. Same thing happens here. I went back to 1.0.7. Stefan Gofferje wrote: > Hi folks, > > I used to have some constructions like > > exten => number/callerid,1,Goto(somewhere) > > After updating to 1.0.8 those does not work any more. > Any hints? > > Regards, > Stefan >

Hello, I am trying to set the tcpIpApp.sntp.gmtOffset="0" setting on the polycom phone provisioning template and I see that it has a variable for ${DSTOFFSET} in the template I tried adding "dstoffset = -28800" to users.conf and sip.conf under both general and individual users but can't get the setting to set. Where/How do I set the variable? Thanks, Bryan --------------

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the

Hello list, Has anyone on list had experience with getting the 7914 module working with Asterisk, probably using the chan-sccp drivers? I have 7912 and 7940 phones working OK, I need something for reception, and I was thinking a 7960 with two of these modules would handle it, but need to know that someone has got it working somewhere (seeing lines that are free, busy etc) Mikel

Hello I need someone to install Chan_SCCP for me and get it working on Elastix with Cisco 7937 Interested party please msn me on sam__tam AT hotmail DOT com or email me back -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090207/58b10073/attachment.htm

Hi all, we are happy to announce the new test-release of chan_sccp. The Cisco 7920 support is working now, however, some call handling stuff is hardcoded for testing. Also some more in-deep-knowledge of Skinny was archieved, how they handle calls. Please test it with your 7940/7960 and 30VIP and report any problems using our bug tracking tool. The latest tarball is available at: