similar to: Cisco 7941G-GE with Asterisk and CTPSEP odyssee

Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEP<mac>.tlv file. Most of the howtos say something about an empty file but

I've got a Cisco 7970 that's not completing its network registration to Asterisk. The "Registering" message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes "Updating CTL" then "Loading...", then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed

I'm difficulty diagnosing a tftp-hpa problem. Specfically, when performing a PXE boot, the machine hangs at the "TFTP...." and times out. This is only occuring at one network location (we'll call it network A). Now on network B, the computers on network can tftp boot no problem. If I run a tftp client from my workstation to connect to the tftp-hpa server on network A, I can

Hi, does anyone of you have made it to get the ANI also picked up? I mean: if I fetch a foreign call to me by using the pickup application I want to see the callerID/ANI of the caller to the foreign extension. Is that possible and if yes - how do I achieve that? Regards, Christophorus

Hi list, is it possible to get in the running dialplan the status of (SIP) lines without using AGI or anything like that? What I want is a stepwise calling: I have several SIP lines (let's say they are three) which I want to dial to alternatingly. But I do not want to dial to a already busy line and catch the busy. Instead I do not want to dial to that peer but to the next one. I want to have

hi list, I set up a new asterisk machine with asterisk 1.2.13 and misdn 0.3.1rc27. I use an e1 card with sip clients. My extensions look like this: [E1] <snip>...<snip> exten => 33006733,1,Set(CALLED=${EXTEN}) exten => 33006733,2,Dial(SIP/1@192.168.0.23) exten => 33006733-ANSWER,3,Answer() [SIP] exten => _X.,1,Noop() exten =>

Hi there, I want to set up an extension set that acts different depending on the count of used lines. I have a EuroISDN E1 board with mISDN and I only want to offer 10 lines. Therefore I set up a global variables LINES in the general section of extensions.conf and instantiate it with 0. I a call is incoming I check the LINES variable wether is 10 or more. If so I make a call transfer. If not

Hi list, I am searching for a possibility to do a certain call transfer method which is called "path replacement" in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and

Hi on the list, does anyone of you have experience with asterisk 1.6 and mISDN, pri primarily? Thanks in advance & Regards, Christophorus

Hi, I read that installing asterisk on Solaris is supported. Does anyone of you actually have experiences with that? And especially, does anyone of you have experiences in runnning asterisk with misdn unter Solaris? Thanks and regards, Christophorus

John; I'm trying to flash 7.1. What I followwd was the wiki page you refer to. My real question is do I need the .load file tftp debug shows the 7960G is asking for? Voipsupply didn't supply it. I'm trying to verify that's what I need befor I start hounding them. -JB Hawaii John Bittner wrote: >John, > >You have to go to 7.1 before 7.3 >Also look at this.

Hi, we are testing a new AD domain that will replace our old NT4 one, and we are setting up a new cifs vserver of our Netapp filer (running Clustered Dataontap 9.2). The new AD domain was a clean deployment created using "samba-tool domain provision --server-role=dc --use-rfc2307 ...". All seems to work well and the Netapp filer joins the domain without errors and seems to run fine.

2 new Cisco 7960 phones are requesting a CTLSEP file, seems like I triggered the universal application loader. I want to load the sip image 7.2 According to this Cisco information: http://www.cisco.com/en/US/customer/products/sw/voicesw/ps4967/products_upgr ade_guides09186a008022a968.html#wp1047292 If the CTLSEP MAC file is not present or is empty, the phone proceeds in nonsecure mode with the

Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091021/56254a0e/attachment.htm

Does anyone have the 7970 work with sccp? I have used the example configurations from the wiki. I can see that the phone is failing to retrieve the CTLSEP<MAC>.tlv file and then is able to retrieve the SEP<MAC>.cnf.xml file. The phone is hanging with the message "updating ctl". Thanks, John SEP<MAC>.cnf.xml: <device> <devicePool>

Hi, Does anyone know how to make this thing (7940) work with asterisk (chan_sccp module) ? I've set the configuration according to the wiki and now the phone just keep asking for CTLSEP<xxx>.tlv from my tftp server. In the cisco's web interface, I found this in the device logs : 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 ...

hi, I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.loads (I understand that is for 7911 model). In any case the phone reboots well. At this moment I

I am unable to copy a file larger than 4GB to a samba fileserver! My config: - client is windows nt 4.0 sp 5 (I tried sp 6 without any change, I tried w2k, too) - server (now) is samba 3.0.6 on suse linux 9.0 kernel 2.4.21-99-default/I586 - share is on a reiserfs hardware-raid array with plenty of free space (explorer correctly _shows_ >500 GB, mapped as ntfs drive) Every time I try to copy a

Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018

This for the archives in case it may help someone: I was able to upgrade two Cisco 7940's from firmware P0030301MFG2 to SIP 7.1 as follows: 1. Installed 7.1 images from the Cisco zip file to the TFTP server. 2. Specified "image_version: P0S3-07-1-00" in SIP<MAC>.cnf and SIPDefault.cnf 3. For the older of the two phones, renamed P003-07-1-00.bin to P0S3-07-.bin, making it