similar to: cisco ip phne 7911G with asterisk

hello, I'm trying to use a remote SMB/CIFS share to backup a local directory tree. I use rsync "version 3.0.4 protocol version 30" and mount.cifs "1.10" with a kernel 2.6.18. The directory I'm trying to backup is quite big and has a lot of files inside. [root@linux dir]# du -hs . 50G . [root@linux dir]# find . -type f | wc -l 41929 I get the following error when

Hello, Throwing calls from Asterisk to PSTN (via a VoIP gateway) some operators sends an explaining audio, in situations as: The phone number does is not assigned The phone is powered off etc. The audio is sent before the call to be answered. So, in an automatic dialling application I'd like to recognize that audio to know what to do with those calls (queue them to a service, mark as wrong

Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: <loadInformation>SIP11.8-0-4SR1S</loadInformation> ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local

I have just got a Cisco 7941G and am experiencing the exact same problem (phone is requesting .tlv file from TFTP server and never asks for .cnf.xml file). The phone originally had SCCP on it, but I downloaded and flashed with the latest Cisco SIP image (8.4(3) released 2009-01-13). In reading your message below, it looks like you were going to try an incremental upgrade?did you have any

Guys, I assure you this is probably the most interesting and weird problem you have encountered (or definitely up there). I'm using ABE 2.1.2C and roughly 500 or so Cisco 7911G Phones. The following is what happens: When trying to dial a number from the cisco 7911G phone it may randomly get stuck on 'Dialing'. The SIP history on the asterisk end goes like this: 1. Cisco ->

I managed to use Cisco IP phones 7911g with asterisk with Sccp and chan_skinny without any configuration files in tftp. Only settings in dhcpd to indicate the tftp address and skinny.conf settings. the problem that I have is that from 8 phones two of them after working a while now are restarting after splash screen . I' ve made a factory reset. Uploaded a SIP firmware but the same.

I mounted vpopmail/domains in sshfs. root at 192.168.239.205:/mnt/disk-b/domains on /mnt/domains type fuse.sshfs (rw,nos uid,nodev,max_read=65536,allow_other) I used qmail+dovecot+vpopmail+mysql dovecot-2.0.beta4. I mounted sshfs with vpopmail/domains directory. my /usr/local/etc/dovecot/conf.d/mail.conf file mmap_disable = yes dotlock_use_excl = yes mail_nfs_storage = yes

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, 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 ...

I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian

Hello All I try to install net Framework 2.0 but without success. During intallation process Registering file System.EnterpriseServices.dll I:"C:\windows\Microsoft.NET\Framework\v2.0.50727\RegSvcs.exe" /bootstrapi 2: instalation crash > ------------------------------------------------------ > Instaling .net 2.0 runtime. Can take several minutes. See

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

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>

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.

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

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Hello, I have the following scenario: 1. VoIP Gateway G400 connected to PSTN 2. Asterisk server 1 (working as IVR) 3. Asterisk server 2 (working as ACD, with several agents connected) I have incoming calls coming from PSTN through the VoIP Gateway to Asterisk server 1 (IVR). When the IVR ends working with the call, transfers it to the Asterisk server 2 (ACD). In Asterisk server 1

Hello, I'm trying to find more information about this Remote Attended Transfers, as is explained in https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers for Asterisk 12 using pjsip stack Was Remote Attended Transfer implemented in previous versions of Asterisk (versions without PJSIP, Asterisk 11 and previous)? Where can I find configuration examples to do it work

Because OS X is not an ELF target, and support was only implemented in LLVM for making ELFs with debug info. I think the docs I wrote on this may suggest that the FSF gdb 7.0+ will work, but that's not the case. Sent from phne Reid On Jan 4, 2011 10:50 PM, "Yuri" <yuri at rawbw.com> wrote: With the same source code, gdb on Linux shows function names and on MacOS it

Hi, I'm experiencing a problem that existed in gluster 3.7. When I creating a (large) tar archive of brand new files I get a lot of 'file changed as we read it'. Further attempts to create archives from the same source work without error messages. Googling about this I found bug reports for 3.7 and other sites (i.e.