similar to: clicktocall via manager with cisco 7905

Hi all, I have this weird problem. I'm running asterisk 1.0.3 on Debian Sid (official debian package). We have 2 fritz ISDN cards. All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial("SCCP/michiel-00000004",

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

Hi! ~ dovecot --version 2.2.22 (fe789d2) I am wondering if there is a way to set up virtual users with multiple valid passwords. We want to be able to provide users with device/app-specific passwords for their email accounts, as well as being able to create temporary "access tokens" for technical support when required. I quickly found out that passdb using passwd-file or an

hi ---------modem.conf :---------- msn=240862922 incomingmsn=240866365,6365 device => /dev/ttyI2 group=1 device => /dev/ttyI1 ; ttyI3, ttyI4 ---------extensions.conf ;------- [sip] exten => _XXXXXXXXXX,1,Dial,Modem/g1:BYEXTENSION (Sjphpone) Call to : 024076xxxx result : --Executing Dial(Sip/roseau-6163","Modem/g1:BYEXTENION") in new stack -- Called g1:024076xxxx --

Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

Hi ! On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone button (or select line with line button - which automatically put second line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5, chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of 7960 is

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just

Hay guys, got trouble with registration with cisco 7975 Here is the debug : <--- Received SIP request (576 bytes) from UDP:192.168.1.61:49533 ---> REGISTER sip:192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381 From: <sip:111 at 192.168.1.4>;tag=0c8525a68961001f44d503e2-d9359bd3 To: <sip:111 at 192.168.1.4> Call-ID: 0c8525a6-89610004-b972d038-5864c98e

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

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Hey, Can anyone comment on the difference between the 7905 and it's upgrade, the 7905-G ? Has anyone used these phones in a configuration? Is SIP well implemented in the 7905 ? Thanks in advance, Mat

Here's the output: Feb 9 15:51:26 SSI001 kernel: SFW2-INext-ACC-TCP IN=eth0 OUT= MAC=00:0f:ea:73:88:12:00:40:2b:67:5b:a7:08:00 SRC=192.168.1.54 DST=192.168.1.2 LEN=48 TOS=0x00 PREC=0x00 TTL=128 ID=51248 DF PROTO=TCP SPT=1964 DPT=139 WINDOW=65535 RES=0x00 SYN URGP=0 OPT (020405B401010402) Feb 9 15:51:28 SSI001 kernel: SFW2-INext-ACC-TCP IN=eth0 OUT=

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

I wrote a simple c programme and do some dtrace test. the progromme shows below(write.c): /////////////////////////////// #include<stdio.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> char *value="hello,my name is wanjm\n"; void main() { int fd; fd=open("/export/home/wanjm/temp/read.txt",O_RDWR|O_CREAT); while(1) {

Greetings All, I am a Postfix/Courier-IMAP user and am interested in Dovecot. I have install Dovecot via FBSD ports. I have got to the part where I can telnet in to the server. -bash-2.05b$ telnet 192.168.1.61 143 Trying 192.168.1.61... Connected to 192.168.1.61. Escape character is '^]'. * OK dovecot ready. x login pete testing * BAD [ALERT] Plaintext authentication is disabled, but

Smbclient worked fluently to create tar compatible backups of all files on an SMB/CIFS share. For which I used an Ubuntu 8.04 LTS Server Edition configuration with LAMP, OpenSSH, Samba and BackupPC packages installed. Testing with smbclient \\\\win-pc\\share -U administrator -d 3 (debugging level 3) returns: (...) name_resolve_bcast: Attempting broadcast lookup for name win-pc Got a positive

I've got asterisk working great with cisco 7905 sip phones. I've just got one issue that I can't figure out. I have a 5300 connected to the PSTN via PRI. When I send a call from the 5300 to asterisk it will ring the 7905 phone for 4 seconds then drop. This is because the only message asterisk sends to the cisco is the 100 trying message. The 5300 receives that and sends a call

I'm deploying a fairly large number of Cisco 7905 with SIP connected to an asterisk PBX. The problem: the 7905 has this nice "feature": You can set a "voice mail number" in configuration, so that you can listen your voice mail just by hitting the "messages" key on the phone. It just autodials "8500" or whatever else. The customer wants this, so

How to configure cisco 7905 with asterisk ,if you please can send me step by step configuration steps . Thanks Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 ********************************************* No employee or agent is authorized to conclude

Hi, I'm using a Cisco 7905 (with SIP image CP7905010300SIP050414A) on Asterisk 1.07. Whenever the user gets voicemail, the MWI goes on, and when the user listens to the voicemail, the MWI goes off. However, the phone is configured (as in the default config by Cisco) to re-read its configuration from tftp every hour. This 'warm reboot' also turns off the MWI, and it won't