similar to: Re: Double sip logins

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and

Geetings to all. I am having a hell of a time getting incoming SIP connections to work properly, and am hoping that someone can help me. Here is what I am using as a guide (from the wiki): "Incoming SIP Connections When Asterisk receives an incoming SIP call, the SIP Channel Module first tries to find a [user] section matching the caller name (From: username), then tries to find a [peer]

hi i've got an AAstra 9133i ip phone, when i've bought it, i've set it to use a SIP/400 account on my asterisk, then, i've changed settings and i've set set phone to use a SIP/500 account . now, when i connect the phone to tthe network, it register itself on asterisk with both accounts!!! -- Registered SIP '500' at 192.168.100.188 port 5060 expires 120 --

I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering if this is still a valid parameter. If not, does anyone know how I can do this? Thanks, Joe

Hi, I'm still seeing mails being downloaded twice via POP3 in my setup, running rc27. The log file shows: dovecot: Mar 18 22:08:05 Error: POP3(joe): mbox sync: UID inserted in the middle of mailbox /home/joe/Mail/INBOX (442671 > 80, seq=10, idx_msgs=153) dovecot: Mar 18 22:08:05 Info: POP3(joe): Disconnected: Logged out top=0/0, retr=153/1069109, del=153/153, size=1066449 The next POP3

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

Grretings to all, I am having a problem with a customer's queue setup that I don't really understand. Background: Customer has 5+ queues and is using dynamic login to the queues based on SIP/XXX for example. There is a litle script that runs that allows agents to log into particular queues via the keypad. The user can log in to any queue that he wants, including multiple queues. The

I have setup several queues for a customer. Their periodic announcement says please wait for the next available agent, or press * to leave a voicemail. This does not work when the message is playing. The message stops, but the user is left in the queue. Q-exit with * works the rest of the time fine. Has anyone seen this or know if it shoudl actually work differently? Regards to all, Joe

Greetings to all, I ma having a problem with channel variables on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be

Greetings to all. I am hoping someone can help me out with a problem I am having getting my Cisco phones, 7960s and 7940s, to download the appropriate files from our TFTP server. The TFTP server is running on Fedora Core 4. The TFTP server appears to be setup properly: service tftp { socket_type = dgram protocol = udp wait =

Dear all i'm planning a migration to asterisk for a high volume IVR service (from 1000 to 1500 concurrent call) The IVR service is based only on DTMF tones so the features required is - play feature - dtmf detection Asterisk will receive calls via VOIP (SIP with g711 codec) The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in

Anybody know the proceedure to factory reset the a 7960 phone running 6.3 SIP software? I've tried holding # when booting the phone and nothing, i can do that on my 8.2 phone but this phone i just got with 6.3 isnt working. Also **# doesnt work either.. -- ~Shaun

I have a directory on Unix exposed as a Samba share. All the filenames are greater than 8.3 format. When I go to this directory from a DOS prompt and do 'del *.*', I get a bunch of errors 'Unable to find file ...'.. But, if the directory contained 500 files, after the 'del *.*' it contains 400 files. I repeat the del *.*, get a few more errors Unable to find file ...

Greetings to all. I am currently implementing call queues for a customer and have come across several "problems". The customer is an airline representative, and will be using call queues for different airline reservations. The customer requires that any agent be able to login to any number of queues. This means that queue members have to be dynamic, not using "member =>

currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet. here si the sip debug msg, we got a Message type: DISCONNECT (69) and unable to dial any numbers. Jul 12 12:56:26 WARNING[1523]: chan_zap.c:1931 pri_find_dchan: No

We're currently in the process of organising a migration from mbox to maildir mail storage on our server, and are hoping to phase this in gradually using dovecot's capabilities. Procmail has already been configured to deliver mail to either the existing mbox or to a new maildir for the user based on the (non)existence of a .converted file in their home directory. Where we are having

Hi All, This is my first post ... I *HAD* been trying to get RHEL AS 4 onto this Dell PE 6400/700, but it just wouldn't go without some weird tweaking. I am VERY happy to say that CentOS 4.4 installed without a hitch! NICELY done folks! The only real issue for me is that the machine fails to reboot or shutdown correctly. It requires physical interaction with the power switch,

I am trying to ssh from my Windows/Cygwin xterm window into my CentOS52 servers w/o using passwords but my keys seem like they're being ignored. I created the key in cygwin using: $ ssh-keygen -t dsa -f id_dsa And copied id_dsa.pub over to centos:~/.ssh/authorized_keys Ideas? tia, - Joe

On 1/18/2018 9:22 AM, Ken McDonald via samba wrote: > > Hi, thanks for your help. Your suggestion makes sense, however I think there should be some way for users to be able to change an expired password from login dialogue. > Actually I had a problem doing this previously with NT4 style Samba domain and never looked into a resolution. > Now that I've found Samba does AD style

i am developing a web application to manage callcenter, i will shortly release it on sf.net this is my problem: i will grant to users the possibility to transfer calls to other users using a web interface, for example if SIP/200 is talking with SIP/400 who wants to transfer the call to SIP/500 i use this commands with manager API: Action: Redirect\r\n Channel: SIP/200-sads\r\n ExtraChannel: