similar to: CallerID retain on internal transfer

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

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 =

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

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

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 have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten

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

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

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

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 =>

Guys and dolls, Greetings, I hope you all are in good health, great spirits and your glasses never empty. I have a samba, openldap question. I am trying to setup a FC-4 box to be a PDC for a small network of about 150 users. I was following the HOWTO on the SAMBA site. Everything seems to be fine however I cannot join the domain. I get the error "User name could not be found." The

Dear list I am relatively new to networking problems of this kind so apologies for the potentially simple question. I am trying to upgrade an existing network to one using Samba 3 to configure roaming XP profiles on a limited number of clients. I have re-written the smb.conf file to reflect what I think are the appropriate settings, and this passed testparm successfully, but I am unclear

Greetings to All. A little background about what I am trying to do, and please excuse the length of this post. I am setting up a voicemail (VM) system based on Asterisk. From what I've heard Vonage uses Asterisk as their VM platform as well. I am running 1.2beta with a MYSQL backend for extensions and VM user info. All the sound files and vm messages are being stored through an NFS

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I

Hi, I've just updated (using yum under fedora 5) to the latest version of samba and am now encountering a major problem with samba. Whenever I attempt to access shares (don't have printers or other such non disk shares) the access fails (after prompting for password, and performing some level of authentication, it notes if no password is entered that Anonymous login successful).

Hi, How can i access to request.remote_ip in a model? --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFFw@public.gmane.org To unsubscribe from this group, send email to

Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk

Hi Guys, Since the automatic fedora update to 23c I'm having problems with samba crashing when a share access is attempted. However, I also notice that the nmbd daemon isn't being started up properly (though its startup script, through init.d, shows it as starting up OK). The error that is being shown in the logs is: [2006/09/28 20:06:35, 0]

I must be doing something stupid, but I can't figure it out. I have three PRI lines connected to Asterisk, one from the phone company, and two more connected to PBXs. Asterisk uses the incoming DID information to decide which PBX to route the call to. Should be simple. Asterisk is clearly getting the caller id info from the phone company: -- Accepting call from '512345xxxx'