similar to: Incoming SIP connection

geetings, after my MERB app was up for about 48 hours I started to get Mysql::Error: MySQL server has gone away mysql hadn''t gone away, it is still running as the same process as when merb started. any advice on having ActiveRecord reconnect if this happens? thanks - jw

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

Hi all, I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result. Here is the problem: I have a peer -which is peer AND user- setted up like this [MyPeer] ; type=peer host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142 permit=yyy.yyy.yyy.yyy/255.255.255.255 context=from-MyPeer dtfmode=auto disallow=all allow=ulaw,alaw

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

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

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

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

Hello, Looking the asterisk 1.8 API documentation (http://www.asterisk.org/astdocs/api/index.html), I see a lot of new fields for sip register uris: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] But the *peer* is not explained anywhere. What it is for ? Regards, Guillaume Bour. -- Guillaume Bour<gbour at proformatique.com>

Anybody have any experience with setting up dhcpd in failover mode between two servers? I set this up on a couple of servers, and it seems to be working, but I don't think it is working "right". It appears both servers are replying to all requests (which for renewals works okay because they both give the same address, but new requests get two different responses). I thought that

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

Hi In Asterisk 1.6/realtime Mysql, we can't put a username/password in a Dial Command ?: 'Dial', 'SIP/Username:Password at MYPEER/${EXTEN},180,r' Thanks Olivier

Geetings, I have a machine with PAM using LDAP. Some old users do not disappear from: * getent passwd But, they are not in: * /etc/passwd or * LDAP Where are they? Any idea? Please, I am affraid of ghost. :-( TIA -- Cosme Faria Corr?a

Hi all, I have implemented a mail system with postfix and dovecot as LDA. Users who have exceeded their quota when receiving messages dont'see the message bounced. I see this error in maillog: dsn=4.3.0, status=deferred (temporary failure) This is my configuration: protocol imap { mail_plugins = quota imap_quota login_executable = /usr/local/libexec/dovecot/imap-login mail_executable =

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

Hello - I currently have 10 DID's coming into one Asterisk server, I seem to be having some difficulty routing based on the DID dialed and am hoping someone on the list can assist me. Here's the relevant info: Ingress SIP trunk: IP: 123.45.45.3456 DID's XXX-XXX-XX00-XX10 sip.conf: [general] useragent=Asterisk port=5060 context=default tos=lowdelay disallow=all allow=ulaw

hello sir, i build llvm-clang successfully in my pc but while running a pass i am geetting this error praveen at ubuntu:~/Desktop/LLVM/ build/Release$ opt -load /lib/LLVMHello.so --help Error opening '/lib/LLVMHello.so': /lib/LLVMHello.so: cannot open shared object file: No such file or directory -load request ignored. please help me to overcome the error. -------------- next part