search for: authusers

I know some of this doesn't belong on this list, but I am just including it for problem history. I am trying to setup IMAP Voicemail with our email server. We are using a non-standards based groupware server called FirstClass. The server has some built in support for IMAP. My problem seems to be that the authuser flag is not supported. When I use mtest in the imap toolkit to connect to

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

I thought I would give the new IMAP support a spin on my home server, but without much luck so far. Asterisk 1.4.0 Dovecot 0.99.14 Maildir format C-client 2006d The imap server is also the Asterisk server, so connections are on the localhost. The error posted to the logs is: IMAP Error: Can't open mailbox {127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote specification

Hi Tony, Try adding "fromuser=xxxxx", maybe "username=xxxx" isn't enough... Just a guess, it already solved a few problems for me. -Manuel -----Messaggio originale----- Da: Tony Hoyle [mailto:tmh@nodomain.org] Inviato: martedì, 18. maggio 2004 13:03 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

Hi, did anybody manage to configure Asterisk (1.4.8) with imap voicemail storage together with an Microsoft Exchange Server (2003)? I can connect my asterisk to a local dovecot imap server without problems. But if I change the settings to our exchange server I can?t connect... Here is my config: imapserver=exchangeserver imapport=143 expungeonhangup=no imapfolder=INBOX imapflags=notls

Hi, I have two accounts with broadvoice. Now, I want to be able to distinguish between them. I though that this would be simple by adding "/EXTEN" at the end of the register statement. For example: register => num1:pass@sip.broadvoice.com/1000 Unfortunately, this is not working. When I call into my box I hear busy tone. My config looks like this: [root@voip asterisk]# cat sip.conf

Hi all, I have two sipgate accounts (numbers), if I have both accounts register only one will work for incoming calls (which is all i'm interested in). However if I disable either account the other account will work perfectly. Am I missing something obvious? Thanks in advance, Ray. Excerpts from sip.conf - [general] 8<---- SNIP! ---->8 Register => 1212121:aaaaaaaa at

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

Hello, everyone. No need to read this message. I'm posting for documentation for other poor, ignorant slobs like me who are struggling to pull together the many technologies to make converged networks happen. Hopefully, this will help save someone else the time I spent. I started the below email until I realized I had solved multiple parts of a compound problem but not all at the same time.

Well the subject line probably says it all. I have a Quintum D3000 which I'm supposed to be getting connected up to our Asterisk system. No matter what I try, neither username or authuser config works. I've also tried md5auth and it still refuses to register. Any one have a config they could share with me? Any help would be much appreciated. Neil

Hi again. I cant get my Budgetone registered in Asterisk, and I cant find what's wrong... uff. This is my config: This fragment is from my sip.conf: [12345] type=user user=12345 username=12345 secret=12345 authuser=12345 qualify=1000 nat=no host=dynamic dtmfmode=rfc2833 reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw context=sip_default And this is from my

I'm trying to test IMAP in 1.4.17 and it appears to be not working. I've compiled imap-2007 with the following on a CentOS 5 box: make slx EXTRACFLAGS="-I/usr/include/openssl -fPIC" and I've configured and compiled asterisk with the following: ./configure --with-imap=/usr/local/src/imap-2007 The compile and install went just fine, no warnings and no errors that I saw.

Hello List. I have a question regarding connecting two asterisk servers. I'm trying to learn how asterisk comunicates from server to server. I already have a server running smoothly now, I'm installing another one to test it along side the actual one. I would like to run different scenarios: 1. Have one of the boxes at a different location outside the LAN and have them communicate. 2.

Hi, I'm making some tests with Tellfree brazilian provider. I'm using 2 users A and B, one for calling and the other to receive calls. When I make a call I can see (from the CLI console) user A is calling user B but user B does not answer (the phone continues to ring) even if the "sip show registry" command says user B is registered. In my sip.conf I have: register =>

I configured this once now I forgot what I did. Two Broadvoice accounts. Incoming is simple - just use the phone numbers. Outgoing: Dial out on a specific line and/or set up the groups and select the other "line" if the first one is busy? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

IVR Number :17275691533 When I try it from xlite configuring my provider directly, it works perfectly. When I try to dial out from dialer , it doesnt work. [sip8] type=peer username=user fromuser=user authuser=user secret=password host=8.14.146.111 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 What cld be the reason ? --------------

I have tried to setup ssmtp to relay through gmail. root=my gmail address AuthUser=my gmail address AuthPass=my password FromLineOverride=YES mailhub=smtp.gmail.com:587 UseSTARTTLS=YES TLS_CA_FILE=/etc/pki/tls/certs/ca-bundle.crt I get this error: Authorization failed (534 5.7.14 https://support.google.com/mail/answer/78754 145sm16287775qhb.20 - gsmtp) I am not using two step verification at

I don't know what's means about register in sip.conf such as: register => user:secret:authuser@host:port/extension even if I registe a sip proxy , but how use it ? I think : when incoming from sip proxy to asterisk : user a --> sip proxy --> asterisk --> pstn sip proxy : SER in ser.cfg farword ( "192.168.0.10" , 5060 ) this will forward a call to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I don't really understand the extension function on: register => user:secret:authuser@host:port/extension First question - -------------- well, it's only local, or is important for authentication on external sip server? Example: I've one external sip account, the number is the URI also (111), pass 'xxx' I'll