similar to: How to register with TCP transport ?

Hi, Digging on this case : 2009/5/26 Olivier <oza-4h07 at myamail.com> > Hi, > > In my sip.conf, I've got : > [general](+) > ; register=>tcp://trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > register=>trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > > When

Hi, this is the second time that i post this, may be a wasnt clear the first time. Im having problems with an incoming peer after i upgraded asterisk from 1.0 to 1.2.4, in 1.0 i used to configure the incoming peers like this: register => @prepago-in [prepago-in] type=friend host=192.168.10.102 ; this is the cisco's ip context = from-external dtmfmode=rfc2833 insecure=very ; required for

Where in the code does something like: register => user[:secret[:authuser]]@host[:port][/extension] from sip.conf 1) get parsed 2) actually register. I tried looking in channels/chan_sip.c and don't see where that happens. Can someone point me the right file and or function. Thanks, Jerry

Hi, Has anyone met something like this ? dialor*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 7541/7541 (Unspecified) D 0 UNKNOWN 7540/7540 (Unspecified) D 0 UNKNOWN 7534/7534 (Unspecified) D 0 UNKNOWN 7533/7533 (Unspecified)

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:

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?

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.

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 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

I finally decided to get a little source code dirt under my fingernails tonight and dig through chan_sip.c to understand how registrations are currently implemented. The hope is to perhaps at least seed some ideas about how to make registrations to a server name, which resolves to multiple IPs, either attempt each IP in the order they're returned by dns, or, simply attempt to register with

Hello, I want to do comparative study between xen and kvm during live migration of vm.analysis of downtime and also using wireshark and netperf tool to perform analysis, what else can be done? _______________________________________________ Xen-devel mailing list Xen-devel@lists.xen.org http://lists.xen.org/xen-devel

How does one configure asterisk for using TCP transport for SIP and not UDP? Thanks, Jerry ----------------------------------- SIP has the following features: ? Lightweight, in that SIP has only six methods, reducing complexity. ? Transport-independent, because SIP can be used with UDP, TCP, ATM & so on. ?

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 there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Hi, when using some non dynamic host eg. host=192.168.111.153 in sip.conf asterisk is not considering specific peer options eg. directmedia=off, transport=tcp if I set host=dynamic and register the sip phone it works as expected. Is this a bug or feature - I wanna disable the usage of directmedia for some peers with fixed ip but wanna allow it in general. Same with transport=tcp. [97]

Hi, I've stumbled upon a problem which I can't solve easily with the available options in tinc - at least as far as I see. If enlightenment is all I need, I'll happily accept pointers ;) I try to establish a connection between two hosts. Each host has multiple addresses assigned to it's internet interface. A stripped down list would be: Host 1: 2001:780:0:1e::1

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.

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

Hello, I'm tinkering with Dovecot-LDA in order to achieve: + Better intergration into sendmail, and + tweaking its way to act on errors. Attached patch is a "brainstorming" like suggestion: a) when sendmail is dropping a message, it spawns the LDA, captures stderr/stdout and, if the LDA returns an exit code != 0, generates either a DSN or, when EX_TEMPFAIL, queues the mail.