search for: mypeer

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 insecure=port,invite nat=yes canreinvite=no call-limit=15 accountcode=...

I have defined a SIP friend without username and secret, only IP-based. I have also defined an accountcode for that "friend", as follows:   [mypeer] type=friend host=192.168.0.100 port=5060 context=mycontext canreinvite=no accountcode=mypeer   Unfortunately the accountcode for the calls originating from "mypeer" doesn't show up in the log (either CSV or ODBC). All the other "friend"s I have (which authenticate with user...

...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 qualifyfreq=300 insecure=port,invite nat=yes outgoinglimit=4 incominglimit=4 [mypeer](peer) host=10.0.138.226 defaultuser=2155551941 fromuser=2155551941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IP (tos 0x0, ttl 64, id 9242, offset 0, flags [none], proto UDP (17), length 555) 10.0.1.3.5060 > 10.0.138.226.5060: SIP, length: 527 OPTIONS sip:10.0.138.2...

...newals works okay because they both give the same address, but new requests get two different responses). I thought that only one server would reply to a particular request. Also, every DHCPACK is followed by a message like this in the log: Sep 23 15:45:50 rad2 dhcpd: bind update on x.x.x.x from mypeer rejected: incoming update is less critical than outgoing update Any ideas? I subscribed and asked over on the ISC-operated dhcp-users list but haven't had any responses. Google finds others asking about the same log message, and the only responses seem to be "well, if you get it for eve...

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>

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

...--- [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Sending to XXX.XXX.XXX.XXX : 5060 (no NAT) [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Using INVITE request as basis request - 78938c042d374b341c4f1b60071d319f at XXX.XXX.XXX.XXX [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found peer 'mypeer' [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 0 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 3 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 101 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description form...

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]

What is the no authority found problem? And how can I register with * on IAX. It keeps rejecting the request telling that XXX not dynamic host. rejected any idea THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/a8a5907d/attachment.htm

Hello My provider allows to activate/deactivate a forwarding rule by sending a SIP MESSAGE. This is done outside a call. That is, while there is no ongoing call, a SIP client just sends the following message: MESSAGE sip:543951354657 at callfwd.sip.providerx.com SIP/2.0 Call-ID: b9ba106e-613a-46b9-8a4d-0efb4dc0a0f2 CSeq: 1 MESSAGE To: <sip:543951354657 at