similar to: allow sip traffic in peer settings

Displaying 20 results from an estimated 60000 matches similar to: "allow sip traffic in peer settings"

2006 Feb 02
0
Sip - no peer or user found on incoming call
Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I
2009 Jan 11
2
sip peer permit/deny - Need some explanation
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
2011 Nov 17
0
2 same sip extension number on 2 asterisk - call not passing on certain condition
Hi list, something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both having an extension [115], one as type peer (caller side 1.4) and one as friend (callee side 1.8). Phones from both location connect to Asterisk from LAN. Router are Linux boxes. Connection between the 2 sites is done like this: On the callee side [115] ;callee type=friend host=dynamic secret=otherSecret
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (
2009 Jul 28
2
Possibly I don't understand sip peers
I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a
2014 Mar 29
1
additional range parameter for sip peer
Many ITSP are using loadbalancers, so if somebody registers on a sip peer with specific dns host, an incoming call may be received from a different ip and the host value in peer section doesnt match, so it will go to default context. For example Telekom or 1&1, biggest providers in Germany are using too many different addresses that its not practical to define them all (up to 50 hosts
2005 Dec 23
6
SIP permit/deny
I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses. [a00090101] type=friend context=Company1 username=a00090101 ;secret=180 ;insecure=very host=dynamic mailbox=company1@vmusers
2015 Apr 02
3
Update peer IP address
I do not want set allowguest=yes. The problem is, there is no official list with ip addresses of Telekom Germany. But I think all ip addresses comes from the ip range 217.0.0.0/13. I have now the following addition to sip.conf. I think it is the only safe option. Or what would you say? [telekom](!) context=from-trunk type=peer defaultuser= authuser= remotesecret= fromdomain=tel.t-online.de
2006 Nov 07
3
Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the
2010 May 10
1
Dialing a SIP Peer without using register strin
Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd asterisk server. scenario is like this. on my system i am using this peer in sip.conf. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=yes
2015 Apr 02
0
Update peer IP address
Actually, the IP address is still used to identify the incoming invite. With the insecure=port option set, Asterisk will presume the invite to still match the trunk account even if the NAT router has mangled (changed) the port number. My suspicion is that when the new register goes out, it's creating a new state in the firewall, resulting in a new port number, which is why you would have to
2015 Apr 02
0
Update peer IP address
That sounds like asterisk was working 100% correctly. If you receive an INVITE from an unknown IP address, then it should fail. Unless you want to allow anonymous, which is genearlly a very bad idea. If you are registering to IP X, but the provider may be transmitting invites from any number of other IP addresses, then you need a list of IP addresses, and have a trunk configuration set up for
2012 Dec 24
0
How to disable authorization during Incoming calls to asterisk
Hi, List My SIP provider requires no authorization in incoming calls to my asterisk 11.1.0 box. I was sure previously that "insecure=invite,port" disabled authorization request during incoming calls to asterisk. But today I tried to connect to a provider (which has MERA MVTS) but could not disable auth requests in incoming calls from this provider with this option
2007 Jan 04
1
asterisk sip peer/user matching methodsforauthentication backwards?
I have considered opening a bug report on this, but wanted to get some feedback and make sure I am not missing something in the way of a simple work around. What is the scenario in which this impacts your implementation? Ours is the desire to use the same realtime SIP database for many asterisk servers, and route the call based on a "home server" value in the realtime database. The
2012 Apr 26
0
Peer SIP authentication with Taqua switch
I'm using Asterisk 1.8.6.0 on my router talking to my ISP's Taqua 7000 (?) switch. I'm using a config that looks like: [sip_proxy-out] type=peer authuser=208nnnnnnnn remotesecret=xyzzy qualify=100 host=n.n.n.n call-limit=5 nat=no ; sendrpid=yes insecure=no But the Taqua responds to outbound INVITES with 403 Forbidden (oddly, not 401 or 407). Also, from what I can tell, the
2005 Aug 26
1
realtime sip channel configuration -> insecure option
Hi all I'm trying to figure out what values are valid for the "insecure" option in a realtime configuration table. The table field is 4 chars long and the actual valid values for this is longer. Can I modify the field length or has this changed? Below is where I looked, if I'm not looking in the right place please let me know. the field on the table is: ... `insecure`
2015 May 28
0
Peer is UNREACHABLE
I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >
2015 Apr 02
2
Update peer IP address
Okay, Scott, I think we are on the wrong path. Maybe I'm wrong though. I will summarize again briefly the problems together: The peer ip address could be another than the ip address of incoming invites After an re-register the REGISTER is send to the new SIP server, answered with OK. But the peer ip address is still the old one (sip show peers). If now is a INVITE, the request is answered
2005 May 12
0
Asterisk, SIP and NAT: Help needed!
I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP
2008 Mar 25
1
Sip exten matching based on contact: sip header?
Asterisk: 1.4.17 with sip realtime Openser 1.3.x Hi, I had this setup working fine until I try putting OpenSER in the picture as a proxy. Unauthenticated calls go to a PRI based app via a ZAP channel, calls to sip users get send to them etc. Now with a proxy in the picture asterisk asks the incoming calls for authentication "407 Proxy Authentication Required". It seems that the