similar to: rules/routes traversal misunderstanding

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-bounces at lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailinglist at unix-solution.de> > To: asterisk-users at lists.digium.com > Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send

Greetings all, I'd like to start implementing a private DUNDi peering group between one of our asterisk servers hosted at a datacentre and the various asterisk boxes sitting at clients' premises. On most of the clients' boxes the dialplan will have an [in-pstn] section containing the various numbers that should be recognised by that box. Where they're from a VoIP provider they

Hello everyone. I've stumbled upon a strange networking issue with multiple interfaces on CentOS 5. The network setup is just like the diagram in http://lartc.org/howto/lartc.rpdb.multiple-links.html It looks like linux is not routing correctly outgoing packets on interfaces different from the one of the default gateway, but instead broadcasts an ARP request on the link, looking for the

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

Hello In the LARCT how-to subitem: 4.2.2. Load balancing the following phrase says: "" Instead of choosing one of the two providers as your default route, you now set up the default route to be a multipath route. In the default kernel this will balance routes over the two providers. It is done as follows (once more building on the example in the section on split-access): ip

Hi, I put on sip.conf the following line #include "sip.d/*.conf" inside I have files like that provider1.conf provider2.conf But asterisk does not want to load it This is the error Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Not found (No such file or directory) this

Greetings list, Over the last few days I've been gearing up to replace a couple of our servers with 1.6 as something of a testbed, but I'm encountering a few problems, and wondering if anyone can help... In extensions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include =>

Hello, I have the following setup: sip phones <->SER <-> asterisk <-> voip provider1 <-> voip provider2 i got a toll-free DID from voipprovider1 to allow people from outside to call into asterisk, get authenticated, and use voipprovider2 to call out (kind of a primitive calling card app). anyway, voiprovider is giving my

How do you setup the timing in Meetme conference? I have a x100p and tdm4x card. When I dialing to my conference I get a request to schedule in the past error message. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 10, 2004 10:48 AM To:

Hi! I'm really hope you can help me solve a little mystery, the mystery is probably just my misunderstanding ! sorry... I've got an iaxy talking to my * box which connects to two providers. I'm running the stable release of the pbx. The only thing is that when dialling from the iaxy the ringing tone isn't heard while calling someone - you just hear silence then, they either

Using Asterisk 13.3.2 on CentOS7 and pjproject 2.4, I can't make asterisk try to send a register. I have configured my pjsip.conf similar to https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples#res_pjsipConfigurationExamples-ASIPtrunktoyourserviceprovider,includingoutboundregistration my pjsip.conf: http://pastebin.com/raw.php?i=EA0PEcrb using tcpdump, I never even

Hi I''ve just tracked down a wierd error that AFAICT is caused by an error not being raised in the setup: context "An Asset" do setup do @provider = Provider.create(:name => "Provider1") @product = Product.new(:name => "Product1", :provider => @provider) @applicant = Applicant.new(:first_name =>

Ok ive been trying to get this to work for about half a year now. Ive searched all over the internet for a solution for my problem. Ive found some solutions, but they only led me to yet more problems. What we want to do is the following: I live in a student complex with 7 other people. Every room has its own internet connection from the same ISP. Ip, gateway, subnet are asigned through dhcp on

How can one measure the effect of NAT traversal packet loss? We currently have no solution for NAT traversal for our SIP clients. There is no doubt that packets are getting lost. What is not clear is how much damage this does. On the face of it, everything seems fine. Could this be so? Perhaps we're suffering a degradation in quality or our call setup times could be improved. How can we

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

It seems to me that ipsecnat tunnel type is not complete. Latest drafts of ipsec nat-traversal use udp port 4500 for nat-traversal communications. (It''s called port floating). That is needed to get rid of ugly ipsec passthru devices. Now ipsecnat opens port udp/500 from any source port. And I think ipsecnat won''t work at all with gw zone defined? I''m not sure about

Hi List, After some research, it seems the only reasonable thing to do in order to get SIP phones behind NAT working reasonably well without fiddling with the DSL router is to have some kind of far end nat traversal mechanism. Is there any way to do this with open source tools? I've seen somewhere that far end nat traversal can be achieved with SER + nathelper does the job... has anybody

It seems that tab traversal between TextCtrl widgets doesn''t work by default. How do you enable that? -- R. Mark Volkmann Partner, Object Computing, Inc.