similar to: Sip Proxy

Hi, My asterisk's version is 1.6.0.26. I've couple sip providers and I've for new SIP provider I need define outbound proxy. Everything is ok in peer section (outboundproxy=192.0.2.1). But what about SIP REGISTER messages? I need send SIP register messages also via outbound proxy. How to write SIP OUTBOUND call register statement and send this to proxy? If I define in general

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as

Hello List, I have been trying to configure a sip profile ( peer / friend ) for each of my customers behind a sip proxy for some time, but I have had no success, so I would appreciate your help. Customer -> OpenSIPS -> Asterisk -> PSTN The opensips is working as a sip proxy with record route, for billing, load balancing and authentication purposes. I would like to be able to define

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

Dear All. I want to become a wholesale VoIP traffic Provider , and i don't have a experience about the software used this career . I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk telephony server gave me all i need to start my business . thanks -- Best Regards Mohamed Daif -------------- next part -------------- An HTML attachment was scrubbed... URL:

How do you configure an outbound proxy for Asterisk? If the sip call is not local I want everything to go to a designated sip proxy. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/bdd57a91/attachment.htm

Hi, all, I am a beginner of asterisk SIP, now I have 3 pc, one runs asterisk as a SIP proxy, and the other two run softphone(Ubiquity) as User Agents. As below: User Agent <------------> Proxy Server <----------------> User Agent (Ubiquity) Asterisk SIP (Ubiquity) My sip.conf and extensions.conf is as follows: sip.conf [general] port =

Setup is: Asterisk --->NAT--> SIP Proxy I have following entry for SIP Proxy in sip.conf [Proxy] type=peer host=Static IP (NAT Firewalls public IP) username=xxxx secret=xxxxx nat=yes???????????????? canreinvite=no???????? qualify=yes Proxy sends a call and I get this error Found no matching peer or user for <NAT's Public IP:70001 NAT is using 70001 as the source port in the

Hi:I 've a question: I'm using asterisk@home; I've seen the dialparties.agi , I want to do this; I've one softphone and I want register it in 2 different Proxy;only X-lite permitted this, all others no; I want have more proxy with others softphone;I run asterisk - R and I've seen when a softphone is registered(when I run it). My question is: Who is the file who makes the

I checked the app store and haven't found anything promising, but I figured I'd ask here. Does anyone know of a SIP or IAX client for a non-jailbroken iPhone that will communicate directly with a machine running Asterisk? I know that there's at least one offering that seems like it's essentially a proxy (App runs on iPhone, iPhone talks to 3rd party server, 3rd party server talks

I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using "X-Lite" I have no issue with settings as follows: Display Name: Any Name User name: 00575000010XXXX Password: 00575000010XXXX Authorization user name: <blank> Domain: directnationalloan.com Checked "Register with domain" and "Send outbound

Hello, I use both ser/asterisk . In fact i wish asterisk to forward all the sip requests which are not handled by domain=domain.tld in sip.conf Here is a diagram: The sip agents use the Sip proxy as an outbound sip proxy and domain=domain.tld . When the sip agents dial sip:user@otherdomain.tld so the request is sent to sip proxy and so to Asterisk. I wish Asterisk to Look up the

Dear List: Please correct me if I am wrong, but as I understand the requirement to connect an IP-PBX to the PSTN via a SIP trunking service provider (ITSP), a SIP proxy service and a SIP registration service are required local to the IP-PBX. Does Asterisk include this functionality and/or are there other open source projects providing these SIP services? Thanks a bunch! John --------------

Hi, I'm going to forward SIP request to special outbound SIP proxy with none SIP port. I have this in my sip.conf [sip_proxy-out] type=peer ; we only want to call out, not be called username=408 host=192.168.0.95 outboundproxy=192.168.0.74 port=9097 I want a To: 408 at 192.168.0.95 by proxy 192.168.0.74:9097 but it turns out the "To" also has the port To: 408 at

Hi Everyone, I am running multiple instances of Asterisk in Proxmox and so far I had one central Asterisk feeding all others with trunks from one provider. Now, I want to connect each Asterisk server directly to the provider. Based on my understanding, each connection made to the provider port 5060 would be on a port that is unique to that server. And so other connections made to the same

Hi Everyone: I have a THREE questions. What is a sip proxy and what is the benefit of having one with Asterisk? I am well aware that we have a sip channel in Asterisk and that we have SIP registration. I am not sure why you would need a SIP server and OR a registration server. Second question, with Asterisk are you able to do video on VOIP video phones? Last question, does

Hi Everyone, I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs. So, the tunnel is established nicely and everyone can ping others. "sip show peers" shows the local subnet of the SIP Phones registered (192.168.100.0/24 ). But there is the old bad one-way audio. Calls also drop after few seconds. In the SIP

Dear All, I am totally new in this arena and I am still waiting for my installation process on freebsd to finish, but I wanted to make sure of the following: - Call routing between IP telephones can be done regardless of who made the phones? - Asterisk does support SIP V2? - it does act as SIP Proxy and Register? -- Thx MAG -------------- next part -------------- An HTML attachment was

I wanna connect proxy server. my IP Phone -> my asterisk -> service provider's proxy server -> extern PSTN phone but asterisk server can't register to proxy server. I think that configuration is right. When asterisk send to register request, proxy server don't response. I did capture packet. but no response. MY setting sip.conf [kms]

Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP -> VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the