similar to: SIP Header VIA when behind NAT

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

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So

Hi, I 've a proxy on my network where some calls are routed to .... And as well some extensions on my Asterisk Server. What I would like to do is to accept all incoming calls from the proxy, wherever they are coming from or going to ... but, as soon as I receive a call with the same number as one extension defined in Asterisk (but through the Proxy !) , it refuses the call, saying that there

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

Hi All; I succeeded to have a success call from Polycom behind NAT while Asterisk has public IP address, but I was not able to have a succeed call (it was established, but no voice running, and then the call disconnected) if Asterisk behind NAT and Polycom behind NAT. When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to asterisk (at asterisk router) and to Polycom IP

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

Hello, We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. We have the issue with calls to these SIP phones - no audio. It is probably the problem with port forwarding on router - but I am not sure how can I forward same sip ports (5004 to 5100) to two phones (nat addresses?)? Any help appreciated! Z. Zivanovic -------------- next part

Hi all, OK. I've tried trawling the archives, but I'm not getting very far. I've got an Asterisk box behind a NAT which I want to register with a SIP provider. In my sip.conf I have (edited to protect the innocent): ----- [general] port = 5060 bindaddr = 0.0.0.0 disallow = all allow = alaw allow = ulaw allow = gsm context = bogus-calls tos = lowdelay nat = yes register =>

I am having problems in a couple of installations where I have SIP phones (both GS101 and ATA186) connecting to an asterisk box that has a public IP address, where the stations are behind NAT. I'm still testing to make sure I have all the permutations looked at, but from what I can tell, what is happening is that in situations where stations behind the NAT call out, no audio is passed

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

Hi i have to forward a call from my asterisk server on another server but my server is behind nat. How can i setup my extension.conf? Actually i have set up it as follows: exten => 0465666666,1,Dial(SIP/user@dormain) my server has a private ip 192.168.100.249 and doesn't have a public ip If i try to call SIP/user@dormain from an adsl connection (with a modem, without nat) the call is

Hi, I have following setup: BT100 ---- Firewall/nat 1 (www.ipcop.org) ---- Internet ----Firewall/nat2 (Vigor) ---- Asterisk . I'd like to use BT100 as local extension to Asterisk. I've done simple setup and BT100 can call Asterisk and place outgoing calls. However I cannot set him to qualify, cause it is claimed as unreachable. I have port redirection at Firewall 1 (to 5060 and rtp

Hi, I have an asterisk server sitting behind a pfsense firewall, I have successfully configured pfsense for NAT traversal, and clients from the internet can call clients inside the network of asterisk, as well as other clients registered with this asterisk server on the internet. The problem now is when a client from the internet do a call, the call disconnects in 10~20 seconds, but during

My asterisk box is behind a NAT firewall. I have friends that are on Earthlink, Vonage, etc. I'd like to make VOIP calls directly to them rather than going through the PSTN. With Earthlink, I can make this work through FWD peeting numbers, but that's sort of a waste of FWD bandwidth. WIth Vonage, it doesn't work. I suspect this is because of the breakage between FWD and Vonage that

I'm trying to set up * server behind NAT. The box is set up as DMZ in my DSL router, i.e. all incoming connections without explicit port mapping are forwarded to *. So far I'm unable to get this setup to work for either IAX or SIP (tried IAXComm & XLite softphones on public IP address). Data seems to come in fine (IAX/SIP debug shows message interaction taking place), but there is no

I currently do this for about 30 different cisco 79xx's connecting to some hosted Asterisk servers. Asterisk listens by default for any SIP connection on UDP port 5060. And will use RTP UDP port 10000 to 20000 The phones use UDP Port 5061 for incoming connections (from Asterisks or other SIP Devices) and use for RTP, UDP port 10000 to 20000. Now, if you are going to have the two remote

--- Peter Zeltins <peter@fintrading.com> wrote: > > > Well, I happen to be one of those very specific cases... ;) and looks > like > will have experiment with it myself. Although I'd hate to re-invent > the > wheel. > > Peter Checking e-mail this morning it looks like we have two independent "fixes" that both do what has been suggested in this

I have been working with this for a wile and I have been watching the list for about a month on this subject, to no avail. I am wondering if anyone has successfully configured asterisk for clients to connect to it when the clients are behind nat. I mean successfully because I can do everything except for audio, my audio is only one way. I am asking so I can determin if I will be continuing

Hi all, -------- I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a "we do not rely". I tried to forward the SIP Port (5060) UDP via UPnP

I have used Grandstream phones for years, and have just started testing a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling and don't know whether it's just a limitation or something I haven't done correctly. The Asterisk server is directly on the Internet with a public IP. The phones are on a private LAN with a NAT router to the Internet. The sip.conf entries for