similar to: X-lIte behind NAT and Asterisk behind NAT

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending 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

Hi All, I've an Asterisk Server behind a NAT. Using DNAT, I've opened port 5060 and all 10000:20000 udp. Sip configured with externalip and subnet. I've another site, also with NAT, where I map the rtp port (as defined in the client) to map to the local client (DNAT). Using Xlite, this configuration works, it requires using the quality=yes and NAT=yes/always in the sip ext

I?ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to register to an * box behind a Cisco SIP ALG. With known good credentials configured on the phone and in *, I get 403 Bad Auth when trying to register. If I put the phone onto the same LAN as * it works fine without changing any authentication parameters whatsoever. If I make the secret blank (null) on the phone and *,

Hi, I wonder what does this warning 399 mean and how to workaround? "sip show peers" says that sip client is unreachable althought it works with some eexceptions ... I saw posts in this list about setting codec to ilbc, is this right action ? Also, I'm very interested if anyone succeded on Asterisk behind one firewall and Grandstream behing another ? I' almost got it working

hi all, i've been struggling with setting appropriate rules for an SMTP-server behind by NAT'd firewall. it's not that there is too little info on the web -- or here, for that matter -- there's scads of it for seemingly endless configs/req'ts -- none that seem to be exactly my own. bottom line: i'm a bit confused, and looking for some experienced advice. my goals (for

I work at home with standard residential cable Internet service and I wanted to test CPA for use with our dialer solution. The first problem I ran into is that CPA only works with a SIP provider that does IP based authentication opposed to usename/password authentication. After I got an account setup to solve that problem I thought I was on my way to being able to test. No so. I got asterisk

Hi, I have a little situation here :( I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor my router have DynamicDNS. So, if I

Helo to all, I am attempting to establish an IPSEC tunnel to a remote freeswan G/W with my laptop. My laptop sits in behind shorewall at home. From the documentation, this is what I Modified in Shorewall: /etc/shorewall/tunnels: ipsec loc 24.65.x.x /etc/shorewall/policy vpn loc ACCEPT loc vpn ACCEPT My question is, have I left anything out?

Hi, i'm having asterisk with sip working fine, including dundi lookups. The only problem i'm having is that the dundi answer allways contains my internal, private ip. Is there any way to set the targeting ip that is sent out in the dundi answer (to my public ip or any other where i want to receive the call)? Regards, Andreas.

Hello Everyone, I have one machine (asterisk server) that is DMZ behind my nat firewall on my client end (at home) i have a linksys wrt54g with 16384->32766 forwarded to my cisco 7960 (which works fine) and 16000 -> 16383 forwarded to my sipura 2100 (which is set to these ports) For some reason, the sipura only works some of the time. Both devices are set to register, is there anything

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, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

I've tried to register * at FWD but * segfaults, so i guess my register-line in sip.conf isn't correct. It looks like: register => fwd#:pwd@192.246.69.223 should it be different? Chris

I have recently set to * to play with and I'm getting the hang of it.. I've seen various posts about problems where the SIP clients are behind NAT but I anted to ask a slightly different question. At the current CVS can the * SIP client register with a SIP provider (such as ICH) when the * box itself is behind a NAT. I have an ICH account which I can configure fine on an ATA 186 using

I wonder if someone out there could loan me a peek at their sip.conf? I have conflicting advice, for instance, about whether or not to use "nat=1" and also whether or not the ATA should be registering with the instance of asterisk it is going to be using to dial out. Thanks in advance. B.

Hi All, Outbound Iconnecthere calls work without any problem but Inbound calls are very intermittent. It seemed to work for a week or so but over the past week 99% of inbound calls are dropped to ICH voicemail. Would moving the Asterisk box to a public IP resolve the problem or is it just an ICH/Asterisk problem? I am registering against natrelay.deltathree.com. asterisk -vvvc shows an

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 =>