similar to: iax behind a SonicWall

Recently the small business I work at replaced an aging linksys wrt style router with a SonicWall one. Ever since this change I have issues with this node falling back to TCP constantly. TCP for me ends up performing very sub-par, and most of the time unworkable. For some reason too the connection usually ends up being routed through another node. I created a new tinc instance using TAP

Hi all, I currently have an Asterisk test server behind a TZ170 Sonicwall firewall / NAT box, with several DIDs. I've found that inbound IAX2 calls don't work reliably (i.e., I get a busy tone) unless I enable "Use Consistent NAT" in the Sonicwall. This feature is poorly documented by Sonicwall, so I thought I'd pass it along. Has anyone else run into this, or figured

Hi all, I'm having trouble with incoming IAX2 calls on 1.2.7.1 - mostly they work, but sometimes the caller just gets dead air or disconnects. IAX2 debugs show HANGUP and INVALID codes in these cases, rather than a proper RINGING transaction. My firewall is doing NAT, and changing the source port from 4569 to something else - my IAX2 provider suggested this might be a problem. Is it?

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify?

Hello Has anyone else run into a problem with dovecot pop3 and sonicwall? A week after switching to dovecot we had two local customers call and report they could no longer download emails with attachments. It turned out to be the email macro filter on the router. After the first report I thought maybe the customer had just changed something or it was a fluke, but then a day later we had the

Sonicwall now has "SIP transformations" check box in the Access section of the interface - does anyone know how to make sense of this function? Tech support is useless, and the help description is confusing. Using on office network to connect Grandstream and Cisco phones to asterisk PBX at remote location. I hate to use linksys or belkin, but they're (ironically) the only 2 that

I've searched the archives a bit and have not really come up with a good answer to my queries. I have * running on a RH9 box behind a LinkSys NAT box. I can talk with iConnectHere outbound just fine. I am trying to configure an inbound Xten softphone from outside. I have that user set as NAT in sip.conf (seems to help), but I still cannot establish a full session. I think the problem comes

For those that might be using Cisco 7940/7960 sip phones and placing calls across an iax2 link, we think the voice quality problem has been identified and corrected. The dev cvs should be updated as of about 3:30pm CDT today (April 14). History: Calls originating from a Cisco 79x0 sip phone and sent via iax2 link to some distant * machine resulted in very poor quality audio, and in some cases,

Here's a good one for the group, I have 2 Ast servers behind a NAT (Sonicwall :-( ) connecting to the same server outside the NAT. Each of the 2 boxes behind register to the outside server. What I am wondering is, would there be a problem if both servers behind the NAT were listening on port 4569, I realized that the NAT'd port gets changed however I wasn't sure if this would be an

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?

Hi folks, Ok, I've seen this question go unanswered on the mailing list, and I assume it's because no one had the heart to break the bad news to the guy asking, but be honest with me, I can take it. At this time it's flat impossible to have multiple IAX phones behind a NAT without using an * gateway because there's no way to have a client listen on a port besides 4569. Is

Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------

After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX protocol went downhill and many carriers (like VoicePulse) are discontinuing support for IAX. Is this correct? We are all heading for SIP? Thanks, MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hopefully this is really a generic question about IAX and doesn't turn out to be something specific to Teliax, because they haven't been too helpful so far. All they can tell me is that my login shows "status unknown" on their end, which prevents me from receiving inbound calls on my Teliax number. Outbound calls through the same server work fine, which rules out most networking

Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register => coiax:pass1 at 69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:4569

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

Hi All - Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use

Hi All, I am having trouble with receiving calls from FWD via IAX. I know this isn't a FWD support forum, but I suspect the problem is my asterisk setup. The problem is that I can dial out to fwd subscribers, even myself but they can't dial me using my FWD number. I don't know much about IAX, but it would seem to me like a registration problem, but I get no errors or warnings in

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel