similar to: IAX2 peer question

Hi, I've brought this up in the past and there was a good discussion - am wondering if there have been any new developments. Our dialtone service, like I am sure is true for most ITSPs, touts the ability to drop your POTs lines for significant savings. For businesses we have a low-cost Atom based PBX and a "fax relay" setup locally with hylafax/iaxmodem to solve that issue,

Howdy, I am playing around with asterisk within an LXC container on Ubuntu 11.04. I have asterisk (1.4.42) running fine, but want access to dahdi_dummy for timing (meetme). I have dahdi installed on the "host", and dahdi_dummy is loaded: root at astnorth:/# ls -ltr /dev/dahdi total 0 crw-rw---- 1 root root 196, 250 2011-06-08 13:59 transcode crw-rw---- 1 root root 196, 253

asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were "scratchy" and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm trying to call them now, but it's Sunday morning in the sticks, and my chances of

Using zoiper on a nexus 4, asterisk 11.5.1, sometimes we see failed authentication. The secret seems correct, so we can't figure out why we're getting failed authentication. But at the same time the device shows as registered: [Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper: Peer 'n4' is now REACHABLE! Time: 441 [Oct 8 18:15:58] NOTICE[519]:

Hi people, pls does anybody know what "(T)" and "(D)" letter means? server3*CLI> iax2 show peers Name/Username Host Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0

Setup as follows: [private*] - Natting Router - [public*] [private*] cannot register via IAX2 correctly while [public*] is running. Status remains UNKNOWN even after minutes, calls from [public*] to [private*] are not possible. Console output of [public*]: | *CLI> iax2 show peers | Name/Username Host Mask Port Status | iaxtest/iaxtest (Unspecified) (D)

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/E&M Wink. I tried to move one span over one

To answer my own message, I figured out a solution (untested) about 10 minutes after posting and leaving the office. Doh! Anyway, the solution (now tested) was to make the Asterisk server behind the NAT register with its peers. Despite reserving port 4569 in the firewall, that was not enough in this particular NAT firewall - it was only being reserved for one connection. Kind regards, Sebastian

The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though, the non-natted server is almost permanently _reachable_ from the natted server. Details are below

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see

Hi All I'm trying to get two Asterisk servers to talk to each other using IAX(2). I've read the WiKi and the docs and tried the examples..... I can't get it to work (I have 2 x 7960's registering on one server and 1 x 7960 registering on the other). I've set them up as follows... The two servers are set up as friends and have consecutive IP address's. The setup is

Some !@$#@@# in the Czech Republic used one of our SIP accounts to place four thousand calls to what appears to be a toll number in Zimbabwe last night. Filter 82.150.165.5. A more overriding problem for me is how do we know what *destinations* to filter so this idea of war dialing a toll number is something we can cutoff before it gets to our upstream provider? Is there some collected

Hello, I am trying to setup my Asterisk box to accept calls from the FWD network. I've followed all the config advice / samples I've found on the web. Making calls to devices on the FWD network from my Asterisk box works flawlessly, but whenever I try to call my Asterisk box from a FWD client I get a busy signal, and a "Call Disconnected" 486 error. What's odd is that I

Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a full featured BLF showing statuses of Ringing, Inuse and Online and one touch directed call pickup. On the asterisk side all that needs to be done is to add a hint to the extension and enable directed pickup.

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569

Struggling with an IP650 and 7 IP335s this morning. I have the following hints defined (courtesy of FreePBX 2.9): extensions_additional.conf:exten => 300,hint,SIP/300 extensions_additional.conf:exten => 301,hint,SIP/301 extensions_additional.conf:exten => 302,hint,SIP/302 extensions_additional.conf:exten => 303,hint,SIP/303 extensions_additional.conf:exten => 304,hint,SIP/304

Hello, We are toying with setting up a redundant data center for our hosted PBX product, and plan to use the OpenVPN feature of our Yealink phones for connectivity to each data center. The feature has been fantastic with the first data center, allowing us to bypass all SIP NAT issues entirely and allowing remote access to the phones' web interface without having to touch the customers'

Hi all, What does (T) mean on the output of "iax2 show peers"? The following my output. darkstar*CLI> iax2 show peers Name/Username Host Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata 201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0