similar to: SIP - connected line has changed. Saving it until answer for IAX2/iaxy

Hi everyone, I just got myself a IAXy device and am trying to integrate it to our asterisk server. I configured the IAXy and it is registering and I get a dial-tone. If I try calling another SIP device, and I get "can't translate IAX2 to SIP" How can I make my IAX device communicate with a SIP device (and vice-versa)? Here's what the log says: -- Executing

I am having trouble with a DTMF-based application on Asterisk 1.0.3. Specifically, when two IAX2-based devices are talking, when they send DTMF to eachother, the other side only hears clicks, and maybe a millisecond of DTMF tone, but not any real duration. Furthermore, when one IAXy device calls the Echo test program, we can hear our echo, but when we punch DTMF in, we get the same effect

Is the a way for IAX2 adapter to detect when the calling party hangs up the phone? I have two IAX2 adapter, one is Digium IAXY when I call the IAXY extension and hang up the phone the IAXY adapter rings about two or three times after the calling party hangs up the phone. The new adapter AG-188N with IAX2 protocol as well does the same thing, the phone rings 2 or 3 times after the calling party

Just setup *, got a developers kit FXO where the incoming/outgoing pstn is plugged in. I've then got an IAXy that is plugged into a Philips DECT phone. * is setup so that the [bell] section rings the phone - exten => s,1,Dial(IAX2/myuser,30) What's happening when someone calls my number is that the phone rings 3 times before you hear the dect phone ringing. Anybody got any ideas as

I've tried to simplified the dial plan and use "n" instead of numbers but I've noticed it is not executing my voicemail if I substitute number with "n" In the example below when the call is not answered, it does not go to voicemail; call just hangup. exten => 1,1,Playback(transfer) exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw) exten =>

John Novack wrote: >And the only IAX2 box made is the Digium one, with it's current shortcomings ? >From reading through the archives, it seems there is currently no way to reset to factory default, no >written MAC address on an individual box, and some other instabilities requiring frequent resets. Yeah, there's the rub. Dunno if it's worth it, I'm willing to give it

I have somewhat miraculously got my server to stay up for over 24 hours now. I was at my remote location, however, and I can't make calls that used to work find. I get the following messages. I get a brief bit of good sound and about the time I see the message "Ooh, voice format changed to 4" all further sound stops. The machine seems to be stalling, but I have noload on both oss

I just bought a new IAXy box and am only achieving one way calling. Both iax.conf and the IAXy support ulaw and gsm. When I try to call, however i get this error: Feb 11 15:20:32 NOTICE[7963]: channel.c:1893 ast_read: Dropping incompatible voice frame on IAX2/iaxy-2 of format ilbc since our native format has changed to ulaw I added support for ilbc on the IAXy and in iax.conf and am still

Hi everyone, I'm going to install an IAXy in my client's office but there the internet conection has a private IP address, however the Asterisk server has a public IP. Private IP Public IP (IAXy)--------(ISP LAN)-----------((Internet))-----(Asterisk Server) This setup can cause any problems to the comunication process? I'm aware that

Asterisk Project Security Advisory - AST-2008-011 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Traffic amplification in IAX2 firmware | | |

Hi, I've provisioned an IAXy adapter on a network segment local to my asterisk server. Provisioning is fine, as is the registration and use of said device. Since the local address is private address space, I setup the public IP address of my Asterisk server as the alternate. When taking it to another location (NAT again behind a WRT54G), it doesn't seem to work. Here's my current

automon is not working for me with asterisk 1.4.22.1 in extension.conf [globals] DYNAMIC_FEATURES=>automon dial is with "w" feature.conf automon => *1 -- Executing [11 at internal:1] Playback("SIP/218-007556b0", "transfer") in new stack -- <SIP/218-007556b0> Playing 'transfer' (language 'en') -- Executing [11 at internal:2]

iH i have an IAXy which i can make calls from but am unable to call. when i dial the extension assigned, i get the following from the console; -- Executing Dial("SIP/5801-b665", "IAX2/5899@192.168.0.5") in new stack -- Called 5899@192.168.0.5 -- Call accepted by 192.168.0.5 (format ULAW) Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected

Hi, methinks that in the good 3 months since i ordered an IAXy, things have changed so much that now almost anybody out there with a VoIP hardweare website offers complete phones for less money than the IAXy, with support for both IAX2 and SIP in many cases, and fully configurable via its own keypad. What would you recommend as a rugged, small, and easy to configure/use self-contained unit that

MusicOnHold works only when I dial IN from PSTN line (Externally), when I dial local extension there is no music.. exten => 315,1,Dial(${family},25,m(penguin)w) There are no error messages: - Accepting DIAL from 10.0.0.108, formats = 0x4 - Executing [315 at internal:1] Dial("IAX2/iaxy-322-14962", "SIP/218|25|m(penguin)w") in new stack - Called 218 - Started music on hold,

I've got a single IAXy installed in a little office nearby and got a call from someone on site a finew mintues ago. Apparently they couldn't make a call on that extension. They'd pick up the phone and get nothing; no dial-tone. Running "iax2 show peers" showed the unit still registered and I could ping it too. Didn't try to call it while it was doing this; sorry. The

Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive so they can not compete. The adapter is upgraded automaticaly when it is connected to new asterisk version; since this adapter is discontinued will it still work with asterisk 1.6 and beyond or will it be\ just a "door stopper"? -- Joseph

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

Has anyone had stability issues with IAX2. (Asterisk 1.0.5). reddwarf*CLI> iax2 show firmware Device Version Size iaxy 22 39344 I'm asking because in the last three weeks I've noticed the following two issues (on separate occasions): 1) Placed a phone call. Asterisk logs show the phone being answered and various files being Played back. But

Does anybody know why Digium S100I IAXy was discontinued? -- #Joseph