similar to: IAX2 missing?

I just got 2 IpDialog phones for use with my Asterisk system. I have been able to get the phones to just dial local extensions but it is not able to register with my system correctly. I would like to know if someone has set these phones up before and how they did it! Is there any examples for use with Asterisk? They seem simple enough to config with there web interface. Thanks

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

Hi all, DIAX 0.9.3 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. Still just IAX1 is supported (for the moment). What's new in 0.9.3? - accept blank passwords; - accept for registration/calls host names, not only IP Address; - password no

Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf

I need to get 2 Asterisk servers working together. I have been reading and doing just about every example I have been able to find here on the list and the Wiki. It's now gotten to the point that nothing on box2 seems to be working. I seem to have a major problem understanding the format. Here is what I have so far. It's 3 days of hair pulling and nothing seems to work! Asterisk box 1

I would like to know if anyone has come up with a script for 911 dialing rules that put correct information on our locations. We have our office in 3 different building one being our production & shipping dock. It is almost 2 blocks away. We are connected with Ethernet Wireless between the buildings and have Sip phones setup in the other 2 locations. All the phones are working just fine.

Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options

Hello ! Asterisk 1.0.9 running on Linux 2.6.12. I'm not able to call iax2 channels. There can be no translation path found. When I try to call from a ZAP PRI channel the following error occurs: channel.c:1891 ast_request: No translator path exists for channel type IAX2 (native 63488) to 72 dial_exec: Unable to create channel of type 'IAX2' What is wrong ? Here is my iax.conf:

I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten => 1900XXXXXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => 1XXX976XXXX,1,Congestion exten => 91900XXXXXXX,1,Congestion exten =>

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

I am new to this forum. As well as a new user of Asterisk. My vendor installed the system and we are still trying to get all the bugs out of it! I have a few questions about configuration and a program to view who is on what extensions. I am looking for a program that will work on my Receptionist work station. She is running Windows 2000 pro. We have not plans on upgrading to XP pro so

I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup

I have a question on SIP devices that are setup and working but you change the login name and contents to them why does asterisk need to be shut down and restarted for them to work? I have reloaded extensions and done a reload command. But the updated sip phones do not work until I shut down and restart asterisk. Is there any other way to update them without restarting the system? Since the

I have about 5 snom 200 phones working fine with everything. Voicemail, Transfers and all. Except I can't seem to use them to pickup parked calls nor place a call on park. I also have sipura-2000 with analog phones that are able to pickup parked calls and to park them. Most of them are on firmware 2.04g I have upgraded one to 2.05c for testing but this did not fix the problem. I get no error

I am trying to get a building that has 3 company's on one asterisk server. I need to make the IVR via DID take them to there right menu. So far I have everything working except when they goto via standard_marco to an extension and are sent to voicemail they are dropped off in the first menu and not the one they came from. In other word sent to another company's menu. If it happens to be

I have a problem in that when you use IAX2 for trunking and have 2 nics one is used to connect directly to 2nd Asterisk server how do we get the outside Nic card to take IAX connections? Is there any way to get this working via two paths? There is only one bindipaddr=10.1.1.1 for internal trunk but outside address section? -----------------------------------------------------------------

Hi, I'm trying to set up a basic FXO <> SIP gateway. That is, I want calls from my SIP phone to simply be dumped onto the POTS line. My (entire) extensions.conf is: [from-sip] exten => _9NXXXXXX,1,Dial(ZAP/1/${EXTEN}) and my zaptel.conf is: fxsks=1 loadzone=us defaultzone=us and my zapata.conf is: context=incoming signalling=fxs_ks echocancel=yes

Hello list When i'm trying to dial into our pstn the following errors occure: -- Executing Dial("SIP/snomsip-dbd0", "/2100") in new stack Sep 9 10:02:22 WARNING[59409]: channel.c:1901 ast_request: No channel type registered for '' Sep 9 10:02:22 NOTICE[59409]: app_dial.c:715 dial_exec: Unable to create channel of type '' == Everyone is busy/congested