similar to: AG-468 4xFXS - my personal review

Who else makes Analog Telephone Adapters with IAX protocol besides Digium? I've seen Farfon is advertising their unit but I'm not sure if they shipped it or not. Why is so hard for others to support this protocol in their adapters? SIP is totally unsuitable for firewall traversal for security reason. I would like to get min. two port unit. Is it possible to use IAX protocol without

I'm just wondering if anyone knows the story with these... http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5721202362&ssPageName=STRK:MEWA:IT He claims they support IAX2 and SIP... but almost no history on the account selling them. I didn't see anything in the wiki about this company either.. Does anyone have any history with these phones? Thanks, Jared

I received a sample IP/Speakerphone from my friends in China today. Asterisk setup was fairly uncomplicated and I had it running as an extension on my server within a few minutes. Sounds quality of both the receiver and the speakerphone are fine (wife's opinion). Are there any tests I should run with this phone? Following are the specs: - Single line appearance - Alpha display, 2x16 chars -

Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don't have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and

I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid="Conference Room Polycom" extensions.conf exten => 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten => 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper

Hi all, I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028. This IP phone is registered in an Asterisk PBX and I've a problem when I dialing internal number. If I dial an internal number, like for example 102, the IP phone takes 35 seconds to send the number to Asterisk; here below the debug output 192.168.0.75: first 192.168.0.75: 1 192.168.0.75: 10 192.168.0.75: 102

Hi, I've got two pa1688 phones that I want to set up to communicate between branch offices without a gatekeeper. Both phones will be behind a firewall and I want to use port forwarding so the phones can communicate. I tested the phones behind a firewall on the same network segment and there were no problems at all using sip. However, I then moved the phones into situ and port forwarded

I am about to buy AT-320EE phones for use together with asterisk. Has anyone used these phones ? Are there any known problems using them using Asterisk ?

I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [asterisk@jonux h323]# make clean rm -f *.o *.so core.* [root@jonux h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN

I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one

I have a gateway, which I call from my mobile phone (free of charge, since it is the same phone company). This gateway gives me a dial tone. I can than dial to any extension number or even other gateways, .... It is getting more a trouble to remember all the numbers, or to key in all the long phone numbers when you got the dialtone. I was thinking of using for this Sphinx2. How can I

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

This may sound kind of crazy and I maybe missing something. But are you placing the call on hold so you can hear the hold music. This may not be the case but you may have to place the call on hold to here the music. Also you mentioned sound, you do not need a sound card in the asterisk box to use this hold music feature. Hope this helps. -----Original Message----- From:

hi, I'm currently trying to use Speex with Asterisk from my OpenH.323 client. It seems to mismatch the codecs, below is my log from Asterisk. My Openh323 client crashes in responding to a Speex request for bits per frame. I'm guessing it either isn't running the codec correctly or doesn't support the same subset of speex codecs as openh323. (I'm using speex-1.0.1 with

Just looking at the README.mysql file in the CVS.. It contains this line.. "We will, where appropriate, make it available via a separate package which will only be usable when Asterisk is used completely within GPL (i.e. not in conjunction with G.729, OpenH.323, etc)." Does this mean that if G.729 licences are installed and used then the MySQL functions can't be used?? This is

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner

Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. Thanks Tin Trung Nguyen Technical Team Mobile: 084-91.365.4857 website: www.daivietcontrol.net -------------- next part -------------- An HTML attachment was scrubbed... URL:

Guys. Maybe this is asking for a lot :) but is there any web client that can use IAX2 and ilbc? This is for a "call us" web idea.... Any leads?

As the title suggests, I was wondering if there was another source of documentation for Asterisk. Related: If one wanted to contribute to documentation, who would one contact? Thanks! Sean

Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn'