Can Not Get My Eicon Diva Running With Asterisk...


Hi Guys,

asterisk drive me crazy!

Now I have tried to use FreePBX but it require MySQL which I can not install du to a conflict with PostgreSQL.

Does someone know, how to configure FreePBX to use PostgreSQL?

Or does someone know another Asterisk Web-Frontend, without Database?

It is realy not funny, to force users to install this monster on an ARM Microcontroller.

I need only enterprise internal stuff to

1) access my 4 Vodafone EasyBox 803A using ISDN and the Eicon Diva 4port v2 Server Card 2) access a 1port HFC Card to connect some ISDN Telephones 3) access my account (20 Telephone numbers) on

Asterisk Users 3.2 years ago 0 Answers

Remote Party ID - Sort Of Working...



I'm trying to set my system to set a caller id using the diaplan when calling an internal extension. In other words, when I dial Joe Smith's extension I want my own phone to show "Joe Smith 555". I have sort of managed that in the sense that my phone shows Joe Smith's caller id based on his sip.conf callerid. But I need this to be done programmatically through the dial plan (Let's say I want to show "Joe Smith" or just "Joe" based on some condition)

I have this in the relevant dialplan snippet:

exten => 123,1,Verbose(1,Test)

exten => 123,n,Set(CONNECTEDLINE(number,i)="555-555-5555")

exten => 123,n,Set(rclidname="TestingB…

Asterisk Users 3.2 years ago 6 Answers

Asterisk 1.8.13 / Res_fax / Res_fax_digium


We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13

The docs at indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP?

Also, the res_fax.conf.sample does not indicate v34 as a valid modem.

Asterisk Users 3.2 years ago 3 Answers

Using Asterisk 10.6 As A T38 Fax Gateway


Hi all, and thanks for taking the time to read this.

I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am receiving calls through the PSTN and want to send them to my VoIP carriers as T38. This is my dialplan:

[fax] exten => _X.,1,Set(FAXOPT(t38gateway)=yes,20) exten => _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)

I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of these things work. When we send a fax:

1. Asterisk does NOT send a REINVITE with the t38 offered. Reading the documentation, it should detect the fax tone with the audiohook and…

Asterisk Users 3.2 years ago 4 Answers

Total Amount Of Asterisk Installations


Counting any Open Source package is difficult for many reasons. There is probably not a reliable answer to this question since there are at least 4 major “flavors” of Asterisk out there (1.4, 1.6, 1.8, 1.10) and open and commercial source. It is reliably > 10,000 and quite possibly over 100,000 or even over 1 million. The Asterisk folks might be willing to tell you how many downloads have been done from , but that wouldn’t tell you the real number. Maybe a good start point for an estimate would start at 200,000+ if you are including all of the versions and types. But then we might still think…

General 3.2 years ago 0 Answers

Telecom HU Cannot Callforward To External Number


Hi List!

I have a Problem with Telecom Hungary, if I set a callforwarding on the Snom, to an external number (mobile). Versions: Asterisk version 1.4.35, libpri, dahdi 2.6.0, snom-7.7.30

When I call the Snom (Extension 68), it responds with "302 moved temporarily", and Asterisk try to dial out over the LOCAL channel using DAHDI. I get a Congestion back from Telecom. Channel 0/2, span 1 got hangup request, cause 21

Here is cli output:

-- Accepting call from 'callerid' to '68' on channel 0/1, span 1 -- Executing [s@macro-station-fallback-Q-VM:5] Dial("DAHDI/1-1", "SIP/68|15|tTW") in new stack -- Called 68

-- Got SIP…

Asterisk Users 3.2 years ago 2 Answers

Asterisk 1.8 On Solaris/sparc


I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.

The system itself is happy and phone calls (between two parties) seem fine.

Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file.

To eliminate encoding as an issue, I have only codec_ulaw/format_pcm loaded and the recording is ulaw. I've niced down the asterisk process to -19 even though I don't see asterisk taking more than 3%…

Asterisk Users 3.2 years ago 5 Answers