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
4) let me use my CISCO CP-3905 SIP phones on the LAN
5) access the VoIP server of my ISP Alice/Hansenet

I have no customers which must accounted or such…

Thanks, Greetings and nice Day/Evening
Michelle Konzack

–
##################### Debian GNU/Linux Consultant ######################
Development of Intranet and Embedded Systems with Debian GNU/Linux
Internet Service Provider, Cloud Computing


itsystems@tdnet Jabber linux4michelle@jabber.ccc.de Owner Michelle Konzack

Gewerbe Strasse 3 Tel office: +49-176-86004575
77694 Kehl Tel mobil: +49-177-9351947
Germany Tel mobil: +33-6-61925193 (France)

USt-ID: DE 278 049 239

Linux-User #280138 with the Linux Counter, http://counter.li.org/

Hi,



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 <123-444-5555>“)

exten => 123,n,Set(CONNECTEDLINE(pres)=alowed)

exten => 123,n,Set(CONNECTEDLINE(name,i)=”Testing”)

exten => 123,n,Set(CONNECTEDLINE(pres)=alowed)

exten => 123,n,Dial(SIP/joesmithpolycomphone,20)

exten => 123,n,Hangup()



I am always seeing remotepolycomphone’s callerid number and name as entered in sip.conf, not “Testing 555-555-5555″, neither am I seeing “TestingB
<123-444-5555>“.



What am I missing for it for accept my dialplan remote-id name and number?



Regards,



Mike

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

The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf 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.

Hello,

Can someone give me an understanding about E1 with ISUP on CCS 7
signalling? Is it possible with asterisk + digium card and how

Regards,

Ashish

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 then send a REINVITE with t38 capability.

2. Asterisk does not offer t38 in the SDP of the initial INVITE. This is not a problem if it correctly detects and REINVITES for faxes, but our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we do not have that capability.

Obviously I would prefer to just detect the fax myself and have asterisk do the REINVITE.

I have read all of the documentation on the asterisk wiki (which is rather short) and anything else I could find online. Unfortunately most of it is out of date and refers to asterisk versions 1.4 to 1.8, which do not have T38 Gateway capability.

Does anybody have any experience in making this work?

Thank you!

Alex