Can Dial Plan Handle New Proprietary SIP HEADER Fields? How?

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Dear asterisk-users,

I have listened that a diaplan on Asterisk can extract information from proprietary SIP messages header fields. That is, if Asterisk receives a SIP message with a modified HEADER (containing proprietary fields) , is it possible to program the dial plan to make Asterisk extract the values of such fields, being possible to handle such values in diaplan, isn't it?

If it is true, is it also possible to use dial plan to make Asterisk include proprietary SIP HEADER fields in a specific SIP message?

Any hint will be very helpful!

Best regards.

RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55…

Asterisk Users 24 days ago 0 Answer

Am I Cracked?

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Hi list!

Very strange... I ran the Asterisk CLI for other tasks, and suddenly I got this message:

== Using SIP RTP CoS mark 5 -- Executing [000972592603325@default:1] Verbose("SIP/192.168.20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") in new stack == PROXY Call from 0123456 to 000972592603325 -- Executing [000972592603325@default:2] Set("SIP/192.168.20.120-0000002a", "CHANNEL(musicclass)

Asterisk Users 27 days ago 19 Answer

Chan_ooh323 To Sip , No Connected Line Info

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Hello!

We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323.

Connection can be shown as

avaya--PRI-asterisk--h323-avaya

When I do call as avaya user I see name of remote end avay user, i.e. connected line info.

As I see in debug remote side send is as

14:07:29:758 Received H.2250 Message = { 14:07:29:758 protocolDiscriminator = 8 14:07:29:758 callReference = 47 14:07:29:758 from = destination 14:07:29:758 messageType = 7 14:07:29:758 Display IE = { 14:07:29:758 Disa 14:07:29:758 }

over h323.

But now we need to connect to another asterisk over SIP.

in this case we have…

Asterisk Users 2 months ago 1 Answer

What's The Best Average Duration For A SIP Test Call?

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Hi,

What is your experience: if you plan to make a test SIP call to check voice quality overa connection what would be the best call duration? The point is that we should have a call long enough to be able to catch/hear impairments that the connection may have.

This is partly a matter of curiosity, but I believe the roots of this question may be quite important.

Thanks! Valeri on behalf of Sevana

Asterisk Users 3 months ago 0 Answer

Anonymous SIP Calls

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We have a FreePBX-12 / Asterisk-12 setup that supports about 24 extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence. The server host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x) and is up-to-date. Registrations require very long random passwords and registrable devices are further restricted by netblock filters. We have the usual firewall and fail2ban intrusion prevention and detection set-ups in place.

Our connection to the rest of the world is via PSTN.

We do our own DNS, both forward and reverse. We have NAPTR and…

Asterisk Users 3 months ago 7 Answer