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DAHDI Loading Issue On Asterisk

Try starting Asterisk with the -f option. It will NOT fork into the background so you will see all messages on startup (including any that might not end up in the log file). Search for DAHDI errors which will likely be there.

Also, if you configure everything and start DAHDI but don’t start Asterisk and run “dahdi_tool”, is it showing you the circuits in an “OK” state?

Josh

Connecting 2 Asterisks, One With PJSIP And Other SIP Returning 401

It’s my first post here, so I’ll cut to the chase

I have 2 Asterisk servers and want to connect them using sip on one and pjsip on the other one. One is running at home and another at a VPS. The first one will be the client (with dynamic ip) and the 2nd the server.

The client uses sip and the server pjsip.

This is the client’s sip.conf

[general]
context = default allowguest = no realm = myrealm.com udpbindaddr = 0.0.0.0
qualify = yes subscribecontext = default localnet2.168.1.0/255.255.255.0
externhost=myhost.com externrefresh0
dtmfmode = auto canreinvite = no jbenable = no sendrpid = yes trustrpid = no disallow=all allow=ulaw allow=alaw register => myuser:mypass@vpsserver

[vpsserver]
type=friend secret=myuser defaultuser=mypass host=vpsserver.domain.com context=inbound canreinvite=no insecure=port,invite

And this is the server’s pjsip.conf

[transport-udp]
type=transport protocol=udp bind=0.0.0.0

[home]
type=endpoint context=trusted disallow=all allow=ulaw allow=alaw transport=transport-udp auth=home aors=home

[home]
type=auth auth_type=userpass password=mypass username=myuser

[home]
type=aor max_contacts

When I check on the client, executing sip show registry I get

Host dnsmgr Username Refresh State Reg.Time vpsserver:5060 N myuser 104
Registered Tue, 15 Apr 2014 22:57:34

which I guess means everything is ok… on the client side, I have on my extensions.conf

exten => 66,1,Dial(SIP/1@vpsserver)

and on the server’s extensions.conf (in the trusted context) I have

exten => 1,1,Playback(hello-world)

So far so good… but when I dial 66 on my client Asterisk, I see the following SIP dialog on the server… the only weird thing is that I see some From: 192.168.1.112 (that’s my home Asterisk’s internal IP… the externhost works fine for all the providers I’m using, so I doubt that’s an issue)

http://pastebin.com/hkFezB8j

Thanks in advance!

WebRTC And JsSIP

We can’t do much with part of your debug. You’ll want to post a pastebin link to your full SIP trace, and be sure that it includes at least VERBOSE messages turned up to 5.[1]

Work on WebRTC support is on-going, so you’ll want to test in the very latest Asterisk version in your branch (11 or above). That means you need to be on 11.9.0-rc2[2] at this moment.


[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
[2]: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.9.0-rc2.tar.gz

FW: Clients Unable To Auth

Hi Guys,



Just new to Asterisk and am completely stumped. I have created two accounts as instructed. Please see below for the config of the user accounts.



[Peter]

type=friend

host=IP address

disallow=all

allow=ulaw

allow=alaw

callerid=Peter <6004>

secret=XXXXXXX

context

Old Asterisk Release Wanting To Upgrade …

Hello, I have been running Asterisk for the past 5+ years on RedHat and I never upgraded it before. All my Asterisk software is of the following release:
1) Asterisk 1.4.21.2
2) Libpri-1.4.4
3) Zaptel-1.4.11
I would like to move the OS to CentOS and then I thought I can at the same time ponder about upgrading Asterisk releases. However, I am bewildered by the myriad of different releases like 1.6, 1.8, 10.x, 11.x, 12.x, 13.x Can someone please give me some advice as to what release I should upgrade?
Or should I just stick to 1.4.x and just upgrade DAHDI?
Thanks. Regards, John Lee The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

Alembic – Asterisk 11

I’ve had years of experience using ODBC for CDR, SIP, and extensions with Asterisk. One thing that has been problematic in the past is with documentation as far as database tables changing between versions (even within minor releases, though that was back in the 1.4 days). I was excited to see there is a plan for better managing that on Asterisk 12 via Alembic. All that being said, are there any plans to implement that with Asterisk 11, since that is the current LTS release? Or are we pretty sure the table structure won’t be changing within that version through the rest of its lifespan, making such an effort a waste?

Thanks,

Josh