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Contact URI showing sip:s@ipaddress

December 10, 2014, 2:15 pm
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David,

I have used the /extension to change the contact and it does not take effect. The username/extension I am using is quite long (30 chars) and wonder if that is the problem. I have tried friend and peer, but it never changes.

Jon

Statistics : Posted by electroman • on Wed Dec 10, 2014 4:15 pm • Replies 8 • Views 404
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Contact URI showing sip:s@ipaddress

December 10, 2014, 2:15 pm
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They are both in sip.conf.

Statistics : Posted by electroman • on Wed Dec 10, 2014 4:15 pm • Replies 8 • Views 404
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Contact URI showing sip:s@ipaddress

December 10, 2014, 2:15 pm
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Hi David,

Thanks again for the reply. I have also posted in the FreePBX forum.

If I were to make these modifications I would likely try the first method but which .conf file would I need to edit for this? Is this also in the extensions.conf file? The second one looks to be in the extensions.conf file but I was not sure about the first method?

Thanks again!
Darryl

Statistics : Posted by electroman • on Wed Dec 10, 2014 4:15 pm • Replies 8 • Views 404
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Contact URI showing sip:s@ipaddress

December 10, 2014, 2:15 pm
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Either of the two methods described will do that. If your problem is how to get FreePBX to set one of these options, you should ask on http://community.freepbx.org/

Statistics : Posted by electroman • on Wed Dec 10, 2014 4:15 pm • Replies 8 • Views 404
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Contact URI showing sip:s@ipaddress

December 10, 2014, 2:15 pm
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Hi David!

Thanks for the prompt reply - much appreciated. I can go in and manually edit the file to do this but I believe the provider is looking not for the extension but the actual DID number in the Contact URI field. Is there a way to insert / override with this instead?

Sorry for what may seem a novice question - I've used FreePBX for some time with the underlying Asterisk and it runs quite well but every now and then something like this creeps up and I'm a little novice in the code area.

Thanks so much again for the help!

Darryl

Statistics : Posted by electroman • on Wed Dec 10, 2014 4:15 pm • Replies 8 • Views 404
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Contact URI showing sip:s@ipaddress

December 10, 2014, 2:15 pm
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Because that is the default when you don't specify an extension (it should not normally be a problem, as s is valid as an extension).

Specify an extension in the register:

Code: ; Format for the register statement is:
;       register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]

Code: ; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).

or use a callbackextension, instead:

Code: ;callbackextension=123            ; Register with this server and require calls coming back to this extension


Documentation quoted from extensions.conf.sample.

Statistics : Posted by electroman • on Wed Dec 10, 2014 4:15 pm • Replies 8 • Views 404
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Contact URI showing sip:s@ipaddress

December 10, 2014, 2:15 pm
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Hi All,

I am currently running FreePBX with Asterisk 1.8.32.1 installed and it has been working fine for quite some time but recently I noticed my inbound calls were failing.

Upon discussing with the service provider they indicated that the Contact URI is in an incorrect format. In the Registration message being sent to to the provider the header for the From and To addresses are fine in the format of <sip:didnumber@siptrunkprovider.com>.

The Contact URI though in which they key on for delivery of calls is showing:

sip:s@ipaddress
Ex. sip:s@69.165.10.20

My trunk settings are (edited):

type=peer
port=5060
insecure=very
host=sipprovider.ca
disallow=all
context=from-pstn
allow=g729&ulaw
username=1234567890
secret=xxxxxxxx

Register String:
1234567890:xxxxxxxx@sipprovider.ca

Any idea why the "s@" is showing up and how I can get rid of it / change it? My trunk settings are what I thought would correct it but I can't seem to get it to work Image

Thanks,
Darryl

Statistics : Posted by electroman • on Wed Dec 10, 2014 4:15 pm • Replies 8 • Views 404
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Asterisk shared address book

December 19, 2014, 12:15 am
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You have a very interesting blog. I will take a while to read the Asterisk section.. Very interesting articles. You're doing a good job

Statistics : Posted by simone686 • on Fri Dec 19, 2014 2:15 am • Replies 1 • Views 74
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PJ ICE Rx error status code: 370400 'Bad Request'

December 20, 2014, 3:19 am
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hello everybody, i need your help Image Image
i used liphone-android through my asterisk server, but ice show error.
Code: PJ ICE Rx error status code: 370400 'Bad Request'.

the stun server show:
Code: received on A1:P1
Got a request (len=62) from 14.29.92.36:5987
Received stun message: 62 bytes
Message header length doesn't match message size: 724 - 62
Request did not parse
Failed to parse message

Thanks in advance for your help.
%>_<%

Statistics : Posted by bludawn • on Sat Dec 20, 2014 5:19 am • Replies 0 • Views 64
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Pulse Dialling in AsteriskNOW 6.12 on TDM400P

December 20, 2014, 4:30 am
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Hi all,
I'm quite new to asterisk and am building a system to replace a Panasonic KX-TA308.

I've successfully got two sip phones configured and one DTMF handset all working how I wanted, however I have two display phones I would also like to still be able to use which are pulse dialing.

Presently it either doesn't detect pulse signals at all, or picks up on them as flash signals.

I've searched around a bit to what other people have done and most recommend setting pulsedial=yes in zapata.conf, however obviously this no longer exists. I have tried setting this in system.conf and get it as unrecognized keyword.

The card I am using is a TDM400P with 1 FXO and 3 FXS ports.

Sorry if i'm repeating something already tackled, or my own lack of knowledge is the issue, but I couldn't find it anywhere.

Thanks.

Statistics : Posted by shackle • on Sat Dec 20, 2014 6:30 am • Replies 0 • Views 70
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Dialplan stops on Dial()

December 20, 2014, 3:45 am
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g parameter solved my problem.

I want to know if the extension answered the phone.

Thank you for your kind help

Regards,
Vangelis

Statistics : Posted by bonanhel • on Sat Dec 20, 2014 5:45 am • Replies 2 • Views 117
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Dialplan stops on Dial()

December 20, 2014, 3:45 am
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If you want the dialplan to continue after the successful completion of a call, you must use the g option.

If you want to do something in parallel with the call, Asterisk doesn't really support that from the dialplan, as the Dial application is what runs the bridge that connects the two parties. It is a fairly common misunderstanding to believe that Dial exits on answer.

If you want to run something before the call is bridged (and this may also mean before Answer is sent upstream), you can use G, U, or (deprecated) M options, but not with your current dialplan structure.

Statistics : Posted by bonanhel • on Sat Dec 20, 2014 5:45 am • Replies 2 • Views 117
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Dialplan stops on Dial()

December 20, 2014, 3:45 am
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Hello,

i have this dial plan. First CURL working fine and asterisk can dial my returned from curl extension number.

My code stops at line 2 and never goes to Goto() and s-ANSWER.

Can anyone help?

Thanks

Kind regards,
Vangelis


Code: [callin]
exten => _X.,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})

Code: [macro-uridial]
exten => s,1,Set(extredir=${CURL(http://www.xxx.com/getextension.php?action=callin&id=${CALLERID(num)})})
exten => s,n,Dial(SIP/${extredir},30)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-ANSWER,1,${CURL(http://www.xxx.com/getextension.php?action=answer&id=${CALLERID(num)})}


Statistics : Posted by bonanhel • on Sat Dec 20, 2014 5:45 am • Replies 2 • Views 117
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Can't make calls. New install.

December 20, 2014, 6:30 pm
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Although I am not familiar with the Cisco phones, I would imagine that the X means they are NOT registered, as that is about the only thing that the phone would be able to detect when idle.

In any case, without proper debugging output from Asterisk, I don't see how we are going to be able to diagnose this further.

Statistics : Posted by jrothwell1988 • on Sat Dec 20, 2014 8:30 pm • Replies 1 • Views 31
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Can't make calls. New install.

December 20, 2014, 6:30 pm
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Asterisk now version 13. New install. Phones are registered and have dial tone. The correct info is on the display. The icon near the top button lists the extension number with a phone icon with a X over it (attached). I can dial the number but then nothing happens until it times out and displays reorder. If I enter the number and hit dial nothing happens. The phones are cisco 7961.

Attachments
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Statistics : Posted by jrothwell1988 • on Sat Dec 20, 2014 8:30 pm • Replies 1 • Views 31
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compilation error - mISDN.git module

December 21, 2014, 4:45 am
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compilation error - mISDN.git module

December 21, 2014, 4:45 am
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Hi all,

Not sure if this is the right place to ask, but it worth to give it a chance.
When trying to compile mISDN module on Linux Debian, I'm running the following commands:
1. git clone git://git.misdn.eu/mISDN.git/
2. cd mISDN
3. http://forums.asterisk.org/configure
4. cp mISDN.cfg.default standalone/mISDN.cfg
5. make modules
6. make modules_install

After "make modules" I get this output:

Hunk #1 succeeded at 148 with fuzz 2 (offset 8 lines).
Hunk #2 succeeded at 161 (offset 8 lines).
Hunk #3 FAILED at 163.
1 out of 3 hunks FAILED -- saving rejects to file drivers/isdn/mISDN/core.c.rej
Error on /usr/bin/patch -p1 < patches/Drivers-isdn-remove-__dev-attributes.patch
cat .patchOK
cat: .patchOK: No such file or directory
make[1]: *** [patched_tree] Error 1
make[1]: Leaving directory `/home/arik/asterix_openbsc/mISDN/standalone'
make: *** [modules] Error 2

As fur as i read about it, it describes a problem between the core.c file and the patch file. In order to compile it, I should solve the conflict.

Does anybody had to deal such issue and know about a solution for this issue?

Thanks in advanced,
Arik

Statistics : Posted by arikd • on Sun Dec 21, 2014 6:45 am • Replies 1 • Views 99
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Handover with Asterisk

December 21, 2014, 9:53 am
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Does Asterisk support handover (like in GSM)? The problem i have is this:

I have two networks net1(192.168.0.0/24) and net2(192.168.1.0/24). These are connected WiFi routers, and i can route packets from one network to the other.

Asterisk server has an IP 192.168.0.102 and is in net1.

I also have two SIP mobile phones in net1 with addresses in the 192.168.0.0/24 range.

I want to make a call and move with one mobile phone to net2 (it automatically gets an address in the range 192.168.1.0/24), but when the phone changes the IP address the connection is lost. I have to call the other phone again to establish a connection.

Is there a possibility to switch between different networks without disconnecting the call (to perform a handover like in GSM)?

Here is a network diagram of the described problem:
http://postimg.org/image/gc2zoz5jx/

Statistics : Posted by adohus • on Sun Dec 21, 2014 11:53 am • Replies 0 • Views 66
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asterisknow-version rpm over writes issues

December 15, 2014, 2:53 am
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Who can I request this package not be a dependency?

Thanks

Statistics : Posted by londonnet • on Mon Dec 15, 2014 4:53 am • Replies 1 • Views 132
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asterisknow-version rpm over writes issues

December 15, 2014, 2:53 am
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is there a way of preventing the asterisknow-version package from over writing the issues file?

I wouldn't bother installing it at all accept it shows as a dependency for upgrading the other asterisk packages.

Thanks

Statistics : Posted by londonnet • on Mon Dec 15, 2014 4:53 am • Replies 1 • Views 132
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