# SIP phone to Allstar Node

**URL:** <https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799>\
**Category:** AllStarLink\
**Created:** [April 16, 2020, 1:54pm UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799 "2020-04-16T13:54:46Z")\
**Posts on this page:** 9\
**Page:** 1

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**Author:** ![w1crb](https://community.allstarlink.org/letter_avatar_proxy/v4/letter/w/45deac/32.png) [@w1crb](https://community.allstarlink.org/u/w1crb)\
**Post date:** [April 16, 2020, 1:54pm UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799/1 "2020-04-16T13:54:46Z")

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AllStar newbie question. I have AllStarLink running on a PI and have a Polycom SIP phone registered in Asterisk. My Allstar node is configured. What do I need to do in order to use the phone to connect to other nodes? Or even my node? Thanks

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**Author:** ![WD6AWP](https://community.allstarlink.org/user_avatar/community.allstarlink.org/wd6awp/32/5_2.png) [@WD6AWP](https://community.allstarlink.org/u/WD6AWP)\
**Post date:** [April 17, 2020, 3:29am UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799/2 "2020-04-17T03:29:36Z")

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Here’s is how I do it. These are code snips, not complete files. Although I think sip.conf would work as is, extensions.conf not so much. All of this is normal Asterisk PBX. As you already have your Pollycom working all you need is to add the one magic line to the correct context in extensions.conf

Do not forward the sip port to your node unless you like getting hacked.

This is where you sip phone logs in:  
sip.conf

```auto
;
; SIP Configuration example for Asterisk
;
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)

[210]
type=friend
host=dynamic
username=210
secret=<your passwd>
dtmfmode=rfc2833 
context=sip-phones
callerid="Tim Line 1" <210>

```

add something like this to extensions.conf

```auto
[sip-phones]
; Extension 210 - Tim's line 1
exten => 210,1,Dial(SIP/210,60,rT)

; Allow SIP calls to local node - this is the magic part.
exten => 2522,1,rpt(2522|P)

```

I dial 2522 and go off hook… bingo! connected to node. Now I can dial up any other node or function your node just like you do from your radio.

fyi, search `sip phones` on this site. There’s tons of discussion.

edit: \*99 makes ptt and # unkeys.

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**Author:** ![G0TRT](https://community.allstarlink.org/user_avatar/community.allstarlink.org/g0trt/32/1605_2.png) [@G0TRT](https://community.allstarlink.org/u/G0TRT)\
**Post date:** [January 7, 2021, 6:58pm UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799/3 "2021-01-07T18:58:38Z")

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Any thoughts on why this wouldn’t work with an SPA525G attempting to connect to a vanilla ASL RPi image?

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**Author:** ![Mike](https://community.allstarlink.org/user_avatar/community.allstarlink.org/mike/32/1569_2.png) [@Mike](https://community.allstarlink.org/u/Mike)\
**Post date:** [January 7, 2021, 7:31pm UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799/4 "2021-01-07T19:31:49Z")

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Could you say more.  
Have you been able to make it work from a softphone or other voip sip phone ?

If not, you may need to edit modules.conf and enable sip.so

It is not enabled by default with both security and memory use in mind.

Then you must be sure that port 5060 (default) passes through your router and iptables firewall.  
Please read up on security for sip in asterisk.  
The default fail2ban protection (if installed) does not cover all needed to protect your system.

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**Author:** ![G0TRT](https://community.allstarlink.org/user_avatar/community.allstarlink.org/g0trt/32/1605_2.png) [@G0TRT](https://community.allstarlink.org/u/G0TRT)\
**Post date:** [January 8, 2021, 7:59am UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799/5 "2021-01-08T07:59:48Z")

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Mike thanks, you pointed out the problem.

My ASL node is in my LAN, using the Raspberry-Pi image. I’d edited extensions.conf and sip.conf using the examples here but I hadn’t realised that modules.conf needed editing to enable the loading of sip.so

Config changed, module reload didn’t enable the connection but a reboot of the pi did. Now the fun begins. Thanks again.

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**Author:** ![Mike](https://community.allstarlink.org/user_avatar/community.allstarlink.org/mike/32/1569_2.png) [@Mike](https://community.allstarlink.org/u/Mike)\
**Post date:** [January 9, 2021, 5:41pm UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799/6 "2021-01-09T17:41:40Z")

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Glad to help.

Put these 2 lines at the top, above your first extension stanza in the general section of sip.conf

alwaysauthreject=yes  
allowguest=no

## This will limit some hacking. The ‘alwaysauthreject’ will make failed attempts look the same as no extension to the hacker, keeping them from IDing extensions to brute force attack later. Use a password on all extensions. Use this lines deny=0.0.0.0/0.0.0.0 permit=192.168.1.112/255.255.255.255 to allow only the static ip of the phone

permit=192.168.1.0/255.255.255.0 would allow your network but exact ip is advised.  
Every once in a while, using putty etc, run asterisk in the foreground (asterisk -rvvv) and watch the verbose output for bad activity. Most often they will be looking for valid extensions…

I’m hoping to write a better how2 for this stuff and add lots of security xtras.  
Make sure you have installed fail2ban. See it from the security menu in ‘asl-menu’  
Our default fail2ban set-up needs some updating for asterisk but better than nothing.

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**Author:** ![G0TRT](https://community.allstarlink.org/user_avatar/community.allstarlink.org/g0trt/32/1605_2.png) [@G0TRT](https://community.allstarlink.org/u/G0TRT)\
**Post date:** [January 11, 2021, 12:17pm UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799/7 "2021-01-11T12:17:54Z")

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What I missed was this:

> [@](#):
>
> `exten => 2522,1,rpt(2522|P)`

I should have changed 2522 to my node number, and having done that, it works.

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**Author:** ![Mike](https://community.allstarlink.org/user_avatar/community.allstarlink.org/mike/32/1569_2.png) [@Mike](https://community.allstarlink.org/u/Mike)\
**Post date:** [January 13, 2021, 1:27am UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799/8 "2021-01-13T01:27:22Z")

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That is something only you would have known… LOL  
But now that I think about it, that is a very old public node number.

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**Author:** ![system](https://community.allstarlink.org/uploads/default/original/2X/4/42700284e201b031bd667c3f63bbae05a74b3104.png) [@system](https://community.allstarlink.org/u/system)\
**Post date:** [January 16, 2021, 1:27am UTC](https://community.allstarlink.org/t/sip-phone-to-allstar-node/16799/9 "2021-01-16T01:27:24Z")

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