Discussion: Asterisk config sip.conf

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Re: grandstream bt101 no caller id

I agree. If you don't set the fromuser variable, everything else should click into place and work correctly.

by vsod, Tuesday 17 of June, 2014 (16:36:15 UTC)
Asterisk as SIP Client... Unable to make outgoing calls

Hi,

I am trying to register my Asterisk as a Client at a SIP provider which provides PSTN access so that I can dail in and out on PSTN using SIP softphone (X-Lite). Now, I am able to register Asterisk against the SIP provider and get incoming calls on softphone too. But the problem is with outgoing calls. After dailing the PSTN number the PSTN phone rings but even after picking the PSTN phone the softphone displays calling 0xxxxxxxxxx (PSTN number). Then finally the sip softphone displays "Call Failed: Service Unavailable" and you hear the voice "The person you called is unavailable".

The Settings in sip.conf are:

[general]
port = 5060
bindaddr = 0.0.0.0
context = others
sipdebug = no
realm = domain.com
trustrpid = yes
sendrpid = yes

register => uname@domain.com:pwd:authname@IP/46
registertimeout=20
registerattempts=10


[my_provider]
type=peer
fromuser=uname
fromdomain=domain.com
canreinvite=no
secret=pwd
insecure=very
host= ip
qualify=yes
nat=no 


The configuration in extensions.conf is as follows:
exten => _0.,1,Dial(SIP/${EXTEN:1}@my_provider)

The output on Asterisk CLI is:

 Executing [04045834323@tutorial:1] Dial("SIP/alice-c0000a60", "SIP/4045834323@my_provider") in new stack
    -- Called 4045834323@my_provider
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/alice-c0000a60' status is 'CHANUNAVAIL'


Can someone please explain where and what I am doing wrong?

Thanks.

by chawlakunal, Sunday 06 of May, 2012 (20:51:04 UTC)
Asterisk with SIP Proxy

I have a question about the register option that can be put in the General context of sip.conf.

We are in a situation where we are using a Vonage account with our Asterisk system, however, Vonage requires authentication on the sip invites. The only way I can get it working in my test environment is by using the register option in my sip.conf.

In our production environment all of our phones are connected using SIP, does this one register option effect any of the other registrations we have? Thanks.



by Ayth, Wednesday 24 of September, 2008 (20:12:49 UTC)
Asterisk and Ekiga

Hi,,

I am using Ekiga as my sip phone and asterisk as PBX. I was unable to register to asterisk.
In sip.conf i have added this ....

[xxx]
type=friend
secret=yyy
qualify=yes
nat=no
host=dynamic 
canreinvite=no
context=home

In extensions.conf iam giving as
exten => xxx,2,Dial(SIP/xxx)

In Ekiga.
I am giving the IP address of the system where asterisk was loaded ...

But, I am unable to register in that,,,

Please, can any one help me in solving this....

by palvali, Sunday 06 of May, 2012 (20:50:27 UTC)
Re: outbound call with service voicedata.be

I had the same issue on a local provider "netcall.pt" which at the time was using SER. When I changed the sip useragent to something other than "Asterisk PBX" it was solved.

by diogob, Thursday 21 of June, 2007 (01:40:57 UTC)
Multiline phone setup

I have a SPA-841. I want to assign 2 different extensions in extensions.conf to the two extensions on the phone. I have tried to change the port on the second ext, but when dialing the extension I set up for it, asterisk says Everyone is busy/congested at this time. When I dial from either ext on the phone, asterisk sees it as coming from ext 1. Can you have to entries in sip.conf for the same phone with different parameters?I have a SPA-841. I want to assign 2 different extensions in extensions.conf to the two extensions on the phone. I have tried to change the port on the second ext, but when dialing the extension I set up for it, asterisk says Everyone is busy/congested at this time. When I dial from either ext on the phone, asterisk sees it as coming from ext 1. Can you have to entries in sip.conf for the same phone with different parameters?

sip.conf:
[general]
context=internal
srvlookup=yes

[test]
username=test
type=friend
secret=password
host=dynamic
;port=5060
allow=all
qualify=yes
nat=no
mailbox=102@default
callerid="John Smith" <3175554321>

[test2]
type=friend
username=test2
secret=password
host=dynamic
port=5065
allow=all
qualify=yes
callerid="Your Name Here" <5555551234>
mailbox=102@default


by Bytecafe, Sunday 06 of May, 2012 (20:50:09 UTC)
Typo

I was looking for defaultexpiry and maxexpiry information, but there was nothing.
They are all written as "expirey".

by cunyalen, Friday 09 of February, 2007 (01:24:19 UTC)
Using Asterisk together with SER

Has anyone gotten this to work?

I need to get it to work with Asterisk between SER & my SIP provider, handling both incoming and outgoing calls.

by evert, Wednesday 01 of November, 2006 (08:51:55 UTC)
Example - Topex GSM Gateway SIP Connection

One can get a SIP card for the Topex GSM gateways. This saves on the TDM overhead on Asterisk machines, with traditional integration with E1. From the Asterisk perspective it is quite simple, see example below, but from the Topex device it is not. *Get help* from their technicians for this, jumpers need to be changes etc. etc.

Example sip.conf

[topex]
    type=peer
    port = 5060
    host=XXX.XXX.XXX.XXX
    bindaddr=XXX.XXX.XXX.XXX  ; Local interface
    insecure=very
    qualify=no
    canreinvite=no
    disallow=all
    allow=alaw
    allow=ulaw
    register=>:@XXX.XXX.XXX.XXX  ; Local interface

Have fun, cheers... Michael Toop.

by MmmToop, Sunday 06 of May, 2012 (20:49:54 UTC)
SIP bindaddr= confusion

The statement "(0.0.0.0 binds to all)" next to the bindaddr= parameter in the default sip.conf is very misleading.

If you configure bindaddr=0.0.0.0 in sip.conf, SIP will not truly bind to any IP other than the first IP on a particular ethernet interface (or more specifically, the system's ip route to the SIP device, which will be the first IP unless configured otherwise). It will respond to requests sent to any IP on the machine, however those responses will always be sent from the first IP and therefore if the original request was sent to a secondary IP, the response will inevitably be discarded by the SIP device as unrelated.

This is contrary to the behavior of hundreds if not thousands of other types of servers, and in particular the behavior of those which claim to bind to all addresses, but apparently is intended behavior as according to Corydon76 from Digium: "This has been asked many times before, and the answer is still the same. We simply don't support this behavior."

So beware — "you cannot alias your Ethernet interface to multiple addresses and expect it to work."

Of course this means you cannot use systems like heartbeat to transfer a virtual IP between SIP servers to maintain redundancy as a workaround for yet another asterisk weakness (that it will only use the first SRV record, though this weakness is actually well-documented and I believe they intend to fix it).

Hope you have a plan C. :) If it helps, we solved this problem here using SRV on the clients (since they handle it correctly), and DUNDi on the server side.

by jburbage, Saturday 04 of March, 2006 (00:30:05 UTC)