Drop calls cause of missing extensions

General Help
1

Hi all!

I’ve installed asterisk(1.8.10.1)+ newest Freepbx on ubuntu 12.04. Now I made all configs as my provider told me:

1)Made new SIP trunk with this settings:

Trunk Name:Name
Outbound CallerID:2105050
PEER Details:
host=voice.melt.ru
port=5060
type=friend
username=2105050
secret=my password
dtmfmode=info
canreinvite=no
nat=yes
qualify=yes
insecure=port,invite
fromdomain=voice.melt.ru
context=incoming
fromuser=2105050
disallow=all
allow=alaw

Register String:
2105050:my [email protected]/2105050

I didnt touch any other settings in there.

2)Made an extension as a generic SIP device
Display Name:Nik
UserID: 100
Password: made simple pass
In Device Options: context: public
nat: no(rfc3581)
port: 5060
allow: alaw

No more changes since default

3)Made an incoming route
Description: in
Set Destination: Extensions 100

I entered no DID number or CallerID number.

4)Configured softphone:Downloaded X-Lite and made account changess:
Account name: Nik
Password: entered pass from extension
User ID = 100
Domain = 192.168.1.75(FreePBX stands there)
Display name:Nik
Auth. name = 100

Now I connected and everything seems to be OK.

I go to CLI and write sip show registry:

PBX*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
voice.melt.ru:5060 N 2105050 105 Registered Thu, 24 Oct 2013 08:32:19
1 SIP registrations.

then show peers

PBX*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
100/100 192.168.1.80 D A 29102 OK (47 ms)
Melt1/2105050 89.184.0.73 N 5060 OK (5 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

Then when i try to ring on my softphone
[2013-10-24 08:12:53] NOTICE[3386]: chan_sip.c:22622 handle_request_invite: Call from ‘2105050’ (89.184.0.73:5060) to extension ‘2105050’ rejected because extension not found in context ‘incoming’

What can I do to fix this?

2

same problem when try to call from ny cellfone to softphone, except context not incoming but public.

3

You will need to use “contexts” that are part of FreePBX, have you read the wiki yet?

4

Hi! Thanks for your reply. I’ve read the wiki on this site, but mby missed something. You are talking about Custom Contexts? If so - I’ve made a new one, where allowed anything and attached it to my extension(100). After this I’ve tryed to make a call from my softphone and recieved this:

[2013-10-25 05:54:17] WARNING[2299]: chan_sip.c:3670 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2013-10-25 05:54:17] WARNING[3818]: app_dial.c:2218 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)

If I try to make a call from my cellphone to 2105050 I receive same as before:

[2013-10-25 05:55:22] NOTICE[2299]: chan_sip.c:22622 handle_request_invite: Call from ‘2105050’ (89.184.0.73:5060) to extension ‘2105050’ rejected because extension not found in context ‘incoming’.

5

No just contexts, there are a couple of contexts already built into FreePBX from-internal and from-pstn , you should try them.

6

Thx for you reply!

I changed context in trunk Peer details from “incoming” to “from-internal” and now we can receive calls, but! there is no sound, I asked my prov, what codec to use, he say - alaw, I have all dissalowed and allow alaw in settings, but still no sound.

When I try to make a call from softphone to cell phone it still doesnt work and say:
[2013-10-28 08:05:07] WARNING[931]: chan_sip.c:28645 reload_config: No valid transports available, falling back to ‘udp’.
[2013-10-28 08:05:14] WARNING[3006]: file.c:663 ast_openstream_full: File cannot-complete-as-dialed does not exist in any format
[2013-10-28 08:05:14] WARNING[3006]: file.c:958 ast_streamfile: Unable to open cannot-complete-as-dialed (format 0x8 (alaw)): No such file or directory
[2013-10-28 08:05:14] WARNING[3006]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/100-00000017 for silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer
[2013-10-28 08:05:14] WARNING[3006]: file.c:663 ast_openstream_full: File check-number-dial-again does not exist in any format
[2013-10-28 08:05:14] WARNING[3006]: file.c:958 ast_streamfile: Unable to open check-number-dial-again (format 0x8 (alaw)): No such file or directory
[2013-10-28 08:05:14] WARNING[3006]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/100-00000017 for silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer

Please help me to figure it out, its becoming a nice experience for me :slight_smile:

7

Seems like some codec is missing…

8

Perhaps you should try context from-pstn.

9

Make sure when you build asterisk (make menuselet) you choosethe extra sound files appropriate, either use udp on the cell-phone (not recommended) or enable tcp transport on asterisk.

10

Thx for you help guys, I think we will find the reason soon, I’ve tried from-pstn context in trunk options, but when i try to call from my cell phone to softphone it make 3 short rings and falls. And no information in logs.

I also tried to change transport to tcp, but now I receive this messages in sip logs:
[2013-11-08 06:29:12] WARNING[950]: chan_sip.c:28645 reload_config: No valid transports available, falling back to ‘udp’.
[2013-11-08 06:29:12] WARNING[950]: chan_sip.c:27260 build_peer: ‘tcp’ is not a valid transport type when tcpenabled=no. If no other is specified, the defaults from general will be used.
[2013-11-08 06:29:12] WARNING[950]: chan_sip.c:27260 build_peer: ‘tls’ is not a valid transport type when tlsenabled=no. If no other is specified, the defaults from general will be used.

But when I register in my softphone using not my local ip adress, but voice.melt.ru as a domain everything works great. Suddenly its not an option.