I set up a SIP TRUNK in FreePBX/Asterisk that works perfectly for incoming calls. Here' s the relevant configuration:
type=friend
host=201.217.31.10
callerid=mynumber
[email protected]
[email protected]
fromuser=595XXYYZZZZZZ
fromdomain=prepago.com.py
secret=******
dtmfmode=auto
trunkname=covoip
context=from-trunk
hasexten=no
hasiax=no
hassip=yes
registeriax=no
registersip=yes
trunkstyle=voip
nat=force_rport,comedia
insecure=port,invite
disallow=all
allow=alaw,ulaw,gsm
qualify=yes
However, whenever I try to place an outgoing call (through the same trunk) I have a "all lines busy" signal from asterisk. If I enable SIP DEBUG this is what I get (apparently my call is being rejected due to an invalid alias at the other side, which I can't control since it's my VOIP provider):
<--- SIP read from UDP:201.217.31.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.16.50:5061;received=190.128.230.22;branch=z9hG4bK6a440fdb;rport=5061
From: <sip:[email protected]>;tag=as3a625f1c
To: <sip:[email protected]>
Call-ID: 59fbc0e25c141a603114ce2214c9d208@[::1]
CSeq: 180 REGISTER
Contact: <sip:[email protected]:5061>;expires=30
Expires: 30
User-Agent: FPBX-AsteriskNOW-12.0.33(13.0.1)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
[2015-02-19 15:48:50] NOTICE[2015]: chan_sip.c:23725 handle_response_register: Outbound Registration: Expiry for 201.217.31.10 is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog '59fbc0e25c141a603114ce2214c9d208@[::1]' Method: REGISTER
[2015-02-19 15:48:52] WARNING[1833]: func_cdr.c:349 cdr_write_callback: CDR requires a value (CDR(variable)=value)
Audio is at 16688
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 201.217.31.10:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.16.50:5061;branch=z9hG4bK61ad8aec;rport
Max-Forwards: 70
From: <sip:[email protected]:5061>;tag=as23ae8214
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-AsteriskNOW-12.0.33(13.0.1)
Date: Thu, 19 Feb 2015 18:48:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1709304421 1709304421 IN IP4 192.168.16.50
s=Asterisk PBX 13.0.1
c=IN IP4 192.168.16.50
t=0 0
m=audio 16688 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:201.217.31.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.16.50:5061;received=190.128.230.22;branch=z9hG4bK61ad8aec;rport=5061
From: <sip:[email protected]:5061>;tag=as23ae8214
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:201.217.31.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.16.50:5061;received=190.128.230.22;branch=z9hG4bK61ad8aec;rport=5061
From: <sip:[email protected]:5061>;tag=as23ae8214
To: <sip:[email protected]>;tag=b72e12N2654e5f93c-504b
Call-ID: [email protected]
CSeq: 102 INVITE
Reason: Q.850 ;cause=38 ;text="11017 - Invalid alias"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 201.217.31.10:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.16.50:5061;branch=z9hG4bK61ad8aec;rport
Max-Forwards: 70
From: <sip:[email protected]:5061>;tag=as23ae8214
To: <sip:[email protected]>;tag=b72e12N2654e5f93c-504b
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.33(13.0.1)
Content-Length: 0
Any ideas of what might be wrong on my side of things?
If I connect a simple softphone to my VOIP provider, it works flawlessly (incoming and outgoing calls).
