Hi!
I’m struggling configuring my ITSP trunk on ISSABEL. Let me context you and tell you that my provider won’t support anything other than their equipment so it is losing time to call their support.
I’ve the trunk configured and I can receive incoming calls, no problems with that. The problem here is the outgoing calls. Sometimes it works, sometimes it doesn’t.
I tried and a softphone (Zoiper) and inbound and outbound ALLWAYS work fine.
My trunk configs:
username=+35122xxxxxxx
type=friend
secret=MYPASSWORD
registername=+35122xxxxxxx
qualify=yes
outboundproxy=proxy.ims.iptv.telecom.pt:5070
insecure=very
host=sip.sapo.pt
fromuser=+35122xxxxxxx
fromdomain=sip.sapo.pt
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=yes
call-limit=1
allow=alaw&ulaw
Register String: +35122xxxxxxx@sip.sapo.pt:MYPASSWORD@proxy.ims.iptv.telecom.pt:5070/+35122xxxxxxx
I should also say that my provider uses various IP Address, if you try to do a DNS resolution for sip.sapo.pt it can be x.x.x.x and after some minutes y.y.y.y
I captured a working session from Zoiper:
INVITE sip:91XXXXXXX@sip.sapo.pt;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 188.83.8.35:51644;branch=z9hG4bK-524287-1---23c4406f193bcb55;rport
Max-Forwards: 70
Contact: <sip:+35122XXXXXXX@188.83.8.35:51644;transport=UDP>
To: <sip:91XXXXXXX@sip.sapo.pt;transport=UDP>
From: <sip:+35122XXXXXXX@sip.sapo.pt;transport=UDP>;tag=4158534c
Call-ID: VpWFoipO5npYWVQKJtZLSw..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.2.6 rv2.8.63
Allow-Events: presence, kpml, talk
Content-Length: 606
v=0
o=Z 0 0 IN IP4 192.168.101.153
s=Z
c=IN IP4 192.168.101.153
t=0 0
m=audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102
a=rtpmap:106 opus/48000/2
a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
a=rtpmap:111 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:112 speex/32000
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:100 telephone-event/16000
a=fmtp:100 0-16
a=rtpmap:99 telephone-event/32000
a=fmtp:99 0-16
a=rtpmap:102 G726-32/8000
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 188.83.8.35:51644;received=188.83.8.35;branch=z9hG4bK-524287-1---23c4406f193bcb55;rport=51644
To: <sip:91XXXXXXX@sip.sapo.pt;transport=UDP>
From: <sip:+35122XXXXXXX@sip.sapo.pt;transport=UDP>;tag=4158534c
Call-ID: VpWFoipO5npYWVQKJtZLSw..
CSeq: 1 INVITE
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 188.83.8.35:51644;received=188.83.8.35;branch=z9hG4bK-524287-1---23c4406f193bcb55;rport=51644
To: <sip:91XXXXXXX@sip.sapo.pt;transport=UDP>;tag=1987012959-1508093419281
From: <sip:+35122XXXXXXX@sip.sapo.pt;transport=UDP>;tag=4158534c
Call-ID: VpWFoipO5npYWVQKJtZLSw..
CSeq: 1 INVITE
Content-Length: 183
Contact: <sip:91XXXXXXX@213.13.24.225:5060;transport=udp>
Content-Type: application/sdp
Call-Info: <sip:10.102.32.228>;appearance-index=1
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, UPDATE, NOTIFY
Supported: timer
P-Asserted-Identity: <sip:91XXXXXXX@sip.sapo.pt;user=phone>
Session: Media
v=0
o=BroadWorks 515332682 1 IN IP4 213.13.24.243
s=-
c=IN IP4 213.13.24.243
t=0 0
m=audio 28888 RTP/AVP 0 101 8
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 188.83.8.35:51644;received=188.83.8.35;branch=z9hG4bK-524287-1---23c4406f193bcb55;rport=51644
To: <sip:91XXXXXXX@sip.sapo.pt;transport=UDP>;tag=1987012959-1508093419281
From: <sip:+35122XXXXXXX@sip.sapo.pt;transport=UDP>;tag=4158534c
Call-ID: VpWFoipO5npYWVQKJtZLSw..
CSeq: 1 INVITE
Content-Length: 181
Contact: <sip:91XXXXXXX@213.13.24.225:5060;transport=udp>
Content-Type: application/sdp
Call-Info: <sip:10.102.32.228>;appearance-index=1
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, UPDATE, NOTIFY
Supported: timer
P-Asserted-Identity: <sip:91XXXXXXX@sip.sapo.pt;user=phone>
v=0
o=BroadWorks 515332682 1 IN IP4 213.13.24.243
s=-
c=IN IP4 213.13.24.243
t=0 0
m=audio 28888 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
CANCEL sip:91XXXXXXX@sip.sapo.pt;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 188.83.8.35:51644;branch=z9hG4bK-524287-1---23c4406f193bcb55;rport
Max-Forwards: 70
To: <sip:91XXXXXXX@sip.sapo.pt;transport=UDP>
From: <sip:+35122XXXXXXX@sip.sapo.pt;transport=UDP>;tag=4158534c
Call-ID: VpWFoipO5npYWVQKJtZLSw..
CSeq: 1 CANCEL
User-Agent: Z 5.2.6 rv2.8.63
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 188.83.8.35:51644;received=188.83.8.35;branch=z9hG4bK-524287-1---23c4406f193bcb55;rport=51644
To: <sip:91XXXXXXX@sip.sapo.pt;transport=UDP>;tag=1987012959-1508093419281
From: <sip:+35122XXXXXXX@sip.sapo.pt;transport=UDP>;tag=4158534c
Call-ID: VpWFoipO5npYWVQKJtZLSw..
CSeq: 1 CANCEL
SIP/2.0 487 Request terminated
Via: SIP/2.0/UDP 188.83.8.35:51644;received=188.83.8.35;branch=z9hG4bK-524287-1---23c4406f193bcb55;rport=51644
To: <sip:91XXXXXXX@sip.sapo.pt;transport=UDP>;tag=1987012959-1508093419281
From: <sip:+35122XXXXXXX@sip.sapo.pt;transport=UDP>;tag=4158534c
Call-ID: VpWFoipO5npYWVQKJtZLSw..
CSeq: 1 INVITE
Content-Length: 0
ACK sip:91XXXXXXX@sip.sapo.pt;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 188.83.8.35:51644;branch=z9hG4bK-524287-1---23c4406f193bcb55;rport
Max-Forwards: 70
To: <sip:91XXXXXXX@sip.sapo.pt;transport=UDP>;tag=1987012959-1508093419281
From: <sip:+35122XXXXXXX@sip.sapo.pt;transport=UDP>;tag=4158534c
Call-ID: VpWFoipO5npYWVQKJtZLSw..
CSeq: 1 ACK
Content-Length: 0
A working session from ISSABEL:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 19772
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 213.13.24.193:5070:
INVITE sip:91XXXXXXXX@proxy.ims.iptv.telecom.pt:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.103.11:5060;branch=z9hG4bK1522bd94;rport
Max-Forwards: 70
From: <sip:+35122XXXXXXX@sip.sapo.pt>;tag=as03ed39f6
To: <sip:91XXXXXXXX@proxy.ims.iptv.telecom.pt:5070>
Contact: <sip:+35122XXXXXXXXX@192.168.103.11:5060>
Call-ID: 5694994c26570af048e1ed5215aaeca3@sip.sapo.pt
CSeq: 102 INVITE
User-Agent: MediaAccess TG789vac v2 Build 10.5.8.I.BQ CP1629UA7MD
Date: Sun, 15 Oct 2017 19:59:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 196653309 196653309 IN IP4 192.168.103.11
s=Asterisk PBX 11.25.0
c=IN IP4 192.168.103.11
t=0 0
m=audio 19772 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-- Called SIP/MEO/91XXXXXXX
<--- SIP read from UDP:213.13.24.225:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.103.11:5060;received=188.83.8.35;branch=z9hG4bK1522bd94;rport=59198
From: <sip:+35122XXXXXXXXX@sip.sapo.pt>;tag=as03ed39f6
To: <sip:91XXXXXXX@proxy.ims.iptv.telecom.pt:5070>
Call-ID: 5694994c26570af048e1ed5215aaeca3@sip.sapo.pt
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:213.13.24.225:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.103.11:5060;received=188.83.8.35;branch=z9hG4bK1522bd94;rport=59198
From: <sip:+35122XXXXXXXXX@sip.sapo.pt>;tag=as03ed39f6
To: <sip:91XXXXXXX@proxy.ims.iptv.telecom.pt:5070>;tag=1478266561-1508097586194
Call-ID: 5694994c26570af048e1ed5215aaeca3@sip.sapo.pt
CSeq: 102 INVITE
Content-Length: 239
Contact: <sip:91XXXXXXX@213.13.24.225:5060;transport=udp>
Content-Type: application/sdp
Call-Info: <sip:10.102.32.228>;appearance-index=1
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, UPDATE, NOTIFY
Supported: timer
P-Asserted-Identity: <sip:91XXXXXXX@sip.sapo.pt;user=phone>
Session: Media
v=0
o=BroadWorks 515557252 1 IN IP4 213.13.24.243
s=-
c=IN IP4 213.13.24.243
t=0 0
m=audio 27180 RTP/AVP 8 101 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
list_route: hop: <sip:91XXXXXXX@213.13.24.225:5060;transport=udp>
Found RTP audio format 8
Found RTP audio format 101
Found RTP audio format 0
Found audio description format telephone-event for ID 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 213.13.24.243:27180
-- SIP/MEO-00000001 is making progress passing it to SIP/600-00000000
Audio is at 18828
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
And a non working session from ISSABEL:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 13534
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 213.13.24.193:5070:
INVITE sip:91XXXXXXXX@proxy.ims.iptv.telecom.pt:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.103.11:5060;branch=z9hG4bK69e62da2;rport
Max-Forwards: 70
From: <sip:+35122XXXXXXX@sip.sapo.pt>;tag=as6be68d63
To: <sip:91XXXXXXXX@proxy.ims.iptv.telecom.pt:5070>
Contact: <sip:+35122XXXXXXX@192.168.103.11:5060>
Call-ID: 23686d5f394586730a1de0a233756789@sip.sapo.pt
CSeq: 102 INVITE
User-Agent: MediaAccess TG789vac v2 Build 10.5.8.I.BQ CP1629UA7MD
Date: Sun, 15 Oct 2017 20:03:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 738519219 738519219 IN IP4 192.168.103.11
s=Asterisk PBX 11.25.0
c=IN IP4 192.168.103.11
t=0 0
m=audio 13534 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-- Called SIP/MEO/91XXXXXXXX
<--- SIP read from UDP:213.13.24.225:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.103.11:5060;received=188.83.8.35;branch=z9hG4bK69e62da2;rport=59198
From: <sip:+35122XXXXXXX@sip.sapo.pt>;tag=as6be68d63
To: <sip:91XXXXXXXX@proxy.ims.iptv.telecom.pt:5070>
Call-ID: 23686d5f394586730a1de0a233756789@sip.sapo.pt
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:213.13.24.225:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.103.11:5060;received=188.83.8.35;branch=z9hG4bK69e62da2;rport=59198
From: <sip:+35122XXXXXXX@sip.sapo.pt>;tag=as6be68d63
To: <sip:91XXXXXXXX@proxy.ims.iptv.telecom.pt:5070>;tag=aprqngfrt-vq1hsv00000c6
Call-ID: 23686d5f394586730a1de0a233756789@sip.sapo.pt
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
Transmitting (NAT) to 213.13.24.225:5060:
ACK sip:91XXXXXXXX@proxy.ims.iptv.telecom.pt:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.103.11:5060;branch=z9hG4bK69e62da2;rport
Max-Forwards: 70
From: <sip:+35122XXXXXXX@sip.sapo.pt>;tag=as6be68d63
To: <sip:91XXXXXXXX@proxy.ims.iptv.telecom.pt:5070>;tag=aprqngfrt-vq1hsv00000c6
Contact: <sip:+35122XXXXXXX@192.168.103.11:5060>
Call-ID: 23686d5f394586730a1de0a233756789@sip.sapo.pt
CSeq: 102 ACK
User-Agent: MediaAccess TG789vac v2 Build 10.5.8.I.BQ CP1629UA7MD
Content-Length: 0
[2017-10-15 21:03:51] WARNING[9455][C-00000003]: chan_sip.c:23347 handle_response_invite: Received response: "Forbidden" from '<sip:+35122XXXXXXX@sip.sapo.pt>;tag=as6be68d63'
Scheduling destruction of SIP dialog '23686d5f394586730a1de0a233756789@sip.sapo.pt' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
Any help?