Everyone is busy/congested at this time

Asterisk Asterisk General
1

Hi,

In extensions.conf we’ve the following configuration for routing 9XXXXXXXX:

exten => _0091.,6,Dial(${TRUNKTB01}/${EXTEN:2}) -> Trunksip
exten => _0091.,7,Dial(${TRUNKTB02}/${EXTEN:2}) -> Trunksip
exten => _0091.,8,Dial(${TRUNKPSTN}/${EXTEN:2}) -> ISDN

But the calls go always to the third option. Supposedly because the channels are busy/congested in the first two options, and also we’ve got a forbidden message…but we don’t know why…

We can receive incoming calls successfully from both these trunk sips and of course, the status of both trunks is ok…

The configuration f the two Trunksips is as follows:

[TRUNKSIP-TB01]
type=peer
host=10.192.207.56
context=incoming-tb
;disallow=all
allow = all
dtmfmode=inband
canreinvite=no
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
permit=10.192.207.56/255.255.255.255

[TRUNKSIP-TB02]
type=peer
host=10.192.207.55
context=incoming-tb
;disallow=all
allow = all
dtmfmode=inband
canreinvite=no
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
permit=10.192.207.55/255.255.255.255

And what we’ve got when we make a call:

  • Executing [00918912099@fernave_nacional:6] Dial(“SIP/28751-c80b8a28”, “SIP/TRUNKSIP-TB01/918912099”) in new stack
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Using SIP VRTP TOS bits 136
    == Using SIP VRTP CoS mark 4
    – Called TRUNKSIP-TB01/918912099
    [Nov 26 16:01:58] WARNING[9776]: chan_sip.c:16406 handle_response_invite: Received response: “Forbidden” from ‘“Teste” sip:211028751@10.192.230.240;tag=as70e2e118’
    – SIP/TRUNKSIP-TB01-1a5a08c8 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    – Executing [00918912099@fernave_nacional:7] Dial(“SIP/28751-c80b8a28”, “SIP/TRUNKSIP-TB02/918912099”) in new stack
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Using SIP VRTP TOS bits 136
    == Using SIP VRTP CoS mark 4
    – Called TRUNKSIP-TB02/918912099
    [Nov 26 16:01:58] WARNING[9776]: chan_sip.c:16406 handle_response_invite: Received response: “Forbidden” from ‘“Teste” sip:211028751@10.192.230.240;tag=as428802cf’
    – SIP/TRUNKSIP-TB02-1a08da98 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    – Executing [00918912099@fernave_nacional:8] Dial(“SIP/28751-c80b8a28”, “DAHDI/g1d/918912099”) in new stack
    – Requested transfer capability: 0x00 - SPEECH
    – Called g1d/918912099
    == Spawn extension (incoming-ep, 22704, 81) exited non-zero on ‘SIP/TRUNKSIP-EP-c8067318’
    – DAHDI/1-1 is ringing
    – Hungup 'DAHDI/1-1’
    How can I fix this issue, please?
2

Hangup cause 21. Call Rejected

SIP reponse 403 Forbidden

The server understood the request, but is refusing to fulfill it.
Authorization will not help, and the request SHOULD NOT be repeated.

I dont know what version of Asterisk you are running , but you have some options who are deprecated on the current version like :

canreinvite=no
nat=no
also you should disallow=all and then allow the codec present on your system

3

Hi,

Thanks for your help.

How have you reached to this result, please?

"Hangup cause 21. Call Rejected

SIP reponse 403 Forbidden".

I’ve changed the configuration as follows:

[TRUNKSIP-TB01]
type=peer
host=10.192.207.56
context=incoming-tb
disallow=all
allow = ulaw
dtmfmode=inband
canreinvite=no
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
permit=10.192.207.56/255.255.255.255

[TRUNKSIP-TB02]
type=peer
host=10.192.207.55
context=incoming-tb
disallow=all
allow = ulaw
dtmfmode=inband
canreinvite=no
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
permit=10.192.207.55/255.255.255.255

But the result is the same…:frowning:

  • Executing [s@macro-prefixo:130] Set(“SIP/28751-c84f7068”, “CDR(userfield)=call-211028751”) in new stack
    – Executing [00918912099@fernave_nacional:6] Dial(“SIP/28751-c84f7068”, “SIP/TRUNKSIP-TB01/918912099”) in new stack
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Using SIP VRTP TOS bits 136
    == Using SIP VRTP CoS mark 4
    – Called TRUNKSIP-TB01/918912099
    [Nov 26 18:03:41] WARNING[9776]: chan_sip.c:16406 handle_response_invite: Received response: “Forbidden” from ‘“Teste” sip:211028751@10.192.230.240;tag=as2eab6470’
    – SIP/TRUNKSIP-TB01-cc014fa8 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    – Executing [00918912099@fernave_nacional:7] Dial(“SIP/28751-c84f7068”, “SIP/TRUNKSIP-TB02/918912099”) in new stack
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Using SIP VRTP TOS bits 136
    == Using SIP VRTP CoS mark 4
    – Called TRUNKSIP-TB02/918912099
    [Nov 26 18:03:41] WARNING[9776]: chan_sip.c:16406 handle_response_invite: Received response: “Forbidden” from ‘“Teste” sip:211028751@10.192.230.240;tag=as6a939d0b’
    – SIP/TRUNKSIP-TB02-cc01a8c8 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    – Executing [00918912099@fernave_nacional:8] Dial(“SIP/28751-c84f7068”, “DAHDI/g1d/918912099”) in new stack
    – Requested transfer capability: 0x00 - SPEECH
    – Called g1d/918912099
    – B-channel 0/29 successfully restarted on span 1
    – DAHDI/1-1 is ringing

And now what should I try, please?

4

You are getting a 403 reponse from 10.192.230.240

this could be caused for differents reasons , for example :

Auth Mismatch, If they do the authentication based on IP address make sure your IP is whitelisted on your carrier side, Also check your account balance

5

Hi,

Thanks for your reply.

The IP 10.192.230.240 is the IP address of server1. The configuration of this extension in this server is the following:

[28751]
type=friend
callerid=(“Teste” <28751>)
context=fernave_nacional
secret=&&28751$$
host=dynamic
dtmfmode=rfc2833
username=28751
progressinband=no
promiscredir=yes
canreinvite=no
qualify=no
;deny=0.0.0.0/0.0.0.0
;permit=10.192.250.100/255.255.255.255

The carrier isn’t receiving any attempt from the SIP Trunks but only from ISND side.

So the server1 ir rejecting the call…but I can’t understand why…

6

There is no secret on the calling side.

7

But Why? The settings are right, I think so…

8

The settings are wrong, because a secret is specified on one side of the trunk, nut not the other.

Also, rather than letting us guess, as you control both sides of the trunk, you should turn up the logging on the side that is rejecting until it tells you why it is rejecting.

9

Could you be more explicit? Which is in fact the error, please?

10

On the other side we have the carrier that assures us that do not get anything, except when calls come from ISDN trunk.

11

Follow this advice

[quote=“david55”]The settings are wrong, because a secret is specified on one side of the trunk, nut not the other.

Also, rather than letting us guess, as you control both sides of the trunk, you should turn up the logging on the side that is rejecting until it tells you why it is rejecting.[/quote]

use sip set debug on to turn up a full sip trace

12

From the carrier they say It’s all ok. And we’ve another Sip Trunks working well with them.

I don´t undestand the result of the trace. Can you help me:

tel*CLI>
<— Transmitting (no NAT) to 10.192.250.100:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.192.250.100;branch=z9hG4bK82006402CA08B25D;received=10.192.250.100
From: “28751” sip:28751@10.192.230.240;tag=CD6194C5-ACF1F2D0
To: sip:00918912099@10.192.230.240;user=phone;tag=as5d976281
Call-ID: 2b257a11-f432dadc-aef8ffd7@10.192.250.100
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:00918912099@10.192.230.240
Content-Length: 0

<------------>
Really destroying SIP dialog ‘62ea41821cd35cb564c12f20621bf49f@10.192.151.83:5060’ Method: OPTIONS
Really destroying SIP dialog ‘1681abf0563e3d753e4625dd65fe2960@10.192.151.83:5060’ Method: OPTIONS
Really destroying SIP dialog ‘6b9d6c662c37d3bc052df2520b7fe826@10.168.25.251’ Method: REGISTER
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– B-channel 0/23 successfully restarted on span 2
tel*CLI>
<— SIP read from UDP://10.157.178.1:5060 —>

<------------->
Really destroying SIP dialog ‘54477fd51b33d5c7746ba0d0101c0d67@10.252.0.10’ Method: OPTIONS
Really destroying SIP dialog ‘3c00398271358cac723ed1ef5c12bf36@10.252.0.10’ Method: OPTIONS
Really destroying SIP dialog ‘52f4899a77fd96cd126a8d540d103fc1@10.252.0.10’ Method: OPTIONS
tel*CLI>
<— SIP read from UDP://10.169.3.250:5060 —>
OPTIONS sip:10.192.231.240 SIP/2.0
Via: SIP/2.0/UDP 10.169.3.250:5060;branch=z9hG4bK44bd6d21
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.169.3.250;tag=as3d9df460
To: sip:10.192.231.240
Contact: sip:asterisk@10.169.3.250:5060
Call-ID: 1806b4f120e2d5b83e101d9764882b66@10.169.3.250:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.1
Date: Tue, 01 Dec 2015 16:40:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in default (domain 10.192.231.240)
tel*CLI>
<— Transmitting (no NAT) to 10.169.3.250:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.169.3.250:5060;branch=z9hG4bK44bd6d21;received=10.169.3.250
From: “asterisk” sip:asterisk@10.169.3.250;tag=as3d9df460
To: sip:10.192.231.240;tag=as292e5892
Call-ID: 1806b4f120e2d5b83e101d9764882b66@10.169.3.250:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:10.192.230.240
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1806b4f120e2d5b83e101d9764882b66@10.169.3.250:5060’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘548cd699-bf3dd8f0-7997acbf@10.168.242.39’ Method: REGISTER
– B-channel 0/18 successfully restarted on span 1
Really destroying SIP dialog ‘25fcc68333c40a7425f57f4943076ebb@10.168.25.250’ Method: OPTIONS
tel*CLI>
<— SIP read from UDP://10.252.0.10:5060 —>
INVITE sip:61791@10.192.230.240 SIP/2.0
Via: SIP/2.0/UDP 10.252.0.10:5060;branch=z9hG4bK288bd168;rport
From: “Faira Ussene” sip:29493@10.252.0.10;tag=as7083baa2
To: sip:61791@10.192.230.240
Contact: sip:29493@10.252.0.10
Call-ID: 135ea42623346176450300f437974dec@10.252.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 01 Dec 2015 16:51:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 991 991 IN IP4 10.252.0.10
s=session
c=IN IP4 10.252.0.10
t=0 0
m=audio 14696 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
— (14 headers 13 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
Sending to 10.252.0.10 : 5060 (no NAT)
Using INVITE request as basis request - 135ea42623346176450300f437974dec@10.252.0.10
Found peer ‘TRUNKSIP-EP’ for ‘29493’ from 10.252.0.10:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.252.0.10:14696
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.252.0.10:14696
Looking for 61791 in incoming-ep (domain 10.192.230.240)
list_route: hop: sip:29493@10.252.0.10
tel*CLI>
<— Transmitting (no NAT) to 10.252.0.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.252.0.10:5060;branch=z9hG4bK288bd168;received=10.252.0.10;rport=5060
From: “Faira Ussene” sip:29493@10.252.0.10;tag=as7083baa2
To: sip:61791@10.192.230.240
Call-ID: 135ea42623346176450300f437974dec@10.252.0.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:61791@10.192.230.240
Content-Length: 0

<------------>
– Executing [61791@incoming-ep:1] Set(“SIP/TRUNKSIP-EP-cc057d58”, “number=29493”) in new stack
– Executing [61791@incoming-ep:2] Dial(“SIP/TRUNKSIP-EP-cc057d58”, “SIP/TRUNKSIP-MXONE07/4444444461791,200,tT”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
Audio is at 10.192.230.240 port 15830
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 10.192.204.201:5060:
INVITE sip:4444444461791@10.192.204.201 SIP/2.0
Via: SIP/2.0/UDP 10.192.230.240:5060;branch=z9hG4bK7f6d017d;rport
Max-Forwards: 70
From: “Faira Ussene” sip:29493@10.192.230.240;tag=as78beec00
To: sip:4444444461791@10.192.204.201
Contact: sip:29493@10.192.230.240
Call-ID: 4a770f140072022f6c006688786d079f@10.192.230.240
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.6
Remote-Party-ID: “Faira Ussene” sip:29493@10.192.230.240;privacy=off;screen=no
Date: Tue, 01 Dec 2015 16:40:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 913345299 913345299 IN IP4 10.192.230.240
s=Asterisk PBX 1.6.1.6
c=IN IP4 10.192.230.240
t=0 0
m=audio 15830 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called TRUNKSIP-MXONE07/4444444461791

tel*CLI>
<— SIP read from UDP://10.192.204.201:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.192.230.240:5060;branch=z9hG4bK7f6d017d;rport=5060
To: sip:4444444461791@10.192.204.201;tag=4cc81036
From: "Faira Ussene"sip:29493@10.192.230.240;tag=as78beec00
Call-ID: 4a770f140072022f6c006688786d079f@10.192.230.240
CSeq: 102 INVITE
User-Agent: Aastra MX-ONE SN/13.211.3
Content-Length: 0

<------------->
— (8 headers 0 lines) —
tel*CLI>
<— SIP read from UDP://10.192.204.201:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.192.230.240:5060;branch=z9hG4bK7f6d017d;rport=5060
Record-Route: sip:10.192.204.201:5060;lr;transport=UDP
Contact: sip:4444444461791@10.192.204.201
To: sip:4444444461791@10.192.204.201;tag=4cc81036
From: "Faira Ussene"sip:29493@10.192.230.240;tag=as78beec00
Call-ID: 4a770f140072022f6c006688786d079f@10.192.230.240
CSeq: 102 INVITE
Content-Type: application/sdp
User-Agent: Aastra MX-ONE SN/13.211.3
Content-Length: 422

v=0
o=2020010001 6223356085182092373 6223356085182166441 IN IP4 10.192.204.214
s=MX-ONE
c=IN IP4 10.192.204.214
t=0 0
m=audio 25748 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sqn:0
a=cdsc:1 image udptl t38
a=cpar:a=T38FaxVersion:0
a=cpar:a=T38MaxBitRate:14400
a=cpar:a=T38FaxRateManagement:transferredTCF
a=cpar:a=T38FaxMaxBuffer:9772
a=cpar:a=T38FaxMaxDatagram:1472
a=cpar:a=T38FaxUdpEC:t38UDPRedundancy

<------------->
— (11 headers 16 lines) —
Found RTP audio format 0
Peer audio RTP is at port 10.192.204.214:25748
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.192.204.214:25748
tel*CLI>
<— SIP read from UDP://10.192.204.201:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.192.230.240:5060;branch=z9hG4bK7f6d017d;rport=5060
Record-Route: sip:10.192.204.201:5060;lr;transport=UDP
Contact: sip:4444444461791@10.192.204.201
To: sip:4444444461791@10.192.204.201;tag=4cc81036
From: "Faira Ussene"sip:29493@10.192.230.240;tag=as78beec00
Call-ID: 4a770f140072022f6c006688786d079f@10.192.230.240
CSeq: 102 INVITE
Content-Type: application/sdp
User-Agent: Aastra MX-ONE SN/13.211.3
ontent-Length: 422

v=0
o=2020010001 6223356085182092373 6223356085182166441 IN IP4 10.192.204.214
s=MX-ONE
c=IN IP4 10.192.204.214
t=0 0
m=audio 25748 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sqn:0
a=cdsc:1 image udptl t38
a=cpar:a=T38FaxVersion:0
a=cpar:a=T38MaxBitRate:14400
a=cpar:a=T38FaxRateManagement:transferredTCF
a=cpar:a=T38FaxMaxBuffer:9772
a=cpar:a=T38FaxMaxDatagram:1472
a=cpar:a=T38FaxUdpEC:t38UDPRedundancy

<------------->
— (11 headers 16 lines) —
– SIP/TRUNKSIP-MXONE07-17298578 is making progress passing it to SIP/TRUNKSIP-EP-cc057d58
Audio is at 10.192.230.240 port 13664
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
tel*CLI>
<— Transmitting (no NAT) to 10.252.0.10:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.252.0.10:5060;branch=z9hG4bK288bd168;received=10.252.0.10;rport=5060
From: “Faira Ussene” sip:29493@10.252.0.10;tag=as7083baa2
To: sip:61791@10.192.230.240;tag=as6626f4a0
Call-ID: 135ea42623346176450300f437974dec@10.252.0.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:61791@10.192.230.240
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1796492428 1796492428 IN IP4 10.192.230.240
s=Asterisk PBX 1.6.1.6
c=IN IP4 10.192.230.240
t=0 0
m=audio 13664 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– SIP/TRUNKSIP-MXONE07-17298578 is ringing
tel*CLI>
<— Transmitting (no NAT) to 10.252.0.10:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.252.0.10:5060;branch=z9hG4bK288bd168;received=10.252.0.10;rport=5060
From: “Faira Ussene” sip:29493@10.252.0.10;tag=as7083baa2
To: sip:61791@10.192.230.240;tag=as6626f4a0
Call-ID: 135ea42623346176450300f437974dec@10.252.0.10
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:61791@10.192.230.240
Content-Length: 0

<------------>
– SIP/TRUNKSIP-MXONE07-17298578 is making progress passing it to SIP/TRUNKSIP-EP-cc057d58
Really destroying SIP dialog ‘19b45de66554d4b663884e976df6c2dc@10.168.25.250’ Method: OPTIONS
tel*CLI>
<— SIP read from UDP://10.169.3.250:5060 —>
OPTIONS sip:10.192.230.240 SIP/2.0
Via: SIP/2.0/UDP 10.169.3.250:5060;branch=z9hG4bK7f7ab605
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.169.3.250;tag=as00cfa5f1
To: sip:10.192.230.240
Contact: sip:asterisk@10.169.3.250:5060
Call-ID: 1da53b6f5aa6e5bb518c68b957502826@10.169.3.250:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.1
Date: Tue, 01 Dec 2015 16:40:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in default (domain 10.192.230.240)
tel*CLI>
<— Transmitting (no NAT) to 10.169.3.250:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.169.3.250:5060;branch=z9hG4bK7f7ab605;received=10.169.3.250
From: “asterisk” sip:asterisk@10.169.3.250;tag=as00cfa5f1
To: sip:10.192.230.240;tag=as0c15c884
Call-ID: 1da53b6f5aa6e5bb518c68b957502826@10.169.3.250:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:10.192.230.240
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1da53b6f5aa6e5bb518c68b957502826@10.169.3.250:5060’ in 32000 ms (Method: OPTIONS)
– Channel 0/1, span 1 got hangup request, cause 17
– DAHDI/1-1 is busy
– Hungup ‘DAHDI/1-1’
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [00918912099@fernave_nacional:9] Hangup(“SIP/28751-cc010878”, “”) in new stack
== Spawn extension (fernave_nacional, 00918912099, 9) exited non-zero on 'SIP/28751-cc010878’
Scheduling destruction of SIP dialog ‘2b257a11-f432dadc-aef8ffd7@10.192.250.100’ in 32000 ms (Method: INVITE)
tel*CLI>
<— Reliably Transmitting (no NAT) to 10.192.250.100:5060 —>
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 10.192.250.100;branch=z9hG4bK82006402CA08B25D;received=10.192.250.100
From: “28751” sip:28751@10.192.230.240;tag=CD6194C5-ACF1F2D0
To: sip:00918912099@10.192.230.240;user=phone;tag=as5d976281
Call-ID: 2b257a11-f432dadc-aef8ffd7@10.192.250.100
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
tel*CLI>
<— SIP read from UDP://10.192.250.100:5060 —>
ACK sip:00918912099@10.192.230.240:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.192.250.100;branch=z9hG4bK82006402CA08B25D
From: “28751” sip:28751@10.192.230.240;tag=CD6194C5-ACF1F2D0
To: sip:00918912099@10.192.230.240;user=phone;tag=as5d976281
CSeq: 2 ACK
Call-ID: 2b257a11-f432dadc-aef8ffd7@10.192.250.100
Contact: sip:28751@10.192.250.100
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_321-UA/4.0.3.7562
Accept-Language: en
Max-Forwards: 70
Content-Length: 0

13

Trace doesn’t match original problem.

As you control both ends, we need traces from both ends.

14

The another End is the carrier…I only got the trace for the Asterisk side. How can we set a Sip debug for only a especific call, please? Otherwise is confusing…

15

OK. I misinterpreted the information in the 403 error message and thought that the address was that of the peer.

The call is being rejected by TRUNKSIP-TB02/918912099 and only the people who run that will know why they are rejecting it, although likely reasons are that you haven’t registered properly, the password is wrong, or they expect the From header user name to include a particular value, and not the callerID (typical of as service not intended for PABXes).

16

Hi,

This is the reply from our carrier:

i) SDP received by a Trunk SIP that is working well

v=0
o=- 4580 4580 IN IP4 10.192.231.244
s=-
c=IN IP4 10.192.231.244
t=0 0
m=audio 10754 RTP/AVP 8 0 18 3 4 111 5 10 7 110 97 101
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
m=video 11110 RTP/AVP 26 31 34 103
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF:2
a=rtpmap:103 H263-2000/90000
a=fmtp:103 profile=0;level=10

ii) SDP received by the trunk that isn’t working:

v=0
o=- 2028614935 2028614935 IN IP4 10.192.230.240
s=-
c=IN IP4 10.192.230.240
t=0 0
m=audio 10962 RTP/AVP 0
a=silenceSupp:off - - - -

So can you put the configuration of this TRunk SIP as the first one?

But I ‘ve confirmed that both configurations are equal…

17

A problem with the SDP should produce Not Acceptable, not Forbidden. I would expect Forbidden to be the result of something in the SIP proper, that they didn’t like.

To get close to the SDP shown, you will need to:

  1. set dtmfmode=rfc2833; and

  2. use a videophone that supports all the listed codecs.

(Chances are that the ITSP doesn’t actually support video, but that is in the offer that you want to try to reproduce.)

I’m surprised that Asterisk hasn’t listed all the supported audio codecs, so I wonder if you have built asterisk with limited support.

canreinvite is deprecated and may be ignored in currently supported versions of Asterisk. You should review your sip.conf options against a current list. However directmedia=no is, I think, the default.

18

Hi,

The technical information sent by the carrier is the following:

"We’ve a Trunk Sip that is working weel and we’ve got the following SDP:

v=0
o=- 4580 4580 IN IP4 10.192.231.244
s=-
c=IN IP4 10.192.231.244
t=0 0
m=audio 10754 RTP/AVP 8 0 18 3 4 111 5 10 7 110 97 101
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
m=video 11110 RTP/AVP 26 31 34 103
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF:2
a=rtpmap:103 H263-2000/90000
a=fmtp:103 profile=0;level=10

ii) SDP sent by this problematic one:

v=0
o=- 2028614935 2028614935 IN IP4 10.192.230.240
s=-
c=IN IP4 10.192.230.240
t=0 0
m=audio 10962 RTP/AVP 0
a=silenceSupp:off - - - -

So, the configurations must be different…"

But, I’ve confirmed and they’re equal…

:frowning:

19

Sorry by the last post…is obviously to go to trash…

good news…once I set dtmfmode to rfc2833 it worked!

But in the other Trunksip it works with “inband”…

Thanks a lot!