Can't hear busy tone with SIP Trunks

humber2 · February 11, 2016, 12:32pm

Hi,

I’m running Asterisk 11.2.1. We’ve a Sip Trunk with a Public Softswitch and when the public destinations are busy, our extensions can’t hear the busy tone.

The configuration of an extension:

[29149]
type=friend
callerid=(“Teste 1” <29149>)
context=infrap_internac
secret=&?#29149
host=dynamic
dtmfmode=rfc2833
username=29149
progressinband=yes
promiscredir=yes
canreinvite=yes
qualify=yes
;deny=0.0.0.0/0.0.0.0
;permit=10.201.8.20/255.255.255.255

And the configuration of SIP Trunk:

[TRUNKSIP-TB01]
type=peer
host=10.192.207.56
context=incoming-tb
;disallow=all
allow = all
dtmfmode=rfc2833
canreinvite=yes
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
permit=10.192.207.56/255.255.255.255
sendrpid = yes
trustrpid = yes
;directrtpsetup=yes

The CLI outputs no clue…as you can see above:

-- Executing [00211020097@infrap_internac:1] Dial("SIP/29149-0001abe5", "SIP/TRUNKSIP-TB01/211020097") in new stack

== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Called SIP/TRUNKSIP-TB01/211020097
– SIP/TRUNKSIP-TB01-0001abe6 is making progress passing it to SIP/29149-0001abe5

ambiorixg12 · February 11, 2016, 6:05pm

it seems early media issue, try using progress() before Dial()

humber2 · February 11, 2016, 6:18pm

Hi! Thanks for your suggestion.

I’ve the following line:

exten => _002XXXXXXXX,1,Dial(${TRUNKTB01}/${EXTEN:2},120,tT)

How can I put this option you I’ve suggested, please?

humber2 · February 11, 2016, 6:36pm

Hi,

Forget the my last post please…

With the following configuration, the result is the same:

exten => _002XXXXXXXX,1,Progress()
exten => _002XXXXXXXX,2,Dial(${TRUNKTB01}/${EXTEN:2},120,tT)

I still hear nothing. It’s supposed to hear busy tone when the remote party is busy…

david551 · February 11, 2016, 10:48pm

You need to use sip debug to get a protocol trace, to see if there is any hope of receiving the call progress signal.

Also, please review your sip.conf. Although not relevant to the question, some of the options are so old that they may have changed from deprecated to removed. Also, it is generally better to use type=peer for everything, unless you have multiple lines with the same IP address.

johnnyasterisk · February 11, 2016, 11:36pm

What handsets / softclients are you using? Do they show busy detection on the display when you ring a number that is busy? Also a sip trace would be good. I saw a similar issue before with Snoms getting service unavailable.

humber2 · February 12, 2016, 10:23am

Hi,

Thanks. I´ve changed the configuration of the extension 29149 to “peer” instead of “friend”.

The trace gave the following information:

— (10 headers 15 lines) —
Transmitting (no NAT) to 10.192.207.56:5060:
ACK sip:211020097@10.192.207.56:5060 SIP/2.0
Via: SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK0b821b5f
Max-Forwards: 70
From: “Teste 1” sip:29149@10.252.2.100;tag=as4d1b45c0
To: sip:211020097@10.192.207.56;tag=DFFE3030383334350516F5D8
Contact: sip:29149@10.252.2.100:5060
Call-ID: 610020d434945b28090b43112e8f7dac@10.252.2.100:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0

Scheduling destruction of SIP dialog ‘610020d434945b28090b43112e8f7dac@10.252.2.100:5060’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.192.250.237:5060 —>
ACK sip:00211020097@10.252.2.100:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bK28100bacFEC1AE41
From: “29149” sip:29149@10.252.2.100;tag=D906AC99-C456FA6E
To: sip:00211020097@10.252.2.100;user=phone;tag=as2178cb0d
CSeq: 2 ACK
Call-ID: 4291836d-48474642-9c0fae57@10.192.250.237
Contact: sip:29149@10.192.250.237
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Accept-Language: pt-pt,pt;q=0.9,en;q=0.8
Max-Forwards: 70
Content-Length: 0

The originator phone is a Polycom IP331 and the destination is a normal POTS Phone.

From another Polycom phones registered in other Asterisks, it works. So, it seems to be a configuration problem in this asterisk or a network data problem. Both configuration of the SIP Trunk with the public softswitch and the extension configuration are equal in the asterisks (but, I’ve changed the configuration for extension 29149 from type=friend to type=peer, as you’ve suggested).

david551 · February 12, 2016, 11:19am

The ACK contains no useful information. You need to provide all the previous responses, in particular 180, 183, 200 and 4xx responses, and the SDP associated with the 1xx and 200 responses.

humber2 · February 12, 2016, 12:29pm

First of all, is there a way to make the SIP debug only to peer 29149?
How I did was use the command - sip set debug on. But it just appear all calls and the result of the trace is somewhat confusing.
Could you see, now, if there are below all the necessary information for proper analysis of this problem, please?

And…thanks!

<— SIP read from UDP:10.192.250.237:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK5f3b462c
From: “asterisk” sip:asterisk@10.252.2.100;tag=as02d77b88
To: “29149” sip:29149@10.192.250.237;tag=9F745304-17C84B99
CSeq: 102 OPTIONS
Call-ID: 44ed77654ffc20b4453e07d8656aa815@10.252.2.100:5060
Contact: sip:29149@10.192.250.237
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Accept-Language: pt-pt,pt;q=0.9,en;q=0.8
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0

<— SIP read from UDP:10.192.250.237:5060 —>
INVITE sip:00211020097@10.252.2.100:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bK5907eaacF5814D41
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone
CSeq: 1 INVITE
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
Contact: sip:29149@10.192.250.237
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Accept-Language: pt-pt,pt;q=0.9,en;q=0.8
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 227

<------------->
— (15 headers 10 lines) —
Sending to 10.192.250.237:5060 (no NAT)
Using INVITE request as basis request - 52694516-de1222b-2d5e3480@10.192.250.237
Found peer ‘29149’ for ‘29149’ from 10.192.250.237:5060

<— Reliably Transmitting (no NAT) to 10.192.250.237:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bK5907eaacF5814D41;received=10.192.250.237
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone;tag=as0fa348dc
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
CSeq: 1 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“Jar209Reftel.refertelecom.pt”, nonce="54c892db"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘52694516-de1222b-2d5e3480@10.192.250.237’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.192.250.237:5060 —>
ACK sip:00211020097@10.252.2.100:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bK5907eaacF5814D41
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone;tag=as0fa348dc
CSeq: 1 ACK
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
Contact: sip:29149@10.192.250.237
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Accept-Language: pt-pt,pt;q=0.9,en;q=0.8
Max-Forwards: 70
Content-Length: 0

<------------->
— (12 headers 0 lines) —

<— SIP read from UDP:10.192.250.237:5060 —>
INVITE sip:00211020097@10.252.2.100:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bKb2a9cc1570E138EA
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone
CSeq: 2 INVITE
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
Contact: sip:29149@10.192.250.237
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Accept-Language: pt-pt,pt;q=0.9,en;q=0.8
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Authorization: Digest username=“29149”, realm=“Jar209Reftel.refertelecom.pt”, nonce=“54c892db”, uri=“sip:00211020097@10.252.2.100:5060;user=phone”, response=“5fc36ca9087c9f855d3cada1f13e470a”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 1455278821 1455278821 IN IP4 10.192.250.237
s=Polycom IP Phone
c=IN IP4 10.192.250.237
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 8 0 127
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
<------------->
— (16 headers 10 lines) —
Sending to 10.192.250.237:5060 (no NAT)
Using INVITE request as basis request - 52694516-de1222b-2d5e3480@10.192.250.237
Found peer ‘29149’ for ‘29149’ from 10.192.250.237:5060
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 127
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 127
Capabilities: us - (gsm|ulaw|alaw|g729|h261|h263|h263p), 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 10.192.250.237:2222
Peer doesn’t provide video
Looking for 00211020097 in infrap_internac (domain 10.252.2.100)
list_route: hop: sip:29149@10.192.250.237

<— Transmitting (no NAT) to 10.192.250.237:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bKb2a9cc1570E138EA;received=10.192.250.237
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
CSeq: 2 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:00211020097@10.252.2.100:5060
Content-Length: 0

<------------>
– Executing [00211020097@infrap_internac:1] Dial(“SIP/29149-0001b576”, “SIP/TRUNKSIP-TB01/211020097,120,tT”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
We think we can do text
Audio is at 17842
Video is at 10.252.2.100:17288
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100002 (gsm) to SDP
Adding video codec 200001 (h261) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100028 (speex32) to SDP
Adding video codec 200004 (h264) to SDP
Adding video codec 200005 (mpeg4) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.192.207.56:5060:
INVITE sip:211020097@10.192.207.56 SIP/2.0
Via: SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab
Max-Forwards: 70
From: “Teste 1” sip:29149@10.252.2.100;tag=as1e91da78
To: sip:211020097@10.192.207.56
Contact: sip:29149@10.252.2.100:5060
Call-ID: 41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2.1
Date: Fri, 12 Feb 2016 12:13:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “Teste 1” sip:29149@10.252.2.100;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 1687

v=0
o=root 248769938 248769938 IN IP4 10.252.2.100
s=Asterisk PBX 11.2.1
c=IN IP4 10.252.2.100
b=CT:384
t=0 0
m=audio 17842 RTP/AVP 8 0 18 3 4 112 5 7 110 97 111 9 102 115 116 117 96 100 107 108 10 118 119 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:100 SILK/12000
a=fmtp:100 maxaveragebitrate=12000
a=fmtp:100 usedtx=0
a=fmtp:100 useinbandfec=1
a=rtpmap:107 SILK/16000
a=fmtp:107 maxaveragebitrate=20000
a=fmtp:107 usedtx=0
a=fmtp:107 useinbandfec=1
a=rtpmap:108 SILK/24000
a=fmtp:108 maxaveragebitrate=30000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 17288 RTP/AVP 31 34 98 99 104
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:104 MP4V-ES/90000
a=sendrecv

-- Called SIP/TRUNKSIP-TB01/211020097

<— SIP read from UDP:10.192.207.56:5060 —>
SIP/2.0 100 Trying
From:"Teste 1"sip:29149@10.252.2.100;tag=as1e91da78
To:sip:211020097@10.192.207.56
Call-ID:41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq:102 INVITE
Server:TB008345/2.1
Via:SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab;received=10.252.2.100
Content-Length:0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.192.207.56:5060 —>
SIP/2.0 183 Session Progress
Content-Type:application/sdp
Contact:sip:211020097@10.192.207.56:5060
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY
Date:Fri, 12 Feb 2016 07:57:19 GMT
From:"Teste 1"sip:29149@10.252.2.100;tag=as1e91da78
To:sip:211020097@10.192.207.56;tag=FDC030303833343505180DFA
Call-ID:41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq:102 INVITE
Server:TB008345/2.1
Via:SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab;received=10.252.2.100
Content-Length:334

<— Transmitting (no NAT) to 10.192.250.237:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bKb2a9cc1570E138EA;received=10.192.250.237
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone;tag=as407f7f22
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
CSeq: 2 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:00211020097@10.252.2.100:5060
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1732830041 1732830041 IN IP4 10.252.2.100
s=Asterisk PBX 11.2.1
c=IN IP4 10.252.2.100
t=0 0
m=audio 16012 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

<------------->
— (15 headers 10 lines) —
Sending to 10.192.250.237:5060 (no NAT)
Using INVITE request as basis request - 52694516-de1222b-2d5e3480@10.192.250.237
Found peer ‘29149’ for ‘29149’ from 10.192.250.237:5060

<— Reliably Transmitting (no NAT) to 10.192.250.237:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bK5907eaacF5814D41;received=10.192.250.237
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone;tag=as0fa348dc
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
CSeq: 1 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“Jar209Reftel.refertelecom.pt”, nonce="54c892db"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘52694516-de1222b-2d5e3480@10.192.250.237’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.192.250.237:5060 —>
ACK sip:00211020097@10.252.2.100:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bK5907eaacF5814D41
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone;tag=as0fa348dc
CSeq: 1 ACK
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
Contact: sip:29149@10.192.250.237
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Accept-Language: pt-pt,pt;q=0.9,en;q=0.8
Max-Forwards: 70
Content-Length: 0

<------------->
— (12 headers 0 lines) —

<— SIP read from UDP:10.192.250.237:5060 —>
INVITE sip:00211020097@10.252.2.100:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bKb2a9cc1570E138EA
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone
CSeq: 2 INVITE
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
Contact: sip:29149@10.192.250.237
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Accept-Language: pt-pt,pt;q=0.9,en;q=0.8
Supported: 100rel,replaces
Allow-Events: conference,talk,hold
Authorization: Digest username=“29149”, realm=“Jar209Reftel.refertelecom.pt”, nonce=“54c892db”, uri=“sip:00211020097@10.252.2.100:5060;user=phone”, response=“5fc36ca9087c9f855d3cada1f13e470a”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 227

v=0
o=- 1455278821 1455278821 IN IP4 10.192.250.237
s=Polycom IP Phone
c=IN IP4 10.192.250.237
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 8 0 127
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
<------------->
— (16 headers 10 lines) —
Sending to 10.192.250.237:5060 (no NAT)
Using INVITE request as basis request - 52694516-de1222b-2d5e3480@10.192.250.237
Found peer ‘29149’ for ‘29149’ from 10.192.250.237:5060
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 127
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 127
Capabilities: us - (gsm|ulaw|alaw|g729|h261|h263|h263p), 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 10.192.250.237:2222
Peer doesn’t provide video
Looking for 00211020097 in infrap_internac (domain 10.252.2.100)
list_route: hop: sip:29149@10.192.250.237

<— Transmitting (no NAT) to 10.192.250.237:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bKb2a9cc1570E138EA;received=10.192.250.237
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
CSeq: 2 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:00211020097@10.252.2.100:5060
Content-Length: 0

<------------>
– Executing [00211020097@infrap_internac:1] Dial(“SIP/29149-0001b576”, “SIP/TRUNKSIP-TB01/211020097,120,tT”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
We think we can do text
Audio is at 17842
Video is at 10.252.2.100:17288
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100002 (gsm) to SDP
Adding video codec 200001 (h261) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100028 (speex32) to SDP
Adding video codec 200004 (h264) to SDP
Adding video codec 200005 (mpeg4) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.192.207.56:5060:
INVITE sip:211020097@10.192.207.56 SIP/2.0
Via: SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab
Max-Forwards: 70
From: “Teste 1” sip:29149@10.252.2.100;tag=as1e91da78
To: sip:211020097@10.192.207.56
Contact: sip:29149@10.252.2.100:5060
Call-ID: 41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2.1
Date: Fri, 12 Feb 2016 12:13:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “Teste 1” sip:29149@10.252.2.100;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 1687

v=0
o=root 248769938 248769938 IN IP4 10.252.2.100
s=Asterisk PBX 11.2.1
c=IN IP4 10.252.2.100
b=CT:384
t=0 0
m=audio 17842 RTP/AVP 8 0 18 3 4 112 5 7 110 97 111 9 102 115 116 117 96 100 107 108 10 118 119 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:100 SILK/12000
a=fmtp:100 maxaveragebitrate=12000
a=fmtp:100 usedtx=0
a=fmtp:100 useinbandfec=1
a=rtpmap:107 SILK/16000
a=fmtp:107 maxaveragebitrate=20000
a=fmtp:107 usedtx=0
a=fmtp:107 useinbandfec=1
a=rtpmap:108 SILK/24000
a=fmtp:108 maxaveragebitrate=30000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 17288 RTP/AVP 31 34 98 99 104
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:104 MP4V-ES/90000
a=sendrecv

-- Called SIP/TRUNKSIP-TB01/211020097

<— SIP read from UDP:10.192.207.56:5060 —>
SIP/2.0 100 Trying
From:"Teste 1"sip:29149@10.252.2.100;tag=as1e91da78
To:sip:211020097@10.192.207.56
Call-ID:41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq:102 INVITE
Server:TB008345/2.1
Via:SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab;received=10.252.2.100
Content-Length:0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.192.207.56:5060 —>
SIP/2.0 183 Session Progress
Content-Type:application/sdp
Contact:sip:211020097@10.192.207.56:5060
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY
Date:Fri, 12 Feb 2016 07:57:19 GMT
From:"Teste 1"sip:29149@10.252.2.100;tag=as1e91da78
To:sip:211020097@10.192.207.56;tag=FDC030303833343505180DFA
Call-ID:41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq:102 INVITE
Server:TB008345/2.1
Via:SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab;received=10.252.2.100
Content-Length:334

<— Transmitting (no NAT) to 10.192.250.237:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bKb2a9cc1570E138EA;received=10.192.250.237
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone;tag=as407f7f22
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
CSeq: 2 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:00211020097@10.252.2.100:5060
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1732830041 1732830041 IN IP4 10.252.2.100
s=Asterisk PBX 11.2.1
c=IN IP4 10.252.2.100
t=0 0
m=audio 16012 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

<------------->
— (11 headers 0 lines) —
Sending to 10.192.250.237:5060 (no NAT)

<— Reliably Transmitting (no NAT) to 10.192.250.237:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bKb2a9cc1570E138EA;received=10.192.250.237
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone;tag=as407f7f22
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
CSeq: 2 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

<— Transmitting (no NAT) to 10.192.250.237:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bKb2a9cc1570E138EA;received=10.192.250.237
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone;tag=as407f7f22
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
CSeq: 2 CANCEL
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060’ in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.192.207.56:5060:
CANCEL sip:211020097@10.192.207.56 SIP/2.0
Via: SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab
Max-Forwards: 70
From: “Teste 1” sip:29149@10.252.2.100;tag=as1e91da78
To: sip:211020097@10.192.207.56
Call-ID: 41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0

Scheduling destruction of SIP dialog ‘41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060’ in 6400 ms (Method: INVITE)
== Spawn extension (infrap_internac, 00211020097, 1) exited non-zero on ‘SIP/29149-0001b576’

<— SIP read from UDP:10.192.207.56:5060 —>
SIP/2.0 200 OK
From:"Teste 1"sip:29149@10.252.2.100;tag=as1e91da78
To:sip:211020097@10.192.207.56;tag=FDC030303833343505180DFA
Call-ID:41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq:102 CANCEL
Server:TB008345/2.1
Via:SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab;received=10.252.2.100
Content-Length:0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.192.207.56:5060 —>
SIP/2.0 487 Request Terminated
Content-Type:application/sdp
Reason:Q.850;cause=31
From:"Teste 1"sip:29149@10.252.2.100;tag=as1e91da78
To:sip:211020097@10.192.207.56;tag=FDC030303833343505180DFA
Call-ID:41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq:102 INVITE
Server:TB008345/2.1
Via:SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab;received=10.252.2.100
Content-Length:334

v=0
o=- 42899809 1 IN IP4 10.192.207.56
s=-
c=IN IP4 10.192.207.56
t=0 0
m=audio 12298 RTP/AVP 8 0 4 18 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=yes
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:13 CN/8000
<------------->
— (10 headers 15 lines) —
Transmitting (no NAT) to 10.192.207.56:5060:
ACK sip:211020097@10.192.207.56:5060 SIP/2.0
Via: SIP/2.0/UDP 10.252.2.100:5060;branch=z9hG4bK4a5c7eab
Max-Forwards: 70
From: “Teste 1” sip:29149@10.252.2.100;tag=as1e91da78
To: sip:211020097@10.192.207.56;tag=FDC030303833343505180DFA
Contact: sip:29149@10.252.2.100:5060
Call-ID: 41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0

Scheduling destruction of SIP dialog ‘41ebcdb444ce6c5e66368d6e5801965d@10.252.2.100:5060’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.192.250.237:5060 —>
ACK sip:00211020097@10.252.2.100:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.192.250.237;branch=z9hG4bKb2a9cc1570E138EA
From: “29149” sip:29149@10.252.2.100;tag=2F7FA542-F12FCD57
To: sip:00211020097@10.252.2.100;user=phone;tag=as407f7f22
CSeq: 2 ACK
Call-ID: 52694516-de1222b-2d5e3480@10.192.250.237
Contact: sip:29149@10.192.250.237
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Accept-Language: pt-pt,pt;q=0.9,en;q=0.8
Max-Forwards: 70
Content-Length: 0

david551 · February 12, 2016, 1:13pm

Does the calling system support early media (i.e. what happens if you replace Progress with Answer?

humber2 · February 12, 2016, 1:18pm

With this configuration the result is the same, i.e. I only hear silence…

exten => 00211020097,1,Answer()
exten => 00211020097,2,Dial(${TRUNKTB01}/${EXTEN:2},120,tT)

david551 · February 12, 2016, 1:21pm

In that case, I would say that problem lies on the called party side. They are claiming to send in band call progress but are not actually doing so. You should get a wireshark trace and confirm this.

humber2 · February 12, 2016, 1:22pm

With tcdump commnad, right?

david551 · February 12, 2016, 1:23pm

That’s how we get pcap files, but I believe wireshark will run directly on Linux.

humber2 · February 12, 2016, 1:25pm

I’m not a expert one using this kind of tools…but I will try to do it. Thanks.

humber2 · February 16, 2016, 2:28pm

Now I’ve a more seriously problem…I have no audio in communications…The configuration I’ve is the following:

Sip Trunk:

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

Example of an extension:

[29149]
type=peer
callerid=(“Teste 1” <29149>)
context=infrap_internac
secret=&?#%#29149
host=dynamic
dtmfmode=inband
username=29149
progressinband=yes
promiscredir=yes
canreinvite=yes
qualify=yes
;deny=0.0.0.0/0.0.0.0
;permit=10.201.8.20/255.255.255.255

And I don’t know why…

david551 · February 16, 2016, 9:54pm

I would turn off directrtpsetup. As far as I know it never got beyond experimental. Otherwise the most common causes of no media are firewalls and NAT. People often forget the firewall on Linux itself.

(Also canreinvite is an obsolete name. Please check your configuration against the latest documentation.)

ambiorixg12 · February 17, 2016, 1:53am

You don’t have the necessary settings to handle a NAT environment.

nat=yes is deprecated, use nat=force_rport,comedia instead.

**externaddr =**X.X.X.X ;your public IP

localnet=192.168.1.0/255.255.255.0 ;your localnet

qualify=yes ;to keep the nat session open

rtpkeepalive= ; Send keepalives in the RTP stream to keep NAT open default is off - zero

directmedia=no

canreinvite= was renamed to directmedia= in Asterisk 1.6.2 to more accurately describe what this setting does.

also change this

;disallow=all
allow = all

for

disallow=all
allow=ulaw
allow=alaw
or any other codec supported by your system, to avoid a posible SIP 488 Not Acceptable Here

david551 · February 17, 2016, 7:55am

They didn’t have anything that suggested they were actually in a NAT environment in the initial post.

Whilst qualify can keep NAT open, it can also make the trunk unusable, if the remote side or router are broken in such a way that they ignore it.

nat= is supposed to be used on an as needed basis, not as a recipe. That is almost certainly why nat=yes was removed.

humber2 · February 17, 2016, 11:17am

What is annoying is that the customer’s data network is not managed by me, and the manager of this network asserts that there is no firewalls or Nat configuration on your side.

With this configuration the result is unfortunately the same:

[29149]
type=peer
callerid=(“Teste 1” <29149>)
context=infrap_internac
secret=&?#%#29149
host=dynamic
dtmfmode=inband
username=29149
progressinband=yes
promiscredir=yes
canreinvite=no
qualify=no
;deny=0.0.0.0/0.0.0.0
;permit=10.201.8.20/255.255.255.255

And the sip trunk:

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

I’ve tried a lot of combinations of these parameters, but the result is always the same…no audio in communications.

Is there any way to provide the tcdump that I’ve got?

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