No sound on sip phone?

parisien9999 · October 21, 2012, 7:44pm

Hello.

i am connecting a siemens gigaset A580 IP to my asterisk server.

the server is at my office and the siemens gigaset is at my home (not the same network)

I have put that in the sip.conf :

[code][general]
canreinvite=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
register = xxxxxxxx@freephonie.net

[210]
type=friend
;auth=md5
username=210
secret=motdepasse123
callerid=“210” <0033183644865>
host=dynamic
context=interne
language=fr
insecure=port
nat=yes
canreinvite=no
dtmfmode=auto
video=no
restrictcid=no
amaflags=default
mailbox=1000@maison[/code]

and i have that in my extension.conf

[code][interne]
exten => 1,1,Answer()
same => n,Set(TIMEOUT(digit)=2)
same => n,Wait(1)
same => n(menuprompt),Background(main-menu)
same => n,WaitExten(40)

[/code]

when i call from the phone and do the 1 extension, i get this on asterisk :

-- Executing [1@interne:1] Answer("SIP/210-0000002b", "") in new stack -- Executing [1@interne:2] Set("SIP/210-0000002b", "TIMEOUT(digit)=2") in new stack -- Digit timeout set to 2.000 -- Executing [1@interne:3] Wait("SIP/210-0000002b", "1") in new stack -- Executing [1@interne:4] BackGround("SIP/210-0000002b", "main-menu") in new stack -- <SIP/210-0000002b> Playing 'main-menu.alaw' (language 'fr') -- Executing [1@interne:5] WaitExten("SIP/210-0000002b", "40") in new stack [Oct 21 02:22:33] WARNING[19760]: chan_sip.c:3694 retrans_pkt: Retransmission timeout reached on transmission 886565065@192_168_1_75 for seqno 3 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6720ms with no response [Oct 21 02:22:33] WARNING[19760]: chan_sip.c:3723 retrans_pkt: Hanging up call 886565065@192_168_1_75 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). == Spawn extension (interne, 1, 5) exited non-zero on 'SIP/210-0000002b'

here is the debug :

zz.zz.zzz.zzz is the ip of the network where the hardphone is behind a nat
yy.yyy.yyy.yyy is the ip of the network where is the server asterisk (behind a nat)
domaine.fr is the domaine name of the network where is the asterisk server.

192.168.1.75 is the internal ip of the sip hardphone
192.168.0.41 is the internal ip of asterisk.

thank you for your help

[code]
localhostCLI> amaflags=defaultrestrictcid=novideo=nodtmfmode=autocanreinvite=nonat=yesinsecure=portlanguage=frcontext=internehost=dynamiccallerid=“210” <0033183644865>secret=motdepasse123;md5918908995ebcc366f28fb0651a6eded9username=test;auth=md5type=friend[210]sip set debug offncore stop nowsip set debug on
localhostCLI>
SIP Debugging enabled

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
<— SIP read from UDP:zz.zz.zzz.zzz:5060 —>
INVITE sip:1@domaine.fr;user=phone SIP/2.0
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bK4b481aa542e330345a5f0e498a8b6ce;rport
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone
Call-ID: 4002489461@192_168_1_75
CSeq: 2 INVITE
Contact: sip:210@zz.zz.zzz.zzz:5060
Max-Forwards: 70
User-Agent: A580 IP/022270000000
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 368

v=0
o=- 5008 66 IN IP4 zz.zz.zzz.zzz
s=Mapping
c=IN IP4 zz.zz.zzz.zzz
t=0 0
m=audio 5008 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->

localhost*CLI>
— (14 headers 16 lines) —

localhost*CLI>
Sending to zz.zz.zzz.zzz:5060 (NAT)

localhost*CLI>
Using INVITE request as basis request - 4002489461@192_168_1_75

localhost*CLI>
Found peer ‘210’ for ‘210’ from zz.zz.zzz.zzz:5060

localhost*CLI>
<— Reliably Transmitting (NAT) to zz.zz.zzz.zzz:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bK4b481aa542e330345a5f0e498a8b6ce;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as185d3e7c
Call-ID: 4002489461@192_168_1_75
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5db65c0a"
Content-Length: 0

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

localhost*CLI>
Scheduling destruction of SIP dialog ‘4002489461@192_168_1_75’ in 32000 ms (Method: INVITE)

localhost*CLI>
<— SIP read from UDP:zz.zz.zzz.zzz:5060 —>
ACK sip:1@domaine.fr;user=phone SIP/2.0
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bK4b481aa542e330345a5f0e498a8b6ce;rport
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as185d3e7c
Call-ID: 4002489461@192_168_1_75
CSeq: 2 ACK
Contact: sip:210@zz.zz.zzz.zzz:5060
Max-Forwards: 70
User-Agent: A580 IP/022270000000
Content-Length: 0

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

localhost*CLI>
<— SIP read from UDP:zz.zz.zzz.zzz:5060 —>
INVITE sip:1@domaine.fr;user=phone SIP/2.0
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;rport
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Contact: sip:210@zz.zz.zzz.zzz:5060
Authorization: Digest username=“210”, realm=“asterisk”, algorithm=MD5, uri="sip:1@domaine.fr;user=phone", nonce=“5db65c0a”, response="b3ea109507f2b980187b455d031a6cc3"
Max-Forwards: 70
User-Agent: A580 IP/022270000000
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 368

v=0
o=- 5008 66 IN IP4 zz.zz.zzz.zzz
s=Mapping
c=IN IP4 zz.zz.zzz.zzz
t=0 0
m=audio 5008 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->

localhost*CLI>
— (15 headers 16 lines) —
Sending to zz.zz.zzz.zzz:5060 (NAT)
Using INVITE request as basis request - 4002489461@192_168_1_75
Found peer ‘210’ for ‘210’ from zz.zz.zzz.zzz:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 9

localhost*CLI>
Found RTP audio format 0

localhost*CLI>
Found RTP audio format 8

localhost*CLI>
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 96
Found audio description format AAL2-G726-32 for ID 97
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x191c (ulaw|alaw|g726|g729|g726aal2|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port zz.zz.zzz.zzz:5008
Looking for 1 in interne (domain domaine.fr)
list_route: hop: sip:210@zz.zz.zzz.zzz:5060

<— Transmitting (NAT) to zz.zz.zzz.zzz:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060
Content-Length: 0

<------------>
– Executing [1@interne:1] Answer(“SIP/210-00000002”, “”) in new stack
Audio is at 16214
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to zz.zz.zzz.zzz:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

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

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #1 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060

localhost*CLI>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
– Executing [1@interne:2] Set(“SIP/210-00000002”, “TIMEOUT(digit)=2”) in new stack

localhost*CLI>
– Digit timeout set to 2.000

localhost*CLI>
– Executing [1@interne:3] Wait(“SIP/210-00000002”, “1”) in new stack

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK76cfdfad;rport
Max-Forwards: 70
From: “asterisk” sip:0000000000@192.168.0.41;tag=as50cf77ce
To: sip:freephonie.net
Contact: sip:0000000000@192.168.0.41:5060
Call-ID: 39120b6b35c8f1c626e2580e770a3206@192.168.0.41:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.17.0
Date: Sun, 21 Oct 2012 19:27:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
SIP/2.0 501 Not Implemented Yet
Call-ID: 39120b6b35c8f1c626e2580e770a3206@192.168.0.41:5060
CSeq: 102 OPTIONS
From: “asterisk” sip:0000000000@192.168.0.41;tag=as50cf77ce
To: sip:freephonie.net;tag=00-30195-08c512b3-7bcdb2441
Via: SIP/2.0/UDP 192.168.0.41:5060;received=yy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK76cfdfad
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘39120b6b35c8f1c626e2580e770a3206@192.168.0.41:5060’ Method: OPTIONS

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #2 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060

localhost*CLI>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
– Executing [1@interne:4] BackGround(“SIP/210-00000002”, “main-menu”) in new stack

localhost*CLI>
– <SIP/210-00000002> Playing ‘main-menu.ulaw’ (language ‘fr’)

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
– Executing [1@interne:5] WaitExten(“SIP/210-00000002”, “40”) in new stack

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #3 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060

localhost*CLI>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #4 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060

localhost*CLI>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #5 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060

localhost*CLI>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #6 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060

localhost*CLI>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #7 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060

localhost*CLI>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
<— SIP read from UDP:zz.zz.zzz.zzz:5060 —>

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

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #8 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060

localhost*CLI>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #9 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060

localhost*CLI>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Retransmitting #10 (NAT) to zz.zz.zzz.zzz:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP zz.zz.zzz.zzz:5060;branch=z9hG4bKc30475f1f7d6ad604beb4060e58a999d;received=zz.zz.zzz.zzz;rport=5060
From: “210” sip:210@domaine.fr;tag=4015411108
To: sip:1@domaine.fr;user=phone;tag=as27a17dbb
Call-ID: 4002489461@192_168_1_75
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1@192.168.0.41:5060
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 689142286 689142286 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 16214 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=ptime:20
a=sendrecv

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
[Oct 21 21:27:44] WARNING[10562]: chan_sip.c:3694 retrans_pkt: Retransmission timeout reached on transmission 4002489461@192_168_1_75 for seqno 3 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

localhost*CLI>
[Oct 21 21:27:44] WARNING[10562]: chan_sip.c:3723 retrans_pkt: Hanging up call 4002489461@192_168_1_75 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

localhost*CLI>
== Spawn extension (interne, 1, 5) exited non-zero on ‘SIP/210-00000002’

localhost*CLI>
Scheduling destruction of SIP dialog ‘4002489461@192_168_1_75’ in 32000 ms (Method: INVITE)

localhost*CLI>
set_destination: Parsing sip:210@zz.zz.zzz.zzz:5060 for address/port to send to

localhost*CLI>
set_destination: set destination to zz.zz.zzz.zzz:5060

localhost*CLI>
Reliably Transmitting (NAT) to zz.zz.zzz.zzz:5060:
BYE sip:210@zz.zz.zzz.zzz:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK00369c31;rport
Max-Forwards: 70
From: sip:1@domaine.fr;user=phone;tag=as27a17dbb
To: “210” sip:210@domaine.fr;tag=4015411108
Call-ID: 4002489461@192_168_1_75
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.17.0
Proxy-Authorization: Digest username=“test”, realm=“asterisk”, algorithm=MD5, uri=“sip:domaine.fr”, nonce="", response="8e29234f153bf860347319586b8eebce"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

localhost*CLI>
<— SIP read from UDP:zz.zz.zzz.zzz:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK00369c31;rport=5060;received=yy.yyy.yyy.yyy
From: sip:1@domaine.fr;user=phone;tag=as27a17dbb
To: “210” sip:210@domaine.fr;tag=4015411108
Call-ID: 4002489461@192_168_1_75
CSeq: 102 BYE
Contact: sip:210@zz.zz.zzz.zzz:5060
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

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

localhost*CLI>
— (10 headers 0 lines) —

localhost*CLI>
SIP Response message for INCOMING dialog BYE arrived

localhost*CLI>
Really destroying SIP dialog ‘4002489461@192_168_1_75’ Method: INVITE

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI> sip set debug on
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI> sip set debug onamaflags=default
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI> amaflags=default
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI> amaflags=defaultsip set debug on
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI> sip set debug o
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI> sip set debug off
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhostCLI> sip set debug off
localhostCLI>
SIP Debugging Disabled

localhost*CLI> [/code]

everything seem to work but i can not hear the main-menu text as i know it exist. I don’t hear anything.

but when i call from external nimber and i transfer the call to the siemens gigaset, it work.

do you know why i don’t hear nothing when i call from the siemens gigaset phone ?

thank you.

david55 · October 21, 2012, 8:44pm

Asterisk is sending a local address in its SDP. That suggests that you haven’t set any of the options for Asterisk being behind NAT. nat=yes only deals with the phone being behind NAT, and might not he needed.

You need to identify the local networks, and provide some means for Asterisk to find its own external address.

parisien9999 · October 21, 2012, 8:55pm

[quote=“david55”]Asterisk is sending a local address in its SDP. That suggests that you haven’t set any of the options for Asterisk being behind NAT. nat=yes only deals with the phone being behind NAT, and might not he needed.

You need to identify the local networks, and provide some means for Asterisk to find its own external address.[/quote]

hello and thank you for your answer.

i just added Externip=xx.xxx.xxx.xxx (ip of the asterisk network) on the general section of the sip.conf and made a sip reload and i still have the problem.

what can i do for asterisk to find its own external address ???

(my asterisk network has a permanent ip address)

EDIT :
I just added localnet=192.168.0.0/255.255.255.0
and it is working.

really thank you for your help, i wouldn’t find without it.

david55 · October 21, 2012, 10:14pm

externaddr externhost or stunaddr. (And with localnets, to define the exclusions.)

Is it still giving the wrong address in the SDP?

parisien9999 · October 23, 2012, 11:13am

[quote=“david55”]externaddr externhost or stunaddr. (And with localnets, to define the exclusions.)

Is it still giving the wrong address in the SDP?[/quote]

Hello.

I don’t know as it is now working for sound coming from asterisk to my local phone.

But when i call outside, i don’t hear anything.

I will make a new thread for that !!!

thank you for your help.