No sound when I make the call to another SIP

ajaysaini · November 21, 2013, 2:52pm

I have setup the asterisk. I’m not able to hear two users sound each other. Both are on different network and Asterisk is on public ip.

I have disabled the firewall but still no sound between two sip users.

Please help what can be issue…

[general]
context=internal
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
alwaysauthreject=yes
canreinvite=no
nat=yes
session-timers=refuse
localnet=0.0.0.0/0.0.0.0
;203.100.77.54/255.255.255.0
;externip=MY_PUBLIC_IP -----------I commented here…is it causing issue…

[7001]
type=friend
host=dynamic
secret=7001
context=internal
mailbox=7001@vm-demo

[7002]
type=friend
host=dynamic
secret=7002
context=internal
mailbox=7002@vm-demo

[2002]
type=friend
host=dynamic
secret=2002
context=internal
;nat=yes
mailbox=2002@vm-demo

Hope to hear from you all soon!!

david55 · November 21, 2013, 3:19pm

How are the separate networks connected to the public internet?

You will need to provide traces.

I believe canreinvite is no longer recognized in current versions of Asterisk and has been deprecated for many years. Please check you configuration options against current documentation, not just cook books.

ajaysaini · November 21, 2013, 3:25pm

I have zoiper on my home wifi and my friend is on his home (different network) and asterisk is our office. Asterisk server is on the public ip.

When I and my friend tries to call each other ring appear but no sound after ring.

Somethings causing this i could not trace it.

Hope to hear from you soon!!

david55 · November 21, 2013, 5:02pm

Most likely there is a problem with your friends’ NAT router settings.

However, to be sure to disable direct media on current versions you must use the new name for canreinvite, and nat=yes is deprecated, so you should read the documentation and make the specific settings that you need.

Please remember to always identify the version of Asterisk.

ajaysaini · November 21, 2013, 5:10pm

I and My friend are using the MTS Data card. There is no hardware router between them. We are able to register the sip users from the zoiper but no sound after accept the call. One side sound appear but not now.

I’m using the Asterisk 11.6.0

Please suggest what i write instead of canreinvite and nat.

Hope to hear from you soon!!

navaismo · November 21, 2013, 11:38pm

As david said you need to provide logs/trace so enable the sip debug do a call and paste the complete output of the call.

ajaysaini · November 22, 2013, 6:19am

Even One side ring sound not there.

Logs—

Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Sent RTP packet to Got RTP packet from Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Sent RTP packet to == Spawn 203.100.77.51:60860 (type 00, seq 001156, ts 3798011841, len 000160)
192.168.1.55:60860 (type 00, seq 062078, ts 184160, len 000160)
203.100.77.51:60860 (type 00, seq 001157, ts 3798012001, len 000160)
192.168.1.55:60860 (type 00, seq 062079, ts 184320, len 000160)
203.100.77.51:60860 (type 00, seq 001158, ts 3798012161, len 000160)
192.168.1.55:60860 (type 00, seq 062080, ts 184480, len 000160)
203.100.77.51:60860 (type 00, seq 001159, ts 3798012321, len 000160)
192.168.1.55:60860 (type 00, seq 062081, ts 184640, len 000160)
203.100.77.51:60860 (type 00, seq 001160, ts 3798012481, len 000160)
192.168.1.55:60860 (type 00, seq 062082, ts 184800, len 000160)
203.100.77.51:60860 (type 00, seq 001161, ts 3798012641, len 000160)
192.168.1.55:60860 (type 00, seq 062083, ts 184960, len 000160)
203.100.77.51:60860 (type 00, seq 001162, ts 3798012801, len 000160)
192.168.1.55:60860 (type 00, seq 062084, ts 185120, len 000160)
203.100.77.51:60860 (type 00, seq 001163, ts 3798012961, len 000160)
192.168.1.55:60860 (type 00, seq 062085, ts 185280, len 000160)
203.100.77.51:60860 (type 00, seq 001164, ts 3798013121, len 000160)
192.168.1.55:60860 (type 00, seq 062086, ts 185440, len 000160)
203.100.77.51:60860 (type 00, seq 001165, ts 3798013281, len 000160)
192.168.1.55:60860 (type 00, seq 062087, ts 185600, len 000160)
203.100.77.51:60860 (type 00, seq 001166, ts 3798013441, len 000160)
192.168.1.55:60860 (type 00, seq 062088, ts 185760, len 000160)
203.100.77.51:60860 (type 00, seq 001167, ts 3798013601, len 000160)
192.168.1.55:60860 (type 00, seq 062089, ts 185920, len 000160)
203.100.77.51:60860 (type 00, seq 001168, ts 3798013761, len 000160)
192.168.1.55:60860 (type 00, seq 062090, ts 186080, len 000160)
203.100.77.51:60860 (type 00, seq 001169, ts 3798013921, len 000160)
192.168.1.55:60860 (type 00, seq 062091, ts 186240, len 000160)
203.100.77.51:60860 (type 00, seq 001170, ts 3798014081, len 000160)
192.168.1.55:60860 (type 00, seq 062092, ts 186400, len 000160)
203.100.77.51:60860 (type 00, seq 001171, ts 3798014241, len 000160)
192.168.1.55:60860 (type 00, seq 062093, ts 186560, len 000160)
203.100.77.51:60860 (type 00, seq 001172, ts 3798014401, len 000160)
192.168.1.55:60860 (type 00, seq 062094, ts 186720, len 000160)
203.100.77.51:60860 (type 00, seq 001173, ts 3798014561, len 000160)
192.168.1.55:60860 (type 00, seq 062095, ts 186880, len 000160)
203.100.77.51:60860 (type 00, seq 001174, ts 3798014721, len 000160)
192.168.1.55:60860 (type 00, seq 062096, ts 187040, len 000160)
203.100.77.51:60860 (type 00, seq 001175, ts 3798014881, len 000160)
192.168.1.55:60860 (type 00, seq 062097, ts 187200, len 000160)
203.100.77.51:60860 (type 00, seq 001176, ts 3798015041, len 000160)
192.168.1.55:60860 (type 00, seq 062098, ts 187360, len 000160)
203.100.77.51:60860 (type 00, seq 001177, ts 3798015201, len 000160)
192.168.1.55:60860 (type 00, seq 062099, ts 187520, len 000160)
203.100.77.51:60860 (type 00, seq 001178, ts 3798015361, len 000160)
192.168.1.55:60860 (type 00, seq 062100, ts 187680, len 000160)
192.168.1.55:60860 (type 00, seq 062101, ts 187840, len 000160)
203.100.77.51:60860 (type 00, seq 001179, ts 3798015521, len 000160)
203.100.77.51:60860 (type 00, seq 001180, ts 3798015681, len 000160)
192.168.1.55:60860 (type 00, seq 062102, ts 188000, len 000160)
203.100.77.51:60860 (type 00, seq 001181, ts 3798015841, len 000160)
192.168.1.55:60860 (type 00, seq 062103, ts 188160, len 000160)
203.100.77.51:60860 (type 00, seq 001182, ts 3798016001, len 000160)
192.168.1.55:60860 (type 00, seq 062104, ts 188320, len 000160)
203.100.77.51:60860 (type 00, seq 001183, ts 3798016161, len 000160)
192.168.1.55:60860 (type 00, seq 062105, ts 188480, len 000160)
203.100.77.51:60860 (type 00, seq 001184, ts 3798016321, len 000160)
192.168.1.55:60860 (type 00, seq 062106, ts 188640, len 000160)
192.168.1.55:60860 (type 00, seq 062107, ts 188800, len 000160)
extension (internal, 7001, 2) exited non-zero on ‘SIP/7002-00000006’

navaismo · November 22, 2013, 6:58am

Now the sip debug without rtp debug.

ajaysaini · November 22, 2013, 8:16am

—Logs–

One side ring not there also

Call-ID: MTUxODg0OTVkM2IzMDM0OGFiOGRjODM3NjkwMjM4ODk
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.5 stamp 71236
Content-Length: 212

v=0
o=- 13029581675084070 2 IN IP4 192.168.1.55
s=X-Lite 4 release 4.5.5 stamp 71236
c=IN IP4 192.168.1.55
t=0 0
m=audio 50584 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (12 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.55:50584
set_destination: Parsing sip:7002@203.100.77.51:6336 for address/port to send to
set_destination: set destination to 203.100.77.51:6336
Transmitting (NAT) to 203.100.77.51:6336:
ACK sip:7002@203.100.77.51:6336 SIP/2.0
Via: SIP/2.0/UDP 203.100.77.54:5060;branch=z9hG4bK66d50343;rport
Max-Forwards: 70
From: sip:7001@203.100.77.54;tag=as3126341d
To: "Ajay Saini"sip:7002@203.100.77.54;tag=59176942
Contact: sip:7001@203.100.77.54:5060
Call-ID: MTUxODg0OTVkM2IzMDM0OGFiOGRjODM3NjkwMjM4ODk
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.6.0
Content-Length: 0

<— SIP read from UDP:203.100.77.51:1031 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.100.77.54:5060;branch=z9hG4bK5461af3c;rport=5060
Contact: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP
To: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP;tag=d6141840
From: "Ajay Saini"sip:7002@203.100.77.54;tag=as764f6673
Call-ID: 41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.14091
Allow-Events: presence, kpml
Content-Length: 186

v=0
o=Z 0 3 IN IP4 192.168.1.121
s=Z
c=IN IP4 192.168.1.121
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.121:10000
set_destination: Parsing sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP for address/port to send to
set_destination: set destination to 192.168.1.121:5060
Transmitting (NAT) to 203.100.77.51:1031:
ACK sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 203.100.77.54:5060;branch=z9hG4bK09b8ee8e;rport
Max-Forwards: 70
From: “Ajay Saini” sip:7002@203.100.77.54;tag=as764f6673
To: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP;tag=d6141840
Contact: sip:7002@203.100.77.54:5060
Call-ID: 41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.6.0
Content-Length: 0

<— SIP read from UDP:203.100.77.51:6336 —>
BYE sip:7001@203.100.77.54:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.55:6336;branch=z9hG4bK-d8754z-0b5a020226bdfd38-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:7002@203.100.77.51:6336
To: sip:7001@203.100.77.54;tag=as3126341d
From: "Ajay Saini"sip:7002@203.100.77.54;tag=59176942
Call-ID: MTUxODg0OTVkM2IzMDM0OGFiOGRjODM3NjkwMjM4ODk
CSeq: 3 BYE
User-Agent: X-Lite release 4.5.5 stamp 71236
Authorization: Digest username=“7002”,realm=“asterisk”,nonce=“50ef600e”,uri=“sip:7001@203.100.77.54:5060”,response=“434ff21e8c0c05e2526acdddd367f9ac”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 203.100.77.51:6336 (NAT)
Scheduling destruction of SIP dialog ‘MTUxODg0OTVkM2IzMDM0OGFiOGRjODM3NjkwMjM4ODk’ in 32000 ms (Method: BYE)

<— Transmitting (NAT) to 203.100.77.51:6336 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.55:6336;branch=z9hG4bK-d8754z-0b5a020226bdfd38-1—d8754z-;received=203.100.77.51;rport=6336
From: "Ajay Saini"sip:7002@203.100.77.54;tag=59176942
To: sip:7001@203.100.77.54;tag=as3126341d
Call-ID: MTUxODg0OTVkM2IzMDM0OGFiOGRjODM3NjkwMjM4ODk
CSeq: 3 BYE
Server: Asterisk PBX 11.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
set_destination: Parsing sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP for address/port to send to
set_destination: set destination to 192.168.1.121:5060
Audio is at 18892
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.100.77.51:1031:
INVITE sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 203.100.77.54:5060;branch=z9hG4bK6486930c;rport
Max-Forwards: 70
From: “Ajay Saini” sip:7002@203.100.77.54;tag=as764f6673
To: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP;tag=d6141840
Contact: sip:7002@203.100.77.54:5060
Call-ID: 41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 280576517 280576519 IN IP4 203.100.77.54
s=Asterisk PBX 11.6.0
c=IN IP4 203.100.77.54
t=0 0
m=audio 18892 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Scheduling destruction of SIP dialog ‘41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060’ in 32000 ms (Method: INVITE)
== Spawn extension (internal, 7001, 2) exited non-zero on ‘SIP/7002-00000023’

<— SIP read from UDP:203.100.77.51:1031 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.100.77.54:5060;branch=z9hG4bK6486930c;rport=5060
Contact: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP
To: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP;tag=d6141840
From: "Ajay Saini"sip:7002@203.100.77.54;tag=as764f6673
Call-ID: 41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060
CSeq: 104 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.14091
Allow-Events: presence, kpml
Content-Length: 186

v=0
o=Z 0 4 IN IP4 192.168.1.121
s=Z
c=IN IP4 192.168.1.121
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.121:10000
set_destination: Parsing sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP for address/port to send to
set_destination: set destination to 192.168.1.121:5060
Transmitting (NAT) to 203.100.77.51:1031:
ACK sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 203.100.77.54:5060;branch=z9hG4bK585f2df0;rport
Max-Forwards: 70
From: “Ajay Saini” sip:7002@203.100.77.54;tag=as764f6673
To: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP;tag=d6141840
Contact: sip:7002@203.100.77.54:5060
Call-ID: 41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 11.6.0
Content-Length: 0

set_destination: Parsing sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP for address/port to send to
set_destination: set destination to 192.168.1.121:5060
Reliably Transmitting (NAT) to 203.100.77.51:1031:
BYE sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 203.100.77.54:5060;branch=z9hG4bK6e904fcc;rport
Max-Forwards: 70
From: “Ajay Saini” sip:7002@203.100.77.54;tag=as764f6673
To: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP;tag=d6141840
Call-ID: 41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060
CSeq: 105 BYE
User-Agent: Asterisk PBX 11.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Scheduling destruction of SIP dialog ‘41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:203.100.77.51:1031 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.100.77.54:5060;branch=z9hG4bK6e904fcc;rport=5060
Contact: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP
To: sip:7001@192.168.1.121:5060;rinstance=5f28380e09cab788;transport=UDP;tag=d6141840
From: "Ajay Saini"sip:7002@203.100.77.54;tag=as764f6673
Call-ID: 41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060
CSeq: 105 BYE
User-Agent: Zoiper rev.14091
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘41768a100e0b2f0b4d28d3165beb7651@203.100.77.54:5060’ Method: INVITE

<— SIP read from UDP:203.100.77.51:6336 —>

<------------->
Really destroying SIP dialog ‘MDJhNzE4Y2ZiYmFkNTgzMTFjZTkxM2MwYTE0NzFhNjg’ Method: REGISTER
server2*CLI>

david55 · November 22, 2013, 11:30am

You need to provide the sip trace for the complete call.

However, I do see an external bridge in there. I think that is because you are still ignoring my advice to read the documentation on the configuration options you are using, and possibly because the default setting has changed.

I am fairly sure that canreinvite has changed from deprecated (which it has been for the best part of five years, to un-recognized. It may well be that directmedia=yes is now the default.

ajaysaini · November 22, 2013, 11:46am

I used the directmedia.

I dial the 100 for the IVR but no sound there.

context=internal
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
alwaysauthreject=yes
nat=yes
session-timers=refuse
directmedia=yes
localnet=192.168.1.0/255.255.255.0

—Logs–

-- <SIP/7002-00000005> Playing '/tmp/test1.slin' (language 'en')

== Spawn extension (internal, 100, 7) exited non-zero on ‘SIP/7002-00000005’
== Using SIP RTP CoS mark 5
– Executing [100@internal:1] Answer(“SIP/7002-00000006”, “”) in new stack
> 0x7f196c01f740 – Probation passed - setting RTP source address to 203.100.77.51:54804
– Executing [100@internal:2] System(“SIP/7002-00000006”, “rm -rf /tmp/test.wav”) in new stack
– Executing [100@internal:3] System(“SIP/7002-00000006”, “/opt/swift/bin/swift -o /tmp/test.wav -p audio/sampling-rate=8000,audio/channels=1 “Welcome to Infoshore Software Private Limited.””) in new stack
– Executing [100@internal:4] Playback(“SIP/7002-00000006”, “/tmp/test”) in new stack
– <SIP/7002-00000006> Playing ‘/tmp/test.slin’ (language ‘en’)
– Executing [100@internal:5] System(“SIP/7002-00000006”, “rm -rf /tmp/test1.wav”) in new stack
– Executing [100@internal:6] System(“SIP/7002-00000006”, “/opt/swift/bin/swift -o /tmp/test1.wav -p audio/sampling-rate=8000,audio/channels=1 “Press 1 to tranfer the call to technical department. Press 2 to transfer the call to sales department. Press 3 to transfer the call to HR Department””) in new stack
– Executing [100@internal:7] BackGround(“SIP/7002-00000006”, “/tmp/test1”) in new stack
– <SIP/7002-00000006> Playing ‘/tmp/test1.slin’ (language ‘en’)
server2*CLI>

david55 · November 22, 2013, 12:06pm

If you need directmedia, it needs to be set to no.

I need all four SDP payloads to see if you have a NAT issue.

ajaysaini · November 22, 2013, 12:43pm

What you mean of here “four SDP payloads to see if you have a NAT issue.”

Please let me know how can i get those

david55 · November 22, 2013, 1:02pm

When you connect a call through Asterisk, there will be an INVITE, 200 OK and ACK for the incoming leg and also one of each for the outgoing leg. Attached to both 200 OKs, and the outgoing leg INVITE, there will be SDP (Session Description Protocol), that’s three. On the incoming leg, there will be SDP attached to either the INVITE or the ACK. That’s a total of four.

The SDP tells the various parties where to send their RTP packets. If you have a NAT problem, that is resulting on no audio, or one way audio, one of those RTP destination addresses may look wrong.

One big mistake people make when asked for traces is trying to take them from their terminal emulator. History buffers are not normally long enough. You need to enable Asterisk’s full log (logger.conf) and use that.

ajaysaini · November 22, 2013, 1:05pm

Please don’t take the call between two users right now. First priority No voice on IVR once dial. I attached the logs in my above messages.

Please help me for this. This is on priority.