No audio

lmergen · October 14, 2005, 1:35pm

Hello,

First of all, I’m new to asterisk. What I’m trying to setup is making an automated outbound call service, possibly more enhancements in the future.

At the moment, I’m trying to just get a test service running… I want to connect an IAX2 client with a voip provider. For the configuration, I’ve added these lines to iax.conf:

[test] type=friend username=test secret=test fromuser=test host=dynamic context=test

These lines to extensions.conf:

[code][test]
exten => 1111,1,Dial(IAX2/test)
exten => 9876,1,Dial(IAX2/gogh)
exten => 1234,1,Dial(SIP/myphonenumber@discount-out)

[discount-out]
exten => _9.,1,Dial(SIP/${EXTEN:1}@discount-out,30,r)[/code]

And these lines to sip.conf:

[code]register => user:pass:user@sip.sipdiscount.com
[discount-out]
type=peer
secret=pass
username=user
host=sip.sipdiscount.com
fromuser=user
nat=yes

[test]
type=friend
username=test
fromuser=test
secret=test
host=dynamic
context=test[/code]

Now, when I log in with for example the DIAX windows client, loggin in goes great… when I dial 1234, my phone rings, I pick up the phone, the DIAX client tells me a conversation has been established but… no sound is coming thru! I have no microphone connected to my computer, but do have a soundcard and well… basically there’s no sound during the conversation.

Any tips on where to look what the problem might be ?

Thanks in advance!

lmergen · October 14, 2005, 1:48pm

In case there is any interrest, here’s the debug log when enabling sip debug on discount-out:

[code] – Accepting AUTHENTICATED call from 81.171.14.12, requested format = 2, actual format = 2
– Executing Dial(“IAX2/test@test/2”, “SIP/0031619620839@discount-out”) in new stack
We’re at 192.168.3.222 port 17352
Answering/Requesting with root capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:0031619620839@sip.sipdiscount.com SIP/2.0
Via: SIP/2.0/UDP 192.168.3.222:5060;branch=z9hG4bK0d5f75fd;rport
From: “diax0.9.15a” sip:ewaldb@192.168.3.222;tag=as0074d1c4
To: sip:0031619620839@sip.sipdiscount.com
Contact: sip:ewaldb@192.168.3.222
Call-ID: 2b9ca49a5beeb53460ac36237e1eae31@192.168.3.222
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 14 Oct 2005 13:40:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 2731 2731 IN IP4 192.168.3.222
s=session
c=IN IP4 192.168.3.222
t=0 0
m=audio 17352 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 213.61.187.147:5060
– Called 0031619620839@discount-out

Sip read:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.3.222:5060;branch=z9hG4bK0d5f75fd;received=217.119.224.127;rport=5060
From: “diax0.9.15a” sip:ewaldb@192.168.3.222;tag=as0074d1c4
To: sip:0031619620839@sip.sipdiscount.com;tag=as529f5b3b
Call-ID: 2b9ca49a5beeb53460ac36237e1eae31@192.168.3.222

CSeq: 103 INVITE
User-Agent: SipProxy
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: sip:0031619620839@213.61.187.147
Content-Length: 0

10 headers, 0 lines

Sip read:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.222:5060;branch=z9hG4bK03ad6cfc;received=217.119.224.127;rport=5060
From: “diax0.9.15a” sip:ewaldb@192.168.3.222;tag=as0074d1c4
To: sip:0031619620839@sip.sipdiscount.com;tag=as53cf7943
Call-ID: 2b9ca49a5beeb53460ac36237e1eae31@192.168.3.222
CSeq: 103 INVITE
User-Agent: SipProxy
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: sip:0031619620839@213.61.187.147
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 18130 18130 IN IP4 213.61.187.147
s=session
c=IN IP4 213.61.187.147
t=0 0
m=audio 51246 RTP/AVP 4 3 0 8 111 5 10 7 18 110 97 101
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

11 headers, 20 lines
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 10
Found RTP audio format 7
Found RTP audio format 18
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 213.61.187.147:51246
Found description format G723
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format DVI4
Found description format L16
Found description format LPC
Found description format G729
Found description format speex
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
– SIP/discount-out-4ebf is making progress passing it to IAX2/test@test/2

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.222:5060;branch=z9hG4bK557d9c88;received=217.119.224.127;rport=5060
From: “diax0.9.15a” sip:ewaldb@192.168.3.222;tag=as5e005afe
To: sip:0031619620839@sip.sipdiscount.com;tag=as1fe28d26
Call-ID: 568cec3a5ec197716751277913188433@192.168.3.222
CSeq: 103 INVITE
User-Agent: SipProxy
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: sip:0031619620839@213.61.187.147
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 18130 18131 IN IP4 213.61.187.147
s=session
c=IN IP4 213.61.187.147
t=0 0
m=audio 53444 RTP/AVP 4 3 0 8 111 5 10 7 18 110 97 101
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

11 headers, 20 lines
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 10
Found RTP audio format 7
Found RTP audio format 18
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 213.61.187.147:53444
Found description format G723
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format DVI4
Found description format L16
Found description format LPC
Found description format G729
Found description format speex
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
list_route: hop: sip:0031619620839@213.61.187.147
set_destination: Parsing sip:0031619620839@213.61.187.147 for address/port to send to
set_destination: set destination to 213.61.187.147, port 5060
Transmitting:
ACK sip:0031619620839@sip.sipdiscount.com SIP/2.0
Via: SIP/2.0/UDP 192.168.3.222:5060;branch=z9hG4bK519dfa68;rport
From: “diax0.9.15a” sip:ewaldb@192.168.3.222;tag=as5e005afe
To: sip:0031619620839@sip.sipdiscount.com;tag=as1fe28d26
Contact: sip:ewaldb@192.168.3.222
Call-ID: 568cec3a5ec197716751277913188433@192.168.3.222
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

(NAT) to 213.61.187.147:5060
– SIP/discount-out-e104 answered IAX2/test@test/4

Sip read:
INVITE sip:ewaldb@192.168.3.222 SIP/2.0
Via: SIP/2.0/UDP 213.61.187.147:5060;branch=z9hG4bK31eea9c8;rport
From: sip:0031619620839@sip.sipdiscount.com;tag=as1fe28d26
To: “diax0.9.15a” sip:ewaldb@192.168.3.222;tag=as5e005afe
Contact: sip:0031619620839@213.61.187.147
Call-ID: 568cec3a5ec197716751277913188433@192.168.3.222
CSeq: 102 INVITE
User-Agent: SipProxy
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 18130 18132 IN IP4 213.61.187.133
s=session
c=IN IP4 213.61.187.133
t=0 0
m=audio 19120 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

11 headers, 10 lines
Using latest request as basis request
Sending to 213.61.187.147 : 5060 (NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 213.61.187.133:19120
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
We’re at 192.168.3.222 port 19942
Answering/Requesting with root capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.61.187.147:5060;branch=z9hG4bK31eea9c8;received=213.61.187.147;rport=5060
From: sip:0031619620839@sip.sipdiscount.com;tag=as1fe28d26
To: “diax0.9.15a” sip:ewaldb@192.168.3.222;tag=as5e005afe
Call-ID: 568cec3a5ec197716751277913188433@192.168.3.222
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:ewaldb@192.168.3.222
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 2731 2733 IN IP4 192.168.3.222
s=session
c=IN IP4 192.168.3.222
t=0 0
m=audio 19942 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

to 213.61.187.147:5060

Sip read:
ACK sip:ewaldb@192.168.3.222 SIP/2.0
Via: SIP/2.0/UDP 213.61.187.147:5060;branch=z9hG4bK7a893672;rport
From: sip:0031619620839@sip.sipdiscount.com;tag=as1fe28d26
To: “diax0.9.15a” sip:ewaldb@192.168.3.222;tag=as5e005afe
Contact: sip:0031619620839@213.61.187.147
Call-ID: 568cec3a5ec197716751277913188433@192.168.3.222
CSeq: 102 ACK
User-Agent: SipProxy
Content-Length: 0

9 headers, 0 lines
– Registered ‘test’ (AUTHENTICATED) at 81.171.14.12:4569[/code]

dufus · October 14, 2005, 3:52pm

No audio generally means that the real time protocols are not passing through your firewall.

Did you setup your firewall rules correctly?

lmergen · October 14, 2005, 6:25pm

[quote=“dufus”]No audio generally means that the real time protocols are not passing through your firewall.

Did you setup your firewall rules correctly?[/quote]

Aha! This could very well be it… I’ve set up the following incoming rules:

voip-info.org/wiki/view/Aste … wall+rules

I figured that would be enough… where can i find information on other ports to forward ?

dufus · October 14, 2005, 7:09pm

The realtime protocol uses UDP ports 10000 through 20000.

You can adjust that using rtp.conf so you don’t have to leave open so many ports.

lmergen · October 17, 2005, 1:02pm

Hello,

Well, as described on that page, my network administrator added the UDP ports:

RTP - the media stream

iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT

So I assume my network setup is sufficient… couldn’t it possibly be some sort of codec problem, or am I thinking wrong there ?

dufus · October 17, 2005, 1:16pm

Since you are passing through a firewall, I’m assuming that there is NAT involved.

SIP and NAT don’t play well together, but can still be made to work.

Check here for more information:

www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions

Also, since you’re using IAX, there’s a different port involved. Port 4569.

Click here for information about IAX trunking.

voip-info.org/wiki-IAX