Re: [Asterisk-Users] X-Lite to Zap, no Voice on other phone!

Etienne Pretorius Mon, 04 Apr 2005 09:55:03 -0700

Etienne Pretorius wrote:

Wilson Pickett wrote:

The problem is - and i was wandering if anyone knows the solution - is
that When I dial from my windows machine,
to an external phone line through Zap, then the receiving party does not
hear my voice - but when the receiving party

Do you have Transmit Silence=YES on the X-Lite? It's under audio settings.
What is the NAT situation?
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No, I just checked it - for good measure though.

No NAT or Firewall, iptables is flushed and policy of ACCEPT. Windows Firewall Disabled.

SIP Proxy (*) set to no NAT'ing for SIP clients.

Kind Regards
Etienne

Technical Support
Kingsley Technologies

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I have disabled all codec's except for gsm. (Problem still persists).
The problem is only when call is made from SIP (X-Lite) to land line (POTS)
then there is no voice output on the receiving side. If I use the standard phone,
then no problem is apparent. I am going to try a IP phone later... and see what happens.

Find Attached the "sip debug ip:192.168.5.71" - * server
also the debug of the "sip debug ip:192.168.5.39" - myself calling.
scenario 1: To cell phone (Voice works 100%)
scenario 2: To land line (Voice does not work at all)

If there is anything else that I should supply, please let me know.

I am grateful for any help - and thank you Wilson Pickett for the reply.
o No Firewall, (Iptables on * server clean and Accept policy on all)
o Windows Firewall on client machine disabled.
o X-Lite not sending silent, Transmit Silent = No.

    -- Executing Dial("SIP/Reception-a39c", "Zap/g2/0836851650") in new stack
    -- Called g2/0836851650
We're at 192.168.5.71 port 12192
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:[EMAIL PROTECTED]>;tag=2918742336
To: <sip:[EMAIL PROTECTED]>;tag=as692cd138
Call-ID: [EMAIL PROTECTED]
CSeq: 1371 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 213


v=0p*CLI>
o=root 1994 1994 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 12192 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 
192.168.5.39:5060;rport;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:[EMAIL PROTECTED]>;tag=2918742336
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 1371 CANCEL
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 192.168.5.39 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:[EMAIL PROTECTED]>;tag=2918742336
To: <sip:[EMAIL PROTECTED]>;tag=as692cd138
Call-ID: [EMAIL PROTECTED]
CSeq: 1371 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0

voip*CLI>
---
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:[EMAIL PROTECTED]>;tag=2918742336
To: <sip:[EMAIL PROTECTED]>;tag=as692cd138
Call-ID: [EMAIL PROTECTED]
CSeq: 1371 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


---
    -- Hungup 'Zap/3-1'
  == Spawn extension (sip, 0836851650, 1) exited non-zero on 
'SIP/Reception-a39c'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 
192.168.5.39:5060;rport;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:[EMAIL PROTECTED]>;tag=2918742336
To: <sip:[EMAIL PROTECTED]>;tag=as692cd138
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 1371 ACK
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 
192.168.5.39:5060;rport;branch=z9hG4bK312DCC013E1E4DBA95E6C1104E7C527D
From: none <sip:[EMAIL PROTECTED]>;tag=2918742336
To: <sip:[EMAIL PROTECTED]>;tag=as692cd138
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 1371 ACK
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---

    -- Executing Dial("SIP/Reception-4949", "Zap/g2/0217619930") in new stack
    -- Called g2/0217619930
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
We're at 192.168.5.71 port 13200
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bKF7B88DB1B9E242C0AC8BBD5F5ABAB233
From: none <sip:[EMAIL PROTECTED]>;tag=488136580
To: <sip:[EMAIL PROTECTED]>;tag=as4153afdd
Call-ID: [EMAIL PROTECTED]
CSeq: 10930 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 213

v=0p*CLI>
o=root 1999 1999 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 13200 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

    -- Executing Dial("SIP/Reception-5cf7", "Zap/g2/0828000495") in new stack
    -- Called g2/0828000495
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
We're at 192.168.5.71 port 15180
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bK6F44B5DB845640A18B3AFAE9073643DE
From: none <sip:[EMAIL PROTECTED]>;tag=2938417032
To: <sip:[EMAIL PROTECTED]>;tag=as7a0a3237
Call-ID: [EMAIL PROTECTED]
CSeq: 53237 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 1963 1963 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 15180 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Zap/3-1 answered SIP/Reception-5cf7
We're at 192.168.5.71 port 15180
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bK6F44B5DB845640A18B3AFAE9073643DE
From: none <sip:[EMAIL PROTECTED]>;tag=2938417032
To: <sip:[EMAIL PROTECTED]>;tag=as7a0a3237
Call-ID: [EMAIL PROTECTED]
CSeq: 53237 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 1963 1964 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 15180 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 
192.168.5.39:5060;rport;branch=z9hG4bKF7CCFB94A67F42669B3C8683544CDCCA
From: none <sip:[EMAIL PROTECTED]>;tag=2938417032
To: <sip:[EMAIL PROTECTED]>;tag=as7a0a3237
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 53237 ACK
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
    -- Hungup 'Zap/3-1'
  == Spawn extension (sip, 0828000495, 1) exited non-zero on 
'SIP/Reception-5cf7'
set_destination: Parsing <sip:[EMAIL PROTECTED]:5060> for address/port to send 
to
set_destination: set destination to 192.168.5.39, port 5060
Reliably Transmitting (no NAT) to 192.168.5.39:5060:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.71:5060;branch=z9hG4bK1d37c962;rport
From: <sip:[EMAIL PROTECTED]>;tag=as7a0a3237
To: none <sip:[EMAIL PROTECTED]>;tag=2938417032
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0


---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.5.71:5060;branch=z9hG4bK1d37c962;rport
From: <sip:[EMAIL PROTECTED]>;tag=as7a0a3237
To: none <sip:[EMAIL PROTECTED]>;tag=2938417032
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
Server: X-Lite release 1103m
Content-Length: 0


--- (9 headers 0 lines)---
Response message is BYE
Destroying call '[EMAIL PROTECTED]'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:

--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 
192.168.5.39:5060;rport;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:[EMAIL PROTECTED]>;tag=1058526819
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 23215 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 195

v=0
o=reception 1251812 1251843 IN IP4 192.168.5.39
s=X-Lite
c=IN IP4 192.168.5.39
t=0 0
m=audio 8000 RTP/AVP 3 101
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (11 headers 9 lines)---
Using latest request as basis request
Sending to 192.168.5.39 : 5060 (non-NAT)
Found user 'Reception'
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.5.39:8000
Found description format gsm
Found description format telephone-event
Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing), 
combined - 0x2 (gsm)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 
(g723)
Looking for 0217619930 in sip
list_route: hop: <sip:[EMAIL PROTECTED]:5060>
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:[EMAIL PROTECTED]>;tag=1058526819
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 23215 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


---
    -- Executing Dial("SIP/Reception-33d9", "Zap/g2/0217619930") in new stack
    -- Called g2/0217619930
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
We're at 192.168.5.71 port 11850
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:[EMAIL PROTECTED]>;tag=1058526819
To: <sip:[EMAIL PROTECTED]>;tag=as6093bccf
Call-ID: [EMAIL PROTECTED]
CSeq: 23215 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 1958 1958 IN IP4 192.168.5.71
s=session
c=IN IP4 192.168.5.71
t=0 0
m=audio 11850 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:


--- (0 headers 0 lines) Nat keepalive ---
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 
192.168.5.39:5060;rport;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:[EMAIL PROTECTED]>;tag=1058526819
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 23215 CANCEL
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 192.168.5.39 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:[EMAIL PROTECTED]>;tag=1058526819
To: <sip:[EMAIL PROTECTED]>;tag=as6093bccf
Call-ID: [EMAIL PROTECTED]
CSeq: 23215 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
voip*CLI>

---
Transmitting (no NAT) to 192.168.5.39:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.5.39:5060;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:[EMAIL PROTECTED]>;tag=1058526819
To: <sip:[EMAIL PROTECTED]>;tag=as6093bccf
Call-ID: [EMAIL PROTECTED]
CSeq: 23215 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0


---
    -- Hungup 'Zap/3-1'
  == Spawn extension (sip, 0217619930, 1) exited non-zero on 
'SIP/Reception-33d9'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 
192.168.5.39:5060;rport;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:[EMAIL PROTECTED]>;tag=1058526819
To: <sip:[EMAIL PROTECTED]>;tag=as6093bccf
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 23215 ACK
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
voip*CLI>
<-- SIP read from 192.168.5.39:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 
192.168.5.39:5060;rport;branch=z9hG4bK500FA7CB11E54933A783E8EB8A14ED00
From: none <sip:[EMAIL PROTECTED]>;tag=1058526819
To: <sip:[EMAIL PROTECTED]>;tag=as6093bccf
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 23215 ACK
Max-Forwards: 70
Content-Length: 0

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Re: [Asterisk-Users] X-Lite to Zap, no Voice on other phone!

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