[asterisk-users] Problems with Analog - SIP phone conversations

[Timothy Smith]

Hi,

Could someone please help me with this?

I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on
an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS,
where an incoming line is plugged and also analog phone plugged to the
FXS port. Am faced with the problems below.

- For conversations between analog phone and sip phone, Analog phone
can't here the SIP user but Sip user hears.
- Calling the PSTN from the Analog phone,  still the analog phone
can't hear but the PSTN user hears him saying "hello." repeatedly.

Any help appreciated?


I attempted a SIP debug and this is a sample out out:

<-- SIP read from 192.168.209.1:48099:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.209.253:5060;branch=z9hG4bK661b7c81;rport=5060;received=
192.168.209.253
From: "asterisk" <sip:asterisk at 192.168.209.253<sip%3Aasterisk at 192.168.209.253>
>;tag=as7b41af2a
To: "Ananth" <sip:Ananth2 at 192.168.102.10 <sip%3AAnanth2 at 192.168.102.10>
>;tag=2bb81ff3969
Call-ID: 7758606e479584bf2c20a80d792841f5 at 192.168.209.253
CSeq: 102 NOTIFY
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)

-----
--- (11 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.102.10:49166
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:Ananth2 at 192.168.102.10 <sip%3AAnanth2 at 192.168.102.10>>
set_destination: Parsing
<sip:Ananth2 at 192.168.102.10<sip%3AAnanth2 at 192.168.102.10>>
for address/port
to send to
set_destination: set destination to 192.168.102.10, port 5060
Transmitting (NAT) to 192.168.209.1:48099:
ACK sip:Ananth2 at 192.168.102.10 <sip%3AAnanth2 at 192.168.102.10> SIP/2.0
Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK4162221c;rport
From: "analog-phone"
<sip:asterisk at 192.168.209.253<sip%3Aasterisk at 192.168.209.253>
>;tag=as2b73e0bc
To: <sip:Ananth2 at 192.168.102.10 <sip%3AAnanth2 at 192.168.102.10>
>;tag=2ae01fe36af
Contact: <sip:asterisk at 192.168.209.253 <sip%3Aasterisk at 192.168.209.253>>
Call-ID: 09727d336de2434a2a1043436322c03b at 192.168.209.253
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Regards,
Tim
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