I’m getting disconnects on outbound numbers that require I enter in multiple DTMF entries. This is the log of when the call disconnects:
[Oct 31 20:17:53] WARNING[1079]: chan_sip.c:2303 retrans_pkt: Maximum retries exceeded on transmission CALL_ID22_00179AACFA5C_T1209263556@192.168.0.197 for seqno 337453400 (Critical Response)
[Oct 31 20:17:53] WARNING[1079]: chan_sip.c:2330 retrans_pkt: Hanging up call CALL_ID22_00179AACFA5C_T1209263556@192.168.0.197 - no reply to our critical packet.
-- Hungup 'Zap/11-1'
== Spawn extension (from-pstn, 18004284322, 2) exited non-zero on 'SIP/adam-0880e000'
The confusing part to me is the SIP debug info. Here is a snippet:
[code]— (10 headers 0 lines) —
Handling message for SAPI/TEI=0/0
Retransmitting #6 (no NAT) to 192.168.0.197:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.197:5060;branch=z9hG4bK_00179AACFA5C_T3A95544A;received=192.168.0.197
From: “Anonymous” sip:adam@xxx.xxx.xxx.xxx:5060;tag=00179AACFA5C_T356113001
To: sip:xxxxxxxxx@xxx.xxx.xxx.xxx:5060;tag=as476fc785
Call-ID: CALL_ID22_00179AACFA5C_T1209263556@192.168.0.197
CSeq: 337453400 INVITE
User-Agent: Asterisk PBX SVN-trunk-r84726M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:1xxxxxxxxxx@192.168.0.1
Content-Type: application/sdp
Content-Length: 280
v=0
o=root 145910542 145910543 IN IP4 192.168.0.1
s=Asterisk PBX SVN-trunk-r84726M
c=IN IP4 192.168.0.1
t=0 0
m=audio 12514 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
call*CLI>
<— SIP read from 192.168.0.197:5060 —>
ACK sip:18004284322@192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.197:5060;branch=z9hG4bK_00179AACFA5C_T476F556D
From: “Anonymous” sip:adam@xxx.xxx.xxx.xxx:5060;tag=00179AACFA5C_T356113001
To: sip:xxxxxxxxx@xxx.xxx.xxx.xxx:5060;tag=as476fc785
Call-ID: CALL_ID22_00179AACFA5C_T1209263556@192.168.0.197
CSeq: 337453400 ACK
User-Agent: DPH-14001.01.08
Contact: sip:adam@192.168.0.197:5060
Max-Forwards: 70
Content-Length: 0[/code]
I’m assuming the "Critical Packet in question is the invite sent, and the confusing part is that the phone replies with an ACK, but * appears to ignore the ACK and continues sending invites.
We use two types of phones: the dlink DPH-140S and the SNOM 360. The call works perfectly with the SNOM. Every single time it fails with the dlink. Does anyone have insight into this behavior?