Daniel - Asterisk
2015-Aug-14 12:54 UTC
[asterisk-users] chan_sip.c: Retransmission timeout reached on transmission
Hello friends:
I am facing cutoffs randomly when negotiating calls.
The PBX dials the destination, the provider (softswitch) receives the
request *[1]* and sudenly the PBX hangs up the call* [2]* while the
provider is still dialing it, as a consequence the remote peer receives a
ghost call. Along the atempt I could see six times a messages regarding NAT
isuues *[3]*
I hope anyone can give me an idea to solve this issue. Softswitch is using
an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with
Asterisk 1.8.11.0
Thanks in advance
Elder D. Arohuanca
Lima - Peru
*[1]*
[Aug 12 19:21:05] VERBOSE[17115] app_dial.c: -- Called
SIP/SIP-PROVIDER/965034648
*[2]*
[Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout reached
on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for seqno
103 (Critical Request) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 8832ms with no response
[Aug 12 19:21:14] WARNING[3477] chan_sip.c: Hanging up call
0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* - no reply to our critical
packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Aug 12 19:21:14] VERBOSE[17115] app_dial.c: == Everyone is
busy/congested at this time (1:0/0/1)
[Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing
[s at macro-dialout-trunk:20] NoOp("SIP/143-000001d8", "Dial
failed for some
reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack
[Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing
[s at macro-dialout-trunk:21] Goto("SIP/143-000001d8",
"s-CHANUNAVAIL,1") in
new stack
*[3]*
Retransmitting #3 (no NAT) to PROVIDER-IP:5060:
INVITE sip:dialed_number at PROVIDER-IP SIP/2.0
Via: SIP/2.0/UDP PBX-PUBLIC_IP:5060;branch=z9hG4bK06c2c701
Max-Forwards: 70
From: "PBX-DID" <sip:outbound-trunk at
PROVIDER-IP>;tag=as27ef83ae
To: <sip:dialed_number at PROVIDER-IP>
Contact: <sip:outbound-trunk at PBX-PUBLIC_IP:5060>
Call-ID: 6b9ad82d4673fdab722f9e53411a767d at PROVIDER-IP
CSeq: 103 INVITE
User-Agent: FPBX-2.8.1(1.8.11.0)
Proxy-Authorization: Digest username="outbound-trunk",
realm="SoftSwitch",
algorithm=MD5, uri="sip:dialed_number at PROVIDER-IP",
nonce="d1b5806808a0888112190722408572932332",
response="40c94f3c04e87e3382c7652d1f012dc9"
Date: Thu, 13 Aug 2015 00:56:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Remote-Party-ID: "PBX-DID" <sip:PBX-DID at
PROVIDER-IP>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 502733417 502733418 IN IP4 PBX-PUBLIC_IP
s=Asterisk PBX 1.8.11.0
c=IN IP4 PBX-PUBLIC_IP
t=0 0
m=audio 13042 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
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Sam Basan
2015-Aug-14 13:33 UTC
[asterisk-users] chan_sip.c: Retransmission timeout reached on transmission
Hi, It's looks like you are having NAT problem. Packets from the provider fail reaching your box. ???? ?????? ???? ?????? 14 ????' 2015 15:56,? "Daniel - Asterisk" <earohuanca at gmail.com> ???:> Hello friends: > > I am facing cutoffs randomly when negotiating calls. > > The PBX dials the destination, the provider (softswitch) receives the > request *[1]* and sudenly the PBX hangs up the call* [2]* while the > provider is still dialing it, as a consequence the remote peer receives a > ghost call. Along the atempt I could see six times a messages regarding NAT > isuues *[3]* > > I hope anyone can give me an idea to solve this issue. Softswitch is using > an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with > Asterisk 1.8.11.0 > > Thanks in advance > > Elder D. Arohuanca > Lima - Peru > > > *[1]* > [Aug 12 19:21:05] VERBOSE[17115] app_dial.c: -- Called > SIP/SIP-PROVIDER/965034648 > > > *[2]* > [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout reached > on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for seqno > 103 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > Packet timed out after 8832ms with no response > [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Hanging up call > 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* - no reply to our critical > packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > ). > [Aug 12 19:21:14] VERBOSE[17115] app_dial.c: == Everyone is > busy/congested at this time (1:0/0/1) > [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing > [s at macro-dialout-trunk:20] NoOp("SIP/143-000001d8", "Dial failed for some > reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack > [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing > [s at macro-dialout-trunk:21] Goto("SIP/143-000001d8", "s-CHANUNAVAIL,1") in > new stack > > *[3]* > Retransmitting #3 (no NAT) to PROVIDER-IP:5060: > INVITE sip:dialed_number at PROVIDER-IP SIP/2.0 > Via: SIP/2.0/UDP PBX-PUBLIC_IP:5060;branch=z9hG4bK06c2c701 > Max-Forwards: 70 > From: "PBX-DID" <sip:outbound-trunk at PROVIDER-IP>;tag=as27ef83ae > To: <sip:dialed_number at PROVIDER-IP> > Contact: <sip:outbound-trunk at PBX-PUBLIC_IP:5060> > Call-ID: 6b9ad82d4673fdab722f9e53411a767d at PROVIDER-IP > CSeq: 103 INVITE > User-Agent: FPBX-2.8.1(1.8.11.0) > Proxy-Authorization: Digest username="outbound-trunk", realm="SoftSwitch", > algorithm=MD5, uri="sip:dialed_number at PROVIDER-IP", > nonce="d1b5806808a0888112190722408572932332", > response="40c94f3c04e87e3382c7652d1f012dc9" > Date: Thu, 13 Aug 2015 00:56:40 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Remote-Party-ID: "PBX-DID" <sip:PBX-DID at PROVIDER-IP > >;party=calling;privacy=off;screen=no > Content-Type: application/sdp > Content-Length: 260 > > v=0 > o=root 502733417 502733418 IN IP4 PBX-PUBLIC_IP > s=Asterisk PBX 1.8.11.0 > c=IN IP4 PBX-PUBLIC_IP > t=0 0 > m=audio 13042 RTP/AVP 18 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150814/1a5d2564/attachment.html>
Daniel - Asterisk
2015-Aug-14 19:11 UTC
[asterisk-users] chan_sip.c: Retransmission timeout reached on transmission
Hello Sam, Do you have any recommendation to overcome these NAT issues? On 8/14/15, Sam Basan <sbasan at bluebe.net> wrote:> Hi, > > It's looks like you are having NAT problem. > Packets from the provider fail reaching your box. > > ???? ?????? ???? > ?????? 14 ????' 2015 15:56,? "Daniel - Asterisk" <earohuanca at gmail.com> > ???: > >> Hello friends: >> >> I am facing cutoffs randomly when negotiating calls. >> >> The PBX dials the destination, the provider (softswitch) receives the >> request *[1]* and sudenly the PBX hangs up the call* [2]* while the >> provider is still dialing it, as a consequence the remote peer receives a >> ghost call. Along the atempt I could see six times a messages regarding >> NAT >> isuues *[3]* >> >> I hope anyone can give me an idea to solve this issue. Softswitch is >> using >> an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with >> Asterisk 1.8.11.0 >> >> Thanks in advance >> >> Elder D. Arohuanca >> Lima - Peru >> >> >> *[1]* >> [Aug 12 19:21:05] VERBOSE[17115] app_dial.c: -- Called >> SIP/SIP-PROVIDER/965034648 >> >> >> *[2]* >> [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout >> reached >> on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for seqno >> 103 (Critical Request) -- See >> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions >> Packet timed out after 8832ms with no response >> [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Hanging up call >> 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* - no reply to our critical >> packet (see >> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions >> ). >> [Aug 12 19:21:14] VERBOSE[17115] app_dial.c: == Everyone is >> busy/congested at this time (1:0/0/1) >> [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing >> [s at macro-dialout-trunk:20] NoOp("SIP/143-000001d8", "Dial failed for some >> reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack >> [Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing >> [s at macro-dialout-trunk:21] Goto("SIP/143-000001d8", "s-CHANUNAVAIL,1") in >> new stack >> >> *[3]* >> Retransmitting #3 (no NAT) to PROVIDER-IP:5060: >> INVITE sip:dialed_number at PROVIDER-IP SIP/2.0 >> Via: SIP/2.0/UDP PBX-PUBLIC_IP:5060;branch=z9hG4bK06c2c701 >> Max-Forwards: 70 >> From: "PBX-DID" <sip:outbound-trunk at PROVIDER-IP>;tag=as27ef83ae >> To: <sip:dialed_number at PROVIDER-IP> >> Contact: <sip:outbound-trunk at PBX-PUBLIC_IP:5060> >> Call-ID: 6b9ad82d4673fdab722f9e53411a767d at PROVIDER-IP >> CSeq: 103 INVITE >> User-Agent: FPBX-2.8.1(1.8.11.0) >> Proxy-Authorization: Digest username="outbound-trunk", >> realm="SoftSwitch", >> algorithm=MD5, uri="sip:dialed_number at PROVIDER-IP", >> nonce="d1b5806808a0888112190722408572932332", >> response="40c94f3c04e87e3382c7652d1f012dc9" >> Date: Thu, 13 Aug 2015 00:56:40 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH >> Supported: replaces, timer >> Remote-Party-ID: "PBX-DID" <sip:PBX-DID at PROVIDER-IP >> >;party=calling;privacy=off;screen=no >> Content-Type: application/sdp >> Content-Length: 260 >> >> v=0 >> o=root 502733417 502733418 IN IP4 PBX-PUBLIC_IP >> s=Asterisk PBX 1.8.11.0 >> c=IN IP4 PBX-PUBLIC_IP >> t=0 0 >> m=audio 13042 RTP/AVP 18 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >