Seemingly unable to call callcenters

xanmanza · October 23, 2015, 7:53am

Hey guys.
So on my setup, I can make normal outbound calls to landlines or cellphone numbers, but I am unable to make calls to what seems to be Call centers.
So numbers I for example cannot get through to:
0861 300 900 (Axxess ADSL Support)
0860 067 327 (NSFAS)
081180 (Telkom support)
These above numbers are South African numbers.

If I connect Zoiper directly to my SIP provider I can call the numbers just fine.
I have no outbound prefixing, the number is sent through as is.
My outbound trunk config:

username=**username** type=friend secret=**secret** insecure=very host=vphone.co.za dtmfmode=rfc2833 disallow=all allow=ulaw&alaw&gsm canredirect=no canreinvite=no qualify=yes
The SIP error I get just reads “Destination not answering”

Regards

david55 · October 23, 2015, 8:50am

An ancient way of implementing free numbers was to not answer the call (which started the metering). The whole call was early media. Originally this was mainly used for recorded announcements, which only require one way audio.

For Asterisk, you must call Progress(), before DIal(), for early media to be passed through. However, I’m not sure that it is passed through in both directions, or just towards the caller.

The other possibility is that the the call is still ringing whilst in the call centre queue, which can be good practice for chargeable calls, and something is timing out.

Asterisk will only time out if you include an explicit time out in the Dial application call.

xanmanza · October 23, 2015, 3:24pm

Hi David
Thanks for your prompt reply.
I am still struggling to solve this problem, the only output I am getting that seems to be pointing to something is the following:

[quote]<— SIP read from UDP:10.0.0.150:59449 —>
PUBLISH sip:25@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—b7679a9041fe0c3a
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: "Xan"sip:25@10.0.0.155;transport=UDP
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=4d55dc32
Call-ID: S2LS45UvnQ7Wc3LohR3xpg…
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 260

<?xml version="1.0" encoding="UTF-8"?>

open On the phone

<------------->
— (16 headers 3 lines) —
Sending to 10.0.0.150:59449 (NAT)

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—b7679a9041fe0c3a;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=4d55dc32
To: "Xan"sip:25@10.0.0.155;transport=UDP;tag=as6f68c4ed
Call-ID: S2LS45UvnQ7Wc3LohR3xpg…
CSeq: 1 PUBLISH
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘S2LS45UvnQ7Wc3LohR3xpg…’ Method: PUBLISH

<— SIP read from UDP:10.0.0.150:59449 —>
SUBSCRIBE sip:25@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—ad7d51d612009026
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: "Xan"sip:25@10.0.0.155;transport=UDP
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=b4606e08
Call-ID: _GuIV-87Q8Dz8UdCE6T70A…
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Sending to 10.0.0.150:59449 (NAT)
Creating new subscription
Sending to 10.0.0.150:59449 (NAT)
list_route: hop: sip:25@105.210.167.170:59449;transport=UDP
Found peer ‘25’ for ‘25’ from 10.0.0.150:59449

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—ad7d51d612009026;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=b4606e08
To: "Xan"sip:25@10.0.0.155;transport=UDP;tag=as68300a28
Call-ID: _GuIV-87Q8Dz8UdCE6T70A…
CSeq: 1 SUBSCRIBE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="744a7aa6"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘_GuIV-87Q8Dz8UdCE6T70A…’ in 6400 ms (Method: SUBSCRIBE)

<— SIP read from UDP:10.0.0.150:59449 —>
SUBSCRIBE sip:25@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—9659183a32ae3e8c
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: "Xan"sip:25@10.0.0.155;transport=UDP
From: “Xan"sip:25@10.0.0.155;transport=UDP;tag=b4606e08
Call-ID: _GuIV-87Q8Dz8UdCE6T70A…
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username=“25”,realm=“asterisk”,nonce=“744a7aa6”,uri="sip:25@10.0.0.155;transport=UDP”,response=“d4d7595b91422deec142a46c8ede527d”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to 10.0.0.150:59449 (NAT)
Found peer ‘25’ for ‘25’ from 10.0.0.150:59449

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—9659183a32ae3e8c;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=b4606e08
To: "Xan"sip:25@10.0.0.155;transport=UDP;tag=as68300a28
Call-ID: _GuIV-87Q8Dz8UdCE6T70A…
CSeq: 2 SUBSCRIBE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘_GuIV-87Q8Dz8UdCE6T70A…’ Method: SUBSCRIBE

<— SIP read from UDP:10.0.0.150:59449 —>
INVITE sip:0861300900@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—2d554ebe156f3ea8
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: sip:0861300900@10.0.0.155;transport=UDP
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=6545bd3a
Call-ID: 4wNovAPcHQOysfNJF6z_ag…
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 216

v=0
o=Z 0 0 IN IP4 105.210.167.170
s=Z
c=IN IP4 105.210.167.170
t=0 0
m=audio 8000 RTP/AVP 8 0 3 97 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
— (14 headers 11 lines) —
Sending to 10.0.0.150:59449 (NAT)
Sending to 10.0.0.150:59449 (NAT)
Using INVITE request as basis request - 4wNovAPcHQOysfNJF6z_ag…
Found peer ‘25’ for ‘25’ from 10.0.0.150:59449

<— Reliably Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—2d554ebe156f3ea8;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=6545bd3a
To: sip:0861300900@10.0.0.155;transport=UDP;tag=as3255c22c
Call-ID: 4wNovAPcHQOysfNJF6z_ag…
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="26555898"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘4wNovAPcHQOysfNJF6z_ag…’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.0.0.150:59449 —>
ACK sip:0861300900@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—2d554ebe156f3ea8
Max-Forwards: 70
To: sip:0861300900@10.0.0.155;transport=UDP;tag=as3255c22c
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=6545bd3a
Call-ID: 4wNovAPcHQOysfNJF6z_ag…
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.0.0.150:59449 —>
INVITE sip:0861300900@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—883552051a086404
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: sip:0861300900@10.0.0.155;transport=UDP
From: “Xan"sip:25@10.0.0.155;transport=UDP;tag=6545bd3a
Call-ID: 4wNovAPcHQOysfNJF6z_ag…
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username=“25”,realm=“asterisk”,nonce=“26555898”,uri="sip:0861300900@10.0.0.155;transport=UDP”,response=“4aa6177d0aac6383899f978a59a6d124”,algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 216

v=0
o=Z 0 0 IN IP4 105.210.167.170
s=Z
c=IN IP4 105.210.167.170
t=0 0
m=audio 8000 RTP/AVP 8 0 3 97 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
— (15 headers 11 lines) —
Sending to 10.0.0.150:59449 (NAT)
Using INVITE request as basis request - 4wNovAPcHQOysfNJF6z_ag…
Found peer ‘25’ for ‘25’ from 10.0.0.150:59449
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw|alaw|ilbc)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 105.210.167.170:8000
Looking for 0861300900 in from-internal (domain 10.0.0.155)
list_route: hop: sip:25@105.210.167.170:59449;transport=UDP

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—883552051a086404;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=6545bd3a
To: sip:0861300900@10.0.0.155;transport=UDP
Call-ID: 4wNovAPcHQOysfNJF6z_ag…
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:0861300900@10.0.0.155:5060
Content-Length: 0

<------------>
We think we can do text
Audio is at 12842
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100028 (speex32) to SDP
Adding codec 100030 (opus) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 196.3.177.10:5060:
INVITE sip:0861300900@vphone.co.za SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK6ed23f71
Max-Forwards: 70
From: “CallQ” sip:Unknown@10.0.0.155;tag=as52c1d523
To: sip:0861300900@vphone.co.za
Contact: sip:Unknown@10.0.0.155:5060
Call-ID: 0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Date: Fri, 23 Oct 2015 15:01:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1430

v=0
o=root 1723808280 1723808280 IN IP4 10.0.0.155
s=Asterisk PBX 11.17.1
c=IN IP4 10.0.0.155
t=0 0
m=audio 12842 RTP/AVP 8 3 0 4 112 5 7 18 110 97 111 9 102 115 116 117 96 108 109 113 10 118 119 107 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:108 SILK/12000
a=fmtp:108 maxaveragebitrate=12000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:109 SILK/16000
a=fmtp:109 maxaveragebitrate=20000
a=fmtp:109 usedtx=0
a=fmtp:109 useinbandfec=1
a=rtpmap:113 SILK/24000
a=fmtp:113 maxaveragebitrate=30000
a=fmtp:113 usedtx=0
a=fmtp:113 useinbandfec=1
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:107 opus/48000/2
a=maxptime:60
a=fmtp:107 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK6ed23f71;rport=5060
From: “CallQ” sip:Unknown@10.0.0.155;tag=as52c1d523
To: sip:0861300900@vphone.co.za
Call-ID: 0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060
CSeq: 102 INVITE

<------------->
— (6 headers 0 lines) —

<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK6ed23f71;rport=5060
From: “CallQ” sip:Unknown@10.0.0.155;tag=as52c1d523
To: sip:0861300900@vphone.co.za;tag=SDuc6bf99-
Call-ID: 0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm=“10.17.180.204”,nonce="896bba33b93b198be7eaaea356570d247bb9"
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Transmitting (no NAT) to 196.3.177.10:5060:
ACK sip:0861300900@vphone.co.za SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK6ed23f71
Max-Forwards: 70
From: “CallQ” sip:Unknown@10.0.0.155;tag=as52c1d523
To: sip:0861300900@vphone.co.za;tag=SDuc6bf99-
Contact: sip:Unknown@10.0.0.155:5060
Call-ID: 0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

We think we can do text
Audio is at 12842
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100005 (g726aal2) to SDP
Adding codec 100006 (adpcm) to SDP
Adding codec 100007 (lpc10) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100013 (siren7) to SDP
Adding codec 100014 (siren14) to SDP
Adding codec 100015 (g719) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100017 (testlaw) to SDP
Adding codec 100018 (silk8) to SDP
Adding codec 100018 (silk12) to SDP
Adding codec 100018 (silk16) to SDP
Adding codec 100018 (silk24) to SDP
Adding codec 100019 (slin) to SDP
Adding codec 100020 (slin12) to SDP
Adding codec 100021 (slin16) to SDP
Adding codec 100022 (slin24) to SDP
Adding codec 100023 (slin32) to SDP
Adding codec 100024 (slin44) to SDP
Adding codec 100025 (slin48) to SDP
Adding codec 100026 (slin96) to SDP
Adding codec 100027 (slin192) to SDP
Adding codec 100028 (speex32) to SDP
Adding codec 100030 (opus) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 196.3.177.10:5060:
INVITE sip:0861300900@vphone.co.za SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK3a7e0cc3
Max-Forwards: 70
From: “CallQ” sip:Unknown@10.0.0.155;tag=as52c1d523
To: sip:0861300900@vphone.co.za
Contact: sip:Unknown@10.0.0.155:5060
Call-ID: 0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username=“27878089890”, realm=“10.17.180.204”, algorithm=MD5, uri="sip:0861300900@vphone.co.za", nonce=“896bba33b93b198be7eaaea356570d247bb9”, response="6a1562dc06a7a71102b67d3e3fd89662"
Date: Fri, 23 Oct 2015 15:01:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1430

v=0
o=root 1723808280 1723808281 IN IP4 10.0.0.155
s=Asterisk PBX 11.17.1
c=IN IP4 10.0.0.155
t=0 0
m=audio 12842 RTP/AVP 8 3 0 4 112 5 7 18 110 97 111 9 102 115 116 117 96 108 109 113 10 118 119 107 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 usedtx=0
a=fmtp:96 useinbandfec=1
a=rtpmap:108 SILK/12000
a=fmtp:108 maxaveragebitrate=12000
a=fmtp:108 usedtx=0
a=fmtp:108 useinbandfec=1
a=rtpmap:109 SILK/16000
a=fmtp:109 maxaveragebitrate=20000
a=fmtp:109 usedtx=0
a=fmtp:109 useinbandfec=1
a=rtpmap:113 SILK/24000
a=fmtp:113 maxaveragebitrate=30000
a=fmtp:113 usedtx=0
a=fmtp:113 useinbandfec=1
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:107 opus/48000/2
a=maxptime:60
a=fmtp:107 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK3a7e0cc3;rport=5060
From: “CallQ” sip:Unknown@10.0.0.155;tag=as52c1d523
To: sip:0861300900@vphone.co.za
Call-ID: 0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060
CSeq: 103 INVITE

<------------->
— (6 headers 0 lines) —
Reliably Transmitting (no NAT) to 196.3.177.10:5060:
OPTIONS sip:vphone.co.za SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK16929c60
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.0.0.155;tag=as08bef8b2
To: sip:vphone.co.za
Contact: sip:Unknown@10.0.0.155:5060
Call-ID: 3cc807f3287217a20cd189796c0489e8@10.0.0.155:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Fri, 23 Oct 2015 15:01:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK16929c60;rport=5060
From: “Unknown” sip:Unknown@10.0.0.155;tag=as08bef8b2
To: sip:vphone.co.za;tag=SDne8pc99-d110e763
Call-ID: 3cc807f3287217a20cd189796c0489e8@10.0.0.155:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘3cc807f3287217a20cd189796c0489e8@10.0.0.155:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 10.0.0.150:59449:
OPTIONS sip:25@105.237.127.217:59449;rinstance=8cfb6471b64d11f0;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK48aa9ef1;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.0.0.155;tag=as18bbd0e6
To: sip:25@105.237.127.217:59449;rinstance=8cfb6471b64d11f0;transport=UDP
Contact: sip:Unknown@10.0.0.155:5060
Call-ID: 6bc2c02a0a5e1c747842bf8b21d75713@10.0.0.155:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Fri, 23 Oct 2015 15:01:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0[/quote]

Please tell me if I need to supply more info or debug output, pulling my hair out in frustration here!

david55 · October 23, 2015, 3:59pm

The trace doesn’t become useful until the final occurrence of

Reliably Transmitting (no NAT) to 196.3.177.10:5060:
INVITE sip:0861300900@vphone.co.za SIP/2.0

and cuts off before anything useful. The SUBSCRIBE and OPTIONS transactions add no useful information. You need to take the trace through until there is an incoming response with an INVITE CSEQ and a status code >= 200.

It is better to use the log files, as they contain timestamps, even though you may need to strip them of escape sequences. Also, please use code blocks for traces, as it makes scrolling through postings a lot easier.

xanmanza · October 23, 2015, 5:04pm

Sorry and thank you, I will keep the practices in mind.
Here is the output following the statement you spoke of, until the line drops with “503” on Zoiper and a spoken message “Destination not answering”:

[code]<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK16929c60;rport=5060
From: “Unknown” sip:Unknown@10.0.0.155;tag=as08bef8b2
To: sip:vphone.co.za;tag=SDne8pc99-d110e763
Call-ID: 3cc807f3287217a20cd189796c0489e8@10.0.0.155:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘3cc807f3287217a20cd189796c0489e8@10.0.0.155:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 10.0.0.150:59449:
OPTIONS sip:25@105.237.127.217:59449;rinstance=8cfb6471b64d11f0;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK48aa9ef1;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.0.0.155;tag=as18bbd0e6
To: sip:25@105.237.127.217:59449;rinstance=8cfb6471b64d11f0;transport=UDP
Contact: sip:Unknown@10.0.0.155:5060
Call-ID: 6bc2c02a0a5e1c747842bf8b21d75713@10.0.0.155:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Fri, 23 Oct 2015 15:01:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.0.0.150:59449 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK48aa9ef1;rport=5060
Contact: sip:10.0.0.150:59449
To: sip:25@105.237.127.217:59449;rinstance=8cfb6471b64d11f0;transport=UDP;tag=04356e3e
From: “Unknown” sip:Unknown@10.0.0.155;tag=as18bbd0e6
Call-ID: 6bc2c02a0a5e1c747842bf8b21d75713@10.0.0.155:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘6bc2c02a0a5e1c747842bf8b21d75713@10.0.0.155:5060’ Method: OPTIONS

<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK3a7e0cc3;rport=5060
From: “CallQ” sip:Unknown@10.0.0.155;tag=as52c1d523
To: sip:0861300900@vphone.co.za;tag=SDuc6bf99-ifevhxd7krtdnkpj.i
Call-ID: 0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060
CSeq: 103 INVITE
Server: Sippy
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (no NAT) to 196.3.177.10:5060:
ACK sip:0861300900@vphone.co.za SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK3a7e0cc3
Max-Forwards: 70
From: “CallQ” sip:Unknown@10.0.0.155;tag=as52c1d523
To: sip:0861300900@vphone.co.za;tag=SDuc6bf99-ifevhxd7krtdnkpj.i
Contact: sip:Unknown@10.0.0.155:5060
Call-ID: 0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Scheduling destruction of SIP dialog ‘0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060’ in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog ‘0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060’ in 6400 ms (Method: INVITE)
Audio is at 18684
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—883552051a086404;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=6545bd3a
To: sip:0861300900@10.0.0.155;transport=UDP;tag=as5f30d3c1
Call-ID: 4wNovAPcHQOysfNJF6z_ag…
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:0861300900@10.0.0.155:5060
Content-Type: application/sdp
Require: timer
Content-Length: 277

v=0
o=root 866353856 866353856 IN IP4 10.0.0.155
s=Asterisk PBX 11.17.1
c=IN IP4 10.0.0.155
t=0 0
m=audio 18684 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<— Reliably Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—883552051a086404;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=6545bd3a
To: sip:0861300900@10.0.0.155;transport=UDP;tag=as5f30d3c1
Call-ID: 4wNovAPcHQOysfNJF6z_ag…
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------>
[2015-10-23 13:02:28] WARNING[3039][C-00000006]: channel.c:4862 ast_prod: Prodding channel ‘SIP/25-0000000c’ failed

<— SIP read from UDP:10.0.0.150:59449 —>
ACK sip:0861300900@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—883552051a086404
Max-Forwards: 70
To: sip:0861300900@10.0.0.155;transport=UDP;tag=as5f30d3c1
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=6545bd3a
Call-ID: 4wNovAPcHQOysfNJF6z_ag…
CSeq: 2 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘4wNovAPcHQOysfNJF6z_ag…’ Method: ACK

<— SIP read from UDP:10.0.0.150:59449 —>
PUBLISH sip:25@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—6ba9de196052b055
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: "Xan"sip:25@10.0.0.155;transport=UDP
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=453a5e33
Call-ID: qQ2_yhCz1KND9ZwYvpqLZw…
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 254

<?xml version="1.0" encoding="UTF-8"?>

open Online

<------------->
— (16 headers 3 lines) —
Sending to 10.0.0.150:59449 (NAT)

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—6ba9de196052b055;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=453a5e33
To: "Xan"sip:25@10.0.0.155;transport=UDP;tag=as647f0545
Call-ID: qQ2_yhCz1KND9ZwYvpqLZw…
CSeq: 1 PUBLISH
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘qQ2_yhCz1KND9ZwYvpqLZw…’ Method: PUBLISH

<— SIP read from UDP:10.0.0.150:59449 —>
SUBSCRIBE sip:25@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—3367cec3ba1f5fc9
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: "Xan"sip:25@10.0.0.155;transport=UDP
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=47715132
Call-ID: MMArsbtN49imcSaVbBYS9A…
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Sending to 10.0.0.150:59449 (NAT)
Creating new subscription
Sending to 10.0.0.150:59449 (NAT)
list_route: hop: sip:25@105.210.167.170:59449;transport=UDP
Found peer ‘25’ for ‘25’ from 10.0.0.150:59449

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—3367cec3ba1f5fc9;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=47715132
To: "Xan"sip:25@10.0.0.155;transport=UDP;tag=as278fe6d2
Call-ID: MMArsbtN49imcSaVbBYS9A…
CSeq: 1 SUBSCRIBE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="585125b7"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘MMArsbtN49imcSaVbBYS9A…’ in 6400 ms (Method: SUBSCRIBE)

<— SIP read from UDP:10.0.0.150:59449 —>
SUBSCRIBE sip:25@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—8069b917c519adbc
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: "Xan"sip:25@10.0.0.155;transport=UDP
From: “Xan"sip:25@10.0.0.155;transport=UDP;tag=47715132
Call-ID: MMArsbtN49imcSaVbBYS9A…
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username=“25”,realm=“asterisk”,nonce=“585125b7”,uri="sip:25@10.0.0.155;transport=UDP”,response=“d5b52908783160dd855f08397ae3951e”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to 10.0.0.150:59449 (NAT)
Found peer ‘25’ for ‘25’ from 10.0.0.150:59449

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—8069b917c519adbc;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=47715132
To: "Xan"sip:25@10.0.0.155;transport=UDP;tag=as278fe6d2
Call-ID: MMArsbtN49imcSaVbBYS9A…
CSeq: 2 SUBSCRIBE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘MMArsbtN49imcSaVbBYS9A…’ Method: SUBSCRIBE
Really destroying SIP dialog ‘0463c2540613b22b3890ce6d6779f0b3@10.0.0.155:5060’ Method: INVITE
localhost*CLI> sip set debug off
SIP Debugging Disabled[/code]

david55 · October 23, 2015, 5:35pm

There is a response to a cancel, but there is no cancel… You are still missing the middle of the trace.

xanmanza · October 23, 2015, 6:22pm

Hi David
When you say “Cancel” do you mean the “487 Request terminated” response?
I made sure after your reply and I think the debug I gave was correct. Just to make double sure, I ran the debug and the call again. Here is my output from the first “103 INVITE” response I receive from my provider.

[code]<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK2cfbcaff;rport=5060
From: “CallQ” sip:Unknown@10.0.0.155;tag=as1955c77e
To: sip:0861300900@vphone.co.za
Call-ID: 111f65a438c9441908982a976c5dc971@10.0.0.155:5060
CSeq: 103 INVITE

<------------->
— (6 headers 0 lines) —
[2015-10-23 16:10:27] NOTICE[2585]: chan_sip.c:15175 sip_reregister: – Re-registration for 27878089890@vphone.co.za
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 196.3.177.10:5060:
REGISTER sip:vphone.co.za SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK209cd112
Max-Forwards: 70
From: sip:27878089890@vphone.co.za;tag=as2e09fb09
To: sip:27878089890@vphone.co.za
Call-ID: 4dbe9b101141c25a10f72b982ca9da8d@127.0.0.1
CSeq: 297 REGISTER
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username=“27878089890”, realm=“vphone.co.za”, algorithm=MD5, uri=“sip:vphone.co.za”, nonce=“1445623722:f71a896a11047a24ebc004cd638b8b77”, response="c423283cba5e747fc95ac98018a80bc4"
Expires: 120
Contact: sip:s@10.0.0.155:5060
Content-Length: 0

<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK209cd112;rport=5060
From: sip:27878089890@vphone.co.za;tag=as2e09fb09
To: sip:27878089890@vphone.co.za;tag=SDgk4jb99-cf1a7d4b
Call-ID: 4dbe9b101141c25a10f72b982ca9da8d@127.0.0.1
CSeq: 297 REGISTER
WWW-Authenticate: Digest nonce=“1445623827:319ab58a36fe64139893d2c0aced3cea”,algorithm=MD5,realm="vphone.co.za"
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Responding to challenge, registration to domain/host name vphone.co.za
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 196.3.177.10:5060:
REGISTER sip:vphone.co.za SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK4e421e5c
Max-Forwards: 70
From: sip:27878089890@vphone.co.za;tag=as2e09fb09
To: sip:27878089890@vphone.co.za
Call-ID: 4dbe9b101141c25a10f72b982ca9da8d@127.0.0.1
CSeq: 298 REGISTER
User-Agent: FPBX-2.11.0(11.17.1)
Authorization: Digest username=“27878089890”, realm=“vphone.co.za”, algorithm=MD5, uri=“sip:vphone.co.za”, nonce=“1445623827:319ab58a36fe64139893d2c0aced3cea”, response="d07c4ad8ae49068dcbf470021e15ab85"
Expires: 120
Contact: sip:s@10.0.0.155:5060
Content-Length: 0

<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK4e421e5c;rport=5060
From: sip:27878089890@vphone.co.za;tag=as2e09fb09
To: sip:27878089890@vphone.co.za;tag=SDgk4jb99-a40f0012
Call-ID: 4dbe9b101141c25a10f72b982ca9da8d@127.0.0.1
CSeq: 298 REGISTER
Contact: sip:s@10.0.0.155:5060;expires=120
Contact: sip:27878089890@105.237.127.217:59449;rinstance=926d66cb5e25f41f;transport=UDP;expires=2954
Date: Fri, 23 Oct 2015 18:10:27 GMT
Content-Length: 0

<------------->
— (10 headers 0 lines) —
[2015-10-23 16:10:27] NOTICE[2585]: chan_sip.c:23737 handle_response_register: Outbound Registration: Expiry for vphone.co.za is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog ‘4dbe9b101141c25a10f72b982ca9da8d@127.0.0.1’ Method: REGISTER

<— SIP read from UDP:196.3.177.10:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.0.155:5060;received=105.210.167.170;branch=z9hG4bK2cfbcaff;rport=5060
From: “CallQ” sip:Unknown@10.0.0.155;tag=as1955c77e
To: sip:0861300900@vphone.co.za;tag=SDd2jpf99-6c4rk4xtfo3pfdyx.i
Call-ID: 111f65a438c9441908982a976c5dc971@10.0.0.155:5060
CSeq: 103 INVITE
Server: Sippy
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (no NAT) to 196.3.177.10:5060:
ACK sip:0861300900@vphone.co.za SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK2cfbcaff
Max-Forwards: 70
From: “CallQ” sip:Unknown@10.0.0.155;tag=as1955c77e
To: sip:0861300900@vphone.co.za;tag=SDd2jpf99-6c4rk4xtfo3pfdyx.i
Contact: sip:Unknown@10.0.0.155:5060
Call-ID: 111f65a438c9441908982a976c5dc971@10.0.0.155:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.17.1)
Content-Length: 0

Scheduling destruction of SIP dialog ‘111f65a438c9441908982a976c5dc971@10.0.0.155:5060’ in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog ‘111f65a438c9441908982a976c5dc971@10.0.0.155:5060’ in 6400 ms (Method: INVITE)
Audio is at 18394
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—8873a7390dbb44fa;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=3f510304
To: sip:0861300900@10.0.0.155;transport=UDP;tag=as7434459c
Call-ID: WXOriETny91FIeAW5abaMA…
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:0861300900@10.0.0.155:5060
Content-Type: application/sdp
Require: timer
Content-Length: 279

v=0
o=root 1006832802 1006832802 IN IP4 10.0.0.155
s=Asterisk PBX 11.17.1
c=IN IP4 10.0.0.155
t=0 0
m=audio 18394 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<— Reliably Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—8873a7390dbb44fa;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=3f510304
To: sip:0861300900@10.0.0.155;transport=UDP;tag=as7434459c
Call-ID: WXOriETny91FIeAW5abaMA…
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------>
[2015-10-23 16:10:51] WARNING[3838][C-00000007]: channel.c:4862 ast_prod: Prodding channel ‘SIP/25-0000000e’ failed

<— SIP read from UDP:10.0.0.150:59449 —>
ACK sip:0861300900@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—8873a7390dbb44fa
Max-Forwards: 70
To: sip:0861300900@10.0.0.155;transport=UDP;tag=as7434459c
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=3f510304
Call-ID: WXOriETny91FIeAW5abaMA…
CSeq: 2 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘WXOriETny91FIeAW5abaMA…’ Method: ACK

<— SIP read from UDP:10.0.0.150:59449 —>
PUBLISH sip:25@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—42807cb2eb0bacd8
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: "Xan"sip:25@10.0.0.155;transport=UDP
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=b22f022d
Call-ID: ZNcdpoCLsiW3AFUj_jBwMQ…
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence
Allow-Events: presence, kpml
Content-Length: 254

<?xml version="1.0" encoding="UTF-8"?>

open Online

<------------->
— (16 headers 3 lines) —
Sending to 10.0.0.150:59449 (NAT)

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—42807cb2eb0bacd8;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=b22f022d
To: "Xan"sip:25@10.0.0.155;transport=UDP;tag=as7ad11ff8
Call-ID: ZNcdpoCLsiW3AFUj_jBwMQ…
CSeq: 1 PUBLISH
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘ZNcdpoCLsiW3AFUj_jBwMQ…’ Method: PUBLISH

<— SIP read from UDP:10.0.0.150:59449 —>
SUBSCRIBE sip:25@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—e3df37cf0c6fcb75
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: "Xan"sip:25@10.0.0.155;transport=UDP
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=47457338
Call-ID: m-YgLGN8ydW0eWIZPqUNnQ…
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Sending to 10.0.0.150:59449 (NAT)
Creating new subscription
Sending to 10.0.0.150:59449 (NAT)
list_route: hop: sip:25@105.210.167.170:59449;transport=UDP
Found peer ‘25’ for ‘25’ from 10.0.0.150:59449

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—e3df37cf0c6fcb75;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=47457338
To: "Xan"sip:25@10.0.0.155;transport=UDP;tag=as1affa822
Call-ID: m-YgLGN8ydW0eWIZPqUNnQ…
CSeq: 1 SUBSCRIBE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3b4ecc69"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘m-YgLGN8ydW0eWIZPqUNnQ…’ in 6400 ms (Method: SUBSCRIBE)

<— SIP read from UDP:10.0.0.150:59449 —>
SUBSCRIBE sip:25@10.0.0.155;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—c92bdc2228aa5ff9
Max-Forwards: 70
Contact: sip:25@105.210.167.170:59449;transport=UDP
To: "Xan"sip:25@10.0.0.155;transport=UDP
From: “Xan"sip:25@10.0.0.155;transport=UDP;tag=47457338
Call-ID: m-YgLGN8ydW0eWIZPqUNnQ…
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username=“25”,realm=“asterisk”,nonce=“3b4ecc69”,uri="sip:25@10.0.0.155;transport=UDP”,response=“c14b8b54b545c99ee47d2db103427fea”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to 10.0.0.150:59449 (NAT)
Found peer ‘25’ for ‘25’ from 10.0.0.150:59449

<— Transmitting (NAT) to 10.0.0.150:59449 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 105.210.167.170:59449;branch=z9hG4bK-524287-1—c92bdc2228aa5ff9;received=10.0.0.150;rport=59449
From: "Xan"sip:25@10.0.0.155;transport=UDP;tag=47457338
To: "Xan"sip:25@10.0.0.155;transport=UDP;tag=as1affa822
Call-ID: m-YgLGN8ydW0eWIZPqUNnQ…
CSeq: 2 SUBSCRIBE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘m-YgLGN8ydW0eWIZPqUNnQ…’ Method: SUBSCRIBE
Reliably Transmitting (NAT) to 10.0.0.150:59449:
OPTIONS sip:25@105.237.127.217:59449;rinstance=8cfb6471b64d11f0;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK6100589b;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.0.0.155;tag=as5a3d2bfb
To: sip:25@105.237.127.217:59449;rinstance=8cfb6471b64d11f0;transport=UDP
Contact: sip:Unknown@10.0.0.155:5060
Call-ID: 7ba31a0519d8a7b878590039041ee7fb@10.0.0.155:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.17.1)
Date: Fri, 23 Oct 2015 18:10:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.0.0.150:59449 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK6100589b;rport=5060
Contact: sip:10.0.0.150:59449
To: sip:25@105.237.127.217:59449;rinstance=8cfb6471b64d11f0;transport=UDP;tag=7c7d666a
From: “Unknown” sip:Unknown@10.0.0.155;tag=as5a3d2bfb
Call-ID: 7ba31a0519d8a7b878590039041ee7fb@10.0.0.155:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘7ba31a0519d8a7b878590039041ee7fb@10.0.0.155:5060’ Method: OPTIONS
Really destroying SIP dialog ‘111f65a438c9441908982a976c5dc971@10.0.0.155:5060’ Method: INVITE
localhostCLI> sip set debug off
SIP Debugging Disabled
localhostCLI>[/code]

david55 · October 23, 2015, 9:18pm

487 is an indirect response to a CANCEL. There should be CANCEL, then in some order, OK for the CANCEL, and 487 for the INVITE.

When the CANCEL is sent is when your dialplan chose to give up, presumably because you specified a timeout.

What the 487 shows is that the call was never answered, but you are missing the 183 that I would expect if it selected early media.

I’d just try adding Progress before Dial and seeing if you get at least one way audio.

xanmanza · October 28, 2015, 7:42pm

Hi
So, much to my frustration, I figured out the problem was that I was setting the CallerIDs to the same numbers, which gave an error on their side.
Thanks for the help

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