Re: [ASTPP] Re: Vicidial AUTODIAL mode not working with ASTPP

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Gilbert Arias

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Jul 17, 2017, 11:52:13 AM7/17/17

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if your calls are working fine dialing from another source, try to pay attention to the vicidial asterisk cli/log and try to find the error there.

On Mon, Jul 17, 2017 at 11:42 AM, Farock <farouk....@gmail.com> wrote:

I forgot to tell that in auto dialing mode no RTP traffic, is sent by ASTPP, one single RTP packet is sent by Asterisk.
i uploaded some logs if you can help.

Le lundi 17 juillet 2017 12:34:35 UTC+1, Farock a écrit :
Hello,

I'm new to ASTPP, i'm wondering if anyone have/had an issue with connecting ASTPP to a Vicidial Server.
My scenario is as follow : Vicidial server(asterisk) --> ASTPP ---> Carrier.

I'm able to make a manual call through vicidial server, but when it comes to auto-dialing, call hangup just after being answered and so not served to agent.
The Vicidial server is behind NAT so nat=yes is set in it, but still cant get calls to work, i don't think its a codec problem i tried with both G729 and G711.

Can someone point me to the right direction, i've been searching for 4 days now.

Thank you community.

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CSD REGISTRATION AGENT

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Jul 17, 2017, 1:12:47 PM7/17/17

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can you paste your asterisk log when the vicidial is failing

On Mon, Jul 17, 2017 at 5:57 PM, Farouk Houmaïdi <farouk....@gmail.com> wrote:

Hello, thank's fo your reply.
Manual calls works fine through an extension from the vicidial/asterisk server, the problem is only when i'm launching an auto dialing campaign.
I tried setting up another carrier in the asterisk server, and it worked like a charm.
i found lot of threads in the internet about freeswitch / asterisk connection, NAT / FIREWALL / CALLERID / RTP problems, i tried to found out which one is my problem unsuccessfuly.
i want to know if someone has been able to get vicidial to work.

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Cordialement.
Farouk Houmaidi
Mobile : 0626620627
Siteweb : www.consolus.ma

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CSD REGISTRATION AGENT

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Jul 17, 2017, 5:52:51 PM7/17/17

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if 0033389322145 is the number you are sending from asterisk you need to have 00 as a trunk prefix on ASTPP

On Mon, Jul 17, 2017 at 7:23 PM, Farouk Houmaïdi <farouk....@gmail.com> wrote:

Here's the log of the failure:

[Jul 15 12:12:58] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 15 12:12:58] -- Executing [0033389322145@default:1] AGI("Local/0033389322145@default-00000005;2", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 15 12:12:58] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=CAMPAGNE))
[Jul 15 12:12:58] -- <Local/0033389322145@default-00000005;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 15 12:12:58] -- Executing [0033389322145@default:2] Dial("Local/0033389322145@default-00000005;2", "SIP/0033389322145@consowave,,tTo") in new stack
[Jul 15 12:12:58] == Using SIP RTP CoS mark 5
[Jul 15 12:12:58] Audio is at 15354
[Jul 15 12:12:58] Adding codec 0x4 (ulaw) to SDP
[Jul 15 12:12:58] Adding codec 0x8 (alaw) to SDP
[Jul 15 12:12:58] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 15 12:12:58] Reliably Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
INVITE sip:0033389322145@PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK463683e7;rport
Max-Forwards: 70
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>
Contact: <sip:2469402275@PRIVATE_ASTERISK_IP:5060>
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Date: Sat, 15 Jul 2017 16:12:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "V7151212580000000098" <sip:0000000000@PUBLIC_ASTPP_IP>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 1148006585 1148006585 IN IP4 PRIVATE_ASTERISK_IP
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
c=IN IP4 PRIVATE_ASTERISK_IP
t=0 0
m=audio 15354 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Jul 15 12:12:58] -- Called SIP/0033389322145@consowave
[Jul 15 12:12:58]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK463683e7;rport=33295;received=PUBLIC_ASTERISK_IP
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 102 INVITE
User-Agent: ASTPP
Content-Length: 0

<------------->
[Jul 15 12:12:58] --- (8 headers 0 lines) ---
[Jul 15 12:12:58]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK463683e7;rport=33295;received=PUBLIC_ASTERISK_IP
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>;tag=X7eXtFy1cSNSF
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 102 INVITE
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm="PUBLIC_ASTPP_IP", nonce="187ceb79-71b5-41f5-aa75-553fd2ab1e02", algorithm=MD5, qop="auth"
Content-Length: 0

<------------->
[Jul 15 12:12:58] --- (13 headers 0 lines) ---
[Jul 15 12:12:58] Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
ACK sip:0033389322145@PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK463683e7;rport
Max-Forwards: 70
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>;tag=X7eXtFy1cSNSF
Contact: <sip:2469402275@PRIVATE_ASTERISK_IP:5060>
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Content-Length: 0

---
[Jul 15 12:12:58] Audio is at 15354
[Jul 15 12:12:58] Adding codec 0x4 (ulaw) to SDP
[Jul 15 12:12:58] Adding codec 0x8 (alaw) to SDP
[Jul 15 12:12:58] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 15 12:12:58] Reliably Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
INVITE sip:0033389322145@PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK5f4d679c;rport
Max-Forwards: 70
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>
Contact: <sip:2469402275@PRIVATE_ASTERISK_IP:5060>
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Proxy-Authorization: Digest username="2469402275", realm="PUBLIC_ASTPP_IP", algorithm=MD5, uri="sip:0033389322145@PUBLIC_ASTPP_IP", nonce="187ceb79-71b5-41f5-aa75-553fd2ab1e02", response="1f89f27d5d89bc41f6c0e2326eacff91", qop=auth, cnonce="759e214d", nc=00000001
Date: Sat, 15 Jul 2017 16:12:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "V7151212580000000098" <sip:0000000000@PUBLIC_ASTPP_IP>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 1148006585 1148006586 IN IP4 PRIVATE_ASTERISK_IP
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
c=IN IP4 PRIVATE_ASTERISK_IP
t=0 0
m=audio 15354 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Jul 15 12:12:58]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK5f4d679c;rport=33295;received=PUBLIC_ASTERISK_IP
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 103 INVITE
User-Agent: ASTPP
Content-Length: 0

<------------->
[Jul 15 12:12:58] --- (8 headers 0 lines) ---
[Jul 15 12:12:59]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK5f4d679c;rport=33295;received=PUBLIC_ASTERISK_IP
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>;tag=yg8NvaF591BcB
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 103 INVITE
Contact: <sip:0033389322145@PUBLIC_ASTPP_IP:5060;transport=udp>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 222

v=0
o=FreeSWITCH 1500113503 1500113504 IN IP4 PUBLIC_ASTPP_IP
s=FreeSWITCH
c=IN IP4 PUBLIC_ASTPP_IP
t=0 0
m=audio 21674 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
[Jul 15 12:12:59] --- (15 headers 10 lines) ---
[Jul 15 12:12:59] list_route: hop: <sip:0033389322145@PUBLIC_ASTPP_IP:5060;transport=udp>
[Jul 15 12:12:59] Found RTP audio format 0
[Jul 15 12:12:59] Found RTP audio format 101
[Jul 15 12:12:59] Found audio description format PCMU for ID 0
[Jul 15 12:12:59] Found audio description format telephone-event for ID 101
[Jul 15 12:12:59] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Jul 15 12:12:59] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jul 15 12:12:59] Peer audio RTP is at port PUBLIC_ASTPP_IP:21674
[Jul 15 12:12:59] -- SIP/consowave-00000008 is making progress passing it to Local/0033389322145@default-00000005;2
[Jul 15 12:13:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 15 12:13:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 15 12:13:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 15 12:13:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 15 12:13:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 15 12:13:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 15 12:13:11] Reliably Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
OPTIONS sip:PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK42f31af0;rport
Max-Forwards: 70
From: "asterisk" <sip:2469402275@PRIVATE_ASTERISK_IP>;tag=as2f33cb01
To: <sip:PUBLIC_ASTPP_IP>
Contact: <sip:2469402275@PRIVATE_ASTERISK_IP:5060>
Call-ID: 02f938c30446acf9045b3b767ddddb12@PRIVATE_ASTERISK_IP:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Date: Sat, 15 Jul 2017 16:13:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
[Jul 15 12:13:12] NOTICE[3561]: chan_sip.c:27117 sip_poke_noanswer: Peer 'consowave' is now UNREACHABLE! Last qualify: 143
[Jul 15 12:13:12] Really destroying SIP dialog '02f938c30446acf9045b3b767ddddb12@PRIVATE_ASTERISK_IP:5060' Method: OPTIONS
[Jul 15 12:13:13]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK42f31af0;rport=33295;received=PUBLIC_ASTERISK_IP
From: "asterisk" <sip:2469402275@PRIVATE_ASTERISK_IP>;tag=as2f33cb01
To: <sip:PUBLIC_ASTPP_IP>;tag=1Bm01U1F1ve4D
Call-ID: 02f938c30446acf9045b3b767ddddb12@PRIVATE_ASTERISK_IP:5060
CSeq: 102 OPTIONS
Contact: <sip:gw+net2phone@PUBLIC_ASTPP_IP:5060;transport=udp;gw=net2phone>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

<------------->
[Jul 15 12:13:13] --- (13 headers 0 lines) ---
[Jul 15 12:13:22] Reliably Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
OPTIONS sip:PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK2919eba5;rport
Max-Forwards: 70
From: "asterisk" <sip:2469402275@PRIVATE_ASTERISK_IP>;tag=as68ea1057
To: <sip:PUBLIC_ASTPP_IP>
Contact: <sip:2469402275@PRIVATE_ASTERISK_IP:5060>
Call-ID: 44eaca105be493573eb0d68b04e11279@PRIVATE_ASTERISK_IP:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Date: Sat, 15 Jul 2017 16:13:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
[Jul 15 12:13:23] Really destroying SIP dialog '44eaca105be493573eb0d68b04e11279@PRIVATE_ASTERISK_IP:5060' Method: OPTIONS
[Jul 15 12:13:24]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK2919eba5;rport=33295;received=PUBLIC_ASTERISK_IP
From: "asterisk" <sip:2469402275@PRIVATE_ASTERISK_IP>;tag=as68ea1057
To: <sip:PUBLIC_ASTPP_IP>;tag=2mDS3pjKy54pS
Call-ID: 44eaca105be493573eb0d68b04e11279@PRIVATE_ASTERISK_IP:5060
CSeq: 102 OPTIONS
Contact: <sip:gw+net2phone@PUBLIC_ASTPP_IP:5060;transport=udp;gw=net2phone>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

<------------->
[Jul 15 12:13:24] --- (13 headers 0 lines) ---
[Jul 15 12:13:31] NOTICE[3561]: chan_sip.c:13719 sip_reregister: -- Re-registration for 2469402275@PUBLIC_ASTPP_IP
[Jul 15 12:13:31] REGISTER 11 headers, 0 lines
[Jul 15 12:13:31] Reliably Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
REGISTER sip:PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK29befb47;rport
Max-Forwards: 70
From: <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as4ac5f917
To: <sip:2469402275@PUBLIC_ASTPP_IP>
Call-ID: 79dbc16e2540d34d4095d8a22a118d1...@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Authorization: Digest username="2469402275", realm="PUBLIC_ASTPP_IP", algorithm=MD5, uri="sip:PUBLIC_ASTPP_IP", nonce="17f7d553-d701-4b5e-8ff3-ab36af7fefd9", response="8eb764c0e39a924e2bcb9a63b5ab8810", qop=auth, cnonce="19dc87ad", nc=00000003
Expires: 120
Contact: <sip:s@PRIVATE_ASTERISK_IP:5060>
Content-Length: 0

---
[Jul 15 12:13:31] Retransmitting #1 (NAT) to PUBLIC_ASTPP_IP:5060:
REGISTER sip:PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK29befb47;rport
Max-Forwards: 70
From: <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as4ac5f917
To: <sip:2469402275@PUBLIC_ASTPP_IP>
Call-ID: 79dbc16e2540d34d4095d8a22a118d1...@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Authorization: Digest username="2469402275", realm="PUBLIC_ASTPP_IP", algorithm=MD5, uri="sip:PUBLIC_ASTPP_IP", nonce="17f7d553-d701-4b5e-8ff3-ab36af7fefd9", response="8eb764c0e39a924e2bcb9a63b5ab8810", qop=auth, cnonce="19dc87ad", nc=00000003
Expires: 120
Contact: <sip:s@PRIVATE_ASTERISK_IP:5060>
Content-Length: 0

---
[Jul 15 12:13:32] Retransmitting #2 (NAT) to PUBLIC_ASTPP_IP:5060:
REGISTER sip:PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK29befb47;rport
Max-Forwards: 70
From: <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as4ac5f917
To: <sip:2469402275@PUBLIC_ASTPP_IP>
Call-ID: 79dbc16e2540d34d4095d8a22a118d1...@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Authorization: Digest username="2469402275", realm="PUBLIC_ASTPP_IP", algorithm=MD5, uri="sip:PUBLIC_ASTPP_IP", nonce="17f7d553-d701-4b5e-8ff3-ab36af7fefd9", response="8eb764c0e39a924e2bcb9a63b5ab8810", qop=auth, cnonce="19dc87ad", nc=00000003
Expires: 120
Contact: <sip:s@PRIVATE_ASTERISK_IP:5060>
Content-Length: 0

---
[Jul 15 12:13:33] Reliably Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
OPTIONS sip:PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK6dc3fe37;rport
Max-Forwards: 70
From: "asterisk" <sip:2469402275@PRIVATE_ASTERISK_IP>;tag=as5adf8ba3
To: <sip:PUBLIC_ASTPP_IP>
Contact: <sip:2469402275@PRIVATE_ASTERISK_IP:5060>
Call-ID: 6041fcdb2d0dfcda7c348f8424745d47@PRIVATE_ASTERISK_IP:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Date: Sat, 15 Jul 2017 16:13:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
[Jul 15 12:13:33]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK29befb47;rport=33295;received=PUBLIC_ASTERISK_IP
From: <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as4ac5f917
To: <sip:2469402275@PUBLIC_ASTPP_IP>;tag=3X6H5H3pUeU9m
Call-ID: 79dbc16e2540d34d4095d8a22a118d1...@127.0.0.1
CSeq: 105 REGISTER
Contact: <sip:s@PRIVATE_ASTERISK_IP:5060>;expires=120
Date: Sat, 15 Jul 2017 16:13:31 GMT
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Content-Length: 0

<------------->
[Jul 15 12:13:33] --- (12 headers 0 lines) ---
[Jul 15 12:13:33] Scheduling destruction of SIP dialog '79dbc16e2540d34d4095d8a22a118d...@127.0.0.1' in 32000 ms (Method: REGISTER)
[Jul 15 12:13:33] NOTICE[3561]: chan_sip.c:21597 handle_response_register: Outbound Registration: Expiry for PUBLIC_ASTPP_IP is 120 sec (Scheduling reregistration in 105 s)
[Jul 15 12:13:33]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK6dc3fe37;rport=33295;received=PUBLIC_ASTERISK_IP
From: "asterisk" <sip:2469402275@PRIVATE_ASTERISK_IP>;tag=as5adf8ba3
To: <sip:PUBLIC_ASTPP_IP>;tag=46Za7cmtrQHvg
Call-ID: 6041fcdb2d0dfcda7c348f8424745d47@PRIVATE_ASTERISK_IP:5060
CSeq: 102 OPTIONS
Contact: <sip:gw+net2phone@PUBLIC_ASTPP_IP:5060;transport=udp;gw=net2phone>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

<------------->
[Jul 15 12:13:33] --- (13 headers 0 lines) ---
[Jul 15 12:13:33] Really destroying SIP dialog '6041fcdb2d0dfcda7c348f8424745d47@PRIVATE_ASTERISK_IP:5060' Method: OPTIONS
[Jul 15 12:13:35]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK5f4d679c;rport=33295;received=PUBLIC_ASTERISK_IP
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>;tag=yg8NvaF591BcB
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 103 INVITE
Contact: <sip:0033389322145@PUBLIC_ASTPP_IP:5060;transport=udp>
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 222

v=0
o=FreeSWITCH 1500113503 1500113504 IN IP4 PUBLIC_ASTPP_IP
s=FreeSWITCH
c=IN IP4 PUBLIC_ASTPP_IP
t=0 0
m=audio 21674 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
[Jul 15 12:13:35] --- (14 headers 10 lines) ---
[Jul 15 12:13:35] list_route: hop: <sip:0033389322145@PUBLIC_ASTPP_IP:5060;transport=udp>
[Jul 15 12:13:35] set_destination: Parsing <sip:0033389322145@PUBLIC_ASTPP_IP:5060;transport=udp> for address/port to send to
[Jul 15 12:13:35] set_destination: set destination to PUBLIC_ASTPP_IP:5060
[Jul 15 12:13:35] Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
ACK sip:0033389322145@PUBLIC_ASTPP_IP:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK2af94b64;rport
Max-Forwards: 70
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>;tag=yg8NvaF591BcB
Contact: <sip:2469402275@PRIVATE_ASTERISK_IP:5060>
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Content-Length: 0

---
[Jul 15 12:13:35] -- SIP/consowave-00000008 answered Local/0033389322145@default-00000005;2
[Jul 15 12:13:35] > Channel Local/0033389322145@default-00000005;1 was answered.
[Jul 15 12:13:35] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 15 12:13:35] -- Executing [8368@default:1] Playback("Local/0033389322145@default-00000005;1", "sip-silence") in new stack
[Jul 15 12:13:35] Sent RTP packet to PUBLIC_ASTPP_IP:21674 (type 00, seq 021451, ts 000160, len 000160)
[Jul 15 12:13:35] -- <Local/0033389322145@default-00000005;1> Playing 'sip-silence.gsm' (language 'en')
[Jul 15 12:13:35] -- Executing [8368@default:2] AGI("Local/0033389322145@default-00000005;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 15 12:13:35] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=CAMPAGNE))
[Jul 15 12:13:35] -- <Local/0033389322145@default-00000005;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 15 12:13:35] -- Executing [8368@default:3] AGI("Local/0033389322145@default-00000005;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Jul 15 12:13:35] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jul 15 12:13:36] -- <Local/0033389322145@default-00000005;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jul 15 12:13:36] -- Executing [8368@default:4] AGI("Local/0033389322145@default-00000005;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Jul 15 12:13:36] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jul 15 12:13:37] -- <Local/0033389322145@default-00000005;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jul 15 12:13:37] -- Executing [8368@default:5] Hangup("Local/0033389322145@default-00000005;1", "") in new stack
[Jul 15 12:13:37] == Spawn extension (default, 8368, 5) exited non-zero on 'Local/0033389322145@default-00000005;1'
[Jul 15 12:13:37] -- Executing [h@default:1] AGI("Local/0033389322145@default-00000005;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jul 15 12:13:38] -- <Local/0033389322145@default-00000005;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Jul 15 12:13:38] -- Executing [h@default:1] AGI("Local/0033389322145@default-00000005;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----40-----3") in new stack
[Jul 15 12:13:39] -- <Local/0033389322145@default-00000005;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----40-----3 completed, returning 0
[Jul 15 12:13:39] Scheduling destruction of SIP dialog '2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP' in 9152 ms (Method: INVITE)
[Jul 15 12:13:39] set_destination: Parsing <sip:0033389322145@PUBLIC_ASTPP_IP:5060;transport=udp> for address/port to send to
[Jul 15 12:13:39] set_destination: set destination to PUBLIC_ASTPP_IP:5060
[Jul 15 12:13:39] Reliably Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
BYE sip:0033389322145@PUBLIC_ASTPP_IP:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK7d835927;rport
Max-Forwards: 70
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>;tag=yg8NvaF591BcB
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Proxy-Authorization: Digest username="2469402275", realm="PUBLIC_ASTPP_IP", algorithm=MD5, uri="sip:0033389322145@PUBLIC_ASTPP_IP:5060", nonce="187ceb79-71b5-41f5-aa75-553fd2ab1e02", response="6657535d43c5bb75823ca83d4c37ecd5", qop=auth, cnonce="00d773e5", nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
[Jul 15 12:13:39] == Spawn extension (default, 0033389322145, 2) exited non-zero on 'Local/0033389322145@default-00000005;2'
[Jul 15 12:13:39]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK7d835927;rport=33295;received=PUBLIC_ASTERISK_IP
From: "V7151212580000000098" <sip:2469402275@PUBLIC_ASTPP_IP>;tag=as35dfba01
To: <sip:0033389322145@PUBLIC_ASTPP_IP>;tag=yg8NvaF591BcB
Call-ID: 2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP
CSeq: 104 BYE
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Content-Length: 0

<------------->
[Jul 15 12:13:39] --- (10 headers 0 lines) ---
[Jul 15 12:13:39] Really destroying SIP dialog '2f1afeff1be508481a2c5db470ff9745@PUBLIC_ASTPP_IP' Method: INVITE
[Jul 15 12:13:43] Reliably Transmitting (NAT) to PUBLIC_ASTPP_IP:5060:
OPTIONS sip:PUBLIC_ASTPP_IP SIP/2.0
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK01cea903;rport
Max-Forwards: 70
From: "asterisk" <sip:2469402275@PRIVATE_ASTERISK_IP>;tag=as03d973fa
To: <sip:PUBLIC_ASTPP_IP>
Contact: <sip:2469402275@PRIVATE_ASTERISK_IP:5060>
Call-ID: 121c706d3b51a2972d9b418e4d3e16d6@PRIVATE_ASTERISK_IP:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by dem...@goautodial.com
Date: Sat, 15 Jul 2017 16:13:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
[Jul 15 12:13:43]
<--- SIP read from UDP:PUBLIC_ASTPP_IP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP PRIVATE_ASTERISK_IP:5060;branch=z9hG4bK01cea903;rport=33295;received=PUBLIC_ASTERISK_IP
From: "asterisk" <sip:2469402275@PRIVATE_ASTERISK_IP>;tag=as03d973fa
To: <sip:PUBLIC_ASTPP_IP>;tag=71BNcy64FjmmK
Call-ID: 121c706d3b51a2972d9b418e4d3e16d6@PRIVATE_ASTERISK_IP:5060
CSeq: 102 OPTIONS
Contact: <sip:gw+net2phone@PUBLIC_ASTPP_IP:5060;transport=udp;gw=net2phone>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

<------------->
[Jul 15 12:13:43] --- (13 headers 0 lines) ---
[Jul 15 12:13:43] NOTICE[3561]: chan_sip.c:21647 handle_response_peerpoke: Peer 'consowave' is now Reachable. (98ms / 300ms)
[Jul 15 12:13:43] Really destroying SIP dialog '121c706d3b51a2972d9b418e4d3e16d6@PRIVATE_ASTERISK_IP:5060' Method: OPTIONS

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CSD REGISTRATION AGENT

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Jul 17, 2017, 5:53:58 PM7/17/17

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so that it can be removed before dialing out of ASTPP because net2phone will not accept the call

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Intel Networks

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Jul 18, 2017, 11:15:57 AM7/18/17

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Hi i hope this will help.

Once you are able to call from Vicidial > ASTPP> Carrier = successful

and Autodial>ASTPP>Carrier = Unsuccessful

Possible problem

If you are trying manual from an softphone or extension with the enforce codec it will work however if you are doing an autdo dial from another extension you will end up with codec priority issue. Remember FS do not do translation, only does pass-true. This i find from my experience has been issue in setting up the route on asterisk side, instead of invite you do a peer.

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Farock

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Jul 18, 2017, 1:33:35 PM7/18/17

to ASTPP

Hi msprober,

thank's for replying but type is already set to peer, as for the codec i tried with both g711 and g729, doesnt change the result.
i must be so dumb missing a small details, but god ive searched so much cant figure out why on the autodial there's no RTP traffic from ASTPP.

Anyone here have successfully connected astpp with a vicidial server in autodialing mode? have you ever experienced a problem like mine?

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Intel Networks

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Jul 19, 2017, 9:52:46 AM7/19/17

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i have never used autodial before, i do have alot of asterisk working with FS (ASTPP) and PYfreebilling. the only this is you have to change the port from asterisk to pyfreebilling from 5060 to 5080

maybe you can give a snap shot of your Astpp setting

can you say what error you getting on the rejecting part - 503, 488?

On Tue, Jul 18, 2017 at 1:33 PM, Farock <farouk....@gmail.com> wrote:

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Hi msprober,
thank's for replying but type is already set to peer, as for the codec i tried with both g711 and g729, doesnt change the result.
i must be so dumb missing a small details, but god ive searched so much cant figure out why on the autodial there's no RTP traffic from ASTPP.

Anyone here have successfully connected astpp with a vicidial server in autodialing mode? have you ever experienced a problem like mine?

Le mardi 18 juillet 2017 16:15:57 UTC+1, msprober a écrit :

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Farock

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Jul 20, 2017, 5:26:33 AM7/20/17

to ASTPP

Hello,

i think its a network problem, my vicidial server is behind NAT, my ASTPP have a public ip address

i tried connecting my vicidial without nat, on another network and autodialing worked.

But in some case i'm not able to make modification in routers or try to DMZ the vicidial server.

i'm trying to find a workaround to resolve this nat problem in the ASTPP server, because in the same senario it works with another softswitch so i think i'm missing something in my freeswitch configuration.

thnk you all, and any help would be appreciated.

Uday

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May 24, 2019, 8:03:16 AM5/24/19

to as...@googlegroups.com

I am having same issue. Installed ASTPP on public IP and vicidial behind nat. Manual dialing working perfectly but when doing auto dialing, call dropped after answer on Vicidial. I tried from 2 different network and 2 different location but having same problem.

Can you please suggest me any solution?

Thanks

Uday.

angel_geraldo

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Jun 5, 2019, 5:21:17 PM6/5/19

to ASTPP

Uday, let me ask you a question, can you set any caller id in the vicidial campaigns with ASTPP?

If yes, can you tell me what setup did you did in the ASTPP to allow any caller id?

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