Re: sip-trunk outgoing call problem (2811 CME to SBC )

ronin · ‎03-15-2018

Hi all

I have run a scenario with cisco router 2811 and I need help to solve outgoing problem

2811 (CME)(172.192.16.58)

---->-----sip trunk---->----- SBC(172.123.101.163)

2811 is connected to SBC through interface Fast 0/1 : 172.192.16.58

2811 is connected to local LAN through interface Fast 0/0: 192.168.1.10

extention 200 is trying to call outside (number 09906171493 ) and call drops after 30 sec silence

all incoming calls are ok

outgoing calls have problem ( just silence and then call drops)

here there is ccsip messages debug

***************************************************************************************************

Rotary#debug ccsip messages

*Mar 14 09:53:44.053: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:09906171493@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23:51341;branch=z9hG4bK-d8754z-c71a3a66da62b552-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:200@192.168.1.23:51341>
To: <sip:09906171493@192.168.1.10:5060>
From: "user"<sip:200@192.168.1.10:5060>;tag=af6e1916
Call-ID: MzFmZDU5MGUyMTU1NzM2OWZkNDY3NzBlNzcxMTE4Yjg.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 402

v=0
o=3cxVCE 89254350 17813835 IN IP4 192.168.1.23
s=3cxVCE Audio Call
c=IN IP4 192.168.1.23
t=0 0
m=audio 40028 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40010 RTP/AVP 34
c=IN IP4 192.168.1.23
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv

*Mar 14 09:53:44.085: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.23:51341;branch=z9hG4bK-d8754z-c71a3a66da62b552-1---d8754z-;rport
From: "user"<sip:200@192.168.1.10:5060>;tag=af6e1916
To: <sip:09906171493@192.168.1.10:5060>
Date: Wed, 14 Mar 2018 09:53:44 GMT
Call-ID: MzFmZDU5MGUyMTU1NzM2OWZkNDY3NzBlNzcxMTE4Yjg.
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

*Mar 14 09:53:44.085: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:09906171493@172.123.101.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK90
Remote-Party-ID: "user" <sip:200@172.192.16.58>;party=calling;screen=no;privacy=off
From: "user" <sip:200@172.192.16.58>;tag=1D7F6F1C-13EC
To: <sip:09906171493@172.123.101.163>
Date: Wed, 14 Mar 2018 09:53:44 GMT
Call-ID: 6AAA0238-26A411E8-80DBD83A-D10C00A5@172.192.16.58
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 300
Cisco-Guid: 1789286728-0648286696-2161498170-3507224741
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1521021224
Contact: <sip:200@172.192.16.58:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 444

v=0
o=CiscoSystemsSIP-GW-UserAgent 8548 6848 IN IP4 172.192.16.58
s=SIP Call
c=IN IP4 172.192.16.58
t=0 0
m=audio 17598 RTP/AVP 0 8 100 101
c=IN IP4 172.192.16.58
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 19236 RTP/AVP 34
c=IN IP4 172.192.16.58
b=TIAS:85000
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;MAXBR=850

*Mar 14 09:53:44.581: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:09906171493@172.123.101.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK90
Remote-Party-ID: "user" <sip:200@172.192.16.58>;party=calling;screen=no;privacy=off
From: "user" <sip:200@172.192.16.58>;tag=1D7F6F1C-13EC
To: <sip:09906171493@172.123.101.163>
Date: Wed, 14 Mar 2018 09:53:44 GMT
Call-ID: 6AAA0238-26A411E8-80DBD83A-D10C00A5@172.192.16.58
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 300
Cisco-Guid: 1789286728-0648286696-2161498170-3507224741
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1521021224
Contact: <sip:200@172.192.16.58:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 444

v=0
o=CiscoSystemsSIP-GW-UserAgent 8548 6848 IN IP4 172.192.16.58
s=SIP Call
c=IN IP4 172.192.16.58
t=0 0
m=audio 17598 RTP/AVP 0 8 100 101
c=IN IP4 172.192.16.58
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 19236 RTP/AVP 34
c=IN IP4 172.192.16.58
b=TIAS:85000
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;MAXBR=850

*Mar 14 09:53:45.581: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:09906171493@172.123.101.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK90
Remote-Party-ID: "user" <sip:200@172.192.16.58>;party=calling;screen=no;privacy=off
From: "user" <sip:200@172.192.16.58>;tag=1D7F6F1C-13EC
To: <sip:09906171493@172.123.101.163>
Date: Wed, 14 Mar 2018 09:53:45 GMT
Call-ID: 6AAA0238-26A411E8-80DBD83A-D10C00A5@172.192.16.58
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 300
Cisco-Guid: 1789286728-0648286696-2161498170-3507224741
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1521021225
Contact: <sip:200@172.192.16.58:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 444

v=0
o=CiscoSystemsSIP-GW-UserAgent 8548 6848 IN IP4 172.192.16.58
s=SIP Call
c=IN IP4 172.192.16.58
t=0 0
m=audio 17598 RTP/AVP 0 8 100 101
c=IN IP4 172.192.16.58
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 19236 RTP/AVP 34
c=IN IP4 172.192.16.58
b=TIAS:85000
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;MAXBR=850

*Mar 14 09:53:47.581: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:09906171493@172.123.101.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK90
Remote-Party-ID: "user" <sip:200@172.192.16.58>;party=calling;screen=no;privacy=off
From: "user" <sip:200@172.192.16.58>;tag=1D7F6F1C-13EC
To: <sip:09906171493@172.123.101.163>
Date: Wed, 14 Mar 2018 09:53:47 GMT
Call-ID: 6AAA0238-26A411E8-80DBD83A-D10C00A5@172.192.16.58
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 300
Cisco-Guid: 1789286728-0648286696-2161498170-3507224741
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1521021227
Contact: <sip:200@172.192.16.58:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 444

v=0
o=CiscoSystemsSIP-GW-UserAgent 8548 6848 IN IP4 172.192.16.58
s=SIP Call
c=IN IP4 172.192.16.58
t=0 0
m=audio 17598 RTP/AVP 0 8 100 101
c=IN IP4 172.192.16.58
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 19236 RTP/AVP 34
c=IN IP4 172.192.16.58
b=TIAS:85000
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;MAXBR=850

*Mar 14 09:53:51.581: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:09906171493@172.123.101.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK90
Remote-Party-ID: "user" <sip:200@172.192.16.58>;party=calling;screen=no;privacy=off
From: "user" <sip:200@172.192.16.58>;tag=1D7F6F1C-13EC
To: <sip:09906171493@172.123.101.163>
Date: Wed, 14 Mar 2018 09:53:51 GMT
Call-ID: 6AAA0238-26A411E8-80DBD83A-D10C00A5@172.192.16.58
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 300
Cisco-Guid: 1789286728-0648286696-2161498170-3507224741
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1521021231
Contact: <sip:200@172.192.16.58:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 444

v=0
o=CiscoSystemsSIP-GW-UserAgent 8548 6848 IN IP4 172.192.16.58
s=SIP Call
c=IN IP4 172.192.16.58
t=0 0
m=audio 17598 RTP/AVP 0 8 100 101
c=IN IP4 172.192.16.58
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 19236 RTP/AVP 34
c=IN IP4 172.192.16.58
b=TIAS:85000
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;MAXBR=850

*Mar 14 09:53:52.189: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

*Mar 14 09:53:59.581: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:09906171493@172.123.101.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK90
Remote-Party-ID: "user" <sip:200@172.192.16.58>;party=calling;screen=no;privacy=off
From: "user" <sip:200@172.192.16.58>;tag=1D7F6F1C-13EC
To: <sip:09906171493@172.123.101.163>
Date: Wed, 14 Mar 2018 09:53:59 GMT
Call-ID: 6AAA0238-26A411E8-80DBD83A-D10C00A5@172.192.16.58
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 300
Cisco-Guid: 1789286728-0648286696-2161498170-3507224741
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1521021239
Contact: <sip:200@172.192.16.58:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 444

v=0
o=CiscoSystemsSIP-GW-UserAgent 8548 6848 IN IP4 172.192.16.58
s=SIP Call
c=IN IP4 172.192.16.58
t=0 0
m=audio 17598 RTP/AVP 0 8 100 101
c=IN IP4 172.192.16.58
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 19236 RTP/AVP 34
c=IN IP4 172.192.16.58
b=TIAS:85000
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;MAXBR=850

Rotary(conf-serv-sip)#
Rotary(conf-serv-sip)#
Rotary(conf-serv-sip)#
*Mar 14 09:54:15.581: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:09906171493@172.123.101.163:5060 SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK90
Remote-Party-ID: "user" <sip:200@172.192.16.58>;party=calling;screen=no;privacy=off
From: "user" <sip:200@172.192.16.58>;tag=1D7F6F1C-13EC
To: <sip:09906171493@172.123.101.163>
Date: Wed, 14 Mar 2018 09:54:15 GMT
Call-ID: 6AAA0238-26A411E8-80DBD83A-D10C00A5@172.192.16.58
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 300
Cisco-Guid: 1789286728-0648286696-2161498170-3507224741
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1521021255
Contact: <sip:200@172.192.16.58:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 444

v=0
o=CiscoSystemsSIP-GW-UserAgent 8548 6848 IN IP4 172.192.16.58
s=SIP Call
c=IN IP4 172.192.16.58
t=0 0
m=audio 17598 RTP/AVP 0 8 100 101
c=IN IP4 172.192.16.58
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 19236 RTP/AVP 34
c=IN IP4 172.192.16.58
b=TIAS:85000
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;MAXBR=850

*Mar 14 09:54:16.197: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:09906171493@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23:51341;branch=z9hG4bK-d8754z-c71a3a66da62b552-1---d8754z-;rport
Max-Forwards: 70
To: <sip:09906171493@192.168.1.10:5060>
From: "user"<sip:200@192.168.1.10:5060>;tag=af6e1916
Call-ID: MzFmZDU5MGUyMTU1NzM2OWZkNDY3NzBlNzcxMTE4Yjg.
CSeq: 1 CANCEL
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 0

*Mar 14 09:54:16.205: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.23:51341;branch=z9hG4bK-d8754z-c71a3a66da62b552-1---d8754z-;rport
From: "user"<sip:200@192.168.1.10:5060>;tag=af6e1916
To: <sip:09906171493@192.168.1.10:5060>
Date: Wed, 14 Mar 2018 09:54:16 GMT
Call-ID: MzFmZDU5MGUyMTU1NzM2OWZkNDY3NzBlNzcxMTE4Yjg.
CSeq: 1 CANCEL
Content-Length: 0

*Mar 14 09:54:16.205: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.1.23:51341;branch=z9hG4bK-d8754z-c71a3a66da62b552-1---d8754z-;rport
From: "user"<sip:200@192.168.1.10:5060>;tag=af6e1916
To: <sip:09906171493@192.168.1.10:5060>;tag=1D7FEC94-20A9
Date: Wed, 14 Mar 2018 09:54:16 GMT
Call-ID: MzFmZDU5MGUyMTU1NzM2OWZkNDY3NzBlNzcxMTE4Yjg.
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0

*Mar 14 09:54:16.209: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:09906171493@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.23:51341;branch=z9hG4bK-d8754z-c71a3a66da62b552-1---d8754z-;rport
Max-Forwards: 70
To: <sip:09906171493@192.168.1.10:5060>;tag=1D7FEC94-20A9
From: "user"<sip:200@192.168.1.10:5060>;tag=af6e1916
Call-ID: MzFmZDU5MGUyMTU1NzM2OWZkNDY3NzBlNzcxMTE4Yjg.
CSeq: 1 ACK
Content-Length: 0

Mohammed al Baqari · ‎03-15-2018

Your 2811 is sending INVITE but your SBC never responds. Check reachbility

Saurabh · ‎03-15-2018

Your CME is sending INVITE to SBC but no response from it. Ensure the "sip setting" on SBC, and ensure sip trunk is up between CME and SBC.

ronin · ‎03-18-2018

hi

thanks for your reply

sip service is up :

when I replace the cisco 2811 with elastix , both incoming and outgoing calls are ok

but when I replace elastix with cisco 2811 , only incoming call is ok but outgoing call is not ok ,

please give me a tip to search and find the problem

thanks

Saurabh · ‎03-18-2018

Check the sip trunk between 2811 and SBC. Enable options ping and ensure that you are getting a reply.

Also check the ip address of 2811 is in trusted list for SBC.

R0g22 · ‎03-19-2018

Compare the SIP INVITE from your Elastix against the one from the 2811. See if you can find any noticeable difference. See if there is a firewall or something in between and it's allowing traffic to and from the 2811 IP.

ronin · ‎03-24-2018

There is not any firewall between cisco 2811 and our provider

and here is elastix debug when extention 100 is calling 0990131375

easterisk > sip set debug

#####################################################################

VoIP*CLI> sip set debug peer SIP
SIP Debugging Enabled for IP: 172.123.101.163
VoIP*CLI>
Reliably Transmitting (NAT) to 172.123.101.163:5060:
OPTIONS sip:172.123.101.163 SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK33b3b3a9;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.192.16.58>;tag=as4331f571
To: <sip:172.123.101.163>
Contact: <sip:Unknown@172.192.16.58:5060>
Call-ID: 6239fb0c3c0a5be1668210a91a339611@172.192.16.58:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Wed, 14 Mar 2018 09:27:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:172.123.101.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK33b3b3a9;rport=5060
Call-ID: 6239fb0c3c0a5be1668210a91a339611@172.192.16.58:5060
From: "Unknown"<sip:Unknown@172.192.16.58>;tag=as4331f571
To: <sip:172.123.101.163>;tag=gdvsskvr
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '6239fb0c3c0a5be1668210a91a339611@172.192.16.58:5060' Method: OPTIONS
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0990131375@from-internal:1] Macro("SIP/100-00000006", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/100-00000006", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/100-00000006", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/100-00000006", "1?Set(REALCALLERIDNUM=100)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/100-00000006", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/100-00000006", "AMPUSERCIDNAME=100") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/100-00000006", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/100-00000006", "AMPUSERCID=100") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/100-00000006", "CALLERID(all)="100" <100>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/100-00000006", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/100-00000006", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/100-00000006", "CALLERID(number)=100") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/100-00000006", "CALLERID(name)=100") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/100-00000006", "Using CallerID "100" <100>") in new stack
-- Executing [0990131375@from-internal:2] NoOp("SIP/100-00000006", "Calling Out Route: SIPout") in new stack
-- Executing [0990131375@from-internal:3] Set("SIP/100-00000006", "MOHCLASS=default") in new stack
-- Executing [0990131375@from-internal:4] Set("SIP/100-00000006", "_NODEST=") in new stack
-- Executing [0990131375@from-internal:5] Macro("SIP/100-00000006", "record-enable,100,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/100-00000006", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/100-00000006", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/100-00000006", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/100-00000006", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/100-00000006", "1?MacroExit()") in new stack
-- Executing [0990131375@from-internal:6] Macro("SIP/100-00000006", "dialout-trunk,2,0990131375,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/100-00000006", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/100-00000006", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/100-00000006", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/100-00000006", "DIAL_NUMBER=0990131375") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/100-00000006", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/100-00000006", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/100-00000006", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/100-00000006", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/100-00000006", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/100-00000006", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/100-00000006", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/100-00000006", "0?Set(REALCALLERIDNUM=100)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/100-00000006", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/100-00000006", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/100-00000006", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/100-00000006", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/100-00000006", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/100-00000006", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/100-00000006", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/100-00000006", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/100-00000006", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/100-00000006", "0?sub-flp-2,s,1") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/100-00000006", "OUTNUM=0990131375") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/100-00000006", "custom=SIP/SIP") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-00000006", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/100-00000006", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-00000006", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/100-00000006", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-00000006", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-00000006", "SIP/SIP/0990131375,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 17382
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.123.101.163:5060:
INVITE sip:0990131375@172.123.101.163 SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK58f2ad5c;rport
Max-Forwards: 70
From: "100" <sip:100@172.192.16.58>;tag=as620bcfc1
To: <sip:0990131375@172.123.101.163>
Contact: <sip:100@172.192.16.58:5060>
Call-ID: 5ada52ae79a86ea62eed50712ac5bbb0@172.192.16.58:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Wed, 14 Mar 2018 09:27:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 952672435 952672435 IN IP4 172.192.16.58
s=Asterisk PBX 1.8.20.0
c=IN IP4 172.192.16.58
t=0 0
m=audio 17382 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called SIP/SIP/0990131375

<--- SIP read from UDP:172.123.101.163:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK58f2ad5c;rport=5060
Call-ID: 5ada52ae79a86ea62eed50712ac5bbb0@172.192.16.58:5060
From: "100"<sip:100@172.192.16.58>;tag=as620bcfc1
To: <sip:0990131375@172.123.101.163>
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.123.101.163:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK58f2ad5c;rport=5060
Record-Route: <sip:172.123.101.163:5060;transport=udp;lr;Hpt=8f48_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=2975>
Call-ID: 5ada52ae79a86ea62eed50712ac5bbb0@172.192.16.58:5060
From: "100"<sip:100@172.192.16.58>;tag=as620bcfc1
To: <sip:0990131375@172.123.101.163>;tag=a6rflavq-CC-35
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:172.123.101.163:5060;transport=udp;Hpt=8f48_16;CxtId=3;TRC=ffffffff-ffffffff>
Content-Length: 211
Content-Type: application/sdp

v=0
o=- 147791478 147791478 IN IP4 172.123.101.163
s=SBC call
c=IN IP4 172.123.101.163
t=0 0
m=audio 15508 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
<------------->
--- (11 headers 10 lines) ---
list_route: hop: <sip:172.123.101.163:5060;transport=udp;lr;Hpt=8f48_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=2975>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.123.101.163:15508
-- SIP/SIP-00000007 is ringing
-- SIP/SIP-00000007 is making progress passing it to SIP/100-00000006

<--- SIP read from UDP:172.123.101.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK58f2ad5c;rport=5060
Record-Route: <sip:172.123.101.163:5060;transport=udp;lr;Hpt=8f48_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=2975>
Call-ID: 5ada52ae79a86ea62eed50712ac5bbb0@172.192.16.58:5060
From: "100"<sip:100@172.192.16.58>;tag=as620bcfc1
To: <sip:0990131375@172.123.101.163>;tag=a6rflavq-CC-35
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:172.123.101.163:5060;transport=udp;Hpt=8f48_16;CxtId=3;TRC=ffffffff-ffffffff>
Require: timer
Session-Expires: 300;refresher=uac
Content-Length: 211
Content-Type: application/sdp

v=0
o=- 147791478 147791479 IN IP4 172.123.101.163
s=SBC call
c=IN IP4 172.123.101.163
t=0 0
m=audio 15508 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
<------------->
--- (13 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.123.101.163:15508
list_route: hop: <sip:172.123.101.163:5060;transport=udp;lr;Hpt=8f48_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=2975>
set_destination: Parsing <sip:172.123.101.163:5060;transport=udp;lr;Hpt=8f48_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=2975> for address/port to send to
set_destination: set destination to 172.123.101.163:5060
Transmitting (NAT) to 172.123.101.163:5060:
ACK sip:172.123.101.163:5060;transport=udp;Hpt=8f48_16;CxtId=3;TRC=ffffffff-ffffffff SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK675a65a4;rport
Route: <sip:172.123.101.163:5060;transport=udp;lr;Hpt=8f48_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=2975>
Max-Forwards: 70
From: "100" <sip:100@172.192.16.58>;tag=as620bcfc1
To: <sip:0990131375@172.123.101.163>;tag=a6rflavq-CC-35
Contact: <sip:100@172.192.16.58:5060>
Call-ID: 5ada52ae79a86ea62eed50712ac5bbb0@172.192.16.58:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0

---
-- SIP/SIP-00000007 answered SIP/100-00000006
-- Locally bridging SIP/100-00000006 and SIP/SIP-00000007
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/100-00000006", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000006", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/100-00000006", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/100-00000006", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,28)
-- Executing [s@macro-hangupcall:28] NoOp("SIP/100-00000006", "End of MEETME check") in new stack
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/100-00000006", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] NoOp("SIP/100-00000006", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:35] GotoIf("SIP/100-00000006", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,41)
-- Executing [s@macro-hangupcall:41] NoOp("SIP/100-00000006", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:42] GotoIf("SIP/100-00000006", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,45)
-- Executing [s@macro-hangupcall:45] GotoIf("SIP/100-00000006", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,48)
-- Executing [s@macro-hangupcall:48] GotoIf("SIP/100-00000006", "1?theend") in new stack
-- Goto (macro-hangupcall,s,50)
-- Executing [s@macro-hangupcall:50] AGI("SIP/100-00000006", "hangup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
-- <SIP/100-00000006>AGI Script hangup.agi completed, returning 0
-- Executing [s@macro-hangupcall:51] Hangup("SIP/100-00000006", "") in new stack
== Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/100-00000006' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/100-00000006'
Scheduling destruction of SIP dialog '5ada52ae79a86ea62eed50712ac5bbb0@172.192.16.58:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:172.123.101.163:5060;transport=udp;lr;Hpt=8f48_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=2975> for address/port to send to
set_destination: set destination to 172.123.101.163:5060
Reliably Transmitting (NAT) to 172.123.101.163:5060:
BYE sip:172.123.101.163:5060;transport=udp;Hpt=8f48_16;CxtId=3;TRC=ffffffff-ffffffff SIP/2.0
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK2cdf030a;rport
Route: <sip:172.123.101.163:5060;transport=udp;lr;Hpt=8f48_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=2975>
Max-Forwards: 70
From: "100" <sip:100@172.192.16.58>;tag=as620bcfc1
To: <sip:0990131375@172.123.101.163>;tag=a6rflavq-CC-35
Call-ID: 5ada52ae79a86ea62eed50712ac5bbb0@172.192.16.58:5060
CSeq: 103 BYE
User-Agent: FPBX-2.8.1(1.8.20.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/100-00000006' in macro 'dialout-trunk'
== Spawn extension (from-internal, 0990131375, 6) exited non-zero on 'SIP/100-00000006'

<--- SIP read from UDP:172.123.101.163:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.192.16.58:5060;branch=z9hG4bK2cdf030a;rport=5060
Call-ID: 5ada52ae79a86ea62eed50712ac5bbb0@172.192.16.58:5060
From: "100"<sip:100@172.192.16.58>;tag=as620bcfc1
To: <sip:0990131375@172.123.101.163>;tag=a6rflavq-CC-35
CSeq: 103 BYE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '5ada52ae79a86ea62eed50712ac5bbb0@172.192.16.58:5060' Method: INVITE
VoIP*CLI>
VoIP*CLI>
VoIP*CLI>
VoIP*CLI>

R0g22 · ‎03-24-2018

Comparing IOS and Elastix INVITE, there is nothing that jumps out. There are some additional headers with the IOS but those are nothing that your ITSP might not like or agree upon.
With the IOS SDP, we are sharing video caps as well which we don't with Elastix. Again if the other end does not support video they can just respond with the UDP port for video attribute set to 0. Not something that would prevent your ITSP from responding.

Talk to your ITSP. See what they don't like with our INVITE. If you don't want to talk to your ITSP, well it will be a hit and trial troubleshooting. More frustrating for you.

ronin · ‎03-24-2018

I talk to my ITSP ,

they do not accept any fault from their configuration and they say , if elastix works with their system then it's our configuration in cisco which should be fixed.

and also our ITSP does not support video calls and messaging ....

does this help us to think about setting some more configuration in our sip-trunk to ITSP?

R0g22 · ‎03-25-2018

Did you share the logs with them at least the SIP INVITE with them and have them tell what is wrong with it ?
There might be some changes that we can add or remove but the big question is WHAT ? Who is your ITSP ?