Asterisk Configuration Guide - Cox Communications

Document version: 1.1 

September 8, 2011 

PROPRIETARY & CONFIDENTIAL 

SIP Trunking Configuration Guide 

for 

Asterisk v1.6 / trixbox v2.8 

© 2010, Cox Communications, Inc. All rights reserved. 

This documentation is the confidential and proprietary intellectual property of Cox Communications, Inc. 

Any unauthorized use, reproduction, preparation of derivative works, performance, or display of this 

document, or software represented by this document is strictly prohibited.

Table Of Contents 


Audience .......................................................................................................................................................4 

1 Introduction...........................................................................................................................................4 

2 Asterisk Security ...................................................................................................................................5 

3 Network Topology ................................................................................................................................5 

4 Lab Network Configuration ..................................................................................................................6 

4.1 Hardware Components .......................................................................................................7 

4.2 Software Requirements.......................................................................................................7 

4.3 Example Configuration Information...................................................................................7 

5 Features .................................................................................................................................................8 

5.1 Features Supported .............................................................................................................8 

5.2 Features Not Supported ......................................................................................................8 

6 Caveats and Limitations........................................................................................................................8 

7 Configuration ........................................................................................................................................9 

7.1 trixbox/Asterisk System Configuration Guide....................................................................9 

7.2 PBX Configuration .............................................................................................................9 

7.2.1 Initial IP PBX Web User Interface (UI) Welcome Screen ....................................9 

7.2.2 System > System Info..........................................................................................11 

7.3 Test Lab Notes..................................................................................................................11 

7.4 Summary of Phone Number Utilization ...........................................................................12 

7.5 Configuration/Viewing of Network Interface Provisioning .............................................12 

7.5.1 System > Network > Ethernet 1 ..........................................................................12 

7.5.2 System > Network > Network Parameters ..........................................................12 

7.6 SIP Trunk Configuration ..................................................................................................13 

7.7 Manual Manipulation of Asterisk Files ............................................................................13 

7.8 SIP Trunk Additions .........................................................................................................14 

7.8.1 PBX > PBX Settings > Trunks > Add SIP Trunk ...............................................14 

7.9 PBX SIP Users & Extension Management.......................................................................15 

7.9.1 PBX > PBX Settings > Extensions > Add Extension {also Voicemail}.............15 

7.9.2 PBX > PBX Settings > Extensions > Follow Me................................................17 

7.9.3 PBX > Endpoint Manager ...................................................................................18 

7.10 Disabling Feature Handling in Phones .............................................................................19 

7.11 Inbound Call Handling......................................................................................................21 

7.11.1 PBX > PBX Settings > Inbound Call Control .....................................................21 

7.12 Features Codes in PBX .....................................................................................................22 

7.12.1 PBX > PBX Settings > Feature Codes ................................................................22 

7.13 Analog Fax/Modem ..........................................................................................................23 

7.13.1 PBX > PBX Settings > ZAP Channel DIDs > Add Channel ..............................23 

7.14 Other PBX Feature Notes .................................................................................................25 

7.14.1 PBX > PBX Settings > Ring Group ....................................................................25 

7.14.2 PBX > PBX Settings > IVR ................................................................................27 

8 Appendix.............................................................................................................................................29 

8.1 Appendix A: Troubleshooting ..........................................................................................29 

8.2 ..................................................................................................................................................29 

8.2.1 Troubleshooting Information...............................................................................29 

8.2.2 Troubleshooting Techniques ...............................................................................30 

Appendix B: Tracing SIP Call Processes / Sample Traces from PBX and Edgemarc 4550..........32 

8.2.3 Basic Outbound On-net and Off-net Calls...........................................................32 

8.2.4 Basic Inbound On-net and Off-net Calls .............................................................36 

Property of Cox Communications, Inc. Version 0.1 

Page 2 of 64

Table of Figures 


8.2.5 Outbound and Inbound Anonymous Calls...........................................................41 

8.2.6 Blind Call Transfer of an On-net Caller to an Off-net Destination .....................46 

8.2.7 Attended Call Transfer of an On-net Caller to an Off-net Destination................51 

8.2.8 Unconditional Call Forwarding from an Off-net Caller to an On-net Destination, 

and Vice Versa..................................................................................................................56 

Figure 1 – Cox Fiber Network ......................................................................................................................4 

Figure 2 – Reference Network Architecture .................................................................................................5 

Figure 3 – SIP Trunk Lab Reference Network .............................................................................................6 

Figure 4 – trixbox Welcome .........................................................................................................................9 

Figure 5 – Administration Console.............................................................................................................10 

Figure 6 – System Information ...................................................................................................................11 

Figure 7 – Edit Interface “Ethernet 1” ........................................................................................................12 

Figure 8 – Network Parameters ..................................................................................................................12 

Figure 9 – Edit SIP Trunk...........................................................................................................................14 

Figure 10 – Add Extension .........................................................................................................................15 

Figure 11 – Add Extension Continued........................................................................................................16 

Figure 12 – Edit Extension Follow Me.......................................................................................................17 

Figure 13 – Endpoint Manager Selection ...................................................................................................18 

Figure 14 – Edit Linksys Device.................................................................................................................18 

Figure 15 – Inbound Call Control...............................................................................................................21 

Figure 16 – Feature Code Admin................................................................................................................22 

Figure 17 – Edit Zap Channel.....................................................................................................................23 

Figure 18 – Route Fax In ............................................................................................................................24 

Figure 19 – Ring Group ..............................................................................................................................25 

Figure 20 – Route Hunt Group Calls ..........................................................................................................26 

Figure 21 – Digital Receptionist .................................................................................................................27 

Figure 22 – Route IVR Calls ......................................................................................................................28 

Table of Tables 

Table 1 – IP Addresses .................................................................................................................................7 


Page 3 of 64


Audience 

This document is intended for the SIP Trunk customer’s technical staff and Avaya Value Added Retailer 

(VAR) having installation and operational responsibilities. 

1 Introduction 

This Configuration Guide describes configuration steps for Cox SIP Trunking to a trixbox/Asterisk PBX. 

Cox SIP Trunking is a scalable and efficient IP trunking telecommunication solution for your business 

that provides all the traditional services such as Direct Inward Dialing, Hunting, Calling Name, Calling 

Number, Local/Long Distance and Business Continuity options, including: 

Burstable Trunk Capacity – Dynamically increases call capacity during peak busy periods so your 

customers never receive a busy signal. 

Call Forward Always – On the trunk group pilot number for all calls in case of an outage (flood, 

fire, power outage, etc.). 

Call Forward Not Reachable – On the trunk group pilot number that operates on a per-call 

contingency basis to forward the call to any PSTN number (i.e., call center or alternate office 

location) during temporary call completion impairments. 

Route Exhaustion – Automatic reroute of trunk group calls to any PSTN phone number (i.e., a 

call center) if calls can’t be completed to the PBX. 

Support for geo-redundant PBX deployments and automatic reroute of SIP Trunks to the backup 

customer data center. 

All calls are routed over Cox’s national fiber network with guaranteed Quality of Service (QoS); calls 

never traverse the Internet. 

Figure 1 – Cox Fiber Network 


Page 4 of 64

2 Asterisk Security 


Cox SIP Trunking are strongly urged to review and implement the Seven Steps to Better SIP Security 

with Asterisk (http://blogs.digium.com/2009/03/28/sip-security/) by John Todd, Asterisk Open Source 

Community Director at Digium.com. 

3 Network Topology 

The high level Cox SIP Trunk network architecture is depicted below. The key network elements are: 

IP PBX – Customer PBX for terminating SIP Trunks. 

Cox Enterprise Session Border Controller (E-SBC) – The E-SBC is a smart service demarcation 

device and SIP Application Layer Gateway (ALG) installed and managed by Cox. 

Core Session Border Controllers and Broadsoft SIP Call Server for maximum survivability and 

reliability. 

PSTN Gateway for connections to the Public Switched Telephone Network (PSTN). 

Figure 2 – Reference Network Architecture 

This SIP Trunk network architecture is replicated across the Cox operating regions for scalability and 

operational autonomy. 

Cox will deploy one or more Enterprise Session Border Controllers (E-SBCs) to meet call capacity, 

customer data center geo-redundancy and trunk group requirements. The E-SBC is owned and managed 

by Cox and is the service demarcation point. The E-SBC performs SIP ALG, SIP normalization, NAT, 

security, traffic shaping/prioritization, performance reporting and remote diagnostic functions. 


Page 5 of 64


4 Lab Network Configuration 

The lab network for the SIP Trunk reference configuration is illustrated in Figure 3 below and is 

representative of an Asterisk PBX configuration. 

Fax Machine 

SIP PHone 

192.168.1.150 

SIP Phone 

192.168.1.151 

SIP Phone 

192.168.1.152 

Trixbox/Asterisk 

IP: 192.168.1.10 

DHCP Server 

192.168.1.x/24 

Customer Premise Cox 

Cox E-SBC #1 

LAN IP: 192.168.1.1 

Net Mask: 255.255.255.0 

Figure 3 – SIP Trunk Lab Reference Network 

Session Border 

Controller #1 (SBC) 

Cox 

Network 

Session Border 

Controller #2 (SBC) 

SIP Call Server 

Note: The trixbox/Asterisk PBX provides the DHCP server for dynamic IP address assignment for the 

SIP Phones; however, the Edgemarc requires a static LAN IP address that must be manually assigned by 

the LAN network administrator. 

The lab network consists of the following components: 

Asterisk/trixbox PBX for voice features, SIP proxy and SIP Trunk termination. 

Various SIP phones on the local LAN. 

The Cox E-SBC is the Edgewater Networks (www.edgewaternetworks.com) Edgemarc 4550. 

The Edgemarc is the service demarcation point between customer’s LAN network and Cox’s 

WAN network and provides firewall/NAT traversal, B2BUA and SIP Application-level gateway. 

The Edgemarc has diverse routes to a primary and secondary Acme SBC. 

Acme Packet Net-Net 9200 Session Border Controllers (SBC). In practice, these are deployed in 

high-availability and geo-redundant pairs. 

PSTN 


Page 6 of 64

4.1 Hardware Components 

trixbox/Asterisk Server 

Cisco/Linksys SPA IP Phones 

Analog Fax Machine 

Edgemarc 4550 E-SBC 

4.2 Software Requirements 

trixbox Software release 2.8.0.3 

Asterisk Software release 1.6.0.22 

Cisco/Linksys SPA942 phones (hardware 1.0.0/2622) 

Edgemarc 4550 9.12.1.1 release 


4.3 Example Configuration Information 

The specific values listed in Table 1 below and in subsequent sections are used in this lab configuration 

described in this document, and are for illustrative purposes only. Customers must obtain and use the 

specific values for their own specific configurations. 

Table 1 – IP Addresses 

Component Cox Lab Value Your Value 

Asterisk PBX 

IP Address 192.168.1.10 

Edgemarc E-SBC 

LAN Subnet Mask 255.255.255.0 

LAN IP Address #1 192.168.1.1 


Page 7 of 64

5 Features 

5.1 Features Supported 

Basic Calls using G.711ulaw 

Calling Party Number Presentation and Restriction 

Call Transfer 

Call Forwarding 

RFC2833 transcoding 

Fax and Modem inband G.711 

5.2 Features Not Supported 

CODEC negotiation of G.729, G.726, and others 

T.38 Fax Relay 


6 Caveats and Limitations 

All configurations are done under the trixbox/Asterisk Web based Administration GUI unless exclusively 

stated in this document. 

trixbox doesn’t support sending ‘anonymous’ caller ID in the Request-URI and From SIP 

header. 

trixbox standard configuration does not support the SIP Diversion header; all call forwarding 

scenarios use the Remote-Party-ID header. 

DTMF: 

o PBX originating calls are use RFC 2833 DTMF Relay. 

o PBX terminating calls use in-band DTMF. 


Page 8 of 64

7 Configuration 

7.1 trixbox/Asterisk System Configuration Guide 

Equipment used for configuration setup: 

trixbox Software Release 2.8.0.3 

Linksys SPA942 phones (hardware 1.0.0/2622) 

Edgemarc 4550 Software version 9.12.1.1 

7.2 PBX Configuration 

7.2.1 Initial IP PBX Web User Interface (UI) Welcome Screen 

1. Click switch in upper right corner. 

Figure 4 – trixbox Welcome 

2. Log in with the default username/password of maint/password. 



Page 9 of 64

Figure 5 – Administration Console 



Page 10 of 64

7.2.2 System > System Info 

Figure 6 – System Information 


7.3 Test Lab Notes 

As tested, the E-SBC requires the static IP assignment of 192.168.1.10/24 to the IP PBX. DHCP is not 

used in the test scenario; SIP phones are given static IP addresses in the same LAN subnet. The SIP 

phones register a single line each to the IP PBX, and the IP PBX registers a single pilot number as the 

sole SIP trunk to/through the E-SBC, which also serves as the LAN gateway and NAT firewall. All SIP 

traffic to/from the ITSP and IP PBX passes through the E-SBC at 192.168.1.1. 


Page 11 of 64

7.4 Summary of Phone Number Utilization 

678-239-1110 Registering PBX Pilot Number and AutoAttendant (BroadSoft) 

678-239-1111 Hunt Group (Broadsoft) 

678-239-1112 Alternate Number for 678-239-1115 

678-239-1113 Analog FXS interface/channel #1 (FAX/modem) 

678-239-1114 Polycom IP601 PBX user extension 

678-239-1115 Linksys SPA942 PBX user extension 


678-239-1117 Hunt Group (PBX) or Linksys SPA942 PBX user extension 


678-239-1119 AutoAttendent (PBX) 

7.5 Configuration/Viewing of Network Interface Provisioning 

7.5.1 System > Network > Ethernet 1 

7.5.2 System > Network > Network Parameters 

Figure 7 – Edit Interface “Ethernet 1” 

Figure 8 – Network Parameters 



Page 12 of 64


7.6 SIP Trunk Configuration 

The default ZAP trunk is deleted and replaced with a SIP trunk, as illustrated on the next page in Figure 

9. Some of the details that are difficult to see are listed below. These parameters are set to what tested 

optimally, and include some that set outbound DTMF for RFC2833, but allow inband DTMF on the 

inbound side, due to transcoding done at the Acme Packets SBC 9200. 

PEER Details: type=peer 

host=192.168.1.1 

fromuser=6782391110 

username=6782391110 

fromdomain=coxbusiness.com 

dtmfmode=rfc2833 

disallow=all 

allow=ulaw 

insecure=invite,port 

context=from-trunk 

nat=yes 

canreinvite=no 

secret=password 

USER Details: dtmfmode=inband 

In the Asterisk sip.conf under the [general] section, the registration string format is: 

user[:secret[:authuser]]@host[:port][/extension] 

Which for this example would be: 

6782391110@coxbusiness.com:password:6782391110@siptrunk/6782391110 

The SIP Trunk registration credentials (authuser and password) must match the values set in the Cox 

Broadworks Trunk Group: 

Your Cox Sales Engineer will provide the randomly generated password. 

The user authorization name (authuser) is normally the 10 digit Trunk Group Pilot Number and 

should be verified with the Broadworks Trunk Group setting during the SIP Trunk installation 

process with the Cox. If in doubt, please contact your Cox representative. 

7.7 Manual Manipulation of Asterisk Files 

There are also some manual (via command line, not accessible from Web UI) additions required to a 

configuration file to support Remote-Party-ID and SIP URI “user=phone” requirements on the SIP trunk. 

The additions are shown highlighted in yellow below: 

[trixbox1.localdomain asterisk]# more /etc/asterisk/sip_general_custom.conf 

trustrpid = yes 

sendrpid = yes 

usereqphone = yes 


Page 13 of 64

7.8 SIP Trunk Additions 

7.8.1 PBX > PBX Settings > Trunks > Add SIP Trunk 

Figure 9 – Edit SIP Trunk 



Page 14 of 64


7.9 PBX SIP Users & Extension Management 

When provisioning a PBX SIP user extension, it is important to note that the CID Num Alias, Outbound 

CID, and Add Inbound CID fields are not overlooked, but are set properly with the assigned 10-digit 

number to ensure consistent caller ID on calls. The PBX identifies users with their extension number—in 

this case the last four digits of the assigned telephone number. The call pickup group, if used, should 

specify in the Device Options section in the middle. Voicemail status and password reset are also done 

on this screen. 

7.9.1 PBX > PBX Settings > Extensions > Add Extension {also Voicemail} 

(Continued on next page) 

Figure 10 – Add Extension 


Page 15 of 64

Figure 11 – Add Extension Continued 



Page 16 of 64

7.9.2 PBX > PBX Settings > Extensions > Follow Me 

The extension will also support “find me, follow me” functionality. 

Figure 12 – Edit Extension Follow Me 



Page 17 of 64


7.9.3 PBX > Endpoint Manager 

The following image shows the built-in utility for configuring SIP phone/devices for use with the PBX. 

Note that the extension to be used by the phone is built first, prior to creating the phone profile with the 

phone’s MAC address. 

Figure 13 – Endpoint Manager Selection 

Figure 14 – Edit Linksys Device 


Page 18 of 64


7.10 Disabling Feature Handling in Phones 

By default, many SIP phones will intercept feature access codes and prevent the PBX from being able to 

receive the feature access code. Dial plan and feature access codes need to be modified from defaults to 

support various SIP features, if they are to be handled outside the phone by the PBX, or the ITSP. An 

example of this is shown in the following configuration file that has been edited for a Linksys SPA942 in 

the trixbox test lab: 

[trixbox1.localdomain tftpboot]# pwd 

/tftpboot 

[trixbox1.localdomain tftpboot]# more spa942.cfg 

 

2 

/spa$MA.cfg 

 

192.168.1.10 

192.168.1.10 

*97 

$USER 

(**xxx|**xxxx|*8|*xx.|xxx*|xxx**|xxxx*|xxxx**[3469]11|0|00|[2- 

9]xxxxxx|1xxx[2-9]xxxxxx|101xxxxx.|xxxxxxxxxxxx.) 

GMT-07:00 

trixbox CE 

 

Text Logo 

Text Logo 

 

start=3/8/7/02:0:0;end=11/1/7/02:0:0;save=1 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 


Page 19 of 64

No 

No 

No 

No 

Yes 

No 

 



Page 20 of 64


7.11 Inbound Call Handling 

Inbound routes are built to properly direct incoming calls to phones behind the PBX over the SIP trunk. 

Each one is associated with a destination of a given type (extension, IVR, hunt group, etc.). 

7.11.1 PBX > PBX Settings > Inbound Call Control 

Figure 15 – Inbound Call Control 


Page 21 of 64


7.12 Features Codes in PBX 

As with the phones, feature access codes may be disabled or modified in the PBX to support feature 

control to be passed to the ITSP when desired. Note that defaults codes can be both disabled or changed 

to meet the needs of the application, as shown below with Call Forwarding and Call Waiting. Also notice 

the *8 code used for pickup group testing is displayed in the Asterisk General Call Pickup field. 

7.12.1 PBX > PBX Settings > Feature Codes 

Figure 16 – Feature Code Admin 


Page 22 of 64

7.13 Analog Fax/Modem 

An analog (ZAP) extension is created for use with faxing and modem calls. 

7.13.1 PBX > PBX Settings > ZAP Channel DIDs > Add Channel 

Figure 17 – Edit Zap Channel 


As before, an inbound call handling rule should be created for the number associated with the analog 

interface. 


Page 23 of 64

Figure 18 – Route Fax In 



Page 24 of 64


In order to allow outbound faxing to work on the analog interfaces of the trixbox Sangoma quad port 

card, the “faxdetect” feature had to manually be fully commented out, as shown below. 

[trixbox1.localdomain asterisk]# pwd 

/etc/asterisk 

[trixbox1.localdomain asterisk]# grep faxdetect chan_dahdi.conf 

;faxdetect=both 

;faxdetect=incoming 

;faxdetect=outgoing 

;faxdetect=no 

7.14 Other PBX Feature Notes 

After configuring the Ring Group testing (sequential hunting, simultaneous, etc. – Select desired Ring 

Strategy), an inbound route is added, as with the other destinations. 

7.14.1 PBX > PBX Settings > Ring Group 

Figure 19 – Ring Group 


Page 25 of 64

Figure 20 – Route Hunt Group Calls 



Page 26 of 64


7.14.2 PBX > PBX Settings > IVR 

An IVR is provisioned as a PBX Auto Attendant with options and another inbound route: 

Figure 21 – Digital Receptionist 


Page 27 of 64

Figure 22 – Route IVR Calls 



Page 28 of 64

8 Appendix 

8.1 Appendix A: Troubleshooting 


8.2 

8.2.1 Troubleshooting Information 

The trixbox CE, based on Asterisk (an open source telephony architecture), utilizes a Web-based user 

interface for management and provisioning. This interface generates much of the underlying code and 

manual manipulation of the configuration files is not generally recommended. 

The system is not very context aware and does not handle call forwarding properly in all situations. In 

many cases, manipulation of the rules and macros from the command line does not result in better feature 

handling. This is the case especially with diversion headers and the p-asserted-identity header. Web UI 

changes may overwrite the limited manual provisioning. 

The forced use of PAI did not work better than the setting of caller ID on the pilot number and selecting 

the “never override caller ID” in the Web UI for the SIP trunk configuration. 

Lastly, each release and each variant of Asterisk-based systems vary in their support of SIP standard 

features and interoperability. 


Page 29 of 64


8.2.2 Troubleshooting Techniques 

Troubleshooting techniques are the same as other IP PBXs with regard to the E-SBC Edgemarc 4550. 

In addition, debugging is done from trixbox’s Asterisk command line interface after an SSH login to the 

PBX server. The default username/password of root/trixbox 

An example of this interface and output follows, though greater granularity is possible: 

[trixbox1.localdomain ~]# asterisk -rvvvvv -g -dddddd -c 

Asterisk 1.6.0.22-samy-r60, Copyright (C) 1999 - 2009 Digium, Inc. and others. 

Created by Mark Spencer 

Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. 

This is free software, with components licensed under the GNU General Public 

License version 2 and other licenses; you are welcome to redistribute it under 

certain conditions. Type 'core show license' for details. 

========================================================================= 

== Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf 

== Found 

== Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf 

== Found 

No entry for terminal type "terminator"; 

using dumb terminal settings. 

Connected to Asterisk 1.6.0.22-samy-r60 currently running on trixbox1 (pid = 3079) 

Verbosity is at least 6 

Core debug is at least 6 

[Jul 26 13:39:20] NOTICE[3119]: chan_sip.c:10048 sip_reregister: -- Re-registration for 

6782391110@coxbusiness.com@siptrunk 

REGISTER 13 headers, 0 lines 

Reliably Transmitting (NAT) to 192.168.1.1:5060: 

REGISTER sip:coxbusiness.com SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK31eb3c6a;rport 

Max-Forwards: 70 

From: ;tag=as083022b8 

To: 

Call-ID: 7c007fff4f41ef746580b3bc4d443cf1@coxbusiness.com 

CSeq: 135 REGISTER 

User-Agent: Asterisk PBX 1.6.0.22-samy-r60 

Authorization: Digest username="6782391110", realm="BroadWorks", algorithm=MD5, 

uri="sip:coxbusiness.com", nonce="BroadWorksXgc3kxgc1Ts56qwjBW", 

response="942f405aadedf9ddf22605dcc0cdea25", qop=auth, cnonce="4347f906", nc=00000020 

Expires: 120 

Contact: 

Event: registration 

Content-Length: 0 

--- 

trixbox1*CLI> 

 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK31eb3c6a;rport 

From: ;tag=as083022b8 

To: 

Call-ID: 7c007fff4f41ef746580b3bc4d443cf1@coxbusiness.com 

CSeq: 135 REGISTER 

Contact: ;expires=60 


 

--- (8 headers 0 lines) --- 


Page 30 of 64


Scheduling destruction of SIP dialog '7c007fff4f41ef746580b3bc4d443cf1@coxbusiness.com' in 32000 

ms (Method: REGISTER) 

[Jul 26 13:39:20] NOTICE[3119]: chan_sip.c:16348 handle_response_register: Outbound Registration: 

Expiry for siptrunk is 60 sec (Scheduling reregistration in 45 s) 


Page 31 of 64


Appendix B: Tracing SIP Call Processes / Sample Traces from PBX and Edgemarc 4550 

8.2.3 Basic Outbound On-net and Off-net Calls 

16:19:55.849735 192.168.1.10.5060 > 192.168.1.1.5060: 

>>>>>>>>>>>>>>>sip header start>>>>>>>>>>>>>>>>>>> 

INVITE sip:6782351004@192.168.1.1;user=phone SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK04a65a07;rport 


From: "Phone 4" ;tag=as55eb6717 

To: 

Contact: 

Call-ID: 297b632f6e61d78e1493367e15ac0b7f@coxbusiness.com 

CSeq: 102 INVITE 


Remote-Party-ID: "Phone 4" ;privacy=off;screen=no 

Date: Mon, 26 Jul 2010 16:19:55 GMT 

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO 

Supported: replaces, timer 

Content-Type: application/sdp 


v=0 

o=root 1140218203 1140218203 IN IP4 192.168.1.10 

s=Asterisk PBX 1.6.0.22-samy-r60 

c=IN IP4 192.168.1.10 

t=0 0 

m=audio 19568 RTP/AVP 0 

a=rtpmap:0 PCMU/8000 

a=silenceSupp:off - - - - 

a=ptime:20 

a=sendrecv 

> 

SIP/2.0 100 Trying 

Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK04a65a07;rport=5060 


To: 




> 

SIP/2.0 180 Ringing 


Record-Route: 


To: ;tag=SDfk21199-1797477000-1280161195951 



Contact: 

Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE 


> 

SIP/2.0 200 OK 





Page 32 of 64


To: ;tag=SDfk21199-1797477000-1280161195951 



Contact: 



Accept: multipart/mixed, application/media_control+xml, application/sdp 


v=0 

o=BroadWorks 18155 1 IN IP4 192.168.1.1 

s=- 

c=IN IP4 192.168.1.1 

t=0 0 




a=ptime:20 

a=sendrecv 

> 

ACK sip:192.168.1.1:5060;transport=udp SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK470b15d0;rport 

Route: 



To: ;tag=SDfk21199-1797477000-1280161195951 

Contact: 


CSeq: 102 ACK 




> 

BYE sip:192.168.1.1:5060;transport=udp SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK6e937e52;rport 

Route: 



To: ;tag=SDfk21199-1797477000-1280161195951 


CSeq: 103 BYE 



X-Asterisk-HangupCause: Normal Clearing 

X-Asterisk-HangupCauseCode: 16 


> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK6e937e52;rport=5060 



To: ;tag=SDfk21199-1797477000-1280161195951 


CSeq: 103 BYE 





Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK72816342;rport 


From: "Phone 4" ;tag=as4719bc96 

To: 

Contact: 

Call-ID: 761a72cf4bbb09a96211c7761d5c80f3@coxbusiness.com 









v=0 

o=root 441889011 441889011 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK72816342;rport=5060 


To: 




> 





To: ;tag=SD1cfl099-540367579-1280161742812 



Contact: 



> 

SIP/2.0 200 OK 




To: ;tag=SD1cfl099-540367579-1280161742812 



Contact: 






Page 34 of 64


v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 




a=mptime:20 

a=ptime:20 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK74a2b66b;rport 

Route: 



To: ;tag=SD1cfl099-540367579-1280161742812 

Contact: 


CSeq: 102 ACK 




> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK6ccb16cb;rport 

Route: 



To: ;tag=SD1cfl099-540367579-1280161742812 


CSeq: 103 BYE 






> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK6ccb16cb;rport=5060 



To: ;tag=SD1cfl099-540367579-1280161742812 


CSeq: 103 BYE 


8.2.4 Basic Inbound On-net and Off-net Calls 

16:24:11.339726 192.168.1.1.5060 > 192.168.1.10.5060: 


INVITE sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKpjd465qb8un3rfgs73dst0isk7 


From: "Pepe Lepue" ;tag=SDr4the01-1735382477-1280161451348- 

To: "phone 1118 trixbox" 

Call-ID: SDr4the01-a6ed05f37bcb44d4a461e95d4dfc6dba-vrvvfv3 


Contact: 

Max-forwards: 69 






v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 

m=audio 16392 RTP/AVP 0 101 


a=sendrecv 

a=ptime:20 

a=rtpmap:101 telephone-event/8000/1 

a=fmtp:101 0-15 

> 


Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKpjd465qb8un3rfgs73dst0isk7;received=192.168.1.1 









Contact: 


> 





To: "phone 1118 trixbox" ;tag=as4430f95c 






Contact: 


> 


Page 36 of 64












Contact: 


> 

SIP/2.0 200 OK 










Contact: 



v=0 

o=root 1005121515 1005121515 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 

ACK sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKsn5hilrs1gh7e7jt3edd6t9ma7-70idsli5 





CSeq: 130598571 ACK 

Contact: 



> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK76430512;rport 

Route: 


From: "phone 1118 trixbox" ;tag=as4430f95c 

To: "Pepe Lepue" ;tag=SDr4the01-1735382477-1280161451348- 


Page 37 of 64



CSeq: 102 BYE 





> 

SIP/2.0 200 OK 



From: "phone 1118 trixbox" ;tag=as4430f95c 

To: "Pepe Lepue" ;tag=SDr4the01-1735382477-1280161451348- 


CSeq: 102 BYE 


> 

INVITE sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKerd0lat6gkj7a7fif7pf5kuqk6 


From: "INFOSYSTEST" ;tag=SDqldbd01-462511161-1280161722604- 


Call-ID: SDqldbd01-148d60f09566739157f5885dfbcf82b3-vrvvfv3 


Contact: 

Supported: 100rel 






v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 


a=rtpmap:0 PCMU/8000/1 

a=sqn: 0 

a=cdsc: 1 image udptl t38 

a=mptime:20 

a=ptime:20 


a=fmtp:101 0-15 

> 


Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKerd0lat6gkj7a7fif7pf5kuqk6;received=192.168.1.1 









Contact: 


Page 38 of 64



> 





To: "phone 1118 trixbox" ;tag=as0a738603 






Contact: 


> 











Contact: 


> 

SIP/2.0 200 OK 










Contact: 



v=0 

o=root 470282598 470282598 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 


Page 39 of 64


ACK sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKhn8ti8rdumitth90ckbte16985-7gihtrh5 





CSeq: 130734199 ACK 

Contact: 



> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1c61cd69;rport 

Route: 


From: "phone 1118 trixbox" ;tag=as0a738603 

To: "INFOSYSTEST" ;tag=SDqldbd01-462511161-1280161722604- 


CSeq: 102 BYE 





> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK1c61cd69;rport=5060 


From: "phone 1118 trixbox" ;tag=as0a738603 

To: "INFOSYSTEST" ;tag=SDqldbd01-462511161-1280161722604- 


CSeq: 102 BYE 



8.2.5 Outbound and Inbound Anonymous Calls 

The trixbox does not provide support directly, but it can be configured to pass the dialed string containing 

the prepended FAC to the BroadSoft server. 

16:35:59.832569 192.168.1.10.5060 > 192.168.1.1.5060: 


INVITE sip:*674046691362@192.168.1.1 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK469f91a6;rport 


From: "Phone 4" ;tag=as4845ca7c 

To: 

Contact: 

Call-ID: 2a6d06a02da4cef15c645ce624fff840@coxbusiness.com 









v=0 

o=root 1080051511 1080051511 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK469f91a6;rport=5060 


To: 




> 





To: ;tag=SDsc0c399-24430308-1280162160253 



Contact: 


Content-Length: 

> 

SIP/2.0 200 OK 




To: ;tag=SDsc0c399-24430308-1280162160253 



Page 41 of 64


Contact: 






v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 




a=mptime:20 

a=ptime:20 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3392d0af;rport 

Route: 



To: ;tag=SDsc0c399-24430308-1280162160253 

Contact: 


CSeq: 102 ACK 




> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK661daccc;rport 

Route: 



To: ;tag=SDsc0c399-24430308-1280162160253 


CSeq: 103 BYE 






> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK661daccc;rport=5060 



To: ;tag=SDsc0c399-24430308-1280162160253 


CSeq: 103 BYE 


> 

INVITE sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKl3fm5ad6fs71r77ikfrn59qp67 


Page 42 of 64



From: "Anonymous" ;tag=SD8lhha01-505577546-1280162190436- 


Call-ID: SD8lhha01-f1bfd59d123425846ebb16e034b71115-vrvvfv3 


Contact: 







v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 



a=sqn: 0 


a=mptime:20 

a=ptime:20 


a=fmtp:101 0-15 

> 


Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKl3fm5ad6fs71r77ikfrn59qp67;received=192.168.1.1 









Contact: 


> 





To: "phone 1118 trixbox" ;tag=as74d69abb 






Contact: 


> 








Page 43 of 64






Contact: 


> 

SIP/2.0 200 OK 










Contact: 



v=0 

o=root 1203194919 1203194919 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 

ACK sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKqsuh243d0fijeqkp4da5tfl1f0-7gjlmpg5 





CSeq: 130968115 ACK 

Contact: 



> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK599ee13d;rport 

Route: 


From: "phone 1118 trixbox" ;tag=as74d69abb 

To: "Anonymous" ;tag=SD8lhha01-505577546-1280162190436- 


CSeq: 102 BYE 





> 


Page 44 of 64


SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK599ee13d;rport=5060 


From: "phone 1118 trixbox" ;tag=as74d69abb 

To: "Anonymous" ;tag=SD8lhha01-505577546-1280162190436- 


CSeq: 102 BYE 



8.2.6 Blind Call Transfer of an On-net Caller to an Off-net Destination 

This is performed on the Linksys SPA942 phone with the PBX’s feature access code of ## (see image 

capture of PBX FACs in the previous section). 

16:57:06.440292 192.168.1.1.5060 > 192.168.1.10.5060: 


INVITE sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKl3ai655c7unjqe4tlic6ev9s97 


From: "INFOSYSTEST" ;tag=SD6ldtd01-1471604381-1280163426450- 


Call-ID: SD6ldtd01-c15ce9ddad329d21134e048ee34509c1-vrvvfv3 


Contact: 







v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 



a=sqn: 0 


a=mptime:20 

a=ptime:20 


a=fmtp:101 0-15 

> 


Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKl3ai655c7unjqe4tlic6ev9s97;received=192.168.1.1 









Contact: 


> 





To: "phone 1118 trixbox" ;tag=as1f0bfb8c 







Page 46 of 64


Contact: 


> 











Contact: 


> 

SIP/2.0 200 OK 










Contact: 



v=0 

o=root 615347514 615347514 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 

ACK sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK9u04228jqikdjtpq6drs9b2la7-7olhd5h5 





CSeq: 131586122 ACK 

Contact: 



> 



Page 47 of 64


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4dd17990;rport 


From: "trixbox" ;tag=as4558d10b 

To: 

Contact: 

Call-ID: 2a10aa9f48db622b058557f57b679471@coxbusiness.com 



Remote-Party-ID: "trixbox" ;privacy=off;screen=no 






v=0 

o=root 972150253 972150253 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK4dd17990;rport=5060 


To: 




> 





To: ;tag=SDrid2699-1091180775-1280163441232 



Contact: 



> 

SIP/2.0 200 OK 




To: ;tag=SDrid2699-1091180775-1280163441232 



Contact: 






Page 48 of 64


v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 




a=ptime:20 

a=sendrecv 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3c991946;rport 

Route: 



To: ;tag=SDrid2699-1091180775-1280163441232 

Contact: 


CSeq: 102 ACK 




> 

BYE sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK9u04228jqikdjtpq6drs9b2la7-7olht5l5 





CSeq: 131586123 BYE 



> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK9u04228jqikdjtpq6drs9b2la7- 

7olht5l5;received=192.168.1.1 





CSeq: 131586123 BYE 





> 

BYE sip:6782391110@192.168.1.10 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKrn3n1nvjn6k7k49sec9td05vr0-7olbv7sa 


From: ;tag=SDrid2699-1091180775-1280163441232 

To: "trixbox" ;tag=as4558d10b 


CSeq: 131593487 BYE 


Page 49 of 64




> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKrn3n1nvjn6k7k49sec9td05vr0- 

7olbv7sa;received=192.168.1.1 


From: ;tag=SDrid2699-1091180775-1280163441232 

To: "trixbox" ;tag=as4558d10b 


CSeq: 131593487 BYE 






8.2.7 Attended Call Transfer of an On-net Caller to an Off-net Destination 

This is performed on the Linksys SPA942 phone with the PBX’s feature access code of *2 (see image 

capture of PBX FACs above). 

17:17:38.815185 192.168.1.1.5060 > 192.168.1.10.5060: 


INVITE sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKvre45c17hej7a4nio7o16p2rm6 


From: "Pepe Lepue" ;tag=SD95t5f01-877709452-1280164658823- 


Call-ID: SD95t5f01-141238228dae477105df10756aa12e6e-vrvvfv3 


Contact: 






v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 



a=sendrecv 

a=ptime:20 


a=fmtp:101 0-15 

> 


Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKvre45c17hej7a4nio7o16p2rm6;received=192.168.1.1 









Contact: 


> 





To: "phone 1118 trixbox" ;tag=as16a048ad 






Contact: 



[tos 0x60] 

17:17:39.162415 192.168.1.10.5060 > 192.168.1.1.5060: 












Contact: 


> 

SIP/2.0 200 OK 










Contact: 



v=0 

o=root 698513381 698513381 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 

ACK sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKgu8rrubj0dgftopqs2vabbk8l6-78ov32h5 





CSeq: 132202308 ACK 

Contact: 



> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK6ed202ad;rport 


From: "Phone 4" ;tag=as3f114e2e 


Page 52 of 64


To: 

Contact: 

Call-ID: 4ddb04e115c79fcd1616e1d64b78ba83@coxbusiness.com 









v=0 

o=root 1995445710 1995445710 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK6ed202ad;rport=5060 


To: 




> 





To: ;tag=SD4ecab99-2124353115-1280164676195 



Contact: 



> 

SIP/2.0 200 OK 




To: ;tag=SD4ecab99-2124353115-1280164676195 



Contact: 





v=0 


s=- 

c=IN IP4 192.168.1.1 


Page 53 of 64


t=0 0 




a=mptime:20 

a=ptime:20 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK496623db;rport 

Route: 



To: ;tag=SD4ecab99-2124353115-1280164676195 

Contact: 


CSeq: 102 ACK 




> 

BYE sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKgu8rrubj0dgftopqs2vabbk8l6-78ovj2l5 





CSeq: 132202309 BYE 



> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKgu8rrubj0dgftopqs2vabbk8l6- 

78ovj2l5;received=192.168.1.1 





CSeq: 132202309 BYE 





> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK64a53593;rport 

Route: 



To: ;tag=SD4ecab99-2124353115-1280164676195 


CSeq: 103 BYE 





Page 54 of 64




> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK64a53593;rport=5060 



To: ;tag=SD4ecab99-2124353115-1280164676195 


CSeq: 103 BYE 



8.2.8 Unconditional Call Forwarding from an Off-net Caller to an On-net Destination, and Vice 

Versa 

This is performed after enabling CF-All and setting the destination from the Linksys SPA942. The PBX 

feature access code is customized to *772 from the default *72 to prevent the forwarding from being set 

at the SIP phone handset. Similarly, the feature is disabled with *773, rather than *73. Alternatively, 

these features can be disabled in the phone configuration files to prevent conflict with the trixbox default 

codes (see also in the section describing the management of SIP phones). 

16:42:56.956653 192.168.1.1.5060 > 192.168.1.10.5060: 


INVITE sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKebao90ubc6kjqb0tviamu3itg7 


From: "INFOSYSTEST" ;tag=SDfuudd01-2102450607-1280162576962- 


Call-ID: SDfuudd01-5f5e48ee71d5ff18b1f63c506945e404-vrvvfv3 


Contact: 







v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 



a=sqn: 0 


a=mptime:20 

a=ptime:20 


a=fmtp:101 0-15 

> 


Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKebao90ubc6kjqb0tviamu3itg7;received=192.168.1.1 









Contact: 


> 





To: "phone 1118 trixbox" ;tag=as317ab7dd 


Page 56 of 64







Contact: 


> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK37e21cea;rport 


From: "trixbox" ;tag=as1dd6149c 

To: 

Contact: 

Call-ID: 41ff6def035e29d9707ae3c04953c781@coxbusiness.com 









v=0 

o=root 892099915 892099915 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK37e21cea;rport=5060 


To: 




> 





To: ;tag=SDb7ph899-462052336-1280162577589 



Contact: 



> 




Page 57 of 64










Contact: 


> 

SIP/2.0 200 OK 




To: ;tag=SDb7ph899-462052336-1280162577589 



Contact: 





v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 




a=ptime:20 

a=sendrecv 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK20c2f513;rport 

Route: 



To: ;tag=SDb7ph899-462052336-1280162577589 

Contact: 


CSeq: 102 ACK 




> 

SIP/2.0 200 OK 










Page 58 of 64


Contact: 




v=0 

o=root 601875251 601875251 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 

ACK sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKtvkvi7a2mc2u0fa3ck1bdla8a1-78kr5hg5 





CSeq: 131161378 ACK 

Contact: 



> 

BYE sip:6782391110@192.168.1.10 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKk5dsg03ppmv37j1o83djcvs0q2-78ktl1ta 


From: ;tag=SDb7ph899-462052336-1280162577589 

To: "trixbox" ;tag=as1dd6149c 


CSeq: 131161667 BYE 



> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKk5dsg03ppmv37j1o83djcvs0q2- 

78ktl1ta;received=192.168.1.1 


From: ;tag=SDb7ph899-462052336-1280162577589 

To: "trixbox" ;tag=as1dd6149c 


CSeq: 131161667 BYE 





> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK509ade9e;rport 

Route: 


From: "phone 1118 trixbox" ;tag=as317ab7dd 


Page 59 of 64


To: "INFOSYSTEST" ;tag=SDfuudd01-2102450607-1280162576962- 


CSeq: 102 BYE 





> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK509ade9e;rport=5060 


From: "phone 1118 trixbox" ;tag=as317ab7dd 

To: "INFOSYSTEST" ;tag=SDfuudd01-2102450607-1280162576962- 


CSeq: 102 BYE 


> 

INVITE sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKibgmm0ic56o7agor0re2u5aro7 


From: "Pepe Lepue" ;tag=SD7lrde01-1499106432-1280162906597- 


Call-ID: SD7lrde01-622c11a1f3ee93bdda02517327f7f834-vrvvfv3 


Contact: 






v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 



a=sendrecv 

a=ptime:20 


a=fmtp:101 0-15 

> 


Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKibgmm0ic56o7agor0re2u5aro7;received=192.168.1.1 









Contact: 


Page 60 of 64



> 





To: "phone 1118 trixbox" ;tag=as17d46528 






Contact: 


> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4a6f687e;rport 


From: "trixbox" ;tag=as6620c7cb 

To: 

Contact: 

Call-ID: 71b6f9ff76be86ba343d420d02fcf02f@coxbusiness.com 









v=0 

o=root 1828163393 1828163393 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK4a6f687e;rport=5060 


To: 




> 





To: ;tag=SDjvesa99-1932241099-1280162907894 


Page 61 of 64




Contact: 



> 











Contact: 


> 

SIP/2.0 200 OK 




To: ;tag=SDjvesa99-1932241099-1280162907894 



Contact: 





v=0 


s=- 

c=IN IP4 192.168.1.1 

t=0 0 




a=mptime:20 

a=ptime:20 

> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4f85e320;rport 

Route: 



To: ;tag=SDjvesa99-1932241099-1280162907894 

Contact: 


CSeq: 102 ACK 






SIP/2.0 200 OK 










Contact: 



v=0 

o=root 943248746 943248746 IN IP4 192.168.1.10 


c=IN IP4 192.168.1.10 

t=0 0 




a=ptime:20 

a=sendrecv 

> 

ACK sip:6782391118@192.168.1.10:5060 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKnn61l9istqlju6q8meiq5189u3-7ok1upj5 





CSeq: 131326195 ACK 

Contact: 



> 

BYE sip:6782391110@192.168.1.10 SIP/2.0 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK7gtp1rnq2njjbpsruejkgm5o54-7ok5eosa 


From: ;tag=SDjvesa99-1932241099-1280162907894 

To: "trixbox" ;tag=as6620c7cb 


CSeq: 131326512 BYE 



> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK7gtp1rnq2njjbpsruejkgm5o54- 

7ok5eosa;received=192.168.1.1 


From: ;tag=SDjvesa99-1932241099-1280162907894 

To: "trixbox" ;tag=as6620c7cb 


CSeq: 131326512 BYE 



Page 63 of 64





> 


Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3fe4bf87;rport 

Route: 


From: "phone 1118 trixbox" ;tag=as17d46528 

To: "Pepe Lepue" ;tag=SD7lrde01-1499106432-1280162906597- 


CSeq: 102 BYE 





> 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP 192.168.1.10:5060;received=10.0.150.178;branch=z9hG4bK3fe4bf87;rport=5060 


From: "phone 1118 trixbox" ;tag=as17d46528 

To: "Pepe Lepue" ;tag=SD7lrde01-1499106432-1280162906597- 


CSeq: 102 BYE

Asterisk Configuration Guide - Cox Communications

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