sip_route
Handles SIP route interface data.
- sip_route:
- -Print -profile list (offered in your market): sip_route -print -profile
- -Print profile list (offered in all markets): sip_route -print -profile -all-markets
- -Print profile data: sip_route -print -profile a
- -Print route data: sip_route -print -route [ -short]
- -Remove route data: sip_route -remove -route
- sip_route (Outgoing traffic):
-set -route [ -profile a] [ -service s] [ -remoteport y] [ -protocol z] [ -proxyip x] [ -proxyport u] [ -routeset v] [-addheader a] [ -uristring0 "s"] [ -uristring1 "s"] [ -uristring2 "s"] [ -uristring3 "s"] [ -uristring4 "s"] [ -uristring5 "s"] [ -uristring6 "s"] [ -uristring7 "s"] [ -fromuri0 "f" ] [ -fromuri1 "f"] [ -fromuri2 "f" ] [ -fromuri3 "f"] [ -fromuri4 "f" ] [ -fromuri5 "f"] [ -fromuri6 "f" ] [ -fromuri7 "f"] [ -remotetelip "r"] [ -rexfromuri "e"] [ -rexstring "e"]
- sip_route (Incoming traffic):
-set -route [ -profile a] [ -service s] [ -accept x] [ -match x] [ -priority p] [ -contextb0 "p"] [ -contextb1 "p"] [ -contextb2 "p"] [ -contextb3 "p"] [ -contextb4 "p"] [ -contextb5 "p"] [ -contextb6 "p"] [ -contextb7 "p"] [ -contexta0 "p"] [ -contexta1 "p"] [ -contexta2 "p"] [ -contexta3 "p"] [ -contexta4 "p"] [ -contexta5 "p"] [ -contexta6 "p"] [ -contexta7 "p"] [ -mwinumber "n"] [ -handleasexn "h"] [ -challenge "c"]
- sip_route (Emergency call data):
-set -route -accept EMERGENCY -match "x" -sosanumber "d" -uristring0 "sip:?@" [ -priority pp]
- sip_route (Third-party registration):
-set -route [ -register l] [ -timer t] [ -numbers 123...321] [ -registerstring "s"] [ -registerport p] [ -proxyip x] [ -proxyport u] [ -routeset v] [ -addheader a] [ -supervise s] [ -supervisetime t] [-localdomain "l"]
- sip_route (Set Digest Credentials):
-set -route [ -realm "r" -authname "a" -password "w" ]
- sip_route (Set Trusted network interoperability):
-set -route -trusted t
- sip_route (Set codec filtering):
-set -route -codecs c
Function
The sip_route command is used to administrate route data specific to SIP trunks. The data is in addition to the traditional route data. When initiating a SIP trunk, you need to use sip_route and then ROCAI, RODAI and ROEQI. Changes to sip_route data can be made without removing RO-data.
- sip_route -set [-profile <trunk profile name>] -route Y -uristring0 "sip:?@<SIPreqURI>",[other sip_route parameters needed or required by the profile -Profile include parameters-]
- ROCAI:ROU=Y, SIG={D11=A for SIP route}, other service parameters.
- RODAI:ROU=Y,TYPE=TL66,VARI=00000000,VARC=00000000,VARO=00000000;
If sip_route -profile is set then VARI,VARC,VARO must have zeros. Instead, this configuration is set in the profile listed as -Profile line protocol parameters-.
- ROEQI:ROU=Y,TRU=<lim>->-<first sequence number>&&<lim>-<lastsequence number>
The TRU parameter defines the LIMs and capacity used for SIP signaling for this route.
Example: Set route which limits the route to signal via lim 3 with maximum 60 concurrent calls.ROEQI:ROU=Y,TRU=3-1&&3-60;
Note: The sip route command has to be executed before "ROEQI", which ties equipment to the route.
Define access code which direct outbound calls to this route RODDI:ROU=Y,DEST=<dest-number>
Parameters
- -accept
Type of matching to perform when handling calls.
Values: NOT_USED, CONTACT_DOMAIN, FROM_DOMAIN, REMOTE_IP, EMERGENCY, TRUNK_INFO, TRUNK_USER, PPI_INFO, PAI_INFO and ALL. (ALL must have priority = 255).
Note: To get appropriate matching priorities for the values of the -accept parameter, you have to consider your naming and priorities for each specific customer case, for example change the settings by:
- Adding more characters in you name string, since the matching is done on the entire string.
- Selecting name strings with the same length to get a certain behavior.
- Changing the order the matching is done, by modifying the -priority parameter for a specific name string compared to other strings.
- -addheader
The parameter is a comma-separated list of additional headers used in INVITES, REGISTER, and so on.
Allowed headers are:
- P-Preferred-Identity
- P-Asserted-Identity
- Request-Disposition
- Alert-Info
- Organization or proprietary headers not used by the SIP-stack
Format:
- 'Header: sip:[email protected]'
- 'Header: name<sip:[email protected]>;para'
- 'Header: data=value'
- -all-markets
The parameter controls if all trunk profiles or only the ones valid for the market (application system) configured in the system shall be printed.
The switch takes no arguments.
- -authname
The parameter is used in REGISTER or INVITE as part of Authentication, as response to 401 Unauthorized (see RFC3310). ‘authname’ is sent in SIP header, Authentication: Digest Username=<authname>
Ref: "Set Digest Credential" in the synopsis.
- -challenge
Challenge incoming INVITE on this trunk. (NOT for EMERGENCY trunks).
Values: "yes" or "no".
- -codecs
Comma-separated list of codecs to offer in SDP.
The following codecs are supported:
PCMA, PCMU, G722, G729, AMR.
Note: Any codec name can be stated, even codecs not implemented in GW.
- -contexta0
Matching string to determine if Unknown public number type should be used for the A-number, primarily for incoming "tel" invite.
See notes.
- -contexta1
Matching string to determine if International number type should be used for the A-number, primarily for incoming "tel" invite.
See notes.
- -contexta2
Matching string to determine if National number type should be used for the A-number, primarily for incoming "tel" invite.
See notes.
- -contexta3
Matching string to determine if Network specific number type should be used for the A-number, primarily for incoming "tel" invite.
See notes.
- -contexta4
Matching string to determine if Local public number type should be used for the A-number, primarily for incoming "tel" invite.
See notes.
- -contexta5
Matching string to determine if Unknown private number type should be used for the A-number, primarily for incoming "tel" invite.
See notes.
- -contexta6
Matching string to determine if Local private number type should be used for the A-number, primarily for incoming "tel" invite.
See notes.
- -contexta7
Matching string to determine if Level 1 regional number type should be used for the A-number, primarily for incoming "tel" invite.
See notes.
- -contextb0
Matching string to determine if Unknown public number type should be used for the B-number, primarily for incoming "tel" invite.
See notes.
- -contextb1
Matching string to determine if International number type should be used for the B-number, primarily for incoming "tel" invite.
See notes.
- -contextb2
Matching string to determine if National number type should be used for the B-number, primarily for incoming "tel" invite.
See notes.
- -contextb3
Matching string to determine if Network specific number type should be used for the B-number, primarily for incoming "tel" invite.
See notes.
- -contextb4
Matching string to determine if Local public number type should be used for the B-number, primarily for incoming "tel" invite.
See notes.
- -contextb5
Matching string to determine if Unknown private number type should be used for the B-number, primarily for incoming "tel" invite.
See notes.
- -contextb6
Matching string to determine if Local private number type should be used for the B-number, primarily for incoming "tel" invite.
See notes.
- -contextb7
Matching string to determine if Level 1 regional number type should be used for the B-number, primarily for incoming "tel" invite.
See notes.
- -fromuri0
String used to create the "from" field in the invite message for number type Unknown public number. See notes.
If no fromuri data is present "uristring0" is used.
The A-number is inserted at the "?" position, A rexstring may be inserted at a "!" position, format is "sip:[email protected]"
The format is sip:[email protected].
- -fromuri1
String used to create the "from" field in the invite message for number type International number. See notes.
If no fromuri data is present "uristring0" is used.
The A-number is inserted at “?” position. A rexstring may be inserted at a ”!” position.
The format is sip:[email protected].
- -fromuri2
String used to create the "from" field in the invite message for number type National number. See notes.
If no fromuri data is present "uristring0" is used.
The A-number is inserted at “?” position. A rexstring may be inserted at a ”!” position.
The format is sip:[email protected].
- -fromuri3
String used to create the "from" field in the invite message for number type Network specific number. See notes.
If no fromuri data is present "uristring0" is used.
The A-number is inserted at “?” position. A rexstring may be inserted at a ”!” position.
The format is sip:[email protected].
- -fromuri4
String used to create the "from" field in the invite message for number type Local public number.
If no fromuri data is present "uristring0" is used.
The A-number is inserted at “?” position. A rexstring may be inserted at a ”!” position.
The format is sip:[email protected].
- -fromuri5
String used to create the "from" field in the invite message for number type Unknown private number. See notes.
If no fromuri data is present "uristring0" is used.
The A-number is inserted at “?” position. A rexstring may be inserted at a ”!” position.
The format is sip:[email protected].
- -fromuri6
String used to create the "from" field in the invite message for number type Local private number. See notes.
If no fromuri data is present "uristring0"is used.
The A-number is inserted at “?” position. A rexstring may be inserted at a ”!” position.
The format is sip:[email protected].
- -fromuri7
String used to create the "from" fields in the invite message for number type Level 1 regional number. See notes.
If no fromuri data is present "uristring0" is used.
The A-number is inserted at “?” position. A rexstring may be inserted at a ”!” position.
The format is sip:[email protected].
- -handleasexn
This setting is used to enable usage of trunk data for incoming extension calls. From field number still needs to match a (pre-) registered extension.
The values are "yes" or "no".
- -localdomain
String used to create the “contact” field in the registration message.
The format is my_company.com
- -match
The IP address, the list of URL addresses, or the emergency numbers to match in the incoming call.
EMERGENCY matches the called/dialed number.
CONTACT_DOMAIN matches the host part of contact header.
FROM_DOMAIN matches the host part of from header.
FROM_USER matches the user (and host) part of from header.
TRUNK_INFO matches a string in request header,
ex: "tgrp=..."
TRUNK_USER matches the user part of in request header.
-If wildcard "*" is used match is done on first part.
-If "@" is used host part is also checked.
REMOTE_IP matches the last/proxy sender of the invite message.
PAI_INFO matches the P-Asserted-Identity header.
PPI_INFO matches the P-Preferred-Identity header.
- -mwinumber
Number to use as message waiting system number when a route is used for incoming NOTIFY from voice mail.
- -numbers
The number range to handle for a registered trunk (see -register).
- -parse
Parsable format using colon notation for easier parsing.
Using print, all sip_route parameters are listed. Only active parameters have values (the same as for normal print).
- -password
The password switch is used in REGISTER or INVITE as part of Authentication, as response to 401 Unauthorized (see RFC3310) in a registered trunk (see -register). password is input to SIP header, Authentication: response=<MD5 hashed password>.
Ref: "Set Digest Credential" in the synopsis.
- -print
Print data route or profile.
- -priority
Priority when matching incoming call data to route data.
Values: Integer 0-255, lowest priority = 255.
Default = 255.
- -profile
- List/print the trunk profiles, "-print -profile" offered in your market.
- List/print all trunk profiles, "-print -profile -all-markets".
- List a profile's parameters, "-print -profile <trunk profile name>".
The profiles are market dependent and parsed from files stored in the server at /etc/opt/eri_sn/sip_trunk_profiles.
Set a route based on trunk profile, "-set -profile
<trunk profile name> -route <route number>...".
List a profile's parameters, "-print -profile <trunk profile name>".
The heading, -Profile includes parameters-, shows required parameters in -set.
The heading, -Profile exclude parameters-, shows excluded parameters in -set; may be fixed parameter value (<sip_route_param>=<param_value>) or blocked parameter (<sip_route_param>). "-register" and "-trusted" are listed as blocked as these are replaced by line protocol parameters (see below)
The heading -Profile line protocol parameters-. These parameters extend and replace RODAI parameters.Each profile file may contain a number of profiles.
SIPLP must be restarted (using the command 'start --system') to trigger parsing of new or changed profile files. Each profile file may contain a number of profiles.
Read the following documents for guidance on profile parameters and the impact on the SIP protocol:
/etc/opt/eri_sn/sip_trunk_profiles/trunk_profiles.template
/etc/opt/eri_sn/sip_trunk_profiles/SIP_interop_readme.txt
See also the DESCRIPTION for how to initiate a route.
- -protocol
Protocol to use for SIP signalling (call setup).
Restart of SIPLP is required if protocol is changed to/from tls in an active route.
Values: udp, tcp or tls.
Default: udp
- -proxyip
IP address or FQDN (DNS name) for outbound proxy. If set, SIP messages are sent to ‘proxyip’ and ‘proxyport’.
The proxy will receive a request URI according to the content of ‘uristringX’. If registered trunk (set by ‘-register’) is used, the proxy receives REGISTER with request URI by the host portion of the registerstring.
- -proxyport
Proxy host port. Range: 0..65535.
If no value entered in proxyport, "default" is assigned. "Default" may also be set. Default will result in a port value according to the protocol used. TCP/UDP use port 5060 and TLS use port 5061.
If the port number is set to 0 (zero) an SRV query is performed.
If proxyip is an IP address, proxyport is used with the stated value. If default, the port according to protocol is selected.
If proxyip is a DNS name, the sipstack will make DNS queries in the following order:
If proxyport=0:
DNS SRV Query. If the response is positive, the SRV list contains a list. Each item has A Record or IP address with a port. If A-Record is found a new query is done to obtain the A-records result list. For each list item a call setup attempt will be done.
The port returned from DNS is used.
If proxyport=default.
DNS A-record Query. If the response is positive, the SRV answer is a result list. Each item has one or several IP address(es). For each list item a call setup attempt will be done.
The protocol default port is used.
If proxyport=1..65535:
DNS A-record Query. If the response is positive, the SRV answer is a result list. Each item has one or several IP address(es). For each list item a call setup attempt will be done.
The stated proxyport is used.
See notes for more information on DNS lookup.
- -realm
This parameter is used in Authentication as response to 401 Unauthorized (see RFC3310) in a registered trunk (see ‘-register’). The realm is sent in the SIP header, Authentication: realm=<realm>.
Ref: "Set Digest Credential" in the synopsis.
- -register
Register the -numbers to the remote system. The switch is only used for “-profile Default”.
Register the "-numbers" to the remote system.NO_REG
Do nothing.
ALL_NUMBERS
All in -numbers. Normally run in LIM 1.
ALL_EXT
All extensions in -numbers. Normally run in LIM 1.
LIM_EXT
LIM extensions in -numbers. Run in all the LIMs in parallel.
TISPAN_BT
Register PBX as TISPAN Business Trunk (No: TS 182 025).
Parameter numbers is not required.
Parameter -registerstring is used to assert PBX identity in calls. (Includes -trusted ID_ASSERT).
Broadworks
Register PBX as Broadworks trunk. Parameter -numbers is not required. Parameter -registerstring is used to assert the PBX identity in calls. (Includes -trusted ID_ASSERT).
SIP_DDI
Register the PBX as a SIP-DDI 1.0 trunk.
Parameter -numbers is not required.
Parameter -registerstring is used to assert PBX identity in calls. (Includes -trusted ID_ASSERT).
SIP_EBT
Register PBX as Enhanced Business Trunking
Parameter numbers is not required.
Parameter -registerstring is used to assert PBX identity in calls. (Includes -trusted ID_ASSERT).
RFC6140
Register PBX according to RFC6140, chapter 5,6,8,9.1,9.2.1. Parameter numbers is not required. Parameter -registerstring is used to assert PBX identity in calls. (trusted network is not part of RFC6140. For SIPConnect 1.1, use -trusted ID_ASSERT).
See also TL66 Parameter description, INDDAT parameter.
- -registerport
Registration host port. Range: 0..65535 (0 being 'default'). registerport is disregarded if -register NO_REG (default), or if proxyip and proxyport is used.
See the -remoteport description on how to interpret 'default', except that the term hostport in this context refers to -registerstring "sip:?<hostport>"
- -registerstring
registerstring is sent in the To-header of a SIP REGISTER. The host portion is also used in the request URI.
The destination of SIP REGISTER is resolved (according to methods described in proxyport) from registerstring unless ’-proxyip’ is used.
Format for ‘-register’ [ALL_NUMBERS,ALL_EXT,LIM_EXT], where ‘-numbers’ is inserted at the "?" position, is "sip:[email protected]". In this case, the PBX represents all numbers individually.
Format for the other ‘-register’ options, which assumes one registered PBX identity, is "sip:[email protected]".
registerstring is disregarded if "-register NO_REG(default)".
- -remoteport
Remote host port. Range:0..65535.
If proxyip is used the host ip-address or DNS name and ports are forwarded to the proxy/GW. No DNS lookup is then done in the MX-ONE. This require the proxy/GW to do the DNS lookup instead.
If the port number is set to 0 (zero) an SRV query is performed. If no value is entered in remoteport, "default" is assigned. "Default" as value may also be set. Default will result in a port value according to the protocol used.
TCP/UDP use port 5060 and TLS use port 5061.
If the port number is set to 0 (zero) an SRV query is performed.
- If host port is an IP address, remoteport is used with the stated value. If default, the port according to protocol is selected.
- If hostip is a DNS name, the sipstack will make DNS queries in the following order:
If remoteport=0:
DNS SRV Query. If the response is positive, the SRV list contains a list. Each item has A-Record or IP address with a port. If A-Record is found a new query is done to obtain the A-records result list. For each list item a call setup attempt will be done.
The port returned from DNS is used.
If proxyport=default
DNS A-record Query. If the response is positive, the SRV answer is a result list. Each item has one or several IP address(es) For each list item a call setup attempt will be done.
The protocol default port is used.
If remoteport=1..65535:
DNS A-record Query. If the response is positive, the SRV answer is a result list. Each item has one or several IP address(es). For each list item a call setup attempt will be done.
The stated remoteport is used.
See notes for more information on DNS lookup.
- -remotetelip
Remote host address. Used when uristrings start with "tel:".
If address is in the form of "enum:e164_search_suffix", then ENUM lookup will be enabled.
- -remove
Remove data for this route.
- -rexfromuri
String used to create the "from" ,"diverted" and "history" fields in the invite when internal users call to the remote extension. If no rexfromuri data is present "fromstringx" or "uristring0" is used.
The A-number is inserted at the "?" position, A rexstring may be inserted at a "!" position, format is "sip:[email protected]".
- -rexstring
String used to extend or prepend the "fromstringx" used, so that the provider will accept the A-number to be presented at the remote extension.
The format is "079999" or any context string manually agreed upon.
- -route
Route number. Set or remove: single value integer.
Print: single value integer or "all".
- -routeset
routeset is a comma-separated list of URI:s used to route the outgoing messages after the outgoing proxy. The setting will populate the "Route:" header in outgoing requests if proxyip is used.
Syntax: List of URI (only host part is used).
- -service
Service/protocol level (License level). PUBLIC Public trunk. PRIVATE Tie-line. PRIVATE_SERVICES Enhanced tie-line.
- -set
Set or change data for this route.
- -short
Print short format, by omitting data that is not set.
- -sosanumber
Numbers to be presented at the emergency center as the A-number (caller identity) when a SIP phone makes an emergency call, and no domain data for emergency calls are found.
The sosanumber should be the same as the destination number for the emergency route.
- -supervise Type of supervision used for this route.
NO_SUPERVISION
No supervision used.
ACTIVE_SUPERVISION
Sends OPTIONS waits for 200OK.
RELAXED_SUPERVISION
Sends OPTIONS and waits for "any" response.
PASSIVE_SUPERVISION
Expects OPTIONS from other side.
- -supervisetime
Time between heart beat test by OPTIONS.
Default=30 (30s), Range 5-3600 (5s to 1h).
- -timer
Default time before re-registering in seconds.
Default=3600 (1h), Range 300-86400 seconds (5min to 24h).
- -trusted
"-trusted" is only used for "-profile Default". sip route to a trusted network, trusts the route destination with restricted originating party information.
Originating Identification Presentation/Restriction (OIP,OIR) (IMS reference 3GPP TS 24.229):NO_TRUSTED
For OIR, from:"Anonymous" <[email protected]> restricts the originating party(RFC3261)
USER
OIR is indicated by Privacy:user. from: and contact: show originating party for OIR and OIP(RFC3323) [Ericsson IMT + VPN-BT 1.0]
ID_ASSERT
OIR is indicated by Privacy:id. from:"Anonymous" <[email protected]>. P-Asserted-Identity shows originating party for OIR and OIP(RFC3325)
ID_PREFERRED
OIR is indicated by Privacy:id. from:"Anonymous"<[email protected]>. P-Preferred-Identity shows originating party for OIR and OIP(RFC3325)
Default=NO_TRUSTED
- -uristring0
String used to create the request uri in SIP requests for type of number Unknown public number. See notes.
Request uri string in SIP requests (ex SIP INVITE). The destination of the request is resolved from the uristring and remoteport (see 'remoteport' description).
The "?" position substitutes B-number Unknown public number type.
Format is sip:[email protected] or tel:?
See chapter .
- -uristring1
String used to create the request uri in SIP requests for type of number International number. See notes.
See 'uristring0' for basic description. The "?" position substitutes B-number International number type.
Format is sip:[email protected] or tel:+?
See notes.
- -uristring2
String used to create the request uri in SIP requests for type of number National number.
See 'uristring0' for basic description. The "?" position substitutes B-number National number type.
Format is tel:?;phone-context=sub.company.com or tel:?;phone-context=+1-1234-555-1000
See notes.
- -uristring3
String used to create the request uri in SIP requests for type of number Network specific number.
See 'uristring0' for basic description. The "?" position substitutes B-number Network specific number type.
Format is tel:?;phone-context=sub.company.com or tel:?;phone-context=+1-1234-555-2000
See notes.
- -uristring4
String used to create the request uri in SIP requests for type of number local public number.
See 'uristring0' for basic description. The "?" position substitutes B-number Local public number type.
Format is tel:?;phone-context=sub.company.com or tel:?;phone-context=+1-1234-555-2000
See notes.
- -uristring5
String used to create the request uri in SIP requests for type of number Unknown private number.
See 'uristring0' for basic description. The "?" position substitutes B-number Unknown private number type.
Format is tel:?;phone-context=sub.company.com or tel:?;phone-context=+1-1234-555-2000
See notes.
- -uristring6
String used to create the request uri in SIP requests for type of number Local private number.
See 'uristring0' for basic description. The "?" position substitutes B-number Local private number type.
Format is tel:?;phone-context=sub.company.com or tel:?;phone-context=+1-1234-555-2000
See notes.
- -uristring7
String used to create the request uri in SIP requests for type of number Local regional number.
See 'uristring0' for basic description. The "?" position substitutes B-number Level 1 regional number type.
Format is tel:?;phone-context=sub.company.com or tel:?;phone-context=+1-1234-555-2000
See notes.
Examples
- Print data for route 1 only showing active parameters.
sip_route -print -route 1 -short
- End route 1.
sip_route -remove -route 1
- Initiate route 1 for outgoing calls only.
sip_route -set -route 1 -protocol udp -realm company.com password secret -uristring0 "sip:[email protected]"
- Initiate route 1 for incoming calls from a remote end and check the originators "from" domain address. For registration of numbers in the own LIM to the remote system, add prefix "23" as this is our own exchange number.
sip_route -set -route 1 -password secret -realm company.com -match company.com -accept FROM_DOMAIN -priority 1 -register LIM_EXT -numbers 322...399 -registerstring "sip:23?company.com"
- Initiate route 5, using the tel protocol with three number types.
sip_route -set -route 5 -protocol tcp -remotetelip remote.company.com -uristring0 "tel:?" -uristring1 "tel:+?" -uristring2 "tel:?;phone-context=remote.company.com" sip_route -set -route 5 -accept REMOTE_IP -match 10.0.1.10,10.0.1.11,10.0.1.12 -contextb2 "phone-context=local.company.com"
Note: Additional commands are needed in LCR and in number_conversion.
- Initiate route 6, using the sip protocol and trunk groups, outgoing hk123, and incoming site321.
sip_route -set -route 6 -uristring0 "sip:?;tgrp=hk123;[email protected]" -uristring1 "sip:+?;tgrp=hk123;[email protected]" sip_route -set -route 6 -accept TRUNK_INFO -match "tgrp=site321" -contextb2 "trunk-context=site.company.com"
Note: Additional commands are needed in LCR and in number_conversion.
- Initiate route 7, using sip protocol and identification, of trunk with PPI, in both directions.
% sip_route -set -route 7 -uristring0 "sip:[email protected]" -uristring1 "sip:[email protected]" sip_route -set -route 7 -accept PPI_INFO -match sip:[email protected]> -addheader 'P-Preferred-Identity: sip:[email protected]'
- Initiate route 8, using public sip trunk through an outbound proxy (corporate Session Border Controller (SBC))
% sip_route -set -route 8 -profile <profile> -proxyip <sbc-inside-ip> -uristring0 "sip:[email protected]" -uristring1 "sip:[email protected]" -accept FROM_DOMAIN -match "siptrunkservice.com"
Note: MX-ONE sends SIP request to -proxyip with the SIP reqURI equal to -uristringX.
- Print what profiles are available.
sip_route -print -profile
- Print a specific profile.
sip_route -print -profile Lync_TLS
Notes
Sip routes can be configured just to register subscribers in remote system without traffic configuration.
For information how to use type of number conversion: see commands "RODDI:ADC=..", "LCDDI:BTON..." and "number_conversion_initiate".
context:
ContextaX and contextbX is normally used to match "phone-context=..." or "trunk-context=...".
ContextaX and contextbX is setting type of number (TON) to "International number" if "+" is found, or if the contextX1 is matched. "Unknown public number" is set if string contextX0 is matching or if no other contextxX string is matching.
fromuri:
If no fromuri data is present for the used type of number, then "fromuri0" is used if configured. If "fromuri0" is empty, then "uristring0" is used.
When "uristring0" is used and registration is not used then the host portion is replaced with the server FQDN.
The A-number is inserted at the "?" position, or a rexstring (see 'rexstring' description) may be inserted at a "!" position. Format is "sip:[email protected]".
An ipv6 address must be sourrounded by brackets: "sip:?@[2001:DB8::2C]". The default type of number for an internal party making an outgoing call is "Unknown private number" but this can be changed depending on other configuration in system (; for example, number conversion).
uristring:
Request uri in SIP requests (that is, INVITE).
The destination of the request is resolved from the uristring and remoteport (see 'remoteport' description). The "?" position is substituted with the B-number. An ipv6 address must be sourrounded by brackets: "sip:?@[2001:DB8::2C]". If no uristring data is present for the used type of number then "uristring0" will be used.
Format is "sip:[email protected]"
"tel:?"
"tel:?;phone-context=+1-1234-555-1000"
"tel:?;phone-context=sub.company.com"
A special format is "sip:?@(host1|host2|host3|...)"
It will cause each telephony server to select one of the hosts in a round-robin fashion.
DNS lookup:
MX-ONE supports multiple A-Records in a DNS SRV lookup if port "0" is used. If -records are returned a second lookup to the DNS is done with the A-Record.
The result of the A-record may contain more than one IP address. The order of the list is the priority by which MX-ONE will attempt to send INVITE until the call is successful. No answer or 503 Service Unavailable will trigger MX-ONE to try the next entry.