![\"\"](\"https:\/\/nramkumar.org\/tech\/wp-content\/uploads\/2026\/05\/AsteriskNetworkDiagram-1024x559.png\")
<\/figure>\n\n\n\nvoip.ms Setup<\/h2>\n\n\n\n
Before we start on our Asterisk setup, we need to enable SMS on the voip.ms DID. Click DID > Manage DID > Edit. Then enable Message Service (SMS\/MMS) and Link the SMS received to this DID to a SIP Account. Select the sub-account that you have configured as part of your voip.ms trunk in your Asterisk server.<\/p>\n\n\n\n
Message Threading<\/h2>\n\n\n\n
One of the key issues to address is message threading – we want messages to be threaded properly by the clients. voip.ms will put the sender information as <number>@<pop server> – this can be made to work, but a more reliable approach is to normalize all addresses as <number>@asterisk.lan (where this is the hostname of the LAN Asterisk server) on inbound\/outbound path and do rewrites as necessary on the Asterisk server. This works better with the clients and they all see the consistent path and information for inbound and outbound messages.<\/p>\n\n\n\n
Routing To All Known Endpoints<\/h2>\n\n\n\n
We can use voip.ms puts the To address (your DID) in voip.ms sends a SIP MESSAGE to your server’s IP with the destination DID in a custom header:<\/p>\n\n\n\n Two things to note: the Add to The client threads the conversation under Add to For Inbound, we need to add the context to the voip.ms trunk in For outbound, we need to add the context to our extension’s configuration. In I found linphone adopts the Use In this post, we will cover configuring a local Asterisk server to route Inbound and Outbound SMS using a voip.ms trunk. This will also cover routing inbound messages to all known clients at the time the message is received so having multiple clients (like Jami, Linphone) registered to your Asterisk server for the inbound extension…<\/p>\n","protected":false},"author":1,"featured_media":0,"comment_status":"open","ping_status":"open","sticky":false,"template":"","format":"standard","meta":{"footnotes":""},"categories":[1],"tags":[],"class_list":["post-694","post","type-post","status-publish","format-standard","hentry","category-uncategorized"],"_links":{"self":[{"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/posts\/694","targetHints":{"allow":["GET"]}}],"collection":[{"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/posts"}],"about":[{"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/types\/post"}],"author":[{"embeddable":true,"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/users\/1"}],"replies":[{"embeddable":true,"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/comments?post=694"}],"version-history":[{"count":1,"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/posts\/694\/revisions"}],"predecessor-version":[{"id":695,"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/posts\/694\/revisions\/695"}],"wp:attachment":[{"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/media?parent=694"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/categories?post=694"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/nramkumar.org\/tech\/wp-json\/wp\/v2\/tags?post=694"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}PJSIP_DIAL_CONTACTS(<extension>)<\/code> to get all the registered contacts for the extension. Then we can loop through each and send the message to each on inbound path ensuring it is delivered to the registered endpoints for the extension.<\/p>\n\n\n\nExtracting the To Address<\/h2>\n\n\n\n
X-SMS-To<\/code> header. This can be accessed using the MESSAGE_DATA<\/code> function in Asterisk since it is a custom header.<\/p>\n\n\n\nInbound SMS: voip.ms \u2192 LAN Clients<\/h2>\n\n\n\n
How voip.ms Delivers Inbound SMS<\/h3>\n\n\n\n
MESSAGE sip:s@<asterisk_lan_address>:5061;transport=TLS SIP\/2.0\nFrom: \"5555555555\" <sip:5555555555@pop_server_address>\nTo: <sip:s@asterisk_lan_address:5061;transport=TLS>\nX-SMS-To: <DID_Number><\/code><\/pre>\n\n\n\nRequest-URI<\/code> user is s<\/code> (not your DID), and the actual destination DID is in X-SMS-To<\/code>. Your dialplan needs to read X-SMS-To<\/code> to know which number was texted.<\/p>\n\n\n\nDialplan<\/h3>\n\n\n\n
\/etc\/asterisk\/extensions.conf<\/code><\/p>\n\n\n\n[sms-inbound]\nexten => s,1,NoOp(Incoming SMS from ${MESSAGE(from)})\n\n\u00a0; MESSAGE_DATA() reads custom SIP headers \u2014 MESSAGE() is only for core attributes\n\u00a0same => n,Set(TARGET_NUMBER=${MESSAGE_DATA(X-SMS-To)})\n\u00a0same => n,Log(NOTICE, SMS to DID ${TARGET_NUMBER} body: ${MESSAGE(body)})\n\n\u00a0; Extract bare number from '\"<number>\" <sip:<number>@<pop_server>>'\n\u00a0same => n,Set(RAW_FROM=${MESSAGE(from)})\n\u00a0same => n,Set(FROM_NUM=${CUT(CUT(RAW_FROM,@,1),:,2)})\n\n\u00a0; Normalize to asterisk.lan \u2014 threads by number regardless of voip.ms edge server\n\n\u00a0same => n,Set(ACTUAL_FROM=sip:${FROM_NUM}@asterisk.lan)\n\n\u00a0same => n,Log(NOTICE, Normalized sender: ${ACTUAL_FROM})\n\n\u00a0; Set To to the internal extension, not the DID.\n\u00a0; Using the DID here causes clients to adopt it as their own SIP identity,\n\u00a0; which breaks outbound endpoint matching.\n\u00a0same => n,Set(MESSAGE(to)=sip:100@asterisk.lan)\n\n\u00a0; Get all registered contacts for extension 100\n\u00a0same => n,Set(CONTACTS=${PJSIP_DIAL_CONTACTS(100)})\n\u00a0same => n,Set(CONTACT_COUNT=${FIELDQTY(CONTACTS,&)})\n\u00a0same => n,Log(NOTICE, ${CONTACT_COUNT} registered contact(s) for exten 100)\n\u00a0same => n,GotoIf($[${CONTACT_COUNT} = 0]?no_contacts)\n\u00a0same => n,Set(ITER=1)\n\u00a0same => n,While($[${ITER} <= ${CONTACT_COUNT}])\n\u00a0\u00a0same => n,Set(CURRENT_CONTACT=${CUT(CONTACTS,&,${ITER})})\n\u00a0\u00a0; CURRENT_CONTACT = PJSIP\/100\/sip:<client_ip>:<client_port> \u2014 field 3 is the SIP URI\n\u00a0\u00a0same => n,Set(CONTACT_URI=${CUT(CURRENT_CONTACT,\/,3)})\n\u00a0\u00a0same => n,Log(NOTICE, Dispatching to pjsip:100\/${CONTACT_URI} from ${ACTUAL_FROM})\n\u00a0\u00a0same => n,MessageSend(pjsip:100\/${CONTACT_URI},${ACTUAL_FROM})\n\u00a0\u00a0same => n,Log(NOTICE, MessageSend status: ${MESSAGE_SEND_STATUS})\n\u00a0\u00a0same => n,Set(ITER=$[${ITER} + 1])\n\u00a0same => n,EndWhile()\n\u00a0same => n,Hangup()\n\nsame => n(no_contacts),Log(WARNING, SMS from ${FROM_NUM} to DID ${TARGET_NUMBER} dropped \u2014 no registered contacts for exten 100)\nsame => n,Hangup()<\/code><\/pre>\n\n\n\nWhat the Client Receives<\/h3>\n\n\n\n
MESSAGE sip:100@<client_ip>:<client_port> SIP\/2.0\nFrom: <sip:<number>@asterisk.lan>\nTo: \u00a0 <sip:100@<client_ip>><\/code><\/pre>\n\n\n\nsip:<number>@asterisk.lan<\/code>. Every future message from that number \u2014 regardless of voip.ms edge \u2014 matches that thread.<\/p>\n\n\n\nOutbound SMS: LAN Clients \u2192 voip.ms<\/h2>\n\n\n\n
\/etc\/asterisk\/extensions.conf<\/code><\/p>\n\n\n\nDialplan<\/h3>\n\n\n\n
[sms-outbound]\n; Match any 10 or 11 digit number\nexten => _X.,1,NoOp(Outbound SMS from ${MESSAGE(from)} to ${EXTEN})\n\u00a0; Extract internal sender extension from 'sip:100@asterisk.lan'\n\u00a0same => n,Set(RAW_FROM=${MESSAGE(from)})\n\u00a0same => n,Set(SENDER_EXTEN=${CUT(CUT(RAW_FROM,@,1),:,2)})\n\n\u00a0; Set To for the voip.ms trunk \u2014 must use their domain, not asterisk.lan\n\u00a0same => n,Set(MESSAGE(to)=sip:${EXTEN}@<pop_server>)\n\n\u00a0; Trunk identity \u2014 DID at voip.ms domain\n\u00a0same => n,Set(TRUNK_FROM=sip:<DID_NUMBER>@<pop_server>)\n\u00a0same => n,Set(TRUNK_TO=pjsip:voipms\/sip:${EXTEN}@<pop_server>)\n\u00a0same => n,Log(NOTICE, SMS from exten ${SENDER_EXTEN} to ${EXTEN} via ${TRUNK_TO})\n\n\u00a0same => n,MessageSend(${TRUNK_TO},${TRUNK_FROM})\n\n\u00a0same => n,Log(NOTICE, Outbound send status: ${MESSAGE_SEND_STATUS})\n\u00a0same => n,GotoIf($[\"${MESSAGE_SEND_STATUS}\" != \"SUCCESS\"]?failed)\n\u00a0same => n,Hangup()\n\nsame => n(failed),Log(ERROR, Outbound SMS to ${EXTEN} failed: ${MESSAGE_SEND_STATUS})\nsame => n,Hangup()<\/code><\/pre>\n\n\n\nWhat voip.ms Receives<\/h3>\n\n\n\n
MESSAGE sip:<to_number>@<pop_server> SIP\/2.0\nFrom: <sip:<DID_Number>@<pop_server>>\nTo: \u00a0 <sip:<to_number>@<pop_server>><\/code><\/pre>\n\n\n\nConfiguring Inbound And Outbound Contexts<\/h2>\n\n\n\n
\/etc\/asterisk\/pjsip.conf<\/code>. Under the voip.ms endpoint section, add message_context=sms-inbound<\/code>.<\/p>\n\n\n\n\/etc\/asterisk\/pjsip_extensions.conf<\/code> (since I am using the #include<\/code> approach to keep local extension configuration separate from trunk configuration), under the extension endpoint section, add message_context=sms-outbound<\/code><\/p>\n\n\n\nClient Specific Quirks<\/h2>\n\n\n\n
To<\/code> header of the received message as their outbound SIP identity when replying. So if you extract the actual DID number and use it as the To<\/code> address in the inbound message, you will see the same come back as sip:<DID_Number>@<asterisk_lan><\/code> when you reply from the client. This will fail as the <DID_Number><\/code> is not a valid extension. The fix is to simply make the To<\/code> address in inbound messages be sip:<extension>@<asterisk_lan><\/code> – this works fine for both Jami and linphone. Jami works fine without this as well as it uses the configured client identity for the SIP trunk correctly.<\/p>\n\n\n\nTesting And Debugging<\/h2>\n\n\n\n
asterisk -rvv<\/code>v and pjsip set logger on<\/code> to see the actual traffic both from voip.ms and from your clients for inbound and outbound messages to understand what is actually being sent\/received.<\/p>\n","protected":false},"excerpt":{"rendered":"