Archive for November, 2019

Forwarded/diverted calls in Asterisk log

Tuesday, November 26th, 2019
VERBOSE[90578][C-0027e6ee]: app_dial.c:935 in do_forward: \
[2019-11-25 09:54:06] \
-- Now forwarding SIP/2765-00559bac \
to 'Local/3135@office' (thanks to SIP/2559-00559bad)

NOTICE[90578][C-0027e6ee]: app_dial.c:958 in do_forward: \
Not accepting call completion offers from call-forward \
recipient Local/3135@office-0014b4ca;1

VERBOSE[109197][C-0027f144]: app_dial.c:935 in do_forward: \
[2019-11-25 12:20:26] \
-- Now forwarding SIP/1888-0055b018 \
to 'Local/7xxx3063600@office' (thanks to SIP/2813-0055b019)

NOTICE[109197][C-0027f144]: app_dial.c:958 in do_forward: \
Not accepting call completion offers from call-forward \
recipient Local/7xxx3063600@office-0014b7e3;1

“thanks to SIP/2559-00559bad” – this is a name of the channel who forwarded the call, either manually or by settings configured in the UAC.

sudo without pass

Tuesday, November 26th, 2019

Executing apt as an example. Add to /etc/sudoers

lexus ALL=(root) NOPASSWD:/usr/bin/apt

OpenSIPS: Adding CALLERID (display-name) to calls from registered users

Thursday, November 21st, 2019

This is about adding a display name to the calls from registered users. An analogue of Asterisk’s Set(CALLERID(name)=John Doe).

The idea was to move SIP accounts from Asterisk to OpenSIPS.

In case of using Asterisk we would configure something like:

callerid=Alexey Kazantsev <222>

How to achieve the same with OpenSIPS? This is the solution:

  • add desired display-names to ‘rpid’ columns of the ‘subscriber‘ table:
Adding desired callerid to ‘rpid’ column of the SIP-account.
modparam("auth_db", "load_credentials", "$avp(display)=rpid")
  • add ‘attr_avp‘ parameter to ‘registrar’ module settings:
modparam("registrar", "attr_avp", "$avp(display)")
  • and finally some magic in the script, to add the display-name to the INVITE request from the registered user, going through our OpenSIPS SBC:
# call from registered user -> add callerid
# and forward to mediaserver for call recording, etc.
# replace only display and do not touch uri

How it looks like?

This is the INVITE coming to OpenSIPS:

2019/11/21 14:16:55.247856 ->
Via: SIP/2.0/UDP;branch=z9hG4bK-3822b894
From: <>;tag=48b014547f398294o1

And this is the same INVITE leaving OpenSIPS, being modified:

2019/11/21 14:16:55.252518 ->
INVITE sip:111@ SIP/2.0
Record-Route: <sip:;lr;ftag=48b014547f398294o1>
Via: SIP/2.0/UDP;branch=z9hG4bK77e6.9bb3aa72.0
Via: SIP/2.0/UDP;branch=z9hG4bK-3822b894
From: "Alexey Kazantsev" <>;tag=48b014547f398294o1

The information stored in the ‘rpid’ column (in our example, or some custom in your architecture) is fetched to AVP at each REGISTER/save, so you do not need to reload anything to take changes in effect.

The callerid info is seen in console output via ‘opensipsctl fifo ul_dump’ command:

		Contact:: sip:222@ Q=
			ContactID:: 3039507536010050217
			Expires:: 42
			Callid:: 9fdd26c2-6de37105@
			Cseq:: 35746
			User-agent:: Cisco/SPA303-7.6.2c
			State:: CS_NEW
			Flags:: 0
			Socket:: udp:
			Methods:: 5247
			Attr:: Alexey Kazantsev

UAC settings: forming INVITE with RURI containing a domain

Thursday, November 7th, 2019

Some notes how to configure Linphone and Cisco SPA-303 to send INVITE with domain in RURI, assuming that your proxy has a different IP address than the resolved address of the domain.

Linphone: add “Route” parameter.
Cisco SPA-303: Outbound proxy, Use outbound proxy = yes, Use OB proxy id Dialog = yes.
Resulting INVITE to a private IP containing a domain in RURI.

That’s all. Have fun!

Inserting Asterisk in your call flow

Wednesday, November 6th, 2019

Let’s imagine that we’ve upgraded our VoIP network which was formerly based on geographically distributed Asterisks.

We configured OpenSIPS servers as registrars, connected them together in a full-sharing usrloc cluster and now we need to route calls between endpoints not directly, but through Asterisks – to handle our calls in a familiar way (CDR records, call recording via MixMonitor, some AGI scripts, etc).

We have to create something like that:

This is a code snippet of OpenSIPS with IP address

	# initial INVITE

		# call from registered user ->
                # forward to mediaserver for call recording, etc.
			$ru="sip:" + $oU + "@" + "";

		# call from Asterisk? -> change domain part before doing lookup

		$var(lookup_flags) = "m";
		if(cluster_check_addr("1", "$si")) {
			xlog("si: $si . $rm from cluster, doing local lookup only\n");
		} else {
			xlog("si: $si . $rm from outside, doing global lookup\n");
			$var(lookup_flags) = $var(lookup_flags) + "g";

		if(!lookup("location", "$var(lookup_flags)"))
			t_reply("404", "Not Found");

			rtpengine_offer("RTP/AVP replace-origin replace-session-connection ICE=remove");

	} # initial INVITE end


A SIP peer to this OpenSIPS in Asterisk sip.conf looks like this:


And a dialplan for CDR/MixMonitor/etc:

context office
	_XXX =>
		NoOp(imagine this is CDR, MixMonitor, AGI);

UPD: assuming your SIP acoounts DB has moved from Asterisk to OpenSIPS cluster, its desirable not just processing REGISTER requests and doing authentication, but also adding caller ids to your SIP accounts.