Hi,

Thanks for the help, here's what I dug up:

The users are visible in Kamailio, output of kamcmd ul.dump: (here 1.1.1.1 is the public ip of my Kamailio+Asterisk server and 2.2.2.2 is the public ip of my home network)

# kamcmd ul.dump { Domain: location Size: 512 AoRs: { AoR: 661@testers.com HashID: 371821163 Contacts: { Contact: { Address: sip:661@192.168.0.106:5062 ;transport=udp Expires: 3588 Q: -1.000000 Call-ID: 56c288b61a0baf63@192.168.0.106 CSeq: 20002 User-Agent: Grandstream GXP2000 1.2.5.3 Received: sip:2.2.2.2:5062 Path: [not set] State: CS_SYNC Flags: 0 CFlags: 64 Socket: udp:1.1.1.1:5060 Methods: 8159 Ruid: uloc-535276f9-1d83-2 Instance: [not set] Reg-Id: 0 Last-Keepalive: 1397984187 Last-Modified: 1397984187 } } AoR: 660@testers.com HashID: 371820875 Contacts: { Contact: { Address: sip:660@192.168.0.101:53928 ;rinstance=6ca94e284d15eb27;transport=TCP Expires: 3224 Q: -1.000000 Call-ID: YzQ5YzZlMGI4NTVjNmE3Y2JlMWYzNGI0ODhlMDRmYmI. CSeq: 12 User-Agent: Z 3.2.21357 r21367 Received: sip:2.2.2.2:1895;transport=TCP Path: [not set] State: CS_SYNC Flags: 0 CFlags: 64 Socket: tcp:1.1.1.1:5060 Methods: 5087 Ruid: uloc-535276f9-1d9e-2 Instance: [not set] Reg-Id: 0 Last-Keepalive: 1397983823 Last-Modified: 1397983823 } } } Stats: { Records: 2 Max-Slots: 1 } }

Yes, the REGISTER message reaches Asterisk and it responds with 200 OK, here is the SIP trace:

T 2.2.2.2:1895 -> 1.1.1.1:5060 [AP] REGISTER sip:testers.com;transport=TCP SIP/2.0. Via: SIP/2.0/TCP 192.168.0.101:53928 ;branch=z9hG4bK-d8754z-efb8733fbbad4b7c-1---d8754z-. Max-Forwards: 70. Contact: sip:660@192.168.0.101:53928 ;rinstance=2b8a38b85fa9eb69;transport=TCP. To: sip:660@testers.com;transport=TCP. From: sip:660@testers.com;transport=TCP;tag=83057513. Call-ID: OTdkNGY5ZDIxYzk5NjQ2NmNlNzU3MWMzZDIwZGQ0YTY.. CSeq: 1 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE. Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri. User-Agent: Z 3.2.21357 r21367. Allow-Events: presence, kpml. Content-Length: 0. .

T 1.1.1.1:5060 -> 2.2.2.2:1895 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 192.168.0.101:53928 ;branch=z9hG4bK-d8754z-efb8733fbbad4b7c-1---d8754z-;rport=1895;received=2.2.2.2. To: sip:660@testers.com ;transport=TCP;tag=9bf08f91615a4a194285ad1308f058f3.c9e3. From: sip:660@testers.com;transport=TCP;tag=83057513. Call-ID: OTdkNGY5ZDIxYzk5NjQ2NmNlNzU3MWMzZDIwZGQ0YTY.. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="testers.com", nonce="U1OO1VNTjalLQmkJEOAWOaLqbTEALvG7". Content-Length: 0. .

T 2.2.2.2:1895 -> 1.1.1.1:5060 [AP] REGISTER sip:testers.com;transport=TCP SIP/2.0. Via: SIP/2.0/TCP 192.168.0.101:53928 ;branch=z9hG4bK-d8754z-2433219a229ba04d-1---d8754z-. Max-Forwards: 70. Contact: sip:660@192.168.0.101:53928 ;rinstance=2b8a38b85fa9eb69;transport=TCP. To: sip:660@testers.com;transport=TCP. From: sip:660@testers.com;transport=TCP;tag=83057513. Call-ID: OTdkNGY5ZDIxYzk5NjQ2NmNlNzU3MWMzZDIwZGQ0YTY.. CSeq: 2 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE. Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri. User-Agent: Z 3.2.21357 r21367. Authorization: Digest username="660",realm="testers.com ",nonce="U1OO1VNTjalLQmkJEOAWOaLqbTEALvG7",uri="sip:testers.com ;transport=TCP",response="ca517e546b71ac50d94fe34032d36e7d",algorithm=MD5. Allow-Events: presence, kpml. Content-Length: 0. .

T 1.1.1.1:5060 -> 2.2.2.2:1895 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 192.168.0.101:53928 ;branch=z9hG4bK-d8754z-2433219a229ba04d-1---d8754z-;rport=1895;received=2.2.2.2. To: sip:660@testers.com ;transport=TCP;tag=9bf08f91615a4a194285ad1308f058f3.9743. From: sip:660@testers.com;transport=TCP;tag=83057513. Call-ID: OTdkNGY5ZDIxYzk5NjQ2NmNlNzU3MWMzZDIwZGQ0YTY.. CSeq: 2 REGISTER. Contact: sip:660@192.168.0.101:53928 ;rinstance=2b8a38b85fa9eb69;transport=TCP;expires=3600;received="sip:2.2.2.2:1895 ;transport=TCP". Content-Length: 0. .

U 1.1.1.1:5060 -> 127.0.0.1:5070 REGISTER sip:127.0.0.1:5070 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK9b6c.4c9fba53000000000000000000000000.0. To: sip:660@127.0.0.1. From: sip:660@127.0.0.1;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-13ac. CSeq: 10 REGISTER. Call-ID: 7bffedd7-7580@1.1.1.1. Max-Forwards: 70. Content-Length: 0. Contact: sip:660@127.0.0.1:5060. Expires: 3600. .

U 1.1.1.1:5070 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK9b6c.4c9fba53000000000000000000000000.0;received=1.1.1.1. From: sip:660@127.0.0.1;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-13ac. To: sip:660@127.0.0.1;tag=as149e59f0. Call-ID: 7bffedd7-7580@1.1.1.1. CSeq: 10 REGISTER. Server: Asterisk PBX 11.8.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Expires: 3600. Contact: sip:660@127.0.0.1:5060;expires=3600. Date: Sun, 20 Apr 2014 09:04:41 GMT. Content-Length: 0. .

cheers, Olli

2014-04-20 11:30 GMT+03:00 Mikko Lehto mslehto@iki.fi:

2014-04-19 (Sat) 20:46 UTC +0300 Olli Heiskanen < ohjelmistoarkkitehti@gmail.com>:

...

Weird thing is the client looks registered but I'm not sure if it really

is

...

registered. If I'm not mistaken I should see the peers when I issue 'sip show peers' on asterisk cli. Instead I get this:

Hi Olli

Some pointers: Can you see users on Kamailio usrloc? Can you see REGISTER requests arriving to Asterisk? Are there any errors on Asterisk when REGISTER request is handled?

You should add some xlog() to AUTH, REGISTRAR and REGFWD routes in Kamailio.

Also, maybe you could provide these outputs: kamcmd ul.lookup location 660@testers.com kamcmd ul.lookup location 661@testers.com kamcmd ul.dump ngrep -d any -W byline -q port 5060 or port 5070

Regards

Mikko Lehto

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Try to install sngrep In both hosts (Kamailio and asterisk). It will help you figure what is happening more clear.

I'll double check my own config, and Let you know the needed fields, At least for my case.

I used the same integration guide, and splitted the model in 3 servers. One for kamailio, one for databases and one for media server (asterisk).

It's now hosting 350 users with an avg of 15 concurrent calls, planning to take it to 1200 users in a course of 6 months. El abr 23, 2014 8:29 AM, "Olli Heiskanen" ohjelmistoarkkitehti@gmail.com escribió:

Hello,

Gracias Pedro, kiitos Mikko.

It's good to know I have configured Kamailio correctly. I added the type into my table but so far no luck having asterisk see the clients registered, at least on cli. I do see that asterisk adds registration data into the table. I'll work on this for a bit and ask in the asterisk list on more tricks on asterisk side. I'll post back here if I find out what the problem was, in case someone is having similar issues.

Thanks again, Olli

2014-04-22 21:06 GMT+03:00 Pedro Niño nino.pedro@gmail.com:

...

Don't forget to include peer type (friend), and The callbacknumber In The table.

It happened to me and asterisk/kamailio behavior was wayyy to weird until made sure both parameters were there.

---

In this setup I have SIP peers in an asterisk table added like this:

INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', ' testers.com');

---

El abr 19, 2014 1:17 PM, "Olli Heiskanen" < ohjelmistoarkkitehti@gmail.com> escribió:

...

Hello,

One of the tests I've been working with is Asterisk realtime integration according to Daniel's guide here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

Weird thing is the client looks registered but I'm not sure if it really is registered. If I'm not mistaken I should see the peers when I issue 'sip show peers' on asterisk cli. Instead I get this:

*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]

Also, calling between clients will fail; in Asterisk cli I get: *CLI> == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [661@default:1] NoOp("SIP/660-00000000", "Testing: Dialed 661") in new stack -- Executing [661@default:2] Dial("SIP/660-00000000", "SIP/661,3600,rt") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/661 == Everyone is busy/congested at this time (1:0/0/1) -- Executing [661@default:3] Hangup("SIP/660-00000000", "") in new stack == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000000'

In this setup I have SIP peers in an asterisk table added like this: INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', ' testers.com');

I have Kamailio and Asterisk on the same machine where Kamailio listens port 5060 and Asterisk listens 5070. Things that differ from the guide are Kamailio and Asterisk versions, which in my case are newer. Also, for another testing case I have MULTIDOMAIN enabled in Kamailio, does this interfere with the realtime integration? I'm using only one domain though.

Please let me know if any configs or traces I can provide will help figure out what's going on.

cheers, Olli

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Hello,

It took me a while to get forward on this, but I had progress. I've changed my sip.conf back and forth so I can't name the exact cause for my problem, but it may have been the fact that in my asterisk sippeers table the fields permit and deny may have been in the wrong order. And/or some configuration values in sip.conf.

So now clients can register and asterisk 'sip show peer' shows the registered clients.

However, there is still one thing that's probably not quite there yet. I'm using the domain 'testers.com' for my clients, but I can't register them using that domain. I was able to get clients to register and visible to asterisk only by using domain '127.0.0.1', if I try commenting that out, asterisk will say: chan_sip.c:28073 handle_request_register: Registration from '< sip:660@127.0.0.1>' failed for '1.1.1.1:5060' - Not a local domain (where 1.1.1.1 is the public ip of the asterisk+kamailio box)

In my sip.conf I have domains defined like this: autodomain=no domain=127.0.0.1 domain=testers.com

I think this may be the cause for this behavior: In my kamailio.cfg I have asterisk and kamailio bindips defined like this: asterisk.bindip = "127.0.0.1" desc "Asterisk IP Address" asterisk.bindport = "5070" desc "Asterisk Port" kamailio.bindip = "127.0.0.1" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port"

And this route forwards REGISTER messages to asterisk using the 127.0.0.1 as domain:

route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER";

$uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";

if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n";

uac_req_send(); }

So question is, what would be the good-practice way to fix my setup into using the client's domain? I thought about using the domain 'testers.com' in place of kamailio.bindip but was unable to build the sip message and send it to kamailio ip. uac_req_send() seems to send the message to what is defined in the request line of the message so replacing it with 'testers.com' would not work.

cheers, Olli

2014-04-23 17:31 GMT+03:00 Pedro Niño nino.pedro@gmail.com:

Ahhh also, don't forget to place the *rtcachefriends*=*yes* in your sip.conf (asterisk) to show the realtime peers El abr 23, 2014 8:29 AM, "Olli Heiskanen" ohjelmistoarkkitehti@gmail.com escribió:

Hello,

...

Gracias Pedro, kiitos Mikko.

It's good to know I have configured Kamailio correctly. I added the type into my table but so far no luck having asterisk see the clients registered, at least on cli. I do see that asterisk adds registration data into the table. I'll work on this for a bit and ask in the asterisk list on more tricks on asterisk side. I'll post back here if I find out what the problem was, in case someone is having similar issues.

Thanks again, Olli

2014-04-22 21:06 GMT+03:00 Pedro Niño nino.pedro@gmail.com:

...

Don't forget to include peer type (friend), and The callbacknumber In The table.

It happened to me and asterisk/kamailio behavior was wayyy to weird until made sure both parameters were there.

---

In this setup I have SIP peers in an asterisk table added like this:

INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', ' testers.com');

---

El abr 19, 2014 1:17 PM, "Olli Heiskanen" < ohjelmistoarkkitehti@gmail.com> escribió:

...

Hello,

One of the tests I've been working with is Asterisk realtime integration according to Daniel's guide here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

Weird thing is the client looks registered but I'm not sure if it really is registered. If I'm not mistaken I should see the peers when I issue 'sip show peers' on asterisk cli. Instead I get this:

*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]

Also, calling between clients will fail; in Asterisk cli I get: *CLI> == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [661@default:1] NoOp("SIP/660-00000000", "Testing: Dialed 661") in new stack -- Executing [661@default:2] Dial("SIP/660-00000000", "SIP/661,3600,rt") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/661 == Everyone is busy/congested at this time (1:0/0/1) -- Executing [661@default:3] Hangup("SIP/660-00000000", "") in new stack == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000000'

In this setup I have SIP peers in an asterisk table added like this: INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', ' testers.com');

I have Kamailio and Asterisk on the same machine where Kamailio listens port 5060 and Asterisk listens 5070. Things that differ from the guide are Kamailio and Asterisk versions, which in my case are newer. Also, for another testing case I have MULTIDOMAIN enabled in Kamailio, does this interfere with the realtime integration? I'm using only one domain though.

Please let me know if any configs or traces I can provide will help figure out what's going on.

cheers, Olli

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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Hello,

Thanks for your suggestion, unfortunately it had no effect on the outcome.

This (using asterisk-kamailio integration with a domain specified for clients) must have been achieved before, I wonder if I'm doing something wrong here, or is this just not doable?

Thanks, Olli

2014-05-18 21:29 GMT+03:00 VOIP Tests kamailio.fs@gmail.com:

Try updating your /etc/hosts file with the domain 'testers.com'.

Arun

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