sr-users September 2010

sr-users@lists.kamailio.org

58 participants
97 discussions

Re: [SR-Users] Asterisk Load balancing and Session Aware
by Alex Balashov 30 Sep '10

30 Sep '10

Ross, On 09/30/2010 10:14 AM, Ross Beer wrote: > The phone sets up another call using INVITE and then uses REFER when > the transfer takes place. Please try to keep the list copied, as a matter of good practice. If I understood your scenario correctly, you don't have a choice but to send the REFER to the same Asterisk server as the one hosting the original call leg, because that is the only server that is aware of that call leg. Perhaps I have not adequately understood. However, if your question is about whether Kamailio can initiate transfers, it can, sort of. You can initiate REFERs and INVITE-hold-REFER-BYE bridges from route script a la: http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2700697 http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2700815 You can also use the management interface to stimulate Kamailio to initiate a 'bridge' of this nature from an outside script: http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2701383 If there is anything that might serve your needs, I think it is that. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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Asterisk Load balancing and Session Aware
by Ross Beer 30 Sep '10

30 Sep '10

Hi, I would like to know if the following is possible in Kamailio, I've tried with OpenSIPs but I don't think it is ideal for my needs. I would like to load balance multiple asterisk boxes which terminate and originate calls. To transfer calls by attended transfer any new calls originating from a phone need to be sent to the same server as the held call. With the dialogue module I can add the call originating from asterisk to a profile and the new call from the phone can check if the user already belongs to a profile and then send the call to the same gateway. What I would like to know is if there is a better way to do this or if Kamailio can perform the transfer without the need to send the call back to the same asterisk box. I've noticed that the Kamailio dialogue module has a few more features than its OpenSips counterpart. I would be very grateful for any feedback. Regards, Ross

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tm problem with two copies of the same invite
by Juha Heinanen 30 Sep '10

30 Sep '10

there is a broken grandstream sip ua that sometimes sends two initial invites back-to-back. according to wireshark, sr received them about 10 microseconds apart. sr then forwards the request to pstn gw, but i don't have wireshark dump of that side of the traffic. anyway, from the ua side i see that about 10 milliseconds after receiving the invites, sr replies to the ua "503 no more gateways available". in my config that reply comes only from GW_FAILURE failure route, which is set up like this: t_on_failure("GW_FAILURE"); t_set_fr("120000", "5000"); t_relay() so no way should failure route get executed before 5 seconds, but in this back-to-back invite case it somehow gets executed almost immediately. i have not seen this behavior in other situation. this looks like a bug to me. perhaps there is some race condition causing it? version of sr where this appears is 3.0.3. -- juha

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Multiple Servers Serving A Domain
by Jon Farmer 30 Sep '10

30 Sep '10

Hi, I currently have 1 OpenSer server in front of a number of Asterisk boxes. I want to move to a scenario where I have multiple OpenSer boxes serving the same domain name in front of the Asterisk boxes. The UA could be registered on either of the OpenSer via SRV. When Asterisk has a call for the UA is it better to get Asterisk to dial both OpenSER or get one OpenSER to fork the call to the other box? Any opinions please? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

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LCR "weight" column question
by Ricardo Martinez 29 Sep '10

29 Sep '10

Hello. I’m wondering how the “Weight” column in the LCR ‘gw’ table Works. I have 5 gateways defined with different weights. gw_name grp_id ip_addr hostname port uri_scheme transport strip tag weight ping flags GW02 58 10.0.0.71 5060 1 1 4 0258 10 0 0 GW03 58 10.0.0.72 5060 1 1 4 0258 10 0 0 GW05 58 10.0.0.132 5060 1 1 4 0258 20 0 0 GW06 58 10.0.0.133 5060 1 1 4 0258 20 0 0 GW09 58 10.0.0.70 5060 1 1 4 0258 20 0 0 GW10 58 10.0.0.69 5060 1 1 4 0258 20 0 0 If I have 100 calls, I was expecting was to have 20 to GW05, 20 to GW06, 20 to GW09 and 20 to GW10 Finalle 10 to GW02 and 10 to GW02. Is this the correct interpretation of this parameter? Thanks in advance. Regards, Ricardo Martinez.-

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Problem with ser - INVITE size problem tcp->udp?
by Simone Felici 29 Sep '10

29 Sep '10

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). The continuous groving up of our infrastructure and using even more codecs, causes the INVITE (udp) to be over 1500bytes. An external->incoming call to our proxy sip comes in with a size of ~1300 and will be forwarded to the end device (sua) over 1500. Ser is adding more than 200bytes in the udp packet. I can see 'INVITE' from ser to customer, but the INVITE is not answerred due the size. I'm sure about that, because in certain situations, deleting codec list the packet-size goes under the 1500bytes and all works. To main issue is from external calls to ser server to our customers. Calls started from customers to ser and then out are all ok. Now the question: Could the passage from UDP to TCP solve the issue? Could someone give me an example and how implement it into our ser configuration? The atteched configuration, ser on a mysql basis, is working well in expect the issue described. Thank's a lot for your help! Simon

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Dialog timeout route
by Alex Balashov 29 Sep '10

29 Sep '10

Greetings, Since the move to name-based routes in Kamailio v3.0, what implications does this have for the 'timeout_route' dialog context attribute? Has this been updated to reflect that change, or do I still have to use numerical routes for it? Can I now set this as a named constant, e.g. $dlg_ctx(timeout_route) = DIALOG_TIMEOUT_ROUTE; ... route[DIALOG_TIMEOUT_ROUTE] { ... } or is it a string? $dlg_ctx(timeout_route) = "DIALOG_TIMEOUT_ROUTE"; Thanks! -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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sip-router 3.0.99 newbe question
by Rouskol Andrey 28 Sep '10

28 Sep '10

Hello, Could somebody check if default ser.cfg file is correct in the following section: route[AUTHENTICATION] { .... # check if the UID from the authentication meets the From header $authuid = $uid; if (!lookup_user("$fu.uid", "@from.uri")) { xlog("L_INFO","fu.uid lookup failed\n"); del_attr("$uid"); } if ($fu.uid != $fr.authuid) { sl_reply("403", "Fake Identity"); drop; } .. Because it didn't work for me till I've replaced: if (!lookup_user("$fu.uid", "@from.uri")) { with: if (!lookup_user("$fr.uid", "@from.uri")) { Best regards, Andrey

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Siremis v2.0 preview
by Elena-Ramona Modroiu 28 Sep '10

28 Sep '10

Hi, Siremis version 2.0 is planned to work with upcoming Kamailio or SER 3.1 and by now is alpha version. If you want to have a preview, then you can access it at: http://demo.asipto.com/siremis20/ username: guest password: guest v2.0 brings a completely new face and many features, reusing the latest openbiz development framework, among them: - easy 4 step web-based installer - siremis account management (including roles per users) - customizable menu from web interface - customizable views from web interface - rework of many kamailio/ser related components such as subscriber, siptrace, generic and usrloc charts Siremis administration module, user profile management and web based customization elements are disabled in the demo, to avoid breaking the content. Packaging is still under work, but if someone is good at php/xml, likes to dig in by itself to sort out issues and wants to help with testing, I can ship a raw tarball, just drop me an email. Feedback about this new version is also appreciated, in order to improve it. Regards, Ramona

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Kamailio --> Mitel (Not Found) ¿INVITE ISSUES?
by Tincho ylm 27 Sep '10

27 Sep '10

Hi all! I really don't know why "Mitel" rejects my calls. I'm using Kamailio to forward calls to Mitel. A little more graphic: Please see the picture: http://s3.subirimagenes.com:81/otros/5226539form.jpg SIP PHONE (Linksys) ---> Kamailio (1.5.4) ----> Mitel ----> Mitel Phone Mitel rejects my calls with "404 Not Found". Ok, you may think: "the extension that you are calling doesn't exists".. please dont think that. (One more thing: If I try to make the same scene using Asterisk instead Kamailio everything works fine.) So, I made a sip capture to see what happens: Sip Phone -> 100 192.168.10.140 -> Sip Phone 192.168.10.150 -> Kamailio 192.168.10.160 -> Mitel Mitel Phone -> 200 Kamailio U 192.168.10.140:5060 -> 192.168.10.150:5060 INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Max-Forwards: 70. Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. Expires: 240. User-Agent: Linksys/SPA941-5.1.8. Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. U 192.168.10.150:5060 -> 192.168.10.140:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.10.140:5060 ;branch=z9hG4bK-d063d53a;rport=5060;received=192.168.10.140. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Server: Kamailio (1.5.4-notls (i386/linux)). Content-Length: 0. U 192.168.10.150:5060 -> 192.168.10.160:5060 INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0. Via: SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Max-Forwards: 69. Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. Expires: 240. User-Agent: Linksys/SPA941-5.1.8. Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> >;tag=0_4044193584-65506210. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Content-Length: 0. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> >;tag=0_4044193584-65506210. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Contact: <sip:192.168.10.160>. Content-Length: 0. This is my Kamailio code from reenvites.. route[4] { t_relay("udp:192.168.10.160:5060"); t_on_reply("1"); exit; } If you pay attention to INVITES (Kamailio SIP messages) you will see: From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. I think that should be: From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160>>. It could be the reason for Mitel rejects? Can I fix it? I can use TEXTOPS but I cant understand why Mitel rejects the Kamailio INVITES. I will thanks any help!

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sr-users September 2010