June 2012 - sr-users - kamailio.org

Re: [SR-Users] simple test case: add user/delete user on kamailio 3.3 presence server
by Daniel-Constantin Mierla 20 Jun '12

20 Jun '12

Hello, On 6/19/12 4:49 PM, Min Wang wrote: > Hi > > Here is another similar case which does not work. maybe it is related > the above issue as well. > > Jitsi 1.1 client: 101/103 , with xcap as storage > proxy: kamailio 3.3 > ip network is: 192.168.122.0 > > the setup is like the following: > > 101(.224) --- proxy (.32) --- 103 (.1) > > After all presence related db are truncated, kamailio was restart. > then bring 101,103 online. > > On 101 add 103, 103 prompt up authorization window, click ok. > now on 103, I can see 101 online status > > but on 101, I can not see 103 status <----the first trace file: > the jitsi-101-add-103-confirm-2.pcap > > (it is grey, even manually change 103 status from online to away > etc, on 101, the 103 always showed grey. <--- second trace file: > jitsi-101-103-change.pcap) > > Please see the attached pcap files. > > > the watcher table shows: > > id presentity_uri watcher_username watcher_domain > event status reason inserted_time > 1 sip:103@192.168.122.32 101 192.168.122.32 presence > 2 1340114579 > 2 sip:101@192.168.122.32 103 192.168.122.32 presence > 1 1340114582 the first record above shows that the status is 2, which means 'pending'. The subscription of 101 was not accepted yet by presentity 103. For the second record, status is 1, which means active subscription. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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simple test case: add user/delete user on kamailio 3.3 presence server
by Min Wang 20 Jun '12

20 Jun '12

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rtimer question
by Alex Balashov 20 Jun '12

20 Jun '12

Hi, I use 'rtimer' and 'mqueue' to asynchronously defer certain heavy DB tasks on which call processing does not depend. I was hoping to get a clarification on the exact behaviour of 'rtimer' when tasks don't complete as fast as they should. Let's say that I set the rtimer route interval to half a second (500 ms), as afforded by new Kamailio 3.3.0 capabilities. What happens if the tasks I am running inside the rtimer route don't complete in that interval? Is that rtimer thread forceably terminated, or is it allowed to run as long as necessary to complete the task? Is there an absolute timeout? If the latter, I assume that another, parallel rtimer route will be started in the background after 500 ms. I am given to understand that there is a thread pool. How big is it? Can I configure its size anywhere? Is there a practical limit? Finally, is there 100% assurance that 'mqueue' is safe to consume for multiple rtimer processes, possibly ones tripping over each other due to unpredictable delays? Or is it designed only to provide blocking mutex for data passed from a SIP worker thread to a different kind of thread? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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kamailio core request dropping - no reply
by Uri Shacked 20 Jun '12

20 Jun '12

Hi, I am testing kamailio replies when an INVITE or another request arrives with lets say, "VIA" header missing.... The core drops the request. But, there is no reply for the originator (so it keep on resending the request...) Why? Uri

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Siremis routing support does not include drouting
by Yufei Tao 20 Jun '12

20 Jun '12

Hello Seems Siremis 3.2.1 doesn't support configuration of drouting. Any plans that it will in the future? I hope to use it to make my drouting configuration easier. What I want to do is to route prefixed numbers to external number at external domains, depending on the prefix (numbers+alphabets+dot and maybe+underscore+hyphen as well). Maybe I should use PDT module instead as it is supported by Siremis? Thanks! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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Re: [SR-Users] Alias possible for presence...
by Daniel-Constantin Mierla 20 Jun '12

20 Jun '12

Hello, On 6/14/12 10:28 AM, Gertjan Wolzak wrote: > > Hello All, > > I have the following challenge. > > We are using kamailio 3.2 and have enabled presence. Which works fine. > > But we would like to be able to work with aliases, so that the account > name and "presentation" name can be different. > > Any Ideas... Or did I miss the documentation on that? > the account name is public or not? If it is some internal id, eventually is used for authentication, but for presence the contacts should know the public id. Or maybe you can give some example to understand how you use aliases in the context of presence... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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Re: [SR-Users] simple test case: add user/delete user on kamailio 3.3 presence server
by Min Wang 19 Jun '12

19 Jun '12

HI Since my previous email has attachment , it is still waiting for the approval. So here is again the text part ( please see the email below) According to the trace, here is the call flow for this case: 101(.224) --- proxy (.32) --- 103 (.1) 101/103 are jitsi 1.1 nightly build client, the configure is to use SIMPLE/xcap on clients. the call flow for 101 adding 103 as contact 101 proxy 103 | | | user-add-103 | | | | | | xcap put pres/rls | | |--------------------------> | | | | | | SUBSCRIBE (presence/103) | | |---------------------------> | | | 202 ok | | |<----------------------------| | | NOTIFY( presenc/pending) | | |<----------------------------| | | | NOTIFY ( winfo/101 pending) | | |---------------------------------->| | 200 ok | | |--------------------------> | | | | 200 OK | | |<----------------------------------| | | auth window pop | | user accept | | SUBSRIBE(presence/101) | | |<----------------------------------| | | 202 OK | | |---------------------------------->| | NOTIFY (winfo,103 active) | | |<-------------------------- | | | 200 ok | | |--------------------------> | | | | NOTIFY (presence/101) | | |---------------------------------> | | | 200 OK | | |<----------------------------------| | NOTIFY (presence/103 open) | | |<-------------------------- | | ****** ( missing part) | 200 ok | | |--------------------------> | | As on the 103, auth windows pop up, then click ok to accept (thus it will added 101 as contact), in this case it does not do any xcap operation? Is it the correct call flow? Thanks min On 06/19/2012 04:49 PM, Min Wang wrote: > Hi > > Here is another similar case which does not work. maybe it is related > the above issue as well. > > Jitsi 1.1 client: 101/103 , with xcap as storage > proxy: kamailio 3.3 > ip network is: 192.168.122.0 > > the setup is like the following: > > 101(.224) --- proxy (.32) --- 103 (.1) > > After all presence related db are truncated, kamailio was restart. > then bring 101,103 online. > > On 101 add 103, 103 prompt up authorization window, click ok. > now on 103, I can see 101 online status > > but on 101, I can not see 103 status <----the first trace file: the > jitsi-101-add-103-confirm-2.pcap > > (it is grey, even manually change 103 status from online to away > etc, on 101, the 103 always showed grey. <--- second trace file: > jitsi-101-103-change.pcap) > > Please see the attached pcap files. > > > the watcher table shows: > > id presentity_uri watcher_username watcher_domain > event status reason inserted_time > 1 sip:103@192.168.122.32 101 192.168.122.32 > presence 2 1340114579 > 2 sip:101@192.168.122.32 103 192.168.122.32 > presence 1 1340114582 > > the active wacher table show: > > id presentity_uri watcher_username watcher_domain > to_user to_domain event event_id to_tag from_tag > callid local_cseq remote_cseq contact record_route > expires status reason version socket_info > local_contact from_user from_domain updated updated_winfo > 1 sip:101@192.168.122.32 101 192.168.122.32 > 101 192.168.122.32 message-summary > a6a1c5f60faecf035a1ae5b6e96e979a-7e2e 8b6ca8e7 > 9e8288d04d0dab18fda28f5fb24b40d8(a)0.0.0.0 1 2 > sip:101@192.168.122.224:5060;transport=udp;registe... > 1340117922 1 1 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 101 192.168.122.32 -1 -1 > 2 sip:101@192.168.122.32 101 192.168.122.32 > 101 192.168.122.32 presence.winfo > a6a1c5f60faecf035a1ae5b6e96e979a-8a99 9975004d > 9b8cb0735bc559232639aa28c16a6f4b(a)0.0.0.0 2 2 > sip:101@192.168.122.224:5060;transport=udp;registe... > 1340117921 1 2 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 101 192.168.122.32 -1 -1 > 3 sip:103@192.168.122.32 103 192.168.122.32 > 103 192.168.122.32 message-summary > a6a1c5f60faecf035a1ae5b6e96e979a-0117 9d873afb > 30fa1ce8cca7a7c50fea11e688ec3d5e@0:0:0:0:0:0:0:0 1 2 > sip:103@192.168.122.1:5060;transport=udp;registeri... > 1340117959 1 1 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 103 192.168.122.32 -1 -1 > 4 sip:103@192.168.122.32 103 192.168.122.32 > 103 192.168.122.32 presence.winfo > a6a1c5f60faecf035a1ae5b6e96e979a-9a71 101d721c > 148be0ec1a4acc087c0083324398cf14@0:0:0:0:0:0:0:0 2 2 > sip:103@192.168.122.1:5060;transport=udp;registeri... > 1340117959 1 2 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 103 192.168.122.32 -1 -1 > 5 sip:103@192.168.122.32 101 192.168.122.32 > 103 192.168.122.32 presence > a6a1c5f60faecf035a1ae5b6e96e979a-5ab7 e9099546 > 6ed81a0fa2042996295974b671a8074f(a)0.0.0.0 1 2 > sip:101@192.168.122.224:5060;transport=udp;registe... > 1340118179 2 1 udp:192.168.122.32:5060 > sip:192.168.122.32:5060;transport=udp 101 192.168.122.32 -1 -1 > 6 sip:101@192.168.122.32 103 192.168.122.32 > 101 192.168.122.32 presence > a6a1c5f60faecf035a1ae5b6e96e979a-98a8 > > > > > thanks > > min > > On 06/19/2012 01:00 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> you have to share the content of the messages sent, providing just >> the url is not enough to see what happens. >> >> watchers table is for the list of contacts and their global status in >> regard to presentity, active_watchers is for keeping the presence >> dialogs. >> >> Cheers, >> Daniel >> >> On 6/19/12 12:47 PM, Min Wang wrote: >>> Hi >>> >>> (1) I tested more. It is due to the difference between bria 3 and >>> jitsi while handling the adding/removing contact: >>> >>> if contact is added/removed: >>> >>> for jitsi, it send out: >>> >>> PUT /xcap-root/resource-lists/users/sip:101@192.168.122.32/index >>> PUT /xcap-root/pres-rules/users/sip:101@192.168.122.32/presrules >>> >>> >>> for bria 3, it only send out >>> >>> PUT >>> /xcap-root/resource-lists/users/101(a)192.168.122.32/contacts-resource-list.xml >>> HTTP/1.0. >>> PUT >>> /xcap-root/resource-lists/users/101(a)192.168.122.32/contacts-resource-list.xml >>> >>> >>> it does NOT send out the pres-rules. >>> >>> Is it the good behavior ( from RFC point of view) ? >>> should I call the pres_update_watchers even though its is for >>> resource-lists? >>> >>> (2) BTW, what is the exact purpose of watcher table? >>> >>> e.g: on 101, if delete 103, watcher table become: >>> >>> id presentity_uri watcher_username >>> watcher_domain event status reason inserted_time >>> 1 sip:103@192.168.122.32 101 192.168.122.32 >>> presence 1 1340096051 >>> 2 sip:101@192.168.122.32 103 192.168.122.32 >>> presence 3 terminated 1340096181 >>> >>> it is item 2 become terminated instead of item 1. If watcher >>> table is for subscription, then seems item 1 should be terminated? >>> Look at the code, it seems somehow serve as mixed role of >>> subscription/auth/xcap purpose, not for subscription state? >>> >>> and is the active_watcher mainly for the state machine of >>> http://tools.ietf.org/html/rfc3857#section-4.7.1? >>> >>> thanks. >>> >>> >>> min >>> >>> >>> >>> >>> >>> >>> >>> >>> On 06/19/2012 10:41 AM, Daniel-Constantin Mierla wrote: >>>> Hello, >>>> >>>> On 6/18/12 2:11 PM, Andreas Granig wrote: >>>>> Hi, >>>>> >>>>> On 06/15/2012 05:25 PM, Min Wang wrote: >>>>>> | 1 | sip:103@192.168.122.32 | 101 | 192.168.122.32 | >>>>>> presence | 2 | | 1339772803 | >>>>>> >>>>>> then I deleted the 103 from the contact list, the watcher >>>>>> table still >>>>>> shows the same. >>>>> For local storage, I'd expect an unsubscribe (subscribe with >>>>> Expires=0) >>>>> from 101 to 103. The obvious work-flow would be to set it to >>>>> "terminated" in watchers table, and in case of a subsequent >>>>> re-subscribe >>>>> it should be changed back to "pending" state, although the >>>>> state-machine >>>>> doesn't indicate a transition from "terminated" back to "pending". >>>>> Isn't >>>>> this the case? >>>>> >>>>> For xcap storage, there are other ways to block/remove a contact >>>>> on/from >>>>> the list. As Iñaki pointed out in >>>>> http://lists.opensips.org/pipermail/devel/2009-August/003868.html, >>>>> the >>>>> xcap server needs to notify the sip server about the change, which in >>>>> turn will notify the other party (103) that it's no longer allowed to >>>>> see 101's state. If the xcap_server module of kamailio is used, >>>>> there is >>>>> the following code snippet in some examples floating around on the >>>>> internet: >>>>> >>>>> >>>>> switch($rm) { >>>>> case "PUT": >>>>> xcaps_put("$var(uri)", "$hu", "$rb"); >>>>> if($xcapuri(u=>auid)=~"pres-rules") { >>>>> pres_update_watchers("$var(uri)", "presence"); >>>>> pres_refresh_watchers("$var(uri)", "presence", 1); >>>>> } >>>>> >>>>> So, shouldn't this update the watchers accordingly? Anyways, also in >>>>> this case the watcher state should change to "terminated", and in >>>>> case >>>>> of a re-subscribe it should go back to pending if xcap rules are >>>>> allowing this. >>>>> >>>>> Maybe someone with good xcap_server/presence insights could >>>>> elaborate on >>>>> that? >>>> to summarize, you say that when the contact is removed from the >>>> buddy list, the watcher table is not updated to state terminated by >>>> pres_update_watchers(...)? >>>> >>>> Cheers, >>>> Daniel >>>> >>> >> >> -- >> Daniel-Constantin Mierla -http://www.asipto.com >> http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw >

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About the subscribe/200-ok contact address on kamailio 3.3
by Min Wang 19 Jun '12

19 Jun '12

HI I have tried the kamailio 3.3, user 101(10.15.20.131) send subscribe to (10.15.20.137), kamailio return 200 OK, but with contact header as: Contact: <sip:10.15.20.137:5060;transport=udp>. is it correct? Since before subscription expired, the client will send the re-subscribe using that contact address, is it right? Should kamailio return the contact as sip:101@10.15.20.137 in this case? Thanks. min ///////////////////////////// pkt captured //// U 10.15.20.131:54740 -> 10.15.20.137:5060 SUBSCRIBE sip:101@10.15.20.137 SIP/2.0. Via: SIP/2.0/UDP 192.168.122.224:54740;branch=z9hG4bK-d8754z-87f9c58f1c70cbe2-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:101@10.15.20.131:54740>. To: <sip:101@10.15.20.137>. From: "101"<sip:101@10.15.20.137>;tag=4f7671df. Call-ID: MWRjNjU2NGNmMWE0Nzk3ODU2ZTZjZWViYmViYmQ1NmY. CSeq: 2 SUBSCRIBE. Expires: 60. Accept: application/watcherinfo+xml. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Proxy-Authorization: Digest username="101",realm="10.15.20.137",nonce="T9n8u0/Z+4/OOU2Qeo4acn7OE+7pb5A1",uri="sip:101@10.15.20.137",response="93caab1f09daa88a1619bb0613b9a5ff",algorithm=MD5. Supported: eventlist. User-Agent: Bria 3 release 3.4.1 stamp 66953. Event: presence.winfo. Content-Length: 0. . # U 10.15.20.137:5060 -> 10.15.20.131:54740 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.122.224:54740;branch=z9hG4bK-d8754z-87f9c58f1c70cbe2-1---d8754z-;rport=54740;received=10.15.20.131. To: <sip:101@10.15.20.137>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-21e3. From: "101"<sip:101@10.15.20.137>;tag=4f7671df. Call-ID: MWRjNjU2NGNmMWE0Nzk3ODU2ZTZjZWViYmViYmQ1NmY. CSeq: 2 SUBSCRIBE. Expires: 60. Contact: <sip:10.15.20.137:5060;transport=udp>. Server: kamailio (3.3.0-rc0 (x86_64/linux)). Content-Length: 0. . .... U 10.15.20.131:54740 -> 10.15.20.137:5060 SUBSCRIBE sip:10.15.20.137:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.122.224:54740;branch=z9hG4bK-d8754z-59fd0765744f37a3-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:101@10.15.20.131:54740>. To: <sip:101@10.15.20.137>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-21e3. From: "101"<sip:101@10.15.20.137>;tag=4f7671df. Call-ID: MWRjNjU2NGNmMWE0Nzk3ODU2ZTZjZWViYmViYmQ1NmY. CSeq: 8 SUBSCRIBE. Expires: 60. Accept: application/watcherinfo+xml. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Proxy-Authorization: Digest username="101",realm="10.15.20.137",nonce="T9n8u0/Z+4/OOU2Qeo4acn7OE+7pb5A1",uri="sip:10.15.20.137:5060;transport=udp",response="f9ef9d95891bbe5722bb47fbe93b20e9",algorithm=MD5. Supported: eventlist. User-Agent: Bria 3 release 3.4.1 stamp 66953. Event: presence.winfo. Content-Length: 0.

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IP ToS
by Olle E. Johansson 19 Jun '12

19 Jun '12

I know that you can set the TOS for packets sent from Kamailio in the config section. I got the question if you can read the TOS in the IP packet of an incoming response or reply in the config scripts? And can you set it per message/transaction? /O

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install problem
by copycall 19 Jun '12

19 Jun '12

hello, i ran into a problem installing kamailio when i got to #7 "create mysql database": < openser.cfg.5 > /usr/local/share/man//man5/ openser.cfg.5 chmod 644 /usr/local/share/man//man5/openser.cfg.5 sed -e "s#/etc/openser/openser\.cfg#/usr/local/etc/openser/openser.cfg#g" \ -e "s#/usr/sbin/#/usr/local/sbin/#g" \ -e "s#/usr/lib/openser/modules/#/ usr/local/lib64/openser/modules/#g" \ -e "s#/usr/share/doc/openser/#/ usr/local/share/doc/openser/#g" \ < scripts/openserctl.8 > /usr/local/share/man//man8/ openserctl.8 chmod 644 /usr/local/share/man//man8/openserctl.8 sed -e "s#/etc/openser/openser\.cfg#/usr/local/etc/openser/openser.cfg#g" \ -e "s#/usr/sbin/#/usr/local/sbin/#g" \ -e "s#/usr/lib/openser/modules/#/ usr/local/lib64/openser/modules/#g" \ -e "s#/usr/share/doc/openser/#/ usr/local/share/doc/openser/#g" \ < utils/openserunix/openserunix.8 > \ /usr/local/share/man//man8/openserunix.8 chmod 644 /usr/local/share/man//man8/openserunix.8 root@copycall:/usr/local/src/openser-1.3.0/sip-server# echo $PATH /usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin root@copycall:/usr/local/src/openser-1.3.0/sip-server# cd root@copycall:~# /usr/local/etc/kamailio/kamctlrc bash: /usr/local/etc/kamailio/kamctlrc: No such file or directory root@copycall:~# /usr/local/sbin/kamdbctl create bash: /usr/local/sbin/kamdbctl: No such file or directory any solution will be greatly appreciated, please include all the code as kamailio is new to me. thank you in advance, dave

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Releasing v3.3.0 later today
by Daniel-Constantin Mierla 18 Jun '12

18 Jun '12

Hello, quick note about packaging the 3.3.0 today, therefore if anyone has commits to push to GIT branch 3.3 after 12:00GMT, please write first to sr-dev mailing list. Once the announcement about the release is sent out, the commits can be pushed again directly. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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Sipcapture and Siptrace Compile Error
by Akan 18 Jun '12

18 Jun '12

When compiling kamailio v3.3, I am getting this error from sipcapture and siptrace. I am compiling on solaris 10, 64bit Sun server with gnu tools. CC (gcc) [M sipcapture.so] sipcapture.o In file included from sipcapture.c:76:0: sipcapture.h:48:4: error: unknown type name 'u_int32_t' sipcapture.h:49:4: error: unknown type name 'u_int32_t' gmake[1]: *** [sipcapture.o] Error 1 In file included from siptrace.c:54:0: ../../modules/sipcapture/sipcapture.h:48:4: error: unknown type name 'u_int32_t' ../../modules/sipcapture/sipcapture.h:49:4: error: unknown type name 'u_int32_t' gmake[1]: *** [siptrace.o] Error 1 Thanks Nathaniel

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Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Charles Chance 18 Jun '12

18 Jun '12

Hello Daniel, Sorry for the delay! I will get around to testing with the latest master branch sometime over the next couple of days and will report back here. Many thanks, Charles Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, can you try with latest master branch -- it should be fixed by: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=1d89d7b… Cheers, Daniel On 5/29/12 5:52 PM, Daniel-Constantin Mierla wrote: Hello, finally had a chance to look over it. I guess it is some race, resulting in a memory space being overwritten. The 200ok for publish with ringing state is processed more or less at the time when 200ok for invite comes and dialog changes state. Can you re-compile with MEMDBG=1 (edit the Makefile.defs or give it in the command line parameter) and try again? Maybe the memory manager can catch the fault operation and provide more details. Cheers, Daniel On 5/17/12 6:58 PM, Charles Chance wrote: Hi Daniel, I have sent the messages over to you. I look forward to hear what you can find. Best, Charles From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 17 May 2012 15:06 To: Charles Chance Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List' Subject: Re: FW: (Devel) Segmentation fault using pua_dialoginfo Hello, the content of ps->param is pretty much all invalid. So looks like a callback on a freed/invalid parameter, could be a double-callback execution. Set debug=3 and send all the messages for such case -- it might be quite a lot of them, so send them to my email address, the mailing list has a limit of message size. Cheers, Daniel On 5/17/12 2:13 PM, Charles Chance wrote: Hi, Requested output as follows: (gdb) frame 1 #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at send_publish.c:246 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); (gdb) p *ps $2 = {req = 0x0, rpl = 0xb7c30340, param = 0xb3dc3ff4, code = 412, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = {s = 0x0, len = 0}} (gdb) p *(ua_pres_t*)(*ps->param) $3 = {id = {s = 0x20455942 <Address 0x20455942 out of bounds>, len = 980445555}, pres_uri = 0x40323531, event = 875444279, expires = 875703856, desired_expires = 858861105, flag = 808794676, db_flag = 1394618421, cb_param = 0x322f5049, next = 0xa0d302e, ua_flag = 979462486, etag = { s = 0x50495320 <Address 0x50495320 out of bounds>, len = 808333871}, tuple_id = {s = 0x5044552f <Address 0x5044552f out of bounds>, len = 775436064}, body = 0x322e3034, content_type = { s = 0x312e3134 <Address 0x312e3134 out of bounds>, len = 1648047155}, watcher_uri = 0x636e6172, call_id = { s = 0x397a3d68 <Address 0x397a3d68 out of bounds>, len = 1647593320}, to_tag = {s = 0x3435634b <Address 0x3435634b out of bounds>, len = 842149473}, from_tag = { s = 0x63306461 <Address 0x63306461 out of bounds>, len = 808334648}, cseq = 1767246349, version = 1394621025, outbound_proxy = 0x322f5049, extra_headers = 0x552f302e, record_route = { s = 0x31205044 <Address 0x31205044 out of bounds>, len = 825111097}, remote_contact = {s = 0x312e3836 <Address 0x312e3836 out of bounds>, len = 976498224}, contact = { s = 0x36373135 <Address 0x36373135 out of bounds>, len = 1701985073}} Yes, it is a test server so very happy to arrange for remote access if required. In the meantime, I will do a little more digging to try to find out why entity is null. Cheers, Charles -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5004 - Release Date: 05/16/12 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Conditionnal hunting with usrloc?
by Igor Potjevlesch 15 Jun '12

15 Jun '12

Hi all, I'm facing a problem with multiple contact in 'location' table with the same 'username'. The idea is to rewrite R-URI with 1 contact and if there is no answer or SIP 5xx, try with the second. Is that make sense? I use SER0.9.9. Regards, Igor.

3 5

Architecture
by Uriel Rozenbaum 15 Jun '12

15 Jun '12

Hey guys, I'm trying to plan a deployment for a couple thousands of users. As we are facing now some issues for scaling, I'd like to make a modular architecture in order to add a server when needed. I was thinking about N+1 Kamailio Proxies to Register customers and handle the border of the network, 1+1 Redirect servers so internal Network Elements can "find" the right Proxy to terminate a call. Customers would use DNS SRV to decide which Proxy to send a REGISTER and then send INVITES to that same proxy. As we have a private core network addressing, I'm thinking I'll have an Asterisk farm in order to perform media handling. Do you have some recommendation or guidelines for this deployment? Thanks! Uriel

3 3

AMR and AMR-WB support
by Jakson Kalsson 15 Jun '12

15 Jun '12

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the SIPX support these codecs and RFC4867 ? Also, any other Server/PBX which support AMR, AMR-WB recommended are welcome. Best regards,

3 3

add user/delete/add user again on kamailio 3.3 presence server
by Min Wang 15 Jun '12

15 Jun '12

HI I am testing the following case in the kamailio 3.3 presence server, all are in 192.168.122.0 network 101 (bria, ip:.224) ----- kamailio ps( ip:.32) --- 103 ( jitsi, ip:.1) on 101 I added 103 in the contact list, 101 <-> 103 can see each other, Then on 101, I deleted 103, thus on 103, 101 became grey, so far so good. Now I tried to add 103 back to contact list, on 101 I can see 103's status (online), but on jitsi 103, I can not see 101's status. ( grey) The watcher table showed that status terminated. +----+------------------------+------------------+----------------+----------+--------+------------+---------------+ | id | presentity_uri | watcher_username | watcher_domain | event | status | reason | inserted_time | +----+------------------------+------------------+----------------+----------+--------+------------+---------------+ | 1 | sip:103@192.168.122.32 | 101 | 192.168.122.32 | presence | 1 | | 1339752028 | | 2 | sip:101@192.168.122.32 | 103 | 192.168.122.32 | presence | 3 | terminated | 1339752031 | +----+------------------------+------------------+----------------+----------+--------+------------+---------------+ I guess it should be status=1 in this case, but somehow kamailio seems does not update the table? I have a complete packet-trace file if needed. thanks. min

1 0

drouting module: can I use non-digit prefix?
by Yufei Tao 14 Jun '12

14 Jun '12

Hi I'm using drouting module and seems that it only allows digit-only prefixes. Also all the number of digits to be stripped ('strip' defined in dr_gateways) must be all digits too. When I used non-digits in these, Kamailio crashed. Is there any way in drouting module that I can use prefixes with alphabet and dots, e.g. "gateway1."? Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

2 7

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 14 Jun '12

14 Jun '12

Hello, finally had a chance to look over it. I guess it is some race, resulting in a memory space being overwritten. The 200ok for publish with ringing state is processed more or less at the time when 200ok for invite comes and dialog changes state. Can you re-compile with MEMDBG=1 (edit the Makefile.defs or give it in the command line parameter) and try again? Maybe the memory manager can catch the fault operation and provide more details. Cheers, Daniel On 5/17/12 6:58 PM, Charles Chance wrote: > > Hi Daniel, > > I have sent the messages over to you. > > I look forward to hear what you can find. > > Best, > > Charles > > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* 17 May 2012 15:06 > *To:* Charles Chance > *Cc:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) > -Users Mailing List' > *Subject:* Re: FW: (Devel) Segmentation fault using pua_dialoginfo > > Hello, > > the content of ps->param is pretty much all invalid. So looks like a > callback on a freed/invalid parameter, could be a double-callback > execution. > > Set debug=3 and send all the messages for such case -- it might be > quite a lot of them, so send them to my email address, the mailing > list has a limit of message size. > > Cheers, > Daniel > > On 5/17/12 2:13 PM, Charles Chance wrote: > >> Hi, >> >> Requested output as follows: >> >> (gdb) frame 1 >> >> #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) >> >> at send_publish.c:246 >> >> 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); >> >> (gdb) p *ps >> >> $2 = {req = 0x0, rpl = 0xb7c30340, param = 0xb3dc3ff4, code = 412, >> flags = 0, >> >> branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = {s = 0x0, len = 0}} >> >> (gdb) p *(ua_pres_t*)(*ps->param) >> >> $3 = {id = {s = 0x20455942 <Address 0x20455942 out of bounds>, >> >> len = 980445555}, pres_uri = 0x40323531, event = 875444279, >> >> expires = 875703856, desired_expires = 858861105, flag = 808794676, >> >> db_flag = 1394618421, cb_param = 0x322f5049, next = 0xa0d302e, >> >> ua_flag = 979462486, etag = { >> >> s = 0x50495320 <Address 0x50495320 out of bounds>, len = 808333871}, >> >> tuple_id = {s = 0x5044552f <Address 0x5044552f out of bounds>, >> >> len = 775436064}, body = 0x322e3034, content_type = { >> >> s = 0x312e3134 <Address 0x312e3134 out of bounds>, len = 1648047155}, >> >> watcher_uri = 0x636e6172, call_id = { >> >> s = 0x397a3d68 <Address 0x397a3d68 out of bounds>, len = 1647593320}, >> >> to_tag = {s = 0x3435634b <Address 0x3435634b out of bounds>, >> >> len = 842149473}, from_tag = { >> >> s = 0x63306461 <Address 0x63306461 out of bounds>, len = 808334648}, >> >> cseq = 1767246349, version = 1394621025, outbound_proxy = 0x322f5049, >> >> extra_headers = 0x552f302e, record_route = { >> >> s = 0x31205044 <Address 0x31205044 out of bounds>, len = 825111097}, >> >> remote_contact = {s = 0x312e3836 <Address 0x312e3836 out of bounds>, >> >> len = 976498224}, contact = { >> >> s = 0x36373135 <Address 0x36373135 out of bounds>, len = 1701985073}} >> >> Yes, it is a test server so very happy to arrange for remote access >> if required. In the meantime, I will do a little more digging to try >> to find out why entity is null. >> >> Cheers, >> >> Charles >> > > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > > ------------------------------------------------------------------------ > > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.2176 / Virus Database: 2425/5004 - Release Date: 05/16/12 > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu

1 1

Making dialog module work for me !!
by SamyGo 14 Jun '12

14 Jun '12

Hello I've two problems regarding dialog module- used for profiling users that are in-a-call. My topology is quiet simple. SIP-USER-71141---->Kamailio------------->Asterisk[AGI]|| CALLEE<-----------------Kamailio<--------------------------------|| 1- I wish to store my custom SIP header value in profiles. I'm having trouble in figuring out on how to use Dialog_vars table in my kamailio server. I'm using dialog module to profile SIP users that are in call or making a call. It works well and the dialog maintains an entry when the call is established. Now, the difficult part is that I want my custom header to be inserted into either dialog table or dialog_vars table such that when I do a query "*is_in_profile("$avp(my_value)")*" it returns me a result. 2- Alternative to 1, I can log A,B leg separately but it doesn't erase all data on call hangup. Asterisk AGI changes the caller-id of SIP-USERA such that if it was 71141 it would be something like "007" on its way out to destination via kamailio. Now, at this point in kamailio when the call is answered by destination the dialog module saves the call data with values corresponding to "007" and not "71141". I have to disable the "dialog", "detect_spirals" in order to count this one call as two separate entities BUT problem occurs when on hangup the B-Leg gets erased from the table but the A-leg remains in DB for as long as infinity. Please help me in this. Let me know if more details are required from me. Regards, Sammy G.

2 9

Kamailio debian repository changes
by Jon Bonilla 14 Jun '12

14 Jun '12

Hi All With the upcoming release of version 3.3.0, we have updated our kamailio debian repositories. * Debian 5.0 "lenny" will no longer be supported. This means that no more nightly builds for this distribution will be triggered. We'll provide builds for stable releases of 3.0 ,3.1 and 3.2 branches though. * Ubuntu 12.04 "Precise" is now supported for nightly builds and 3.2+ versions. Next time a 3.2.x version is tagged we'll provide the build. Meanwhile you can download the nightly build of 3.2 branch for this distro. * There will be no more nightly builds of 3.1 branch. We'll provide stable builds in case another version is released though. * We now support nightly builds of branch 3.3 (same as 3.2 and master). You can test latest git versions of branch 3.3 until a stable release is built using these repositories: deb http://deb.kamailio.org/kamailio33-nightly squeeze main deb http://deb.kamailio.org/kamailio33-nightly wheezy main deb http://deb.kamailio.org/kamailio33-nightly lucid main deb http://deb.kamailio.org/kamailio33-nightly precise main All the information and repositories can be found in kamailio wiki as usual: http://www.kamailio.org/dokuwiki/doku.php/packages:debs http://www.kamailio.org/wiki/packages/debs

1 0

Dispatcher alg:7 not working accordingly
by SamyGo 14 Jun '12

14 Jun '12

Hello, I'm having trouble using algorithm 7 in dispatcher module. Here is my kamailio version. The problem is that despite multiple dispatcher entries corresponding to one set all calls go only to only one destination. [root@control1 ~]# kamailio -V version: kamailio 3.2.3 (x86_64/linux) 59f87e flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 59f87e compiled on 18:04:08 Apr 19 2012 with gcc 4.1.2 [root@control1 ~]# Here is the dipacther module params. # ------- Load-balancer params ------ modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@109.XXX.2XX.XX") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("dispatcher", "hash_pvar", "$avp(i:274)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "ds_hash_initexpire", 60) Below is the function call based on avp(i:274) avp_db_query("select uid from uid_did_map where accountcode = '$fU'","$avp(i:274)"); xlog("L_INFO","$rm from $fu (IP:$si:$sp) --------- UID:'$avp(i:274)-----------'\n"); ds_load_update(); if(!ds_select_dst("1","7")) { sl_send_reply("500", "No destination available"); xlog("route[MAIN] : $rm : No destinations available for $rd"); exit; } I'm getting two different values for avp(i:274) {26,29} but the destination are not hashed on its base. Please help. Regards, Sammy G.

3 15

Alias possible for presence...
by Gertjan Wolzak 14 Jun '12

14 Jun '12

Hello All, I have the following challenge. We are using kamailio 3.2 and have enabled presence. Which works fine. But we would like to be able to work with aliases, so that the account name and "presentation" name can be different. Any Ideas. Or did I miss the documentation on that? Regards, Gertjan Wolzak

1 0

Lua script calling rewriteuri
by Sharif Tanvir Rahman 14 Jun '12

14 Jun '12

Hello, We plan to use Kamailio as redirect server. Corresponding contact list is to be fetched by a Lua script from DB. The problem is about calling the required 'rewriteuri' function: 1. From lua script. both 'sr.rewriteuri()' and 'sr.modf("rewriteuri", ...)" fails to write the 'contact' field. 2. I tried an alternative way; pushing the value from lua (e.g. sr.pv.sets("contact", "sip:...")) and got the value in the Kamailio.cfg ($var(contact)). But can't call the rewriteuri function with this variable's value as argument. This seems to be a general problem of calling a function from .cfg with a variable's value as parameter. Any help much appreciated. -- Sharif

2 3

topoh - capturing encrypted values
by Alex Balashov 13 Jun '12

13 Jun '12

Given the transparency with which 'topoh' operates, is there any way to capture the new, fake values that it has substituted? In other words, can I, for example, xlog the new Call-ID produced by it? Or does that all take place on a level of abstraction that cannot be intercepted in script? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

About xcap xuid in kamailio
by Min Wang 13 Jun '12

13 Jun '12

Hi I am trying to use kamailio xcap server, saw the sample code: if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; why should the xuid have sip:xx@domain.com format? The xcap standard seems does not enforce the format on the xui part? http://xcap.example.com/root/resource-lists/users/joe |-------------v-------------||------v------||---v---| document root AUID XUI thanks. min

4 14

AVPs - how many values on stack
by Olle E. Johansson 13 Jun '12

13 Jun '12

I got this question today that I can't find the answer on: How many values are there in an avp? If I have pushed X values into the AVP - is there a function that counts them for me? /O

2 1

release of v3.3.0 scheduled for Monday, June 18, 2012
by Daniel-Constantin Mierla 13 Jun '12

13 Jun '12

Hello, as branch 3.3 was created and several web pages updated/created already for v3.3.0, the plan is to make the full release next Monday, on June 18, 2012. Packaging specs look very good thanks to Jon and Peter. The few issues left open should be worked meanwhile. For other opinions, reply with your comments here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

About sdpops
by Grégoire Vandendeurpel 13 Jun '12

13 Jun '12

Hello, I'm using Kamailio 3.2.3 and SDPOPS module. I just have a question about the configuration: What do you mean by "ANY_ROUTE" in configuration ?? Cause I don't know where in the configuration , I have to use the differents functions of SDPOPS. If I do something like this: request_route { if ($fU == "103") { sdp_remove_codecs_by_id("0,3,8"); #sdp_remove_media("audio"); } It does not work . So please help me about where I have to put this in?? Thanks in advance -- *Grégoire Vandendeurpel, * * * *IT Sector*

2 8

sipcapture logging
by Konstantin M. 13 Jun '12

13 Jun '12

Hello, Is it possible to log SIP sessions using sipcapture module to log files by log format instead of database ? Thanks!

3 6

Re: [SR-Users] Kamailio Unexpectedly Terminating
by Nathaniel L. Keeling III 12 Jun '12

12 Jun '12

Yes, This is a 64bit system T-Mobile. America’s First Nationwide 4G Network Jason Penton <jason.penton(a)smilecoms.com> wrote: >from the logs he has sent the mem addresses all look 64 bit. Sparc AFAIK is >only 64 bit anyway > >Cheers >Jason > >On Tue, Jun 12, 2012 at 8:50 AM, Daniel-Constantin Mierla <miconda(a)gmail.com >> wrote: > >> Hello, >> >> is it 64bit architecture? >> >> Cheers, >> Daniel >> >> >> On 6/11/12 9:12 PM, Akan wrote: >> >>> Here is the information requested: >>> >>> System = SunOS >>> Node = -f >>> Release = 5.10 >>> KernelID = Generic_141444-09 >>> Machine = sun4u >>> BusType = <unknown> >>> Serial = <unknown> >>> Users = <unknown> >>> OEM# = 0 >>> Origin# = 1 >>> NumCPU = 1 >>> >>> One machine is a Sun Fire V120, 2g memory, UltraSPARC-IIe 650MHz, >>> UltraAX-i2 >>> The other machine is a Sun Netra T1 200, 1g memory, UltraSparc-IIe >>> 500MHz, UltraAX-i2 >>> >>> Thanks >>> >>> Nathaniel L Keeling >>> >>> >>> >>> On 6/11/2012 8:33 AM, Daniel-Constantin Mierla wrote: >>> >>>> Hello, >>>> >>>> I committed a patch that should make it work when realm_prefix is not >>>> set for registrar module: >>>> >>>> http://git.sip-router.org/cgi-**bin/gitweb.cgi/sip-router/?a=**commit;h= >>>> **d42379da90f2ec87cb5dbb00ebb563**c7528ec910<http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=d42379d…> >>>> >>>> For the moment I have no idea why SIGBUS is triggered, i tried to access >>>> the structure directly, not via defines, and there is still a problem when >>>> realm_prefix is set. It's ok accessing the .len field but not .s of the >>>> same structure. The field .s is the first, making me think it's not >>>> alignment issue, or it is something very specific to this system. >>>> >>>> Overall, I am not familiar with sparc/solaris systems, maybe if you send >>>> here full details about OS version, architecture and CPU type, others using >>>> similar systems can give hints. >>>> >>>> Jason reported that on solaris with x86, this problem was not >>>> encountered. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 6/4/12 8:57 AM, Daniel-Constantin Mierla wrote: >>>> >>>>> Hello, >>>>> >>>>> still strange with the crash on that if condition. One of your previous >>>>> reports showed a different line, when assigning a pointer. Also, I kind of >>>>> understood that save() was ok, but lookup() not. Is not the case anymore as >>>>> the backtrace shows. >>>>> >>>>> SIGBUS can occur because of alignment in memory which is thrown in >>>>> solaris, does not happen in linux flavours. I am thinking of putting the >>>>> prefix field first so the access to it is at beginning of the straucture, >>>>> just to see if that is the problem, although is should not be. >>>>> >>>>> I will add a patch soon, otherwise I am out of ideas for the moment >>>>> with the SIGBUS, I would need access to the machine to do some testing by >>>>> myself. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> On 6/2/12 10:36 PM, Akan wrote: >>>>> >>>>>> Ok, I reloaded the servers with v3.2 from git without performing the >>>>>> checkout and with just the master branch. Reran my tests on 2 servers and >>>>>> Kamailio terminated with a core dump. I have included the full trace of one >>>>>> of the servers. The other trace has the same results. >>>>>> >>>>>> Thanks >>>>>> >>>>>> Nathaniel >>>>>> >>>>>> On 5/30/2012 2:25 PM, Daniel-Constantin Mierla wrote: >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> On 5/30/12 9:10 PM, Akan wrote: >>>>>>> >>>>>>>> This was a re-pull from the git master. I had deleted all of the old >>>>>>>> files and issued these commands: >>>>>>>> >>>>>>>> git clone --depth 1 git://git.sip-router.org/sip-**router<http://git.sip-router.org/sip-router>kamailio >>>>>>>> cd kamailio >>>>>>>> git checkout -b 3.2 origin/3.2 >>>>>>>> >>>>>>>> This is the source that I got. Could this be a problem within git? >>>>>>>> >>>>>>> do not checkout branch 3.2, just try with the master branch, as said, >>>>>>> not all patches are backported to stable 3.2. >>>>>>> >>>>>>> For sake of completeness, here are some guidelines: >>>>>>> http://www.kamailio.org/wiki/**install/devel/git<http://www.kamailio.org/wiki/install/devel/git> >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>> >>>>>> ______________________________**_________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>>> sr-users(a)lists.sip-router.org >>>>>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>>>>> >>>>> >>>>> -- >>>>> Daniel-Constantin Mierla -http://www.asipto.com >>>>> http://twitter.com/#!/miconda -http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda> >>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - >>>>> http://asipto.com/u/katu >>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - >>>>> http://asipto.com/u/kpw >>>>> >>>>> >>>>> >>>> -- >>>> Daniel-Constantin Mierla -http://www.asipto.com >>>> http://twitter.com/#!/miconda -http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda> >>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - >>>> http://asipto.com/u/katu >>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - >>>> http://asipto.com/u/kpw >>>> >>>> >>>> No virus found in this message. >>>> Checked by AVG - www.avg.com <http://www.avg.com> >>>> Version: 2012.0.2178 / Virus Database: 2433/5061 - Release Date: 06/10/12 >>>> >>>> >>> >>> >>> ______________________________**_________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.com >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda> >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - >> http://asipto.com/u/katu >> >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - >> http://asipto.com/u/kpw >> >> >> >> >> ______________________________**_________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >> > >This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer > >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

High availability design with Kamailio
by Reinhard, Carel (GD) 12 Jun '12

12 Jun '12

Dear all, hope to get some information from you guys! For a redundant service we have to build up SIP proxy/registrar server in a high available scenario. Therefore, we will have 2 SIP proxy/registrar on two completely independent servers. They should be SIP dialog stateful and replicate all the session/call states between each other. If one server fails, the other should have all the SIP dialog information for call handling and accounting. How can this done in a stable and reliable way? Is this feature mature enough to support enterprise requirements? The only information I have is to build it up with the DISPATCHER module, but it seems more to be a stateless load-balancer instead of a HA module. Who has experience with HA failover designs with Kamailio and can give me some hints? Thank you very much in advance Kind regards Carel Carel Reinhard Security and VoIP Engineer SRG SSR Switzerland Telefon direkt +41 31 350 94 07 Carel.Reinhard(a)srgssr.ch<mailto:Carel.Reinhard@srgssr.ch>

5 10

Shared memory fragmentation
by Morten Isaksen 12 Jun '12

12 Jun '12

Hi, We have 2 Kamailio 3.0.3 servers that has been running with carrierroute for about 2 years without any problems. They have 128 MB shared memory and modparam("carrierroute", "config_source", "db") modparam("carrierroute", "db_url", "<DBURL>") modparam("carrierroute", "fetch_rows", 500) The carrierroute table is about 91K lines and have been growing slowly. Suddenly we get this ERROR: carrierroute [cr_data.c:585]: could not allocate shared memory from available pool after a few "kamctl cr reload". I increased the shared memory to 256 MB but with the same result. I have now increased it to 512 MB and it seems to work better now. I have noticed this. After a restart the shmem counters is like this: shmem:total_size = 536870912 shmem:used_size = 28486752 shmem:real_used_size = 40147128 shmem:max_used_size = 41135424 shmem:free_size = 496723784 shmem:fragments = 555 And after the first "kamctl cr reload" it is like this: shmem:total_size = 536870912 shmem:used_size = 28619016 shmem:real_used_size = 51842768 shmem:max_used_size = 76993616 shmem:free_size = 485028144 shmem:fragments = 722063 Notice the increase in fragments. Sequentials "kamctl cr reload" does not change the fragments allot. Any ideas? -- Morten Isaksen

3 4

maddr parameter in contact header and fix_nated_contact
by Morten Isaksen 12 Jun '12

12 Jun '12

Hi, I have this setup: Mediation server with internal ip -- NAT firewall -- Kamailio proxy -- PSTN gateway The mediation server sends a contact header with maddr=<internal IP> fix_nated_contact only changes the host part and leaves the maddr intact. When the PSTN gateway sends a PRACK or a BYE it puts the <internal IP> in the RURI, and kamailio then try to forward the message directly to the <internal IP> and not the NAT ip. >From RFC 3261 19.1.1: The maddr parameter indicates the server address to be contacted for this user, overriding any address derived from the host field. When an maddr parameter is present, the port and transport components of the URI apply to the address indicated in the maddr parameter value. [4] describes the proper interpretation of the transport, maddr, and hostport in order to obtain the destination address, port, and transport for sending a request. I can see that OpenSIPS has fixed this. http://sourceforge.net/tracker/?func=detail&atid=1086410&aid=3312423&group_… Would it be possible to get this patch into Kamailio also? -- Morten Isaksen

2 5

Install Kamailio v3.3.x from GIT
by Daniel-Constantin Mierla 12 Jun '12

12 Jun '12

Hello, now that the branch 3.3 is available, I created the tutorial about how to install Kamailio v3.3.x from GIT: * http://www.kamailio.org/wiki/install/3.3.x/git Feedback and improvements are welcome! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

git branch 3.3 created
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, GIT branch 3.3 was created to hold the release series 3.3.x -- version 3.3.0 will be out in max. 1 week, a decision after collecting eventual issues still open to be reported today. Here are few details about how to use it. Download and use branch 3.3 from git: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.3 origin/3.3 Pushing patches: - only bug fixes or documentation improvements have to be committed to branch 3.3 - first commit to master branch (if the bug applies there as well), cherry-pick the commit to branch 3.3 and push to remote repository Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

development for future version 3.4.0 is open
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, short note to let everyone know that new features can be committed again to the master branch -- they will be part of future version 3.4.0 (if no other numbering release will be decided). Testing phase of version 3.3.0 within master is ended, a dedicated branch being created for it (details in a separate message). Happy coding, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Capturing a users call (SIP+RTP+T.38)
by Klaus Darilion 11 Jun '12

11 Jun '12

Hi all! Do you know any comfortable tools to filter out a certain users call? e.g. searching in SIP packets for the user pattern, get the media port out of SDP and capture also RTP+T.38 I found pcapsipdump but have not tried it yet. Any suggestions or do I have to write my own tool? thanks Klaus

4 6

branching 3.3 - to be used for 3.3.x releases
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, I am going to create the GIT branch 3.3 in short time -- this branch will be used for releasing 3.3.x versions. After that master can take again commits of new features, to be part of next major release 3.4.x. Branch 3.3 will take only fixes. From my point of view, I have two issues that I am trying to solve and the 3.3.0 is ready to go out, namely: - a solaris sigbus - pua_dialoginfo issue reported on mailing list Otherwise, nothing else critcal is out there, to my knowledge. Report them if you have other ones. Packaging specs were updated, so we are pretty much ready to release when no relevant issues is open. The wiki page with what is new in this release was updated a bit as well, add if you find something missing at: - http://www.kamailio.org/wiki/features/new-in-devel Also, contribute to the upgrade guidelines: - http://www.kamailio.org/wiki/install/upgrade/3.2.x-to-3.3.0 Cheer, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 2

About kamailio 3.3 msrp module and sylkserver
by Min Wang 11 Jun '12

11 Jun '12

Hi I am wondering what are the major differences between kamailio 3.3 msrp module and sylkserver (http://sylkserver.com/features.phtml ) ? thanks. min

2 2

Dispatcher PSTN
by Kr0m 11 Jun '12

11 Jun '12

Hello I am not able to dial to pstn with kamailio, the call is routed to my pstn-gw(asterisk), but the final phone rings 4 or 5 seconds and then it is hanged up. My outbound route is: route[PSTN] { if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } route(TOASTERISK); exit; return; } route[TOASTERISK] { sl_send_reply("100","Trying"); uac_replace_from("$fn","sip:$fn@$fd"); route(NATMANAGE); ds_select_dst("1","4"); t_on_failure("1"); t_relay(); } failure_route[1] { ds_mark_dst("i"); if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } route(TOASTERISK); } With a traffic capture i can see the traffic returning to my kamailio server. Any suggestion will be appreciated.

2 1

Kamailio and Nonce Count
by patrice.bodeven＠orange.com 11 Jun '12

11 Jun '12

Hello, I try to understand the nonce count handling. By parameter "modparam("auth", "nonce_expire", 21600)" I can define the validity of the Nonce in the time. But I want also to define the validity of the Nonce in the number of use. For exemple, I want to limit the reuse of the nonce at 64 times (including register, Invite) But I can see the definition of the nonce_count for all the SIP users on Kamailio and not user per user. If I understand when I use modparam("auth", "nonce_count", 1) # enable nonce_count support modparam("auth", "nc_array_order", 20) # 1M in-flight nonces, using 1Mb memory modparam("auth", "nid_pool_no", 4) I define a memory to store the nonce values for all the SIP users. So when User A use nonce x for REGISTER and nonce y for INVITE, nonce x is still valid ! My understanding is it correct ? Is anyway to define a nonce validity in number of times per user ? Patrice B _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified. Thank you.

2 1

Kamailio 3.2 + RTP proxy - no errors - no media
by Lukas Lani 10 Jun '12

10 Jun '12

Hi, I am trying to use RTPproxy 1.2.1 in bridge mode between IPv4 and IPv6 together with Kamailio 3.2. I configured and ran it on server together with Kamailio. I start it with parameters: -l 192.168.6.2 -6 /2001:6666:6666:6666::2 -s udp:localhost:7722 SIP messages are correctly exchanged, SDP looks to be translated, but when RTP session starts, I see RTP flow only from UC1 to RTPproxy and nothing more. On RTPproxy server I see 4 open ports for IPv4 address but no one for IPv6 address. When I tried to change start parameters to -6 2001:6666:6666:6666::2 -l /192.168.6.2 -s udp:localhost:7722 I could see 4 open ports for IPv6 address but no one for IPv4 address. Can anybody help? do you know about any known error? can it be related to Kamailio or RTPproxy? thank you, Lukas Lani

2 2

Looking for RTP Proxy in TCP
by Austin Einter 08 Jun '12

08 Jun '12

Hi All Now I am using Kamailio 3.1.5 and RTP proxy 1.1. Looks both are compatible and working fine. The RTP Proxy basically sends/receives RTP packets over UDP. Is there any RTP Proxy available that does send/receive of RTP packets over TCP and also should be compatible with Kamailio 3.1.5. If you have any information in this regard, kindly share. Thanks a lot. Austin

6 7

lcr from_uri not matching
by Ben WIlliams 08 Jun '12

08 Jun '12

Has anyone experienced this problem where lcr only works when from_uri is null. When I change it to .* the match fails. Thanks Ben

3 8

Re: [SR-Users] Issue using Kamailio as a proxy in front of Asterisk
by Daniel-Constantin Mierla 07 Jun '12

07 Jun '12

Do not write private emails, keep the mailing list cc-ed for the topics started there. Cheers, Daniel On 6/7/12 12:22 PM, Dominik Mauritz wrote: > Daniel, Stoyna, > > the solutions you described look very interesting to me. I will have a > look at both and pick the one that fits best. > > Thanks guys. > > Rgs, > Dominik > > > > Am 07.06.12 10:57, schrieb Daniel-Constantin Mierla: >> Hello, >> >> On 6/6/12 9:57 PM, Stoyan Mihaylov wrote: >>> We use Jitsi as SIP client, and openxcap along with camailio to >>> handle presence. Then jitsi know if account is online or offline. >>> Our Asterisk dont know nothing about accounts (it accepts all calls >>> from kamailio). There I run AGI scripts, which can check kamailio >>> tables - and I can know if account is online or offline. Of course >>> this do not work if account is "forced" to offline. >>> But may be there is better solution. >>> >>> On Wed, Jun 6, 2012 at 10:04 PM, Dominik Mauritz >>> <dominik.mauritz(a)web.de <mailto:dominik.mauritz@web.de>> wrote: >>> >>> I have already tried that. I defined SIP-Accounts in Asterisk >>> with host=<Kamailio-IP> (instead of host=dynamic). This solves the >>> described problem but it also has side effects: >>> >>> - You don't have the correct presence status on your phone (e. >>> g. xlite) indicating wether the account is online or offline >>> - Asterisk always sends invites to Kamailio on incoming calls >>> even if there is no phone registered to the account >>> >>> These are not the worst things in the world, but it is maybe not >>> the best solution possible. >>> >>> >>> >>> Am 06.06.12 20:41, schrieb Stoyan Mihaylov: >>> >>> We use also Kamailio in front of Asterisk - but I forward >>> only calls to Asterisk - register/unregister stay in Kamailio. >>> Asterisk dont know which device is registered, and which is not. >>> >>> >>> On Wed, Jun 6, 2012 at 8:20 PM, Dominik Mauritz >>> <dominik.mauritz(a)web.de <mailto:dominik.mauritz@web.de> >>> <mailto:dominik.mauritz@web.de <mailto:dominik.mauritz@web.de>>> wrote: >>> >>> Hi All, >>> >>> some days ago I installed Kamailio as a front end for >>> Asterisk following this tutorial: >>> >>> http://kb.asipto.com/asterisk:__realtime:kamailio-3.1.x-__asterisk-1.6.2-as… >>> <http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb> >>> >>> >>> >>> I added Multi Domain support and adjusted some other things >>> to fit my environment. Almost everything is working perfectly now. >>> One small issue is left: >>> >>> With Kamailio in front of Asterisk I have one feature (next >>> to other cool things) I was missing for years. I am now able to >>> register more than one device on the same SIP account. This is >>> nothing new for Kamailio users but Asterisk doesn’t support this. >>> >>> If I register two phones on one SIP account with Kamailio >>> everything is fine. They are able to make outgoing calls and both >>> ring on incoming calls. But if one phone unregisters Kamailio >>> forwards the unregister request to Asterisk and Asterisk sets this >>> account to offline. Now a call comes in but Asterisk is not sending >>> out an invite because for Asterisk the phone is offline. >>> >>> I wonder if it is possible to forward an unregister request >>> from Kamailio to Asterisk only if the last endpoint registered with >>> one SIP account unregisters. If there is more than one endpoint >>> using the same Account Kamailio should not forward the unregister >>> request to Asterisk. >>> >>> Any idea? >>> >> you can use reg_fetch_contacts(...) in your config to find out how >> many contacts are for an user and based on that do not send the >> un-register to asterisk, see: >> >> http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2498205 >> >> http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2498441 >> >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla -http://www.asipto.com >> http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 >> -http://asipto.com/u/katu >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 >> -http://asipto.com/u/kpw >> >> >> > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 1

record calls audio and video
by Saul Waizer 07 Jun '12

07 Jun '12

Hello List, I am looking for a way to record the audio and video of a call. I am using Kamailio (latest) on Ubuntu. Does anyone know if there's any existing modules that allow for recording or any other solution that can provide recording of calls on Kamailio? Thank you!

4 6

Siremis v3.2.1 Released
by Elena-Ramona Modroiu 07 Jun '12

07 Jun '12

Hi, Siremis v3.2.1 is out, the web management interface for Kamailio SIP Server. It is the last release with new features planned to be compatible with Kamailio v3.2.x. Next major releases for Siremis will target upcoming Kamailio v3.3.x. This time new features relate to web pages that allow management of own profile by SIP users, such as changing SIP password, management of speed dials or views for accounting, cdrs and location records. There was also added a form for doing user registrations (e.g., can be used for allowing public registrations). You can read more details about the release as well as details about installation at: * http://siremis.asipto.com/ Regards, Ramona

1 0

Limiting calls from users in prepaid scenerio
by Aft nix 07 Jun '12

07 Jun '12

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, We are thinking about launching a different "usage policy". The usage policy will be prepaid in some sense. but instead of limiting "number of minutes" we are thinking about limiting "number of calls". That means if you pay this X amount of doller, you can make Y amount of calls. calls can last whatever the user wants given that its under 1 hr. I've seen how to limit concurrent calls using SER, but the usage i have described, is not seen by me. So please help me out with this thing. My initial understanding about how it can be done : 1) when a dialog is confirmed with ACK(we have both callid, from tag, to tag), we can make an entry to the database by increasing "number of calls" field by one. 2) A field in database will be used as flag for usage limit. when "number of calls" exceeds the usage limit, it will flag it as "finished usage limit". 3) when a INVITE arrives, a check will be made with $fU against DB. With this framework in mind, I have read "db_mysql" module. And to my surprise, no direct the database functions are not exported to the cfg. I don't know if any module in kamailio exports mysql_queury() to cfg, which i need for my purpose. And may be there is some built in module which provides a higher level layer which will enable myself achieving my goal. If in the end i need "exported" mysql functions, then i can develop it myself. So which way i should go? Thanks in advance - -- - -aft -----BEGIN PGP SIGNATURE----- Version: OpenPGP.js v0.1 Comment: http://openpgpjs.org wsBcBAEBAgAQBQJPzc6dCRCJVJ6A/SK8awAARAkH/R5rqfiTPqBP+zZr8rgM 0JE42l9jxYq5OteD13U4CvcrUklXZlgDf2jwr1F7ndd1rlV3yKXAT2u61Bz5 LwNneRZydl3xnPmt2+TV6Qkcgy8RXgyRMntP9iLeQOXWJNW7D637h7tLL11d O50s86ofgJT+dfTICbomNZjLZRr1DvYOdcByG3LHcYx033G6bIFlMqJ0ZO+G cBdgtJcLrwo6iPn8Gv+5aJDy2kRUphrF7cVCzFzLPnNVZpvXNNcv7LB9bkUH cClcLDVAbsdw0TPd9Z6DNJSxFlHy2uzwNWB9mhpVJD1toT8Geb4iCOi+ZG8P WT5fgNwSoZlEm0BBBHwMUdk= =fcdD -----END PGP SIGNATURE-----

2 2

Issue using Kamailio as a proxy in front of Asterisk
by Dominik Mauritz 07 Jun '12

07 Jun '12

Hi All, some days ago I installed Kamailio as a front end for Asterisk following this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb I added Multi Domain support and adjusted some other things to fit my environment. Almost everything is working perfectly now. One small issue is left: With Kamailio in front of Asterisk I have one feature (next to other cool things) I was missing for years. I am now able to register more than one device on the same SIP account. This is nothing new for Kamailio users but Asterisk doesn’t support this. If I register two phones on one SIP account with Kamailio everything is fine. They are able to make outgoing calls and both ring on incoming calls. But if one phone unregisters Kamailio forwards the unregister request to Asterisk and Asterisk sets this account to offline. Now a call comes in but Asterisk is not sending out an invite because for Asterisk the phone is offline. I wonder if it is possible to forward an unregister request from Kamailio to Asterisk only if the last endpoint registered with one SIP account unregisters. If there is more than one endpoint using the same Account Kamailio should not forward the unregister request to Asterisk. Any idea? Thanks, Dominik

3 4

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sr-users June 2012