- sr-users - kamailio.org

Bug with server originated call
by David 20 Jun '12

20 Jun '12

Hello, I am using the example that came with my kamailio deb file to launch a call server side. I send a dummy INVITE to the phone, when the phone responds 200 OK, I reply with a REFER to send the phone to the desired number. ( ctd.sh.gz ) My problem is that if I don't answer the ctd call, the phone rings for a long time ( or endlessly on certain devices ). I tried adding an Expires: 10, and it did not make any difference. Is there a way to cancel this call from the ctd script ? David

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Kamailio v3.3.0 Released - new major version is out
by Daniel-Constantin Mierla 20 Jun '12

20 Jun '12

Kamailio v3.3.0 is out – a major release with a very large number of new features and improvements. This release is a result of about 6 months of development and 2 months of testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. Version 3.3.0 comes with 7 brand new modules in addition to a lot of fresh features in core and old modules. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v3-3-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy SIP routing in a secure, flexible and easier way with Kamailio v3.3.0! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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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

kamailio as SBC
by Mino Haluz 06 Jun '12

06 Jun '12

Hi, I know that kamailio is SIP proxy, but is there any way how to implement kamailio as SBC like OpenSIPS with B2BUA module ? I tried OpenSIPS with this module, but it does not work with mediaproxy module. Mino

5 4

Re: [SR-Users] : Capturing a users call (SIP+RTP+T.38)
by Anton Kvashenkin 06 Jun '12

06 Jun '12

You can try homer sip capture server, just google it. 06.06.2012 15:08 пользователь "Klaus Darilion" <klaus.mailinglists(a)pernau.at> написал: > 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 > > ______________________________**_________________ > 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> >

1 0

Kamailio and Netann
by patrice.bodeven＠orange.com 05 Jun '12

05 Jun '12

Hello, I'am working on Kamailio version 3.2.2. I'm doing some test in interaction with Application server and media server. The applicationServer sends a INVITE sip to Media Server via the Kamailio. The R-URI has the format sip:annc@cl1.oms.ser.v0.pftest.net;play=http://1xx.1xx.1xx.1xx:8080/errfile/err.wav;repeat=forever But the parsing done by Kamailio of $ru is trunked to sip:annc@cl1.oms.ser.v0.pftest.net;play=http 1) Could you confirm if Kamailio supports Netann protocol ? 2) If not, how to route the INVITE with the correct R-URI format ? Thanks in advance. PatriceB _________________________________________________________________________________________________________________________ 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.

3 3

Test SIP Servers for Simultaneous calls
by Vineet Menon 05 Jun '12

05 Jun '12

Hi, I want to test kamailio or any other server for it's simultaneous call rates, form which I mean how much calls it can support simultaneously?? I know about SIPp, but that won't fulfill my purpose, because SIPp doesn't offer simultaneity; rather it issues calls serially. BTW, is it possible to do multiple calls from same machine, since ports are consumed in the previous call?? I think the answer is negative.... I think, to do this kind of testing, we need multiple PCs..which call simultaneously... Any light on this?? Regards, Vineet Menon

3 4

Single-quoted vs double-quoted params in Path/Route
by Andreas Granig 04 Jun '12

04 Jun '12

Hi guys, When I implemented the Path module 6 years ago, I chose to enclose the received-URI in double-quotes, like this: Path: <sip:1.2.3.4:5060;received="sip:1.2.3.5:1234;transport=tcp"> This Path value is stored in usrloc by registrar module, and is converted to a Route in a subsequent request. This worked fine so far, because there's the logic in parser/parse_param.c to handle double-quoted parameters. Now while doing some interop tests with another platform, it turned out that double-quotes are not allowed in URI params (see https://lists.cs.columbia.edu/pipermail/sip-implementors/2008-May/019335.ht…, and in the grammar the allowed chars in the "unreserved" definition are "alphanum" and "mark", where "mark" is only "-" / "_" / "." / "!" / "~" / "*" / "'" / "(" / ")" ). Anyways, to fully comply with RFC3261, I'd like to push a bugfix to path module to use single-quotes instead, however it also needs a change in the parser to use parse_quoted_param() also in case of single-quoted params (which hasn't been handled at all until now). My question now is if there are any concerns on your side with handling single-quoted URI params the way we handle double-quoted ones? If we were strict, we'd actually need to remove the handling for double-quoted ones, but for backwards-compatibility I'd rather keep it there. Comments? Andreas

2 2

FS#213 - pua_reginfo: registration state may be incorrect when unregistering single contact
by Gnaneshwar Gatla 01 Jun '12

01 Jun '12

I'm working with pua_reginfo module, I thought everything was working fine. But FS#213, is not resolved yet. @@ -118,9 +111,10 @@ str* build_reginfo_full(urecord_t * record, str uri, ucontact_t* c, int type) { xmlNewProp(contact_node, BAD_CAST "id", BAD_CAST buf); /* Check, if this is the modified contact: */ if (ptr == c) { - if ((type & UL_CONTACT_INSERT) || (type & UL_CONTACT_UPDATE)) + if ((type & UL_CONTACT_INSERT) || (type & UL_CONTACT_UPDATE)) { + reg_active = 1; xmlNewProp(contact_node, BAD_CAST "state", BAD_CAST "active"); - else + } else xmlNewProp(contact_node, BAD_CAST "state", BAD_CAST "terminated"); if (type & UL_CONTACT_INSERT) xmlNewProp(contact_node, BAD_CAST "event", BAD_CAST "created"); else if (type & UL_CONTACT_UPDATE) xmlNewProp(contact_node, BAD_CAST "event", BAD_CAST "refreshed"); from the above code in pua_reginfo_fix_registration_state.patch If(ptr == c) returns false and the server keeps the state to "active" and the reg_info is never set to refreshed/terminated. I was wondering if I could use the following instead. If(strcmp(ptr->c, c->c)), which would compare the contact string, if equal the reg_info is update in the presentity. Regards Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

Re: [SR-Users] SIP Client for iphone
by Daniel-Constantin Mierla 01 Jun '12

01 Jun '12

Please keep the mailing list cc-ed -- private messages from mailing list are going to be ignored. If you go to jitsi.org, in the download page there are references to the mac os x builds. Cheers, Daniel On 6/1/12 9:38 AM, Vijay Thakur wrote: > Thanks for prompt reply. > Please suggest me the link from where i can download jitsi for Mac OS X. > > On Friday 01 June 2012 01:06 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> On 6/1/12 9:06 AM, Vijay Thakur wrote: >>> Dear All, >>> >>> Kindly suggest me the compatible client for iphone and Mac OS Leopard. >>> I have found that Jisti is working very fine for all features on >>> Linux and MS Windows. >>> >> I use jitsi for Mac OS X, iirc, it worked fine on Leopard, now I >> upgraded the OS. For iPhone and even the other OSes, check Linphone. >> >> 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

multiple t_on_branch[]
by Vitaliy Aleksandrov 01 Jun '12

01 Jun '12

Hi all, Does kamailio allow to configure more than one branch_route for an initial INVITE ? I have tried to add: t_on_branch(branch_r1); t_on_branch(branch_r2); before t_relay(). Mentioned branch routes write debug to the kamailio.log and it looks like that only the last branch_route[branch_r2] works.

2 1

SIP Client for iphone
by Vijay Thakur 01 Jun '12

01 Jun '12

Dear All, Kindly suggest me the compatible client for iphone and Mac OS Leopard. I have found that Jisti is working very fine for all features on Linux and MS Windows. -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.thakur(a)loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

4 6

CFGUtils module question
by Fabian Borot 31 May '12

31 May '12

Hello Can somebody help me understand the example given with MI command "is_gflag"? ... $ kamctl fifo set_gflag 1024 $ kamctl fifo is_gflag 1024 TRUE $ kamctl fifo is_gflag 1025 TRUE $ kamctl fifo is_gflag 1023 FALSE $ kamctl fifo set_gflag 0x10 $ kamctl fifo is_gflag 1023 TRUE $ kamctl fifo is_gflag 1007 FALSE $ kamctl fifo is_gflag 16 TRUE This is how I understand it. There are 32 posible flags (0..31), each one can be set independently. The commands "is_flag" and "set_gflag" expect a bitmask as value (decimal or hex) If I set "kamctl fifo set_gflag 1024", this means "100 0000 0000", I expect that the only flag that has been set to 1 is the position 10. How come the command "kamctl fifo is_gflag 1025" which is "100 0000 0001" can yield TRUE?. I expect it to be TRUE only if flags 0 and 10 are both set to 1. Same thing with setting the flag to 0x10 (16 in decimal, "1 0000" in binary), how come the "is_gflag 1023 and is_gflag 1007" can return TRUE? tx a lot fborot

2 2

Fwd: [Solved]Creating RURI ($ru) from Contact ($ct)
by Aft nix 31 May '12

31 May '12

Hi all, I'm posting the script which worked. In future if anybody faces the same problem can get a solution out of this. In request_route: #!ifdef WITH_HASH if (is_method("INVITE") && !has_totag()){ xdbg ("Contact [$ct] with Callid [$ci]"); $sht(a=>$ci) = $ct; } #!endif in route[WITHINDLG] #!ifdef WITH_HASH xdbg ("ruri [$ru]"); if ($(sht(a=>$ci){nameaddr.uri}) != $ru){ xdbg ("hash table entry [$sht(a=>$ci)]"); $ru = $(sht(a=>$ci){nameaddr.uri}); sht_rm_name_re("ha=>$ci"); xdbg ("rewriting ruri to [$ru]"); } #!endif Thanks everyone for the help. I'm marking this thread as solved. -- -aft -- -aft

3 7

SIP-XMPP gateway and not existent recipient
by intel＠intrans.baku.az 31 May '12

31 May '12

I set up sip-xmpp gateway using xmpp module(in component mode). messages from sip client to xmpp delivered, but I have a problem. If recipient is not existent, the sip client don't know nothing about it. How can I inform sip client about such condition? The xmpp server(ejabberd) inform it's component about this error(I've checked it using tcpdump), but kamailio seems ignore this error completely. What can I do in this situation? -- Homo Homini domini est!

1 0

Max registered users
by Vitaliy Aleksandrov 30 May '12

30 May '12

Hi all, I'm trying to find a way to get a number of the registered user locations and if it is bigger that some limit then kamailio will not save new registrations. reg_fetch_contacts() from registrar module doesn't help as it can fetch contacts only for one user. max_contacts parameter of the registrar module limits the number of contacts per AOR, but i need to limit max number of all contacts that proxy will keep in usrloc.

3 3

acc: can't get code for the Source-IP attribute
by Rabary 30 May '12

30 May '12

Hi all, When I start kamailio, I have theses errors in syslog and kamailio failed to start: May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: INFO: usrloc [hslot.c:53]: locks array size 512 May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: INFO: auth [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support disabled May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: acc_radius [acc_radius_mod.c:276]: acc: can't get code for the Source-IP attribute May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: acc_radius [acc_radius_mod.c:289]: failed to init radius May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: acc [acc_mod.c:758]: failed to initialize extra acc engine May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: acc_radius [acc_radius_mod.c:154]: cannot register ACC RADIUS engine May 30 12:02:11 kamailio /usr/sbin/kamailio[5652]: ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module acc_radius (/usr/lib/kamailio/modules_k/acc_radius.so) In radiusclient dictionary.sip, I have this: teddy@kamailio:~$ sudo grep -r -i source-ip /etc/radiusclient-ng/* /etc/radiusclient-ng/dictionary.sip:ATTRIBUTE Sip-Source-IP-Address 108 ipaddr And in dictionary file I include this dictionary.sip like this at the end of the file: $INCLUDE /etc/radiusclient-ng/dictionary.sip There are kamailio and libradiusclient-ng2 version installed: ii kamailio 3.2.3+squeeze1 very fast and configurable SIP proxy ii kamailio-mysql-modules 3.2.3+squeeze1 MySQL database connectivity module for Kamailio ii kamailio-radius-modules 3.2.3+squeeze1 radius modules for Kamailio ii libradiusclient-ng-dev 0.5.6-1.1 Enhanced RADIUS client library development files ii libradiusclient-ng2 0.5.6-1.1 Enhanced RADIUS client library So I don't know where I'm doing wrong, can anyone tell me what am I missing ? Thanks. -- Rabary Teddy Inutile d'imprimer ce mail

1 1

PUA_RegInfo patch
by Gnaneshwar Gatla 30 May '12

30 May '12

Hey, I have fixed the incorrect uri's being generated by this module while publishing user location. This patch also includes some more user record data to the xml during publish. I'm not sure if I have followed the naming convention right. Please find the patch file attached. Regards, Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

2 2

xhttp README
by Klaus Darilion 29 May '12

29 May '12

Hi! The readme should mention that core setting tcp_accept_no_cl=yes is required (at least for Firefox and Chrome) regards Klaus

3 5

Using Sip-Router Logging & Register Problem Problem
by Faisal Yousuf 29 May '12

29 May '12

Hi Experts, I am a newbee with SIP-Router/Kamailio - I installed Kamailio on a CentOS 5.5 VM - following the instructions from the following link http://kb.asipto.com/kamailio:install:3.0.x-from-git-centos5x Kamailio is installed & I can start kamailio deamon from service kamailio start However I also created two subscribers - but when I try to register my SIP Phone (by setting Kamailio as the domain to register to) - I can see my sip-phone sending the Register request to Kamailio (I put a tcpdump on the kamailio machine to see sip-messages reaching the host) but I dont get any response from Kamailio - so my sip phone reaches the 408 Registeration Timeout state. Now, I dont get any logs from Kamailio to know what is happening while my phone is sending the register request - I tried to configure logging following instructions from the following link http://www.kamailio.org/dokuwiki/doku.php/utils:basic-syslog-configuration But I dont get any logs in the /var/log/kamailio log file. To assist in understanding the issue - I am attaching the syslog config , kamailio config file & the kamailio service script in the init.d folder. My first objective will be to enable logging for kamailio & the next objective is to get the basic registration & routing between endpoints - my ultimate objective is to configure SER/Kamailio as a load-balancer for our B2BUA & Media Servers (I found the dispatcher module can help in accomplishing this) - Can anybody help me in identifying why I am not getting any logs ? I have already seen the wiki & have developed some understanding regarding how Kamailio is configured but am I seeking more information regarding the route language - (the state-model + reference of the functions that can be used in the route script etc.) - in this regards if I can get any documentation pointers from the user community that can help me gain a better understanding the use of Kamailio for above it will be great Regards, Faisal Regards, Faisal

4 9

Logging peer status
by Niklas Larsson 29 May '12

29 May '12

Hi, I'm coming from an asterisk background and are now using kamailio to handle registration and load balancing. In the asterisk world I added qualify=yes and listening on the AMI for events when peer is registering / unregistering / peer goes unreachable and reachable - then storing this info in a MySQL db to be displayed on a web page for support etc. This works really nicely and I'm trying to figure out how to do the same with kamailio and are able to generate an event when a peer is registering (in the registry route using mqueue and some lua magic). But can't find a way to get an event when an option ping (nathelper module) is unsuccessful or when an location record has expired(http://lists.kamailio.org/pipermail/users/2009-May/023270.html). So my question is - how are you guys dealing with this? What have I missed? /niklas

1 0

Validate domains, expose sip_hostport2su or similar
by Øyvind Kolbu 29 May '12

29 May '12

Hi, is there a way to validate a domain, e.g. it resolves to something, without calling t_relay() or forward() and checking the return code? In my case I'd like to validate enum results before dispatching it to our outbound proxies. If not valid I must use the pstn servers. In the logs I can see a function named sip_hostport2su does some validation, but it seems to be a c-function only. From the log: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "example.org" ERROR: tm [ut.h:327]: failed to resolve "example.org" -- Øyvind Kolbu

2 2

Address of Record (AOR) parameter
by Bruno Bresciani 29 May '12

29 May '12

Hi all, I can't understand how obtain address of record (aor) to functions lock_udomain and unlock_udomain... What is the meaning of this parameter? In other modules this parameter is obtained by R-URI on SIP messages struct (registrar) or by next value of node.kids mi_root struct (usrloc). Best Regards

1 0

planning the release of v3.3.0
by Daniel-Constantin Mierla 29 May '12

29 May '12

Hello, looking at the volume of reported issues on mailing lists and tracker, upcoming v3.3.0 seems to go well. I have few items on my to-do reported on the mailing list. Otherwise, in my side, I have 3.3.0 running in a production-like environment for about 4 weeks without problems and another one in a pre-production phase. Having in consideration all these, I propose to branch 3.3 on June 11. If everything looks very good at that time (or actually few days before), I think we can do the release also by that date. If not, we can do it a week later. Branching will allow new features development on git master, there are already several contributed patches with new features flowing around on mailing lists and tracker. Therefore, if you are aware of any issue, please report asap on mailing list (sr-dev) or tracker (preferable - http://sip-router.org/tracker/). Other suggestions about planning the release 3.3.0 are welcome, as well. 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

1 0

Creating RURI ($ru) from Contact ($ct)
by Aft nix 28 May '12

28 May '12

On Mon, May 28, 2012 at 4:52 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > On 5/27/12 12:46 AM, Aft nix wrote: >> >> [...] >> >> Hi Aleksandrov >> Thanks for the reply. >> >> Yes my checks should be more specific. >> >> On the side how , i'm facing difficulties to print script variables >> and hashtable entries. >> >> xdbg("my var [$var(temp)]") >> xdbg("my hash entries [$sht(a=>$ci)"]) >> >> Both statements shows parse errors. What's the correct syntax to print >> script >> variables and hash table entries? > > if you copy&paste the lines from your config, then the errors are: > - missing ; at the end of lines > - second xdbg has the double quotes before ], making it invalid syntax Hi Daniel, Thanks, other than the typo, its ok to print stuffs like this, then. > > 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 > > > -- -aft -- -aft

1 0

Installation of Kamailio, Asterisk, siremis in Debain Squeeze
by Vijay Thakur 28 May '12

28 May '12

Dear Experts, I want to install Kamailio, Asterisk, siremis in Debian 6.0 Squeeze. For this purpose i have followed the following document: (1) http://nil.uniza.sk/sip/kamailio/installing-siremis-20-kamailio-web-managem… - Siremis Installation (2) http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour - Kamailio Installation (3) http://nil.uniza.sk/sip/kamailio/installing-siremis-20-kamailio-web-managem… - Siremis Installation I need the following features: 1 User Management 2 Contacts 3 Instant Messaging 4 My Presence (Presence Status) 5 Voice Call (SIP) 6 Conference (SIP Audio Calls) 7 Video calls 8 SMS 9 Outbound Audio Call 10 File Transfer 11 Video calls 12 Share Link 13 Face book Integration 14 Push Notifications (Broadcast Message) According to the the documents, some features are not working, or working features are not working sometime as supposed to work. Kindly suggest me straight forward document or any help to achieve it. -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail :vijay.thakur@loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web:www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

2 1

Sending MSRP message to kamailio/msrprelay causes 501 error
by Reichert Alexander 28 May '12

28 May '12

Hi, We are developing a Java based SIP gateway implementation and are performing some integration tests on a kamailio/msrprelay server (plugin from www.msrprelay.org). The SIP signaling part works so far, but when sending a MSRP message, the client receives a 501 "Unknown method" error. However, it seems the request does not reach the MSRP module on the server as there is no corresponding trace log on the server side. On Protocol level, everything looks fine so far, the sequence executed is as follows: -->SIP REGISTER <-- SIP/2.0 401 Unauthorized -->SIP REGISTER <--SIP/2.0 200 OK -->INVITE <--SIP/2.0 407 Proxy Authentication Required -->ACK -->INVITE <--SIP/2.0 180 Ringing <--SIP/2.0 180 Ringing <--SIP/2.0 200 OK -->ACK The INVITE is processed with the following SDP packet in the request: v=0 o=- 1337788359935 1337788359935 IN IP4 localhost s=MSRP Session c=IN IP4 localhost t=0 0 m=message 2855 TCP/MSRP * a=path:msrp://localhost:2855/U3FuvW00m62UY0020H71;tcp a=accept-types:message/cpim text/* application/im-iscomposing+xml a=accept-wrapped-types:* a=setup:active And response: v=0 o=- 3546777215 3546777216 IN IP4 46.105.78.68 s=sipsimple 0.20.0 c=IN IP4 46.105.78.68 t=0 0 m=message 57233 TCP/MSRP * a=path:msrp://msrphost:2855/5AUNTOt7XlFsK7VuT4SzETEzMzc3ODg0MTUuMDcyOjQ2LjEwNS43OC42OA==;tcp msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp a=accept-types:message/cpim text/* application/im-iscomposing+xml a=accept-wrapped-types:* a=setup:passive But when sending the MSRP message, 501 is returned: MSRP f0CHA1TD SEND To-Path: msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp From-Path: msrp://localhost:52673/b10XadK9;tcp Message-ID: 13377883623135c260a3568cc Byte-Range: 1-*/14 Hello Francois -------f0CHA1TD$ MSRP f0CHA1TD 501 Unknown method: SEND To-Path: msrp://localhost:52673/b10XadK9;tcp From-Path: msrp://msrphost:57233/6bf6dd73b9e4587b65b6;tcp -------f0CHA1TD$ When testing with two sip-simple client endpoints, MSRP messaging works fine, but on protocol traces we cannot identify any differences between sip-simple communication flows and the ones we are implementing. Does anyone have a hint on what is wrong in this message flow? Any advice is appreciated. Thanks, Alex

2 3

Creating RURI ($ru) from Contact ($ct)
by Aft nix 28 May '12

28 May '12

Hi, I'm trying to create a ruri from contact header like following : In request route I've saved contact header in a hashtable using htable module. request_route{ --------------------- --------------------- #!ifdef WITH_HASH if (is_method("INVITE") && !has_totag()){ xdbg ("Contact [$ct] with Callid [$ci]"); $sht(a=>$ci) = $ct; } #!endif ---------------- } And in route[WITHINDLG] route[WITHINDLG] { --------------------------- --------------------------- #!ifdef WITH_HASH xdbg ("ruri [$ru]"); if ($sht(a=>$ci) != $ru){ $ru = ($sht(a=>$ci)); xdbg ("rewriting ruri to [$ru]"); } #!endif But it log shows that its unable to parse the uri : [4039]: DEBUG: <core> [parser/parse_uri.c:1277]: parse_uri: bad uri, state 0 parsed: <<sip> (4) / <<sip:2020@103.23.169.2:64358>> (29) : DEBUG: <core> [parser/parse_uri.c:1327]: ERROR: parse_sip_msg_uri: bad uri <<sip:2020@103.23.169.2:64358>> [4039]: ERROR: pv [pv_core.c:232]: failed to parse the R-URI [4039]: DEBUG: <script>: rewriting ruri to [<null>] I'm not understanding what's wrong with that. May be the port part should be discarded, but how i'm gonna do that? -- -aft

3 3

SER configuration File build system
by G Madhavi 28 May '12

28 May '12

Dear sir, We are searching for *ser.buildsystem.latest.tar.gz* in the internet throughly. But we are unable to find this particular file.so please send me the file or the web links which we can be able to download and configure our IMS network in proper. * * * * Thank you Best Regards, madhavi -- * *

1 0

any documentation around?
by Sherif Omran 26 May '12

26 May '12

Hi guys, is there any documentation for Siremis around? thanks S.

1 0

TLS config file
by Bruno Bresciani 25 May '12

25 May '12

The parameter [server:default] of TLS config file must be declared? I want specify two different interfaces (WAN and LAN) and I don't know which interface will be used as default by TLS module... Someone know some information about my question? Best Regards

2 2

[OT] sems [and kamailio] in risk-of-life service
by Daniel-Constantin Mierla 25 May '12

25 May '12

just forwarding from SEMS mailing list an interesting story of using kamailio and sems in emergency services... -------- Original Message -------- Subject: [Sems] sems in risk-of-life service Date: Thu, 24 May 2012 15:29:38 +0800 From: Jeremy A <jeremy(a)electrosilk.net> To: sems(a)lists.iptel.org Hi, This is a belated report on the use of SEMS in a risk of life service. The system uses kamailio in a distributed architecture of dozens of Fire & Rescue stations. This is heavily based on distributed and replicated DNS. A single '911' style headquarters has duplicate hot swap-over control rooms at other locations. The headquarter and alternate posts have servers that service HQ operator positions with SIP phones. These provides sidecar indication of F&R Station state for up to 64 F&R stations - using BLF. These phones are hooked into an integrated analogue audio management system. Each Fire and Rescue station has an embedded SIP based controller that runs Kamailio and proprietary software to control the F&R station electrical and safety systems as well as provide public address functions to alert the F&R staff of a new emergency. These PA announcements are SIP based using a DSL network and are live from the HQ positions, plus computer synthesized voice, as well as alerting tones. Each station also has multiple SIP phones for in-station and station to station calling. The network is decentralized so failure of the central control system still allows point to point communications between Fire and Rescue stations. The headquarter systems uses SEMS as the primary operator manager to perform multiple simultaneous deployment calls to remote Fire and Rescue stations. SEMS is used to create a dynamic conference between an operator and multiple Fire & Rescue stations. These are automatically initiated by SEMS and answered by the F&R embedded systems. This means an operator can broadcast a deployment message and initiate station control activities at up to five stations (fifth alarm) This is only constrained by the bandwidth available at the headquarters. Our SEMS packages have been designed to handle non-answered calls to the conference and provide operator indication by 'SMS' messages to the handsets and audio feedback. The system provides full forensic recording by using rtpproxy at all locations. These recordings are archived by an out-of-band process. Control of the system is purely SIP based - so every item in the system is a SIP based entity. This includes servers, embedded systems, and phones. The phones are physically integrated into operator positions that also handle PSTN, PBX, and radio traffic. The interface is purely keyboard on the operator phones. Options for integration of the SIP system into CAD (Computer Aided Dispatch) are obvious. The only drawback is the rusty and ancient systems and the unbelievable process required to get approval to integrate. The system as provided provides at least 5 nines reliability. Probably a lot better. The only downside is the DSL network (provided by others at amazing expense) that provides a system with a lousy 2 nines reliability. We are in the process of developing an offering using redundant DLS/3G routing to improve this. The field stations are a hybrid Centos 5/Slax system running out of flash. The HQ systems are straight Centos 5 systems running off disk or off flash. Future versions will be pure Centos out of flash with no fancy memory overlay - flash is well good enough. The system has been live for over a year with no major issues. I can't say how many lives have been saved, but certainly quite a few. At least we haven't been sued yet :-) Cheers.

1 0

Hop by hop CANCEL, wrong to header
by Vitaliy Aleksandrov 25 May '12

25 May '12

Hi all, I have a question about CANCEL message processing. My call sceraio: When an INVITE request comes I need to rewrite a domain part of the several headers(to, from, contact, SDP ip) according to the outgoing interface. I can do it before t_relay(), but when destination user has more then one locations and they are reachable through different interfaces all forked INVITEs will have the same domain. To avoid that problem i have tried to move rewriting part(subst from the textops module) to the branch_route. Unfortunately that solution didn't helped me, because "To" header of the outgoing CANCEL messages is wrong (unchanged. as it was at the moment when transaction was created by the t_relay() ). As i understood from the documentation and mailing list kamailio builds CANCEL based on outgoing INVITE. I did a small research and found that Kamailio really takes outgoing INVITE from each branch (invite_transaction_cell -> uac[b_id].request.buffer) and builds CANCEL (build_local_reparse() from tm/t_msgbuilder.c) based on it. But kamailio does an exception for the To header which is described below: ***cancel_branch* function (from t_cancel.c) calls *build_local_reparse*() and fills one of the parameters with a pointer to unmodified To header. * build_local_reparse*() uses received To header to construct outgoing CANCEL. I have changed: cancel = build_local_reparse(t, branch, &len, CANCEL, CANCEL_LEN, &t->to, reason); to: cancel = build_local_reparse(t, branch, &len, CANCEL, CANCEL_LEN, NULL, reason); and it works for me now. It looks like if E2E_CANCEL_HOP_BY_HOP *e2e_cancel()* (t_fwd.c) will call *e2e_cancel_branch()* which works as i want instead of *cancel_branch()*, but it is just my assumption. Why does kamailio generate CANCEL requests in such a way ? Did i miss something from the RFC3261 or kamailio documentation ? Thanks in advance for any help !

2 2

rtpproxy and timeout notification
by Laurent Schweizer 24 May '12

24 May '12

Hello all, I see the new timeout notification feature from the rtpproxy when no voice traffic is detected. I have 2 questions regarding this timeout notification: 1) When a timeout is detected and kamailio send the BYE is that possible to manipulate the BYE before he is sent to add maybe an extra header that indicate that the call was stopped due to a timeout issue ? 2) If a user put the call on hold (with a=recvonly or maybe old way IP=0.0.0.0.0) what will happens (the music on hold is not generated by rtpproxy but by another server) ? he will detect this as a timeout ? Is that possible to indicate to rtpproxy that for a moment stream will be only in one way ? Best regards Laurent

2 1

Question about more complicated rtpproxy scenario
by Lukas Macura 24 May '12

24 May '12

Hi to all, please could somevody help me with more complicated rtpproxy scenario? I still cannot understand how to make it working. We have SBC based on kamailio 3.2.3. Everything seems to work until I want to translate all SDP messages by SBC IP address (for both sides). Moreover, I need to translate ipv4 to ipv6 and vice versa too. I found some example script how to achieve ipv4 to ipv6 and vice versa and I am tryinf to use it. It is relatively easy to translate all RFC addresses to one public IP, but I need to translate even our public IPS into SBC IP. I made some rtpproxy sets: modparam("rtpproxy", "rtpproxy_sock", "1 == udp:127.0.0.1:7722") (EXTRA_OPTS="-l SBC_IP") for internal net to external modparam("rtpproxy", "rtpproxy_sock", "2 == udp:127.0.0.1:7723") (EXTRA_OPTS="-l /SBC_IP -6 SBC_IP6") for ipv4 to ipv6 modparam("rtpproxy", "rtpproxy_sock", "3 == udp:127.0.0.1:7724") (EXTRA_OPTS="-l SBC_IP -6 /SBC_IP6") for ipv6 to ipv4 modparam("rtpproxy", "rtpproxy_sock", "4 == udp:127.0.0.1:7725") (EXTRA_OPTS="-l SBC_IP/SBC_IP) for external to internal but I cannot make it working. Either there are problems with "incorrect port 0 in reply from rtp proxy" or SDP gets malformed because of more times rtpproxy_manage(). Please can you help me with some basic hint, where into routing logic put rtpproxy_manage() and set_rtpproxy_set() ? Moreover, how to check if rtpproxy was already forced? I could not find any example of forcing RTP proxy for all calls for both sides. I need this because entire RTP traffic has to go across SBC. And my internal IPs are not private. Thank you for any hint, Lukas Macura

2 1

vsf value in Record-Route header
by Robert R 24 May '12

24 May '12

Hi, * * * * We are facing a problem in our opener proxy receiving: *"Message too big" error.* Looking at the INVITE message sent by opener I notice the following in the record-route header: Record-Route: <sip:10.10.10.10;lr=on;ftag=5114A38-F80;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA-> Is this a valid value for vsd? Can we correct the value? Thanks, R

2 1

LinuxTag Today: Workshop build your own UC service with Kamailio and Jitsi
by Daniel-Constantin Mierla 24 May '12

24 May '12

Hello, short message to let you know that today at 12:00 I will do a practical workshop at LinuxTag in Berlin, about how to build your own unified communication service using kamailio as server and jitsi as client. If you visit the event, you can join the workshop in room New York 2.1, more details at: http://www.linuxtag.org/2012/en/program/project-workshops/details.html?no_c… Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

allow_address failed initialize permissions
by Grégoire Vandendeurpel 23 May '12

23 May '12

Hi, I'm using kamailio 3.2.3 and I use permissions.so But when I reload Kamailio , error: failed to initialize the allow_address function 0(3815) ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module permissions (/usr/lib/kamailio/modules_k/permissions.so) ERROR: error while initializing modules I don't use this function in kamailio.cfg.. just allow_trusted !!! Any idea why this error still appears? Best regards -- *Grégoire Vandendeurpel, * * * *IT Sector*

2 1

Detect and Correct wrong BYEs from proxy.
by Aft nix 23 May '12

23 May '12

Hi all, When record routing is enabled, that means, the proxy will act statefull mode, There has to be a mechanism for storing dialogue states. Like the location of the peers (Contact header filed of initial INVITE and 200 OK). And i suspect these states are stored in variables which has a lifespan same as the dialogue itself. I guess important variables (Contact header, local and remote tag etc) are exported to configuration file also. I'm aware of the dialogue module but i'm kind of convinced there has to be a simpler way for achieving my goal. Our Sip gateway provider sending a wrong BYE when PSTN side phone hangs up. Instead of using Contact Header of the initial INVITE, its sending BYEs with its IP as ruri. So the BYE message is looping between proxy and gateway instead of relayed to UAC. What i'm trying to achieve is following : route[RELAY] { if(is_method('BYE')) { <rewrite ruri with contact field of the initial invite> t_relay() } } -- -aft

1 0

Siremis 3.2.1 PDO Issue.
by Jeremy Ardley 23 May '12

23 May '12

Hi, I'm installing a brand-new Kamailio / Siremis system. I have an issue with Siremis failing to load PDO extensions. I have other systems working O.K. on Centos 5.x This is my configuration: Centos 6.2 php-5.3.3-3.el6_2.8.i686.rpm php-cli-5.3.3-3.el6_2.8.i686.rpm php-common-5.3.3-3.el6_2.8.i686.rpm php-mysql-5.3.3-3.el6_2.8.i686.rpm php-pdo-5.3.3-3.el6_2.8.i686.rpm Siremis 3.2.1 php -m [PHP Modules] bz2 calendar Core ctype curl date ereg exif fileinfo filter ftp gettext gmp hash iconv json libxml mysql mysqli openssl pcntl pcre PDO pdo_mysql pdo_sqlite Phar readline Reflection session shmop SimpleXML sockets SPL sqlite3 standard tokenizer xml zip zlib [Zend Modules] The area where it screws up is in util.php while I am setting up Siremis via web-page $status[5]['item'] = 'PDO extensions'; $pdos = array(); if (extension_loaded('pdo')) $pdos[] = "pdo"; if (extension_loaded('pdo_mysql')) $pdos[] = "pdo_mysql"; if (extension_loaded('pdo_mssql')) $pdos[] = "pdo_mssql"; if (extension_loaded('pdo_oci')) $pdos[] = "pdo_oci"; if (extension_loaded('pdo_pgsql')) $pdos[] = "pdo_pgsql"; $status[5]['value'] = implode(", ", $pdos); $status[5]['status'] = $pdos[0]=='pdo' ? 'OK' : 'FAIL - PDO extensions are required.'; I have commented out the test but it fails later saying the PDO extensions are not loaded. I note that the php modules report gives upper case but the test is lower case. (I understand the extension_loaded function is case insensitive) Any assistance appreciated.

1 0

Rtpproxy and UPDATEs
by Spencer Thomason 22 May '12

22 May '12

Hello, I'm working on a residential type application where we are using Kamailio for NAT traversal and Freeswitch as a voicemail and media server. When a UA that is behind NAT sends an INVITE to check voicemail everything works correctly until the user listens to the message. The sdp in the initial INVITE is rewritten and rtp proxy is working but Freeswitch (on a public IP) then sends an UPDATE to display the caller name of the person who left the message. The problem is that the UAC (in this case a Polycom phone) then responds with its private IP in the SDP. Is there a was to handle these UPDATEs? I'm using Kamailio 3.2.3 with a fairly stock config. This is an excerpt of the config file with the NAT handling route: # RTPProxy control route[NATMANAGE] { if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } return; } Thanks, Spencer

3 7

kamailio debug send messages
by Bruno Bresciani 22 May '12

22 May '12

Hi all, Why kamailio send to syslog only received SIP messages? I want see the messages that kamailio send to the UAs, it's possible? On TCP/TLS SIP messages there is a degub of buffer with the message that will be send, but a strange content has been shown... following the log: *tcp_send: buf= ^W^C^A^B0HFCÃ°g^W^R Â£MÃ<8f>C^U<8f>ÃQB^^,Â¼8ÃÃ^KZÃÃ|<97>Â¼zÃÃ¶<82>Ã¬Ã±Ã%sÃÃ¢Â³gÂ¿Ã SÂ©<97>}y<91>.<94>Ã°Â¯Ã´<8f>YÂ¿Â¥Âµf<9d>Ã«<8f><9d>* My kamailio is version 3.1.2 and run with global parameter 'debug=3' and send the syslog messages. Best Regards

7 15

SQLite datatypes
by Sebastian Ferguson 22 May '12

22 May '12

Hi: I'm trying to configure a Kamailio 3.2 server as a Redirect Server. A softswitch we own will send a SIP INVITE for some calls like this "INVITE sip:102@sipwise.telecentro.net.ar SIP/2.0". Once detected the B number "102" (in this example), the kamailio server will search in a sqlite3 database the transltated number the call should be directed. And then the server will send a 300 Redirect with the new request uri. Like this "SIP/2.0 300 Redirect /.../ Contact: sip:63793266@voip.telecentro.net.ar". And the softswitch will let the call go to the right number. The translation desition is made based on the geographic area where the A party is located. If the translated number is like 63793266 or 1557311721 the system is working ok. So the .cfg file seems to be OK. But if the translated number is like 8003330303 the variable holding the SQL return is set to "-486604289". I think this is related to data types. In the logs I can see that kamailio is defaulting datatypes to INT (I've tested varchar, int, and bigint in sqlite3), but I don't know how large could an integer be in Kamailio. According to sqlite3 documentation the number 8003330303 is not long enough and in fact the number is in the database. Has anybody seen something like this? Can anyone help? I can provide the hole script and database if this helps. Regards, Sebastian Ferguson

3 8

Free Tickets for LinuxTag 2012
by Carsten Bock 22 May '12

22 May '12

Hi, we've got some free tickets for LinuxTag 2012, we want to share. If you like to receive one, just send me a short reply. They are given away on a first come, first served basis. Meet us at Hall 7.2b, Booth 278! So long, Carsten P.S.: I'll give a presentation about Kamailio, IMS and Rich Communications on Wednesday, May 23rd, 12:45 - 13:30 (Room Berlin I). -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ -- Meet ng-voice at LinuxTag 2012 in Berlin - May 23rd - 26th, 2012. Save the date!

1 0

Priority of targets when using uac_redirect
by Stefan Prelle 21 May '12

21 May '12

Hi, I am using kamailio 3.2.2 and the uac_redirect module to evaluate the response from a SIP redirect server. My problem is, that kamailio does not pick the target with the highest priority, but the one with the lowest. In the 302 the kamailio is receiving, there is a contact header like this: Contact:<sip:+49xxx@a.a.a.a:5060;user=phone>;q=0.5,<sip: +49xxx@b.b.b.b:5060;user=phone>;q=0.25. In my failure_route block I basically use the following commands get_redirects("1"); t_relay(); The call then gets redirected to b.b.b.b whereas I would expect it to go to a.a.a.a When I change my script to get_redirects("*"); the call is forked to both IPs in parallel (starting with a.a.a.a) Is this a bug in uac_redirect or am I using it in a wrong way? Regards, Stefan

1 0

installing the kamailio-devel problem with db_mysql
by Uri Shacked 21 May '12

21 May '12

Hi, i am trying to install the devel version using git. did the changes on the modules,lst file to include "db_mysql" i have mysql5.5 already installed with the shared components and client... i have no mysql.h file... so, the "make all" of kamailio failes... any ideas ? BR, Uri

3 3

Presence dialoginfo expiry
by Tais Plougmann Hansen 18 May '12

18 May '12

Hi, I have presence/pua_dialoginfo set up and working well with 3.2.3. The only problem I'm facing right now is dialog event expiry from presentity. If there's no call activity for a while, the dialog event will expire and should a client reboot after this, they'll start up with unknown state in BLF subscriptions. I.e. question marks in the display on Aastra phones and will stay that way until there's new call activity. Is there a way to keep dialog events fresh even without call activity? I've tried setting override_lifetime very high (604800) but the events still managed to get removed after an hour or so. -- Tais Plougmann Hansen OSD Consulting ApS

1 0

Re: [SR-Users] Change the Server field on the SIP dialog
by Charles Chance 18 May '12

18 May '12

Actually, for your version the documentation is at http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#server_header but it's the same thing :) Regards, Charles http://www.sipcentric.com -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Charles Chance Sent: 18 May 2012 11:12 To: sal(a)kom-1.com; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER)-Users Mailing List' Subject: Re: [SR-Users] Change the Server field on the SIP dialog Hi, Have you tried setting server_header? See here for more info: http://www.kamailio.org/wiki/cookbooks/3.2.x/core#server_header Regards, Charles http://www.sipcentric.com ________________________________________ From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Sal Aguilar Sent: 17 May 2012 19:00 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Change the Server field on the SIP dialog Hi Kamailio/SIP Router List: I turn to you guys since I could find a straight forward answer (it seems my google skills have dimished) on the wiki. Yesterday evening, my boss asked me about the possibility of replacing the value of the Server field, that our Kamailio 3.1 SIP Dispatcher is sending to customers. I want to know if this can be accomplished by adding a statement into the kamailio.cfg or is there something else needed. Below is a sample of the SIP dialog that we are currently sending to customers. Call-ID: 0a63682b2d021cbf4ddeac4833fe1b10(a)192.168.1.2. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. Your feedback would be greatly appreciated. Salvador Aguilar Email: sal(a)kom-1.com USA: +1.305.523.9581 Colombia: +57.1.381.9533 Mobile: +505.8672.4725 KOM 1 Communication ~more than just talk. www.kom-1.com ________________________________________ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5006 - Release Date: 05/17/12 _______________________________________________ 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 ----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2176 / Virus Database: 2425/5006 - Release Date: 05/17/12

1 0

Change the Server field on the SIP dialog
by Sal Aguilar 18 May '12

18 May '12

Hi Kamailio/SIP Router List: I turn to you guys since I could find a straight forward answer (it seems my google skills have dimished) on the wiki. Yesterday evening, my boss asked me about the possibility of replacing the value of the Server field, that our Kamailio 3.1 SIP Dispatcher is sending to customers. I want to know if this can be accomplished by adding a statement into the kamailio.cfg or is there something else needed. Below is a sample of the SIP dialog that we are currently sending to customers. Call-ID: 0a63682b2d021cbf4ddeac4833fe1b10(a)192.168.1.2. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. Your feedback would be greatly appreciated. Salvador Aguilar Email: <mailto:sal@kom-1.com> sal(a)kom-1.com USA: +1.305.523.9581 Colombia: +57.1.381.9533 Mobile: +505.8672.4725 KOM 1 Communication ~more than just talk. www.kom-1.com <http://www.kom-1.com/>

2 1

sdpops command not working
by Grégoire Vandendeurpel 18 May '12

18 May '12

Hello, I use Kamailio 3.2.3 and I used the command from the sdpops module " sdp_remove_media("video");" But when i restarted kamailio, this error appeared: 0(2623) ERROR: <core> [cfg.y:3393]: cfg. parser: failed to find command sdp_remove_media 0(2623) : <core> [cfg.y:3532]: parse error in config file /etc/kamailio/kamailio.cfg, line 721, column 27: unknown command, missing loadmodule? Do You know why ? Cause I loeded the sdpops module. Thanks in advance for your help -- *Grégoire Vandendeurpel, * * * *IT Sector*

2 13

Question about LCR ping option
by Fabian Borot 17 May '12

17 May '12

Hello I am using kamailio 1.5.2. Our kamailio server has multiple IPs, and 1 of them happens to be a private IP for management purposes. When enabling the "ping" feature of the LCR module the machine sends the OPTIONS messages from the private IP. I need it to use at least any of the public IPs and if I can specify it it would be even better. I tried using the directive "ping from" with one of the public IPs but it only uses it to populate the "From" header, not to send the message from. One option could be to configure the configuration file with the "listen" directive including only the public IPs. However, there are more than 50 IPs, is there a limit to the number of IPs? I tried with a subnet and it is not allowed. Is there any other work around? txs a lot fborot

1 0

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 17 May '12

17 May '12

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

2 1

dlz-ldap-enum - expose LDAP data to Kamailio via ENUM
by Daniel Pocock 17 May '12

17 May '12

I've recently released a dlz ENUM module for the bind9 nameserver: http://www.opentelecoms.org/dlz-ldap-enum Basically, it handles ENUM queries from Kamailio, Asterisk, FreeSWITCH, repro, Lumicall, searches for the phone number in LDAP, and if found, returns the email address as both a SIP address and Jabber address This should make it even easier than ever before to get federated VoIP up and running using email addresses interchangeably with phone numbers. If the data already exists in LDAP as an address book, then just install bind9, install the module and you're up and running. Regards, Daniel

1 0

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 17 May '12

17 May '12

Hello, can you print in frame one: p *ps p *(ua_pres_t*)(*ps->param) I wonder why entity is null Is this a test system where eventually one can get remote access for live troubleshooting while you are testing? Cheers, Daniel On 5/16/12 2:02 PM, Charles Chance wrote: > > Hi Daniel, > > It happens every time but I just can't seem to pinpoint the cause. > Those parameters were initially unset and it still happened -- I added > them later in an attempt to diagnose. > > I removed them and re-ran, which produces a slightly different > backtrace... > > Program terminated with signal 11, Segmentation fault. > > #0 core_hash (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at > ../../hashes.h:277 > > 277 end=s1->s+s1->len; > > (gdb) bt > > #0 core_hash (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at > ../../hashes.h:277 > > #1 publ_cback_func (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at > send_publish.c:246 > > #2 0x005041d6 in run_trans_callbacks_internal (cb_lst=0xb3dc8e78, > type=1024, trans=0xb3dc8e38, params=0xbfcdd5d8) at t_hooks.c:290 > > #3 0x005044d6 in run_trans_callbacks (type=1024, trans=0x200, > req=0x0, rpl=0xb7c30340, code=412) at t_hooks.c:317 > > #4 0x0052db28 in local_reply (t=0xb3dc8e38, p_msg=0xb7c30340, > branch=0, msg_status=412, cancel_data=0xbfcdd86c) at t_reply.c:2001 > > #5 0x00530b45 in reply_received (p_msg=0xb7c30340) at t_reply.c:2350 > > #6 0x0809d5f4 in forward_reply (msg=0xb7c30340) at forward.c:790 > > #7 0x080e11a9 in receive_msg ( > > buf=0x82c4fa0 "SIP/2.0 412 Conditional request failed\r\nVia: > SIP/2.0/UDP xx.xx.xx.xxx;branch=z9hG4bK7058.f06a8876.0\r\nTo: > sip:152@xx.xx.xx.xxx:5060;tag=03373b1b433b65504ea28bd3d9e6db76.331b\r\nFrom: > sip:152@xx.xx.xx."..., len=374, rcv_info=0xbfcdda58) at receive.c:270 > > #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 > > #9 0x080b0426 in main_loop () at main.c:1633 > > #10 0x080b3d02 in main (argc=11, argv=0xbfcddd34) at main.c:2546 > > ...although the root cause appears to be the same: > > (gdb) frame 0 > > #0 core_hash (t=0xb3dc8e38, type=1024, ps=0xbfcdd5d8) at > ../../hashes.h:277 > > 277 end=s1->s+s1->len; > > (gdb) p *s1 > > Cannot access memory at address 0x40323531 > > (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 *hentity > > Cannot access memory at address 0x0 > > The PUBLISH seems to be sent ok for both caller/callee for states > trying/confirmed. On tear-down however, it appears to crash somewhere > after sending state terminated for the callee as demonstrated by the > log... > > May 16 11:48:57 ws1 /usr/local/kamailio-devel/sbin/kamailio[31954]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:000091@xx.xx.xx.xxx:5060 > > May 16 11:48:57 ws1 /usr/local/kamailio-devel/sbin/kamailio[31953]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:000091@ xx.xx.xx.xxx:5060 > > May 16 11:48:57 ws1 /usr/local/kamailio-devel/sbin/kamailio[31953]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:152@ xx.xx.xx.xxx:5060 > > May 16 11:48:57 ws1 /usr/local/kamailio-devel/sbin/kamailio[31953]: > INFO: <core> [mem/f_malloc.c:529]: freeing a free fragment > (0xb3dc7c18/0xb3dc7c20) - ignore > > May 16 11:49:08 ws1 /usr/local/kamailio-devel/sbin/kamailio[31958]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:000091@ xx.xx.xx.xxx:5060 > > May 16 11:49:08 ws1 /usr/local/kamailio-devel/sbin/kamailio[31958]: > INFO: pua_dialoginfo [dialog_publish.c:386]: CALLING dialog_publish > for URI sip:152@ xx.xx.xx.xxx:5060 > > May 16 11:49:08 ws1 /usr/local/kamailio-devel/sbin/kamailio[31953]: > ERROR: presence [presentity.c:784]: No E_Tag match a.1337165329.31958.1.0 > > May 16 11:49:09 ws1 /usr/local/kamailio-devel/sbin/kamailio[31941]: > ALERT: <core> [main.c:785]: child process 31958 exited by a signal 11 > > I've attached the records from the presentity table also, as the > presence module doesn't appear to be matching the PUBLISHes with > existing records. > > Thanks again for your time, > > Charles > > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* 16 May 2012 08:15 > *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, > > hmm, hentity is null even a check for its value is done above ... > > Is this reproducible or happens from time to time? What are the values > you set for $avp(s:puburis_caller) and $avp(s:puburis_callee)? > > Can you try without parameters: > > modparam("pua_dialoginfo", "use_pubruri_avps", 1) > modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") > modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") > > These were added by a recent contribution, I want to see if they are > the reason. > > Cheers, > Daniel > > On 5/16/12 12:00 AM, Charles Chance wrote: > >> Hi Daniel, >> >> I tried to check that already, but it's not there: >> >> (gdb) frame 1 >> >> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) >> >> at send_publish.c:246 >> >> 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); >> >> (gdb) p *hentity >> >> Cannot access memory at address 0x0 >> >> Config parameters are: >> >> modparam("pua", "db_url", "mysql://user:pass@xx.xx.xx.xxx/dbname" >> <mailto:mysql://user:pass@xx.xx.xx.xxx/dbname>) >> >> modparam("pua", "db_table", "kam_pua") >> >> modparam("pua_dialoginfo", "caller_confirmed", 1) >> >> modparam("pua_dialoginfo", "include_localremote", 0) >> >> modparam("pua_dialoginfo", "override_lifetime", 300) >> >> modparam("pua_dialoginfo", "use_pubruri_avps", 1) >> >> modparam("pua_dialoginfo", "pubruri_caller_avp", >> "$avp(s:puburis_caller)") >> >> modparam("pua_dialoginfo", "pubruri_callee_avp", >> "$avp(s:puburis_callee)") >> >> Cheers, >> >> Charles >> >> ------------------------------------------------------------------------ >> >> *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> *Sent:* 15 May 2012 21:58 >> *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, >> >> interesting that the output of bt shows inline core_hash function >> prototype with same parameters as publ_cback_func, not being able to >> spot eventual wrong parameter values. >> >> Anyhow, in the gdb, can you do the commands: >> >> frame 1 >> p *hentity >> >> and send the output? Let's see the value of the pub uri for which >> hashing function crashes. >> >> What are the parameters for pua and pua_dialog info you set in the >> config file? >> >> Cheers, >> Daniel >> >> On 5/15/12 11:53 AM, Charles Chance wrote: >> >>> Hi Daniel, >>> >>> As per your previous reply (sorry, I deleted it accidentally), >>> please find attached output of 'bt full' as requested. >>> >>> Best regards and thanks again, >>> >>> Charles >>> >>> ------------------------------------------------------------------------ >>> >>> *From:*Charles Chance [mailto:charles.chance@sipcentric.com] >>> *Sent:* 13 May 2012 11:21 >>> *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) >>> -Users Mailing List' >>> *Subject:* (Devel) Segmentation fault using pua_dialoginfo >>> >>> Hi, >>> >>> Currently testing devel version and experiencing the following when >>> using pua_dialoginfo: >>> >>> ------------------------------------- >>> >>> Program terminated with signal 11, Segmentation fault. >>> >>> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> ../../hashes.h:279 >>> >>> 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; >>> >>> (gdb) bt >>> >>> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> ../../hashes.h:279 >>> >>> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> send_publish.c:246 >>> >>> #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, >>> type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 >>> >>> #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, >>> req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 >>> >>> #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, >>> branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 >>> >>> #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 >>> >>> #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 >>> >>> #7 0x080e11a9 in receive_msg ( >>> >>> buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP >>> xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: >>> sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom >>> <sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2%5Cr%5CnFrom>: >>> sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, >>> rcv_info=0xbfcc3648) at receive.c:270 >>> >>> #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 >>> >>> #9 0x080b0426 in main_loop () at main.c:1633 >>> >>> #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 >>> >>> (gdb) up >>> >>> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> send_publish.c:246 >>> >>> 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); >>> >>> (gdb) p hentity->pres_uri >>> >>> Cannot access memory at address 0x8 >>> >>> (gdb) p *hentity >>> >>> Cannot access memory at address 0x0 >>> >>> ------------------------------------- >>> >>> Your time is greatly appreciated... >>> >>> 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: 2012.0.2171 / Virus Database: 2425/5000 - Release Date: 05/15/12 >> > > > -- > 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: 2425/5001 - Release Date: 05/15/12 > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

un-suscribe
by Davis Doe 17 May '12

17 May '12

Please i would not like to receive mail from the kamailio group thanks davis

1 0

Setting up siremis 2.0
by pranjal deka 17 May '12

17 May '12

Hi, I have a sip registrar.(kamailio server installed in one server) I want to use the siremis 2.0 to collect management information from the sip registrar. Can you guide me how to make the connection between the sip registrar and the siremis so that the stats can be collected? Thanks

1 0

How to use Async module of Kamailio
by 罗俊明 17 May '12

17 May '12

1 0

Re: [SR-Users] How does RLS work with kamailio.
by Gnaneshwar Gatla 17 May '12

17 May '12

Hello, I have partly fixed this, I was using an incorrect xml format and missing usage of pua module. It took a little more than debugging to find this out. Now I have run into this new issue. I have implemented a presence + rls + xcap server: When client subscribes for RLS, kamailio Notifies with resourcelist for the client. The pua within the server does backend subscribtions. These backend subscribtions are responded with 202 Accepted replies. When the resources in the contact list change their presence information the pua generates NOTIFIY with subscription state: pending. Rls module does not reply back. Kamailio does not return from the following command. if(is_method("NOTIFY")) { xlog("handle notify"); rls_handle_notify(); xlog("notify was handled"); exit; }; Thank you for looking into this. Regards Gnani From: Gnaneshwar Gatla Sent: Thursday, April 19, 2012 4:53 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Cody Herzog Subject: How does RLS work with kamailio. Hello, I'm trying to implement RLS with integrated XCAP server. I have successfully run the Kamailio instance with the following snippet. Since I had no RLS clients available, I have tried to test this with SIPp. The attached pcap trace is the result. Test: 1. Send an XCAP signal over http to Kamailio. 2. Register and subscribe Kamailio. The test does store the xml received over http in the xcap table in the database. But rls module does not store the data neither in rls_presentity or rls_watcher. When I sent the Subscribe with sipp, the server throws error that subscription was not found in the database. I would like to know if I'm doing something wrong to implement this service. Is there any brief tutorial on how to implement this? Kamailio.cfg snippet: ##Presence modparam("presence", "server_address", PRESENCE_IP) modparam("presence", "clean_period", 65) modparam("presence", "db_update_period", 60) modparam("presence", "max_expires", 75) modparam("presence", "fallback2db", 1) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) ##rls modparams modparam("rls", "db_url", DBURL) modparam("rls", "waitn_time", 10) modparam("rls", "integrated_xcap_server", 1) modparam("rls", "max_notify_body_length", 32000) modparam("rls", "to_presence_code", 10) modparam("rls", "server_address", RLS_URI) modparam("rls", "outbound_proxy", PUA_OUTBOUND_PROXY) modparam("rls", "xcap_root", XCAP_ROOT) ##XCAP modparams modparam("xcap_server", "db_url", DBURL) modparam("xcap_client", "db_url", DBURL) modparam("xcap_client", "query_period", 50) route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { $var(ret_code)= rls_handle_subscribe(); xlog("Subscribe: $var(ret_code)"); if($var(ret_code)== 10) handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } In the [xhttp:request]: switch($rm) { case "PUT": xcaps_put("$var(uri)", "$hu", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } else if ($xcapuri(u=>auid)=~"resource-lists" || $xcapuri(u=>auid)=~"rls-services") { xlog("==== xhttp put: refreshing rls watchers $var(uri)\n"); rls_update_subs("$var(uri)", "presence"); } exit; break; Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

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Re: [SR-Users] kamailio Crashes with rls module
by Gnaneshwar Gatla 17 May '12

17 May '12

Oppsie.. forgot to attach the document. Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199 From: Gnaneshwar Gatla Sent: Thursday, April 26, 2012 2:14 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: kamailio Crashes with rls module Hello, I'm experimenting the usage of rls module with Kamailio. I have been able to successfully update xcap documents on Kamailio XCAP server. I have tried to use SIPp to get a NOTIFY for subscribing a rls-service list. Kamailio does get the xml from the xcap table and tries to parse it and crashes. I'm not sure if this is because the format of the XML, I derived the xml format from the RFC 4662 and 4826. I have attached the Kamailio log. I need help with this. Any help is appreciated, thank you very much. Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

2 6

TLS performance testing
by Kristian Kielhofner 16 May '12

16 May '12

Hello everyone, I'm doing some TLS performance testing on Kamailio 3.2.1. Here's my setup: kamailio -V version: kamailio 3.2.1 (x86_64/linux) 31c991 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, USE_STUN, 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: 31c991 compiled on 19:38:03 Dec 20 2011 with gcc 4.4.6 uname -a Linux null.null.com 2.6.32-220.el6.x86_64 #1 SMP Tue Dec 6 19:48:22 GMT 2011 x86_64 x86_64 x86_64 GNU/Linux CentOS 6.2 on a Dell PowerEdge R610 with 24 Intel X5650 Cores at 2.67GHz and 12GB of RAM (I could use more). Kamailio is the default config with a few changes: - WITH_TLS defined - TLS is using self-generated CAs/certs (essentially openvpn easy-rsa) with 1024 bit key size - TLS is *not* configured to verify client OR server certs by default - I'm using TLS v1 (SSL 3.1) - TLS cipher suites are set to any (although my simulated UAs only offer AES 256+SHA) - Various changes to Kamailio children (up to 256 at times) and memory sizes (up to 2048mb and even 4096mb at times) - One DNS based alias added - Maximum TCP connections increased to 65000 - Kamailio is configured to only listen on tested IP (UDP, TCP, TLS sockets active) - Syslog has been configured to log local0 (Kamailio) asynchronously My test rig/call generator is an Ixia Xcellon-Ultra NP load module with IXLoad. My call scenario does the following: - Registers two simulated user agents (100000, 200000) to Kamailio with TLS - Places call from 100000 to 200000 via Kamailio with TLS - Increments both user agents by 1 and continues as quickly (cps) as I like up to a channel limit (also configurable) - The Ixia generates a valid SDP but no RTP is generated (although that's certainly possible at these call levels) Two 1 gig ports on the Ixia are connected to the Broadcom NICs on the Dell R610 via a Cisco Catalyst 4948 switch. One port on the Ixia emulates the 100000 agents (A leg) and the other emulates the 200000 agents (B leg). Of course I can provide more information if needed. Here are some test numbers: With TLS at 20cps, 120 sec calls, up to a total of 2470 calls (4940 registrations) life is good. Very good - call setup time averages 23ms, the cps rate holds indefinitely, and not a single call or registration fails over long term tests. UDP and TCP numbers are excellent (bordering ridiculous) - usually around 500cps with practically no reasonable upper limit on simultaneous calls. This doesn't need any further discussion :). The TLS numbers start falling apart pretty quickly after 20cps, however. If I change the TLS test to 40cps, 120 sec calls, up to a total of 4940 calls (9,880 registrations) Kamailio starts to (seriously) struggle. The rate starts fluctuating all over the place, call setup time averages jump to 8000ms (or more) and things just generally get ugly. Interestingly enough all of the user agents are able to register, the logs look fine (to my eye at this log level) and the system (CPU, network, etc) doesn't appear to be under stress at all. I have a few questions: 1) Is there something obviously wrong or stupid I'm doing here? 2) Why are the TLS tests so much worse than TCP and UDP? Am I missing something here? Thanks (in advance) for any advice anyone might be able to offer! -- Kristian Kielhofner

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Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Marius Zbihlei 16 May '12

16 May '12

On 05/16/2012 03:02 PM, Charles Chance wrote: >>> >>> Currently testing devel version and experiencing the following when >>> using pua_dialoginfo: >>> >>> ------------------------------------- >>> >>> Program terminated with signal 11, Segmentation fault. >>> >>> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >>> ../../hashes.h:279 >>> >>> 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; >>> >>> Hello, Don't know if this is related, but the statement above hides a nasty bug: p[1] is a char (as for the description), so is signed, shifting to the left 16 bits is ok, as an integral promotion takes place. The problem is that if p[1] is negative (-1 is 0xff), then the integral promotion is signed as well and is complementary of 2 (or whatever the implementation considers), and the shift will return another value as expected. For example : #include <stdio.h> int main(){ char b = 0xff; unsigned h = (b<<16); printf("%x\n", h); return 0; } will print ffff0000 instead of ff0000 as expected. Haven't calculated the implication of this related to the bug. An fix would be to cast p[1] to (unsigned char) Cheers, Marius -- Zbihlei Marius Head of Linux Development Services Romania 1&1 Internet Development srl Tel KA: 754-9152 Str Mircea Eliade 18 Tel RO: +40-31-223-9152 Sect 1, Bucuresti mailto: marius.zbihlei(a)1and1.ro 71295, Romania

2 1

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 16 May '12

16 May '12

Hello, hmm, hentity is null even a check for its value is done above ... Is this reproducible or happens from time to time? What are the values you set for $avp(s:puburis_caller) and $avp(s:puburis_callee)? Can you try without parameters: modparam("pua_dialoginfo", "use_pubruri_avps", 1) modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") These were added by a recent contribution, I want to see if they are the reason. Cheers, Daniel On 5/16/12 12:00 AM, Charles Chance wrote: > > Hi Daniel, > > I tried to check that already, but it's not there: > > (gdb) frame 1 > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) > > at send_publish.c:246 > > 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); > > (gdb) p *hentity > > Cannot access memory at address 0x0 > > Config parameters are: > > modparam("pua", "db_url", "mysql://user:pass@xx.xx.xx.xxx/dbname") > > modparam("pua", "db_table", "kam_pua") > > modparam("pua_dialoginfo", "caller_confirmed", 1) > > modparam("pua_dialoginfo", "include_localremote", 0) > > modparam("pua_dialoginfo", "override_lifetime", 300) > > modparam("pua_dialoginfo", "use_pubruri_avps", 1) > > modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") > > modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") > > Cheers, > > Charles > > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* 15 May 2012 21:58 > *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, > > interesting that the output of bt shows inline core_hash function > prototype with same parameters as publ_cback_func, not being able to > spot eventual wrong parameter values. > > Anyhow, in the gdb, can you do the commands: > > frame 1 > p *hentity > > and send the output? Let's see the value of the pub uri for which > hashing function crashes. > > What are the parameters for pua and pua_dialog info you set in the > config file? > > Cheers, > Daniel > > On 5/15/12 11:53 AM, Charles Chance wrote: > >> Hi Daniel, >> >> As per your previous reply (sorry, I deleted it accidentally), please >> find attached output of 'bt full' as requested. >> >> Best regards and thanks again, >> >> Charles >> >> ------------------------------------------------------------------------ >> >> *From:*Charles Chance [mailto:charles.chance@sipcentric.com] >> *Sent:* 13 May 2012 11:21 >> *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) >> -Users Mailing List' >> *Subject:* (Devel) Segmentation fault using pua_dialoginfo >> >> Hi, >> >> Currently testing devel version and experiencing the following when >> using pua_dialoginfo: >> >> ------------------------------------- >> >> Program terminated with signal 11, Segmentation fault. >> >> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >> ../../hashes.h:279 >> >> 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; >> >> (gdb) bt >> >> #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >> ../../hashes.h:279 >> >> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >> send_publish.c:246 >> >> #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, >> type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 >> >> #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, >> req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 >> >> #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, >> branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 >> >> #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 >> >> #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 >> >> #7 0x080e11a9 in receive_msg ( >> >> buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP >> xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: >> sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom >> <sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2%5Cr%5CnFrom>: >> sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, >> rcv_info=0xbfcc3648) at receive.c:270 >> >> #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 >> >> #9 0x080b0426 in main_loop () at main.c:1633 >> >> #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 >> >> (gdb) up >> >> #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at >> send_publish.c:246 >> >> 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); >> >> (gdb) p hentity->pres_uri >> >> Cannot access memory at address 0x8 >> >> (gdb) p *hentity >> >> Cannot access memory at address 0x0 >> >> ------------------------------------- >> >> Your time is greatly appreciated... >> >> 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: 2012.0.2171 / Virus Database: 2425/5000 - Release Date: 05/15/12 > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

db_cluster module
by Uri Shacked 16 May '12

16 May '12

Hi, can the db_cluster module be used to all DB connections? if today i set for example "modparam("dialplan", db_url", DBURL)" i will be able to use "modparam("dialplan", db_url", CLUSTER)" ? still, on the #!define DBURL "mysql://.........." i can not use clustering right? BR, Uri

2 1

Re: [SR-Users] Siremis install fail at last page of web install--Solved
by Magnus 16 May '12

16 May '12

Just in case anyone comes up with same issue - post solving the directory/file ownership and repeating the import of db files - Siremis came up as it should Sorry for noise. -----Original Message----- From: Magnus Sent: Tuesday, May 15, 2012 4:35 PM To: sr-users(a)lists.sip-router.org Subject: RE: Siremis install fail at last page of web install "can't open cmp file to write " in the httpd error log I get two lines " mkdir(): No such file or directory in /var/www/html/siremis-3.2.0/openbiz/bin/Resource.php on line 408" and next line of "fopen(/var/www/html/siremis-3.2.0/siremis/files/cache/metadata/CB3F6C48_log Service.cmp): failed to open stream"

2 1

Siremis install fail at last page of web install
by Magnus 16 May '12

16 May '12

Hello all, I have an issue with final stage of installing siremis - at the last screen of the install I get an error in browser of "can't open cmp file to write" in the httpd log I get two lines " mkdir(): No such file or directory in /var/www/html/siremis-3.2.0/openbiz/bin/Resource.php on line 408" and next line of "fopen(/var/www/html/siremis-3.2.0/siremis/files/cache/metadata/CB3F6C48_log Service.cmp): failed to open stream" I have down a chown on the siremis dir to make the web process the owner, and this allowed me to get to the last page of the install, all shows as successful expect instead of a login page I get the error. The server platform is centos if makes any difference. Thanks Magnus

2 2

Re: [SR-Users] FW: (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 15 May '12

15 May '12

Hello, interesting that the output of bt shows inline core_hash function prototype with same parameters as publ_cback_func, not being able to spot eventual wrong parameter values. Anyhow, in the gdb, can you do the commands: frame 1 p *hentity and send the output? Let's see the value of the pub uri for which hashing function crashes. What are the parameters for pua and pua_dialog info you set in the config file? Cheers, Daniel On 5/15/12 11:53 AM, Charles Chance wrote: > > Hi Daniel, > > As per your previous reply (sorry, I deleted it accidentally), please > find attached output of 'bt full' as requested. > > Best regards and thanks again, > > Charles > > ------------------------------------------------------------------------ > > *From:*Charles Chance [mailto:charles.chance@sipcentric.com] > *Sent:* 13 May 2012 11:21 > *To:* 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) > -Users Mailing List' > *Subject:* (Devel) Segmentation fault using pua_dialoginfo > > Hi, > > Currently testing devel version and experiencing the following when > using pua_dialoginfo: > > ------------------------------------- > > Program terminated with signal 11, Segmentation fault. > > #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > ../../hashes.h:279 > > 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; > > (gdb) bt > > #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > ../../hashes.h:279 > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > send_publish.c:246 > > #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, > type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 > > #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, > req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 > > #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, > branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 > > #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 > > #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 > > #7 0x080e11a9 in receive_msg ( > > buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP > xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: > sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom: > sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, > rcv_info=0xbfcc3648) at receive.c:270 > > #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 > > #9 0x080b0426 in main_loop () at main.c:1633 > > #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 > > (gdb) up > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > send_publish.c:246 > > 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); > > (gdb) p hentity->pres_uri > > Cannot access memory at address 0x8 > > (gdb) p *hentity > > Cannot access memory at address 0x0 > > ------------------------------------- > > Your time is greatly appreciated... > > Charles > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

kamdbctl create changing mysql root password
by Tim King 15 May '12

15 May '12

This is not the first time I have had this issue during installation, but this time around I am having difficulty figuring out what I am doing wrong. Mysql is up and working, and client works. I can connect to it with the client and run queries without issue. When I run kamdbctl create I get the following. INFO: test server charset INFO: creating database openser ... ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: YES) ERROR: Creating core tables failed! It creates the database but than fails to create tables. From this point I can no longer connect to mysql with what was my root password. I have to go through the process of resetting my root password before I can log back into mysql. How is this script chaning or corrupting my mysql root password? This is the relevant content of my kamctlrc file: # $Id$ # # The Kamailio configuration file for the control tools. # # Here you can set variables used in the kamctl and kamdbctl setup # scripts. Per default all variables here are commented out, the control tools # will use their internal default values. ## your SIP domain SIP_DOMAIN=sbc.mydomain.net ## chrooted directory # $CHROOT_DIR="/path/to/ chrooted/directory" ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter. DBENGINE=MYSQL ## database host DBHOST=localhost ## database name (for ORACLE this is TNS name) DBNAME=openser # database path used by dbtext, db_berkeley or sqlite # DB_PATH="/usr/local/etc/kamailio/dbtext" ## database read/write user DBRWUSER=root ## password for database read/write user DBRWPW="mypassword" ## database read only user DBROUSER=root ## password for database read only user DBROPW="mypassword" ## database super user (for ORACLE this is 'scheme-creator' user) DBROOTUSER="root"

2 1

kamdbctl create changing mysql root password
by Tim King 15 May '12

15 May '12

This is not the first time I have had this issue during installation, but this time around I am having difficulty figuring out what I am doing wrong. Mysql is up and working, and client works. I can connect to it with the client and run queries without issue. When I run kamdbctl create I get the following. INFO: test server charset INFO: creating database openser ... ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: YES) ERROR: Creating core tables failed! It creates the database but than fails. From this point I can no longer connect to mysql with what was my root password. I have to go through the process of resetting my root password before I can log back into mysql. How is this script chaning or corrupting my mysql root password? This is the relevant content of my kamctlrc file: # $Id$ # # The Kamailio configuration file for the control tools. # # Here you can set variables used in the kamctl and kamdbctl setup # scripts. Per default all variables here are commented out, the control tools # will use their internal default values. ## your SIP domain SIP_DOMAIN=sbc.mydomain.net ## chrooted directory # $CHROOT_DIR="/path/to/chrooted/directory" ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter. DBENGINE=MYSQL ## database host DBHOST=localhost ## database name (for ORACLE this is TNS name) DBNAME=openser # database path used by dbtext, db_berkeley or sqlite # DB_PATH="/usr/local/etc/kamailio/dbtext" ## database read/write user DBRWUSER=root ## password for database read/write user DBRWPW="mypassword" ## database read only user DBROUSER=root ## password for database read only user DBROPW="mypassword" ## database super user (for ORACLE this is 'scheme-creator' user) DBROOTUSER="root"

1 0

Kamailio CS_Dirty
by Albert Petit 15 May '12

15 May '12

Hello, On a Kamailio deployment it seems some users get to a wrong state and permanently keeping in: CS_DIRTY Expires==0 for some long times (even half an hour). When they're in this status seems system cannot reach them. Normally , when working on USRLoc mode 2 if the registration expires usually what I observed is that it just not appears on 'kamctl ul show' The affected users are seen like this: AOR:: blue4 Contact:: sip:Blue4@192.168.101.6:5700 Q= Expires:: deleted Callid:: efc560dc2dbd248d4d1f9d39633773df Cseq:: 49995 User-agent:: PoC-client/OMA1.0 Genaker_X-Platform_IMS_SDK/v2.0 State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:10.133.21.141:6666 Methods:: 8143 The refresh from the clients seem to come intime however some users (and not all) got to this CS_DIRTY state instead of being removed or refreshed. The client refresh looks to arrive in-time so it should be refreshed. Which could be a common cause (misconfiguration?) for this type of problem? If not, could enable debug mode for usrloc or I need to enable for full kamailio? Thanks Albert

1 0

Question about dialog module (state 1)
by Ricardo Martinez 15 May '12

15 May '12

Hello. I’m using the dialog module to keep control of simultaneous calls. In some cases, and I’m still trying to find why this happens, the dialog stays in “STATE:: 1”, which according to the docs is a dialog which no provisional response has been sent yet. Is there a way to eliminate this kind of dialogs??? When I list the dialogs I can see : dialog:: hash=2523:2040584107 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: 25b8e5354ca0850a4d0bd9c43eeb617a(a)pxext.redvoiss.net from_uri:: sip:557100052213@pxext.redvoiss.net from_tag:: as10dc7c85 caller_contact:: sip:557100052213@200.68.19.44:5060 caller_cseq:: 103 caller_route_set:: caller_bind_addr:: udp:64.76.154.110:5060 callee_bind_addr:: to_uri:: sip:0981985480@pxext.redvoiss.net to_tag:: callee_contact:: callee_cseq:: callee_route_set:: when I try to eliminate this dialog : kamctl fifo dlg_terminate_dlg 25b8e5354ca0850a4d0bd9c43eeb617a(a)pxext.redvoiss.net 500 command 'dlg_terminate_dlg' failed Thanks in advance, Regards, Ricardo Martinez.-

2 4

[SIREMIS] send MI Commands to multiple kamailio servers
by SamyGo 15 May '12

15 May '12

Hello all, Following through the manual at : http://kb.asipto.com/siremis:install32x:mi-commands I'm able to send MI commands on external/remote server But next hurdle is how can I define multiple remote servers i.e <Remote name="remote0" address="127.0.0.1" port="8033"/> <Remote name="remote1" address="192.168.2.156" port="8033"/> <Remote name="remote2" address=" 192.168.2.150" port="8033"/> Is there any already working configuration else I think I may need to start a surgery on siremisMICommands.php file and make it loop over the remote tag and disptach commands until loop ends. is there any "MI-proxy" sort of thing available !!? Regards, Sammy.

3 5

Pickup group implementation in Kamailio
by M.C. 15 May '12

15 May '12

Hi all, I want to implement a "group pickup" (asterisk like) with Kamailio and Snom phones. I found in the changelog this comment: "- an useful application - call pickup: - INVITE comes in, callee is in a pickup group - store callid, cseq, etc in database or notify the other phones in the group (easy to do for snom phones and extra programmable buttons) via uac_req_send() - send a remote control command telling which call to pickup (callid, cseq) and where to redirect. In config, call t_cancel_callid(). in failure route for INVITE catch the cancelled transaction (flag is set), get the new destination (e.g., stored in htable) and forward the invite there"" regarding new functions in tmx module, but I have found no examples in kamailio documentation. Is this the way to implement a group pickup in kamailio? Regards, Matteo

2 2

FW: (Devel) Segmentation fault using pua_dialoginfo
by Charles Chance 15 May '12

15 May '12

Hi Daniel, As per your previous reply (sorry, I deleted it accidentally), please find attached output of 'bt full' as requested. Best regards and thanks again, Charles _____ From: Charles Chance [mailto:charles.chance@sipcentric.com] Sent: 13 May 2012 11:21 To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List' Subject: (Devel) Segmentation fault using pua_dialoginfo Hi, Currently testing devel version and experiencing the following when using pua_dialoginfo: ------------------------------------- Program terminated with signal 11, Segmentation fault. #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at ../../hashes.h:279 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; (gdb) bt #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at ../../hashes.h:279 #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at send_publish.c:246 #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 #7 0x080e11a9 in receive_msg ( buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom: sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, rcv_info=0xbfcc3648) at receive.c:270 #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 #9 0x080b0426 in main_loop () at main.c:1633 #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 (gdb) up #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at send_publish.c:246 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); (gdb) p hentity->pres_uri Cannot access memory at address 0x8 (gdb) p *hentity Cannot access memory at address 0x0 ------------------------------------- Your time is greatly appreciated. Charles

1 0

modify connection information field from a 200 OK response
by Openser Kamailio 15 May '12

15 May '12

hi, I try to change the IP address in connection information field from a 200 OK request passing through Kamailio. I want the IP address of the Kamailio in connection information. So, I've tried rtpproxy_answer("","@IP kamailio"), rtppproxy_offer("","@IP kamailio), rtpproxy_manage("","@IP kamailio"). It didn't change anything. Thanks for Help!

2 4

why is tcp_send failing?
by Juha Heinanen 15 May '12

15 May '12

i have been wondering why kamailio refuses to forward bye to $du sip:192.26.99.39:8000;transport=tcp;lr;ftag=a8266461a81c723o0 wireshark shows that it is not even trying to set up tcp connection to the destination and with telnet i can verify that tcp connection can be set up. debug at level 3 is below. -- juha May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: rr [loose.c:776]: URI to be processed: 'sip:192.26.99.39:8000;transport=tcp;lr;ftag=a8266461a81c723o0' May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: rr [loose.c:785]: Next URI is a loose router ... May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=(nil) May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=30919, isACK=0 May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb486de58, callback type 1, id 0 entered May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: d28c87700c1c48b71a3b7f63090bf160 May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: <core> [dns_cache.c:3400]: dns_sip_resolve(192.26.99.39, 0, 0), ip, ret=0 May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: <core> [msg_translator.c:204]: check_via_address(192.98.100.135, 192.168.0.145, 0) May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: ERROR: tm [../../forward.h:171]: msg_send: ERROR: tcp_send failed May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_fwd.c:1365]: t_send_branch: send to 192.26.99.39:8000 (2) failed May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: ERROR: tm [t_fwd.c:1383]: ERROR: t_send_branch: sending request on branch 0 failed May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_funcs.c:361]: ERROR:tm:t_relay_to: t_forward_nonack returned error May 12 15:58:30 sars /usr/sbin/sip-proxy[25453]: DEBUG: tm [t_funcs.c:369]: -477 error reply generation delayed

2 5

Re: [SR-Users] (Devel) Segmentation fault using pua_dialoginfo
by Daniel-Constantin Mierla 15 May '12

15 May '12

Hello, to get some info until I get the time to look more deeper, can you send also the output of 'bt full'? It will print also other variables that may help troubleshooting faster. Cheers, Daniel On 5/13/12 12:21 PM, Charles Chance wrote: > > Hi, > > Currently testing devel version and experiencing the following when > using pua_dialoginfo: > > ------------------------------------- > > Program terminated with signal 11, Segmentation fault. > > #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > ../../hashes.h:279 > > 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; > > (gdb) bt > > #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > ../../hashes.h:279 > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > send_publish.c:246 > > #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, > type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 > > #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, > req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 > > #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, > branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 > > #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 > > #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 > > #7 0x080e11a9 in receive_msg ( > > buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP > xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: > sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom: > sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, > rcv_info=0xbfcc3648) at receive.c:270 > > #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 > > #9 0x080b0426 in main_loop () at main.c:1633 > > #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 > > (gdb) up > > #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at > send_publish.c:246 > > 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); > > (gdb) p hentity->pres_uri > > Cannot access memory at address 0x8 > > (gdb) p *hentity > > Cannot access memory at address 0x0 > > ------------------------------------- > > Your time is greatly appreciated... > > Charles > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Extending app_python (patch)
by Konstantin M. 15 May '12

15 May '12

Hi, As I see, app_python module has almost no logging functionality except LM_ERR. As a python developer and user I need a more python functionality. did a patch (attached) against kamailio-3.2.3 for adding an additional logging and I'd like to share it. Testcase: 0(5403) INFO: app_python [python_iface.c:41]: Loggers.py: LM_GEN1: msg: Test Message 0(5403) WARNING: app_python [python_iface.c:60]: Loggers.py: LM_GEN2: msg: Test Message 0(5403) ALERT: app_python [python_iface.c:76]: Loggers.py: LM_ALERT: msg: Test Message 0(5403) CRITICAL: app_python [python_iface.c:92]: Loggers.py: LM_CRIT: msg: Test Message 0(5403) ERROR: app_python [python_iface.c:108]: Loggers.py: LM_ERR: msg: Test Message 0(5403) WARNING: app_python [python_iface.c:124]: Loggers.py: LM_WARN: msg: Test Message 0(5403) NOTICE: app_python [python_iface.c:140]: Loggers.py: LM_NOTICE: msg: Test Message 0(5403) INFO: app_python [python_iface.c:156]: Loggers.py: LM_INFO: msg: Test Message

2 1

Re: [SR-Users] pua_dialoginfo - publish for caller or callee
by charles.chance＠sipcentric.com 15 May '12

15 May '12

2 1

Re: [SR-Users] pua_dialoginfo - publish for caller or callee
by Daniel-Constantin Mierla 14 May '12

14 May '12

Hello, the upcoming 3.3 has some options to set the URIs for which to send the PUBLISH, see: http://kamailio.org/docs/modules/devel/modules_k/pua_dialoginfo.html#id2534… You can decide in config file how many publishes and where to be sent. But these are additions after 3.2. Cheers, Daniel On 5/12/12 3:38 PM, Charles Chance wrote: > > Hi, > > Quick question relating to pua_dialoginfo (3.2) - is there a way to > specify for which side (caller/callee) a PUBLISH is generated, or must > it always be for both? > > Cheers, > > Charles > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

kamdbctl error in Ubuntu 12.04
by Ryan Gholam 14 May '12

14 May '12

I am trying to install Kamailio with ( Sudo apt-get install ...) without recompiling etc .... , i reach a point where :> /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL > ERROR: database engine not loaded - tried 'MYSQL' any ideas how to solve this prob ? Thank you

2 4

TM Callback Error
by Reda Aouad 14 May '12

14 May '12

Hi, After upgrading from 3.2 to 3.3, I am getting the following error: ERROR: dialog [dlg_handlers.c:937]: registering TMCB to wait for negative ACK This is my version: kamailio -V version: kamailio 3.3.0-pre1 (i386/linux) 275c8a 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, F_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: 275c8a compiled on 13:14:05 May 6 2012 with gcc 4.5.2 And idea? Thanks Reda

3 3

Consulting needed
by Pete Ashdown 14 May '12

14 May '12

I have been trying to accomplish a couple tasks with Kamailio over the past month with no luck. What I need is a bit of one-on-one training with someone who knows the lay of the land. If you do this kind of consulting and can use Skype with possibly a shared-screen terminal, please drop me an email with your rate.

4 5

Kamailio SIP server and Cisco 7971G-GE
by KA Veelenturf 14 May '12

14 May '12

Hello everyone, I am trying to get a new SIP firmware version on our Cisco 7971G-GE. Now the problem is that I don't really know which configuration files have to be included to get everything working. For now on I have the new SIP firmware and SEP<mac>cnf.xml and the XMLDefault.cnf.xml files. Does anybody have a template for the .xml files to get a new firmware version working on our Cisco and also to get it work with Kamailio? With kind regards, Koen.

2 1

(Devel) Segmentation fault using pua_dialoginfo
by Charles Chance 13 May '12

13 May '12

Hi, Currently testing devel version and experiencing the following when using pua_dialoginfo: ------------------------------------- Program terminated with signal 11, Segmentation fault. #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at ../../hashes.h:279 279 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; (gdb) bt #0 core_hash (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at ../../hashes.h:279 #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at send_publish.c:246 #2 0x002861d6 in run_trans_callbacks_internal (cb_lst=0xb3d7e370, type=1024, trans=0xb3d7e330, params=0xbfcc31c8) at t_hooks.c:290 #3 0x002864d6 in run_trans_callbacks (type=1024, trans=0x200, req=0x0, rpl=0xb7bea008, code=200) at t_hooks.c:317 #4 0x002afb28 in local_reply (t=0xb3d7e330, p_msg=0xb7bea008, branch=0, msg_status=200, cancel_data=0xbfcc345c) at t_reply.c:2001 #5 0x002b2b45 in reply_received (p_msg=0xb7bea008) at t_reply.c:2350 #6 0x0809d5f4 in forward_reply (msg=0xb7bea008) at forward.c:790 #7 0x080e11a9 in receive_msg ( buf=0x82c4fa0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP xx.xx.xx.xxx;branch=z9hG4bKc13b.25313b6.0\r\nTo: sip:000094@xx.xx.xx.xxx;tag=03373b1b433b65504ea28bd3d9e6db76.d7f2\r\nFrom: sip:000094@xx.xx.xx.xxx;tag=4cc17c2c75cb82fa"..., len=393, rcv_info=0xbfcc3648) at receive.c:270 #8 0x0817801b in udp_rcv_loop () at udp_server.c:544 #9 0x080b0426 in main_loop () at main.c:1633 #10 0x080b3d02 in main (argc=11, argv=0xbfcc3924) at main.c:2546 (gdb) up #1 publ_cback_func (t=0xb3d7e330, type=1024, ps=0xbfcc31c8) at send_publish.c:246 246 hash_code= core_hash(hentity->pres_uri, NULL, HASH_SIZE); (gdb) p hentity->pres_uri Cannot access memory at address 0x8 (gdb) p *hentity Cannot access memory at address 0x0 ------------------------------------- Your time is greatly appreciated. Charles

1 0

pua_dialoginfo - publish for caller or callee
by Charles Chance 12 May '12

12 May '12

Hi, Quick question relating to pua_dialoginfo (3.2) - is there a way to specify for which side (caller/callee) a PUBLISH is generated, or must it always be for both? Cheers, Charles

1 0

change/replace anything in body message
by Konstantin M. 12 May '12

12 May '12

Hi, Is there a way to change/replace anything in body message ? Used: replace_body_atonce, replace_body_all, replace_all, subst_body. Nothing is working. Thanks,

2 2

(no subject)
by Albert Petit 11 May '12

11 May '12

Hello, On a Kamailio deployment it seems some users get to a wrong state and permanently keeping in: CS_DIRTY Expires==0 for some long times (even half an hour). When they're in this status seems system cannot reach them. Normally , when working on USRLoc mode 2 if the registration expires usually what I observed is that it just not appears on 'kamctl ul show' The affected users are seen like this: AOR:: blue4 Contact:: sip:Blue4@192.168.101.6:5700 Q= Expires:: deleted Callid:: efc560dc2dbd248d4d1f9d39633773df Cseq:: 49995 User-agent:: PoC-client/OMA1.0 Genaker_X-Platform_IMS_SDK/v2.0 State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:10.133.21.141:6666 Methods:: 8143 The refresh from the clients seem to come intime however some users (and not all) got to this CS_DIRTY state instead of being removed or refreshed. The client refresh looks to arrive in-time so it should be refreshed. Which could be a common cause (misconfiguration?) for this type of problem? If not, could enable debug mode for usrloc or I need to enable for full kamailio? Thanks Albert

1 0

Kamailio ignores some ACK
by Efelin Novak 11 May '12

11 May '12

Hi folks, I have a strange problem when Kamailio ignores ACKs in a specific scenario. The call flow is as follows: A -> INVITE -> kamailio -> INVITE -> B [omitting 100 and 180] A <- 200 OK <- kamailio <- 200 OK <- B A -> ACK -> kamailio There are INVITE Xlogs, Reply ROUTE xlogs and media-proxy logs in the syslog. However there is no information about these ACKs. No XLOGs are printed even if there is one on the top of the main route. "tcpdump -A -s0 -i any -n port 5060" receives this message correctly: 14:47:01.246153 IP 111.111.11.11.5060 > 80.80.80.80.60442: SIP, length: 915 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.11.11:5060;rport=60442;x-route-tag="tgrp:A";branch=z9hG4bK1634E6A88 Record-Route: <sip:111.111.11.11;lr;ftag=599248D4-260;vsf=AAAAAAAAAAAAAAAAAAAAW0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6> Contact: <sip:80.80.80.80:65002;transport=udp> To: "test_account"<sip:bob@server.com>;tag=cb7dd641 From: <sip:alice@111.111.11.50>;tag=599248D4-260 Call-ID: 9AFCFC51.11.50 CSeq: 101 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length:263 v=0 o=- 492575093 492575093 IN IP4 111.111.11.60 s=test_device i=(o=IN IP4 192.168.1.10) c=IN IP4 111.111.11.71 t=0 0 m=audio 16416 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 14:47:01.254511 IP 111.111.11.50.60442 > 111.111.11.11.5060: SIP, length: 521 ACK sip:80.80.80.80:65002;transport=udp SIP/2.0 Via: SIP/2.0/UDP 111.111.11.50:5060;x-route-tag="tgrp:A";branch=z9hG4bK1634E7DE8 From: <sip:alice@111.111.11.50>;tag=599248D4-260 To: "test_account"<sip:bob@server.com>;tag=cb7dd641 Call-ID: 9AFCFC51.11.50 Route: <sip:111.111.11.11;lr;ftag=599248D4-260;vsf=AAAAAAAAAAAAAAAAAAAAW0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6> Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 My Kamailio version is kamailio 3.1.0 (i386/linux) 1e204f. Does anybody knows where can be a problem? How can I check whether Kamailio receives something? ... Jan

5 6

error searching pvar "sp."
by Konstantin M. 11 May '12

11 May '12

Hello, When I have the following code: xlog("Dropped INVITE from unregistered location: $si:$sp.\n"); -- I'm getting a parse error: 0(3735) ERROR: <core> [pvapi.c:708]: error searching pvar "sp." 0(3735) ERROR: <core> [pvapi.c:912]: wrong char [./46] in [$sp. ] at [3 (0)] 0(3735) ERROR: xlog [xlog.c:314]: wrong format[Dropped INVITE from unregistered location: $si:$sp. ] When I changed to (removed a point after $sp): xlog("Dropped INVITE from unregistered location: $si:$sp\n"); -- No errors. What I'm doing wrong ? Or it's a parser bug ? Thanks!

2 1

t_continue usage
by Vineet Menon 11 May '12

11 May '12

Hi, the module doc page http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2525893says t_continue() has 3rd parameter which says which route block to execute. SO does that mean t_continue() should be present before routing logic(s)?? Where should be t_continue() be placed in a config?? Regards, Vineet Menon

2 1

Community dinner: Barcelona June 13th
by Olle E. Johansson 10 May '12

10 May '12

Hello! I will be running an Asterisk SIP Masterclass - the last one - in Barcelona in June. During this week, I will organize a dinner for everyone working with or interested in Asterisk, Kamailio and other Open Source platforms for realtime communication. It's Wednesday, June 13th somewhere in Barcelona - location will be announced later. You pay our own dinner (unless we can find sponsors) and enjoy the geeky company for free! To join the event, use this Facebook event https://www.facebook.com/events/307548349321608/ See you in Barcelona! /O ------ http://edvina.net - Open Unified Communication - training, consulting, workshops

1 0

kamailio - mysql master-slave connection question
by Uri Shacked 10 May '12

10 May '12

Hi, In my system configuration, i use 2 DB servers. One will be the master and the other is the slave. In kamailio cfg, I set the mysql server as a host name (not ip). On my DNS server the ip behind the name is the master`s address. I wonder, if I stop the mysqld on the master, and change the ip address in the DNS to the slave`s address. Than i do "kamctl dispatcher reload"... Will kamailio go to the DNS, find the address and connect to the slave? BR, Uri

1 0

[SIREMIS] Multiple kamailio management commands
by SamyGo 10 May '12

10 May '12

Hello list, I'm trying to have my Siremis interface send MI commands to multiple kamailio servers i.e reload dispatcher of all the kamailio servers when I reload from Siremis interface. The issue Im facing is that the commands dont get executed on any other server except localhost/ as mentioned in URL: http://kb.asipto.com/siremis:install32x:mi-commands I've edited the <Remote name="remote" address="127.0.0.1" port="8033"/> to : <Remote name="remote" address="192.168.2.156" port="8033"/> Please guide on how to have multiple kamailio receive command from single Siremis interface. Regards, Sammy.

2 8

Re: [SR-Users] Bug: Pua_reginfo
by Gnaneshwar Gatla 10 May '12

10 May '12

Hello, I have noticed that user records were not being correctly saved in the location table. I have tried to fix this will the attached code. This would fix reginfo_handle_notify() to store the user-record in the right format. This fixes part of the problem, I'm still trying to fix the incorrect uri in the PUBLISH message generated by this module. Regards, Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199 From: Gnaneshwar Gatla Sent: Tuesday, May 08, 2012 4:58 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Carsten Bock (carsten(a)ng-voice.com) Subject: Bug: Pua_reginfo Hello all, There seems to be an deadlock between registrar and pua_reginfo module whenever a SIP client refreshes its registration. 1. The pua_reginfo does not publish Event:reg when a sip client registers for the first time. 2. The pua_reginfo module does not use ul.lock_udomain() to lock the ul table to get the record. 3. The pua_reginfo does not release the record that it accessed from ul.get_urecord() Since the record is not released, when the client re-registers, the registrar module is unable to query for the userlocation and sends a 200 OK response with 0 bindings. Which leads to no contact header in the response message and eventually deletion of the record in the location table(expires). I have tried to fix this using lock_udomain, release_urecord and unlock_udomain, but failed to rectify the error. The deadlock happens in modules_k/usrloc/udomain.c (db_load_urecord()). if (ul_dbf.query(_c, keys, 0, vals, columns, (use_domain)?2:1, 13, order, &res) < 0) My Question: 1. is pua_reginfo supposed to delete record from location database and send the same data to a presence server? a. If above is true, then should the registrar module insert a new record in the location table? b. If above is false, the registrar module should be able to refresh/update the record in the location table. This error is also impacts pua_usrloc module too as it cannot access the record and sends creates an incorrect uri. Help: I'm new and learning how Kamailio works, is there a clean way that pua_reginfo module gets the record? Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

Problem with radius_www_authorize
by Ricardo Martinez 09 May '12

09 May '12

Hello. We have detected a behavior not specified in the docs. I ask for support to solve this issue. We are using a Radius system to authorize our clients, so, for REGISTER we use the command radius_www_authorize(“mydomain.com”) According to the documentation the possible answers are : Negative result codes may be interpreted as follows: -7 (internal error) - some internal error occurred (see syslog); -6 (nonce reused) - nonce is used more than once; -5 (no credentials) - credentials were not found in request; -4 (stale nonce) - stale nonce; -3 (authorization failed) - RADIUS responded with Access Reject which may be, for example, due to user not found or wrong password; -1 (error) - some error occured during authorization (see syslog); Despite of this, for BadPassword or UserNotExist from the radius answer the radius_www_authorize command return “-2” as a retcode. So, is the documentation missing something? Or I’m doing something wrong? Thanks Ricardo Martinez.-

2 2

Config include
by Konstantin M. 09 May '12

09 May '12

Hi, I would like (and a many people here I believe) to have a functional of including a multiple config files like (foe example asterisk's #include "path/to/some/config.conf"). Is it possible to implement a such feature ? Thanks!

4 7

Rtpproxy issue with connection information
by Openser Kamailio 09 May '12

09 May '12

Hi, i'm currently working with kamailio 3.2 and rtpproxy 1.2.1. Both are set up on the same computer. When rtpproxy adds an SDP to an Invite, it adds two IPv4 addresses in owner/creator session and connection information field with an error, i.e: *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 * 172.27.170.984* 172.27.170.98 *Connection Information (c)*: IN IP4 *172.27.170.984* 172.27.170.98 the correct ones should be : *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 172.27.170.98 *Connection Information (c)*: IN IP4 172.27.170.98 Thanks for help!!

3 3

Alias configuration with multidomain
by Arjan Kuiken 09 May '12

09 May '12

Hello SR-users, I want to configure a kamailio 3.2.3. We want to use it to connect multiple SIP servers as users and let the Kamailio work as a Trunk/ Proxy connected to a PSTN gateway. We have multiple domains and aliases per SIP servers. When I register a SIP server to Kamailio everything goes well, but when I try to call an dbalias it doesn't work. It gives 2 errors (trace included). I think something is done wrong in my kamailio.cfg but I don't have a clue anymore. I included the cfg file as attachement. Can you please have a look? Thanks in advance! Kind regards, Arjan Kuiken Technical Specialist [intercity-zakelijk-logo-handetekening (2)] Kruisweg 659, 2132 NC, Hoofddorp T: 020 - 655 3000 F: 020 - 6531534 E: Arjan.kuiken(a)intercityzakelijk.nl<mailto:arjan.kuiken@intercityzakelijk.nl> I: www.intercityzakelijk.nl<http://www.intercityzakelijk.nl/>

2 2

Accounting '180 Ringing' without SDP
by Stephen Dodge (Bistech) 09 May '12

09 May '12

Hello, We are running Kamailio 3.1.5 and using the acc module to generate CDRs and provide reporting information. We have a requirement to report ring time and we do this by accounting on early media for 183 provisional responses. Unfortunately we have a couple of customer PBX's that cannot provide a SDP with the ringing response. i.e they just respond with a '180 ringing'. My understanding of the module is that the "early_media" parameter is looking for a SDP in either a 180 or 183 response before accounting. Although it may be possible to account the 180 without SDP using the acc module I cannot find anything obvious in the documentation. If it is possible could someone point me in the right direction or suggest a creative solution, otherwise can I ask that this functionality be introduced. Many Thanks. Steve Dodge. ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

2 1

detect retransmission messages...
by Vineet Menon 09 May '12

09 May '12

Hi, How do we detect a message as retransmitted message, in a transaction oriented proxy? Regards, Vineet Menon

2 1

Re: [SR-Users] detect retransmission messages...
by Alex Balashov 09 May '12

09 May '12

t_check_trans() generally does the trick. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Vineet Menon <mvineetmenon(a)gmail.com> wrote: >Hi, > >How do we detect a message as retransmitted message, in a transaction >oriented proxy? > >Regards, > >Vineet Menon > >_______________________________________________ >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

kamdbctl
by Klaus Darilion 09 May '12

09 May '12

Hi! I wonder what is the status of kamdbctl tool for 'migrate' command - should it work? The description says: kamdbctl migrate <old_db> <new_db> .....................(migrates DB from 1.2 to 1.3, not implemented yet!) This seems a bit outdated. regards Klaus

3 3

kamdbctl error
by Vineet Menon 09 May '12

09 May '12

Hi, I am compiling sources of kamailio 3.2.3 on ubuntu 12.04. I have used make command > `make group_include="standard mysql" all` > to make the sources. and then > `make install`..... > Now when i do kamdbctl it throws an error, $ kamdbctl create > ERROR: could not load the script in > /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL > ERROR: database engine not loaded - tried 'MYSQL' > Any guess what is the problem??? Regards, Vineet Menon

3 7

Kamailio [No Audio]
by Fred Flintsone 09 May '12

09 May '12

I am attempting to route local registered users to local registered users without going to the media server. I have a media server [*] for PSTN. But it doesn't support all the CODECS i wan't to use. Signalling seems fine, but i get NO AUDIO on either call leg. I have the RTP captures from the local side making the call and the media packets seem to be going exactly where the SDP dictates. So i'm a little confused. Is this where an RTP Proxy would come in handy? I haven't been able to get the audio to work without going to the media server. If an RTP Proxy is the answer, how much overhead does the proxy add to the Kamailio server? Is it something that you don't run on the same machine and use a distributed environment of RTP proxies on other servers? Or is this something that should be fine and working without a media proxy? I really appreciate any help! Thank you, Fred

2 1

Bug: Pua_reginfo
by Gnaneshwar Gatla 08 May '12

08 May '12

Hello all, There seems to be an deadlock between registrar and pua_reginfo module whenever a SIP client refreshes its registration. 1. The pua_reginfo does not publish Event:reg when a sip client registers for the first time. 2. The pua_reginfo module does not use ul.lock_udomain() to lock the ul table to get the record. 3. The pua_reginfo does not release the record that it accessed from ul.get_urecord() Since the record is not released, when the client re-registers, the registrar module is unable to query for the userlocation and sends a 200 OK response with 0 bindings. Which leads to no contact header in the response message and eventually deletion of the record in the location table(expires). I have tried to fix this using lock_udomain, release_urecord and unlock_udomain, but failed to rectify the error. The deadlock happens in modules_k/usrloc/udomain.c (db_load_urecord()). if (ul_dbf.query(_c, keys, 0, vals, columns, (use_domain)?2:1, 13, order, &res) < 0) My Question: 1. is pua_reginfo supposed to delete record from location database and send the same data to a presence server? a. If above is true, then should the registrar module insert a new record in the location table? b. If above is false, the registrar module should be able to refresh/update the record in the location table. This error is also impacts pua_usrloc module too as it cannot access the record and sends creates an incorrect uri. Help: I'm new and learning how Kamailio works, is there a clean way that pua_reginfo module gets the record? Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

SDPOPS in siremis how to?
by Grégoire Vandendeurpel 08 May '12

08 May '12

Hello, Thx already for your first explanation. I'm going to describe more precisaly what is my purpose and what I need: About my final project: On one physical machine, I use 4 virtual machines, which are : one with kamailio 3.2.3, asterisk 1.6.13, odbc and Siremis 3.2 An other one is a first client in an other virtual subnet as kamailio.. which reprisents the "public network" ! The two others virtual machines are in the same subnet as Kamailio which represent the "private network" ( like a company and the public network) So, everything in the private network works well, with TLS, Siremis and asterisk, perfect communication! I still need to make the route between the two subnet but this shoud be easy and what I need precisly is to make a filter on SDP media managed by Siremis like: This is why I need to use the sdpops module of Kamailio, of course. I need to make the possibilities in Siremis to set things like : this subnet or this address -> remove this media like application, video , only audio can be allow ! This is the filter I need to make it work. Of course in Siremis because , I m simulating calls from public network and into the private network of a company . And it as to be user friendly, not spacially for a computer scientist. This is why, I need to make this simple filter on the media of sdp, and user friendly in Siremis. Thank u in advance for your help Best regards -- *Grégoire Vandendeurpel, * * * *IT Sector*

1 0

routing logic
by Vineet Menon 08 May '12

08 May '12

Hi, Where can i find a nice tutorial for routing logic in cfg file for kamailio? i want help on the basic functions like record_route() , sl_send_reply(), t_relay() etc... Regards, Vineet Menon

2 3

Re: [SR-Users] Kamailio LDAP integration
by Reda Aouad 07 May '12

07 May '12

Sorry didn't reply to mailing list before. Emails are below. SHA1 encryption may not encrypt the same way as HA1 (HA1 = MD5 of realm + username + password), so the problem may be here. I suggest you store your passwords as clear text in LDAP for testing first. Reda On Fri, May 4, 2012 at 11:14 PM, Saul Waizer <saulwaizer(a)gmail.com> wrote: > with the variations I get different results: > 4(24126) ERROR: <script>: Password={SHA}v/m3IZiuy+VVizqnt56e2baZsT8= > 4(24126) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=760 a=17 n=if > 4(24126) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=756 a=28 > n=pv_www_authenticate > 4(24126) DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1 > 4(24126) DEBUG: auth [api.c:210]: check_response: Our result = > '3839aa4cae572f5f8b23601a2bb1178f' > 4(24126) DEBUG: auth [api.c:220]: check_response: Authorization failed > > On Fri, May 4, 2012 at 3:11 PM, Saul Waizer <saulwaizer(a)gmail.com> wrote: > >> Also: i used xlog to print out the password and I get the same exact >> password I have on my LDAP server, so it seems something with the decoding >> >> >> On Fri, May 4, 2012 at 3:01 PM, Saul Waizer <saulwaizer(a)gmail.com> wrote: >> >>> Now i got it down to this: >>> >>> 2(23003) INFO: <script>: ldap_search: found [1] entries for >>> (uid=mmiller) 2(23003) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] >>> l=759 a=17 n=if >>> 2(23003) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=755 a=28 >>> n=pv_www_authenticate >>> 2(23003) DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1 >>> 2(23003) DEBUG: auth [auth_mod.c:455]: HA1 string calculated: >>> c69622bbd922ec9321ab1293c226b703 >>> 2(23003) DEBUG: auth [api.c:210]: check_response: Our result = >>> '939676a5591165f1da8ba04562d446b2' >>> 2(23003) DEBUG: auth [api.c:220]: check_response: Authorization failed >>> 2(23003) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=756 a=27 >>> n=www_challenge >>> 2(23003) DEBUG: auth [challenge.c:102]: build_challenge_hf: >>> realm='23.22.35.43' >>> 2(23003) DEBUG: auth [challenge.c:244]: auth: 'WWW-Authenticate: Digest >>> realm="23.22.35.43", nonce="T6Qn/E+kJtAU7IvGh4OLivg7ptLbdida" >>> >>> I have changed the values of: >>> >>> if (!pv_www_authenticate("$td", "$avp(password)", "0")) { >>> www_challenge("$td", "0"); >>> >>> to: >>> >>> if (!pv_www_authenticate("$td", "$avp(password)", "1")) { >>> www_challenge("$td", "0"); >>> >>> because of the password in LDAP is stored as SHA1, and according to >>> the docs, it should be 1. I'm so close it seems :) >>> >>> *flags* - the value of this parameter can be a bitmask of following: >>> >>> - >>> >>> *1* - the value of password parameter is HA1 format >>> >>> >>> >>> On Fri, May 4, 2012 at 2:47 PM, Reda Aouad <reda.aouad(a)gmail.com> wrote: >>> >>>> can you also print the avp(s:password) to log to see what its value is? >>>> use: >>>> xlog('Password=$avp(s:password)') >>>> after ldap_search and you'll see its output in the log file >>>> maybe you're not correctly getting the password from the ldap search >>>> url, avp(s:password) is then null and you get the error that it can't be >>>> converted to string >>>> >>>> Reda >>>> >>>> >>>> >>>> On Fri, May 4, 2012 at 8:40 PM, Reda Aouad <reda.aouad(a)gmail.com>wrote: >>>> >>>>> in the line >>>>> if (!pv_www_authenticate("$td", "$avp(password)", "0")) { >>>>> >>>>> write avp(s:password) instead of avp(password) >>>>> not sure it will solve it though.. if it doesn't, maybe others can >>>>> help you more on this. >>>>> >>>>> Reda >>>>> >>>>> >>>>> >>>>> On Fri, May 4, 2012 at 5:50 PM, Saul Waizer <saulwaizer(a)gmail.com>wrote: >>>>> >>>>>> Hello Reda, >>>>>> >>>>>> Thank you for your feedback, after some further research and testing >>>>>> I got the LDAP search working, I am just having one issue with the password >>>>>> variable: >>>>>> >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=755 >>>>>> a=28 n=pv_www_authenticate >>>>>> 3(22487) ERROR: <core> [sr_module.c:1613]: Could not convert PV to >>>>>> str >>>>>> 3(22487) ERROR: auth [auth_mod.c:569]: failed to get passwd value >>>>>> >>>>>> My relevant configuration: >>>>>> >>>>>> route[AUTH] { >>>>>> #!ifdef WITH_AUTH >>>>>> if (is_method("REGISTER")) >>>>>> { >>>>>> if(is_present_hf("Authorization")) >>>>>> { >>>>>> # ldap search >>>>>> >>>>>> if >>>>>> (!ldap_search("ldap://demo/ou=demo,dc=mydomain,dc=com?uid,userPassword?")) >>>>>> >>>>>> { >>>>>> switch ($retcode) >>>>>> { >>>>>> case -1: >>>>>> # no LDAP entry found >>>>>> sl_send_reply("404", "User Not Found"); >>>>>> exit; >>>>>> case -2: >>>>>> # internal error >>>>>> sl_send_reply("500", "Internal server error"); >>>>>> exit; >>>>>> default: >>>>>> exit; >>>>>> } >>>>>> } >>>>>> ldap_result("uid/$avp(s:username)"); >>>>>> ldap_result("userPassword/$avp(s:password)"); >>>>>> xlog("L_INFO", "ldap_search: found [$retcode] entries for >>>>>> (uid=$fU)"); >>>>>> if (!pv_www_authenticate("$td", "$avp(password)", "0")) { >>>>>> www_challenge("$td", "1"); >>>>>> exit; >>>>>> } >>>>>> sl_send_reply("200", "ok"); >>>>>> exit; >>>>>> } else { >>>>>> www_challenge("$td", "1"); >>>>>> exit; >>>>>> } >>>>>> } else { >>>>>> >>>>>> And the error message: >>>>>> >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=735 >>>>>> a=26 n=ldap_search >>>>>> 3(22487) DEBUG: ldap [ldap_api_fn.c:273]: LDAP URL parsed into >>>>>> session_name [demo], base [ou=demo,dc=mydomain,dc=com], scope [0], filter [] >>>>>> 3(22487) DEBUG: ldap [ldap_api_fn.c:433]: [demo]: performing LDAP >>>>>> search: dn [ou=demo,dc=mydomain,dc=com], scope [0], filter [(null)], >>>>>> client_timeout [5000000] usecs >>>>>> 3(22487) DEBUG: ldap [ldap_api_fn.c:240]: [demo]: [1] LDAP entries >>>>>> found >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=752 >>>>>> a=26 n=ldap_result >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=753 >>>>>> a=26 n=ldap_result >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=754 >>>>>> a=27 n=xlog >>>>>> 3(22487) INFO: <script>: ldap_search: found [-1] entries for >>>>>> (uid=mmiller) 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] >>>>>> l=759 a=17 n=if >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=755 >>>>>> a=28 n=pv_www_authenticate >>>>>> 3(22487) ERROR: <core> [sr_module.c:1613]: Could not convert PV to >>>>>> str >>>>>> 3(22487) ERROR: auth [auth_mod.c:569]: failed to get passwd value >>>>>> 3(22487) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=756 >>>>>> a=27 n=www_challenge >>>>>> 3(22487) DEBUG: auth [challenge.c:102]: build_challenge_hf: >>>>>> realm='ip.of.sip.server' >>>>>> 3(22487) DEBUG: auth [challenge.c:113]: build_challenge_hf: >>>>>> qop='auth' >>>>>> 3(22487) DEBUG: auth [challenge.c:244]: auth: 'WWW-Authenticate: >>>>>> Digest realm="ip.of.sip.server", nonce="T6P5yU+j+J23OE93mPaektZpJszGpt/l", >>>>>> qop="auth" >>>>>> >>>>>> Any help is greatly appreciated! >>>>>> Thanks >>>>>> >>>>>> >>>>>> >>>>>> On Thu, May 3, 2012 at 4:22 PM, Reda Aouad <reda.aouad(a)gmail.com>wrote: >>>>>> >>>>>>> Hi Saul, >>>>>>> >>>>>>> username_avp_spec was previously a AUTH module parameter to specify >>>>>>> a variable that was passed to pv_www_authorize implicitly (the function >>>>>>> doesn't take arguments). Now you should use the new >>>>>>> pv_www_authenticate and pass to it explicitly the credentials as arguments. >>>>>>> >>>>>>> So forget about username_avp_spec since it doesn't exist as module >>>>>>> param anymore (this is why you are getting the error). Store the result of >>>>>>> ldap_search in the avps as in the tutorial using ldap_result, and pass them >>>>>>> to pv_www_authenticate as parameters. pv_www_authenticate takes the >>>>>>> following arguments: >>>>>>> - realm: which you can get from "to domain" using $td >>>>>>> - password: $avp(s:password) >>>>>>> - flag: set it to 0 as a first test >>>>>>> >>>>>>> example: >>>>>>> pv_www_authorize("$td", "$avp(s:password)", 0) >>>>>>> >>>>>>> This function takes the username from the authentication header, so >>>>>>> no need to pass it anymore as argument. >>>>>>> >>>>>>> Reda >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, May 3, 2012 at 8:47 PM, Saul Waizer <saulwaizer(a)gmail.com>wrote: >>>>>>> >>>>>>>> Hello List, >>>>>>>> >>>>>>>> I am trying to incorporate an existing LDAP directory with our >>>>>>>> Kamailio installation for SIP authentication. A good friend suggested to >>>>>>>> checkout this tutorial and adapt it to fit my needs (and current version) >>>>>>>> >>>>>>>> >>>>>>>> http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-auth-ldap >>>>>>>> >>>>>>>> It seems like the AUTH module does not contain the function >>>>>>>> username_spec (which I believe is not used anymore) but the >>>>>>>> username_avp_spec which is not part of the AUTH module but the H350 module >>>>>>>> http://kamailio.org/docs/modules/3.2.x/modules_k/h350.html >>>>>>>> >>>>>>>> I enabled the h350 module and tried setting the params as described >>>>>>>> in the documentation: >>>>>>>> >>>>>>>> modparam("auth", "username_spec", "$avp(s:username)") >>>>>>>> modparam("auth", "password_spec", "$avp(s:password)") >>>>>>>> modparam("auth", "calculate_ha1", 1) >>>>>>>> >>>>>>>> I got the following error after checking the configuration: >>>>>>>> >>>>>>>> ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter >>>>>>>> <username_spec> not found in module <auth> >>>>>>>> >>>>>>>> I am running kamailio 3.2.3 (i386/linux) Ubuntu >>>>>>>> >>>>>>>> Thank you in advance! >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> 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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>> >>>> >>> >> >

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Re: [SR-Users] Remove_hf doesn't work
by Daniel-Constantin Mierla 07 May '12

07 May '12

Hello, you have fix_nated_contact() which does an update to contact address: May 7 15:46:58 localhost /usr/local/sbin/kamailio[15742]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio//kamailio.cfg] l=889 a=25 n=fix_nated_contact If you want to remove the contact, then do not call that function, or use the msg_apply_changes() as mentioned previously. Cheers, Daniel On 5/7/12 4:23 PM, Openser Kamailio wrote: > Hi Daniel, > > Please find my log messages in attachments. > Open my log file with Notepad, it's better. > > Thanks! > > On Mon, May 7, 2012 at 2:41 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > do you use any other function that updates the address in contact > header? > > If you load debugger module, enable cfgtrace option and send the > execution log messages over here, we can tell if there is another > function that may alter contact header. > > One option is to use msg_apply_changes() before and after > remove_hf(), but it is not the optimal way. > > Cheers, > Daniel > > > On 5/7/12 2:13 PM, Openser Kamailio wrote: >> Hi, >> >> I'm currently working with kamailio 3.2. I try to remove the >> Contact Header field with remove_hf("Contact") through the file >> named kamailio.cfg. >> When i apply remove_hf("Contact"), it only removes the word >> Contact, but the contact address is still present. Let me show you: >> >> Without remove_hf: >> Contact: sip:<20505@sipserver.com:5060 >> <http://20505@sipserver.com:5060>>;......... >> >> After remove_hf: >> sip:<20502@sipserver.com:5060 >> <http://20502@sipserver.com:5060>>;............ >> >> I want to remove the whole contact address. >> Thanks for help! >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Problem with adding extra field in acc table.
by Gary Chen 07 May '12

07 May '12

I using Mysql DB. I need to add extra fields in acc table. The field having problem is the last one 'account' Here is my code in kamailio.cfg: ... modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;user_agent=$hdr(User-Agent);account=$var(s:account_number)") ... route { .... dp_translate("3", "$fU/$avp(s:dst)") $v(s:account_number) = $avp(s:dst); .... } By looking at the xlog message, $v(s:account_number) has correct value. But the value in acc table is not always correct. Sometime account field populated with 0 and sometime it populated with wrong account number. It seems that $v(s:account_number) sometime corrupted. Am I doing something wrong? Gary

2 1

Remove_hf doesn't work
by Openser Kamailio 07 May '12

07 May '12

Hi, I'm currently working with kamailio 3.2. I try to remove the Contact Header field with remove_hf("Contact") through the file named kamailio.cfg. When i apply remove_hf("Contact"), it only removes the word Contact, but the contact address is still present. Let me show you: Without remove_hf: Contact: sip:<20505@sipserver.com:5060>;......... After remove_hf: sip:<20502@sipserver.com:5060>;............ I want to remove the whole contact address. Thanks for help!

2 1

Re: [SR-Users] json and jsonrpc-c
by Ron McLeod 06 May '12

06 May '12

I set the PKG_CONFIG_PATH environment variable to /usr/local/lib/pkgconfig . After that it built fine. From: Ron McLeod [mailto:ron.kamailio@mcleodnet.com] Sent: Saturday, May 05, 2012 9:24 PM To: 'sr-users(a)lists.sip-router.org' Subject: json and jsonrpc-c I am trying to build json and jsonrpc-c for kamailio-3.2.3 from a fresh download of the source tarball but not having success. I modified the Makefile to include the two modules in the standard package, and remove the line which excluded them from building. I see the code for the modules in the source. Also, I have already built and installed the two dependencies libevent and libjson. Could someone take a look at the make out and see if there is any indication of what I might be doing wrong? http://www.siplabs.net/tracebin/make_and_install_output.txt Thanks Ron

2 2

siptrace port number error
by Reda Aouad 06 May '12

06 May '12

Hi, I'm trying to setup Kamailio as a Homer capture server itself. Kamailio listens on 2 ports : 53 for SIP users, and 9060 for homer SIP capture. siptrace module duplicates packets to sipcapture module on port 9060. I'm getting the following error : ERROR: siptrace [siptrace.c:1669]: invalid port number; must be in [1024,65536] This is my conf ----------------------- listen=udp:10.238.235.173:53 listen=udp:127.0.0.1:9060 #!ifdef WITH_SIPTRACE modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", FLD_TRACE) modparam("siptrace", "duplicate_uri","sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on",1); modparam("siptrace", "trace_to_database", 0); #!endif Any help? Thanks Reda

3 4

json and jsonrpc-c
by Ron McLeod 06 May '12

06 May '12

I am trying to build json and jsonrpc-c for kamailio-3.2.3 from a fresh download of the source tarball but not having success. I modified the Makefile to include the two modules in the standard package, and remove the line which excluded them from building. I see the code for the modules in the source. Also, I have already built and installed the two dependencies - libevent and libjson. Could someone take a look at the make out and see if there is any indication of what I might be doing wrong? http://www.siplabs.net/tracebin/make_and_install_output.txt Thanks - Ron

1 0

json and jsonrpc-c
by Ron McLeod 06 May '12

06 May '12

I am trying to build json and jsonrpc-c for kamailio-3.2.3 from a fresh download of the source tarball but not having success. I modified the Makefile to include the two modules in the standard package, and remove the line which excluded them from building. I see the code for the modules in the source. Also, I have already built and installed the two dependencies - libevent and libjson. Could someone take a look at the make out and see if there is any indication of what I might be doing wrong? http://www.siplabs.net/tracebin/make_and_install_output.txt Thanks - Ron

1 0

domain kaobona.com
by Thulani Ngwenya 05 May '12

05 May '12

Hi, I registered my SIP domain kaobona.com to be hosted on iptel.org, how do I get users to register accounts using my domain? Kind regards, Thulani

1 0

pua_reginfo: no Username in PUBLISH message
by Gnaneshwar Gatla 04 May '12

04 May '12

Hello, I was looking at the pua_reginfo to make multiple instances of Kamailio be aware of registered users on the network. I'm experimenting with pua_reginfo with Kamailio 3.2.2. I've used the configuration as described in http://kamailio.org/docs/modules/3.2.x/modules_k/pua_reginfo.html#id2551427 I tried an trial run of the Kamailio instance and it gives the following output PUBLISH sip:@64.64.203.108 SIP/2.0 Via: SIP/2.0/UDP 64.64.203.108;branch=z9hG4bKb62c.b8e3d5b5.0 To: sip:@64.64.203.108 From: sip:@64.64.203.108;tag=533cb9e91f4b999cf76861cbb9ed54ed-e085 CSeq: 10 PUBLISH Call-ID: 17dedfff-14418(a)127.0.0.1 Content-Length: 320 Max-Forwards: 70 Event: reg Expires: 3601 Content-Type: application/reginfo+xml <reginfo xmlns="urn:ietf:params:xml:ns:reginfo" version="0" state="full"> <registration aor="sip:@64.64.203.108" id="0xb6ef7740" state="active"> <contact id="0xb4d1f820" state="active" event="registered" expires="60" callid="ZmRjZDg3MmIxMjI5NmU1OWU0YjJlYWUzYzhjZDViM2I." cseq="2"> <uri>sip:ggatla01@192.168.128.108:23840;rinstance=27b18eed9568795e;transport=udp</uri> </contact> </registration> </reginfo> The PUA module does not insert username in the Request, To and From headers while publishing for Event: reg. The xml payload is right but the "aor" does not have the username. Is there anything missing that I need to configure? Lastly, the presence module in the same Kamailio instance is generating "478 Unresolvable destination" for this message. Regards Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

2 2

Kamailio LDAP integration
by Saul Waizer 04 May '12

04 May '12

Hello List, I am trying to incorporate an existing LDAP directory with our Kamailio installation for SIP authentication. A good friend suggested to checkout this tutorial and adapt it to fit my needs (and current version) http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-auth-ldap It seems like the AUTH module does not contain the function username_spec (which I believe is not used anymore) but the username_avp_spec which is not part of the AUTH module but the H350 module http://kamailio.org/docs/modules/3.2.x/modules_k/h350.html I enabled the h350 module and tried setting the params as described in the documentation: modparam("auth", "username_spec", "$avp(s:username)") modparam("auth", "password_spec", "$avp(s:password)") modparam("auth", "calculate_ha1", 1) I got the following error after checking the configuration: ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter <username_spec> not found in module <auth> I am running kamailio 3.2.3 (i386/linux) Ubuntu Thank you in advance!

2 3

sdp parsing error
by RUSHI NAYAK 04 May '12

04 May '12

for the following sdp content i'm getting error. v=0 o=rushi 602301 603670 IN IP4 192.168.1.125 s=- c=IN IPV4 192.168.1.125 t=0 0 m=audio 8192 RTP/AVP 5 0 3 can you give correction or sample sdp string.

2 1

Sending NOTIFY using t_uac_dlg over xmlrpc
by Krishna Kurapati 04 May '12

04 May '12

Hi, I was able to send MESSAGE using xmlrpc (mi and t_uac_dlg) I got the 200 OK response to the message. However, when I try to send NOTIFY, I receive HTTP response but with empty methodResponse. Is NOTIFY not suppored by t_uac_dlg? Thanks Krish Kura POST /RPC2 HTTP/1.0 User-Agent: ruby-sip-client Host: 10.1.0.6 Content-Type: text/xml Connection: keep-alive Content-Length: 650 <?xml version="1.0" ?><methodCall><methodName>mi</methodName><params><param><value><string>t_uac_dlg</string></value></param><param><value><string>NOTIFY</string></value></param><param><value><string> sip:kkurapati_staff_test@10.1.0.6</string></value></param><param><value><string>.</string></value></param><param><value><string>.</string></value></param><param><value><string>From: <sip:kkurapati_provider_test@10.1.0.6> To: <sip:kkurapati_staff_test@10.1.0.6> Max-Forwards: 70 Call-ID: e3f15b5dc200234cff2c7b648d79c49f(a)10.1.0.3 </string></value></param><param><value><string>User added</string></value></param></params></methodCall> ## T 2012/04/27 14:39:11.579281 10.1.0.6:5060 -> 10.1.0.3:44817 [AP] HTTP/1.0 200 OK. Via: SIP/2.0/TCP 10.1.0.3:44817. Server: kamailio (3.3.0-dev7 (x86_64/linux)). Content-Length: 137. . <?xml version="1.0"?> <methodResponse> <params> <param> <value><array><data> </data></array></value> </param> </params> </methodResponse>

2 3

sdpops kamailio ans Siremis
by Grégoire Vandendeurpel 04 May '12

04 May '12

Hi, I'm using Siremis 3.2 ans Kamailio 3.2.3 for my final project of my bachloor degree I would like to know how Siremis and Kamailio work together? Like: If I decide to set a protocol in the acl trusted in Siremis like TLS, ok, it works well !! But I need to know the different links between siremis and kamailio to working well.. Why? Cause I need to implement in Siremis a filter form on the sdpops module, and it has to work well of course with the database of ser! So.. What are the different steps to do this? - Create new table sdpops in ser - create the form in siremis and view .. - what in kamailio.cfg?? I found this: http://kb.asipto.com/siremis:install32x:new-views OK, but in my case, I need to do that for SDPOPS, which is a module of kamailio, whithout any table and components about! I mean, there is no openser.sdpops in the database of kamailio, so create the table ok.. but what are the columns, datas, to set in ?? cause I tried php gen_meta.php Serdb sdpops ser.acl.sdpops of course it does not work --> no openser.sdpops ...... thx in advance for your help Best regards -- *Grégoire Vandendeurpel, * * * *IT Sector*

2 2

SEMS 1.4.3 released
by Stefan Sayer 04 May '12

04 May '12

Hello, the SIP Express Media Server project happily announces the availability of the SEMS 1.4.3 release. This is a SEMS 1.4 series bugfix release and should be a drop-in replacement to 1.4 installations. For details of the changes see the changelog at http://ftp.iptel.org/pub/sems/1.4/1.4.3/CHANGELOG Please find sources at http://ftp.iptel.org/pub/sems/sems-1.4.3.tar.gz binary packages in the usual locations soon, and all documentation and links at http://iptel.org/sems If you are using custom applications which create outbound calls, please specifically test the behavior with the commit 26fdaa8e b/f: stopping failed outbound calls Many thanks to everyone who contributed with issue reports and patches! Best Regards Stefan Sayer SEMS - E stands for Express!

1 0

Need Help please
by Rudi Kusnaka 04 May '12

04 May '12

Hello there, I've installing the Siremis 3.2 on a linux that installed Kamailio 3.2. When i'll try to access everything in SER Menu, i'm getting a error messages that tell me about the table in OPENSER database is missing or can't located. But this is a fresh installed Siremis system. Where i can found the openser database stucture so i can build them again? Best Regards, Rudi Kusnaka

2 1

Siremis on Mysql and Kamailio on Postgresql
by Jonny Klompas 04 May '12

04 May '12

I've setup a siremis installation using Mysql for the siremis tables. Kamailio is installed with postgresql on another machine. I selected Postgresql in seremis setup under SIP DB and correctly inserted the right username/password/host. Connection was successful. I can login to Siremis (ver 3.2.0) and access the Admin menu successfully. However when I access the Ser Menu, all the options give errors: For example, trying to to select Server Services, Domain List gives the following errors: [2012-04-05 18:11:36 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /usr/local/siremis-3.2.0/openbiz/bin/Resource.php:283 *function:* errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/usr/local/siremis-3.2.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /usr/local/siremis-3.2.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/usr/local/siremis-3.2.0/openbiz/bin/Resource.php", 283, Array(11) ) @ *function:* trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /usr/local/siremis-3.2.0/openbiz/bin/Resource.php 283 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /usr/local/siremis-3.2.0/openbiz/bin/BizSystem.php 544 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /usr/local/siremis-3.2.0/siremis/modules/service/userEmailService.php 108 *function:* SystemInternalErrorEmail ( Array(2), Null ) @ /usr/local/siremis-3.2.0/siremis/modules/common/form/ErrorForm.php 34 *function:* Report ( ) @ /usr/local/siremis-3.2.0/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /usr/local/siremis-3.2.0/openbiz/bin/BizController.php 110 *function:* dispatchRequest ( ) @ /usr/local/siremis-3.2.0/openbiz/bin/BizController.php 32 *function:* include_once ( "/usr/local/siremis-3.2.0/openbiz/bin/BizController.php" ) @ /usr/local/siremis-3.2.0/siremis/bin/controller.php 6 ------------------------------ * * Similarly. the all the other optiones under the Ser Menu give errors. My Config.xml (passwords and IPs modified) follows: <?xml version="1.0" standalone="no"?> <Config> <DataSource> <Database Name="Default" Driver="Pdo_Mysql" Server="localhost" Port="3306" DB Name="siremis" User="siremis" Password="siremisrw" Charset="utf8"/> <Database Name="Serdb" Driver="Pdo_Pgsql" Server="x.x.x.x" Port="5432" DBName="dbname" User="dbuser" Password="123456" Charset="utf8"/> </DataSource> </Config> Please advise. Many thanks Jonny * *

4 11

Siremis upload file
by Jesús Pérez Rubio 03 May '12

03 May '12

Hi, I have a problem making some changes on siremis front-end everything was ok except when I try to add a field to upload a file I got this error: #2, fopen(ejemplo.csv): failed to open stream: No such file or directory could anybody help me please? :) Mi configuration: DO: <BizField Name="batch_file" Column="batch_file" Length="64" Required="N" Type="Blob"/> Insert: <Element Name="fld_batch_file" Class="File" FieldName="batch_file" Label="File (.csv)" AllowURLParam="N"/> .php: public function InsertRecord($recArr) { return parent::InsertRecord($recArr); } SQL: mysql> desc batch_provision; +------------+---------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +------------+---------+------+-----+---------+----------------+ | id | int(11) | NO | PRI | NULL | auto_increment | | batch_file | blob | YES | | NULL | | +------------+---------+------+-----+---------+----------------+ 2 rows in set (0.02 sec) Thanks in advance. -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com

2 1

Routing based on gruu
by Krishna Kurapati 03 May '12

03 May '12

Hi, I am trying to use GRUU for routing MESSAGEs. For example if I want to route to specific instance of the registered user, what do I need to send in the MESSAGE? If I know pub-gruu of the instance, where do I insert it so that kamailio can correctly process it and send it to the right instance? Thanks Krish Kura

2 1

kamctl trouble
by Vineet Menon 03 May '12

03 May '12

Hi, I have a kamailio installation running, but in kamctl the moni command works, but stop, restart, start command doesn't work... why is it so?? Regards, Vineet Menon

3 3

AVP assignment question
by Klaus Darilion 03 May '12

03 May '12

Hi! For example: $avp(i:test) = 1; $avp(i:test) = 2; # now: test[0]=2, test[1]=1 I would think that assigning or deleting using an index would overwrite the respective element in the AVP list, but assigning $null always deletes [0] and assigning a value always add another AVP to the top, e.g: $(avp(i:test)[1]) = 3; # I would expect: test[0]=2, test[1]=3 # but it is: test[0]=3, test[1]=2, test[2]=1 $(avp(i:test)[2]) = $null; # I would expect: test[0]=3, test[1]=2 # but it is: test[0]=2, test[1]=1 So, is the current behavior really the correct behavior? It is not what I would expect when using indexes. If it is correct, I will improve the documentation accordingly. regards Klaus

2 2

dialog - keep alives with lower cseq
by Daniel-Constantin Mierla 03 May '12

03 May '12

Hello, several days ago, just before freezing the development for v3.3.0, I added to dialog module the option to send keep alives for ongoing calls in order to detect if caller/callee is gone. SIP specs require to increment the CSeq for requests within dialog, but because the keep alives are sent from the sip server, it would results de-synchronization of the CSeq values hold in phones themselves (e.g., a BYE created by caller/callee after a keep alive will be with lower cseq than the other side would expect and accept). One solution would be to update cseq when BYE passes the server to the right value. This implies more processing, as a call can have many re-INVITEs or other requests within dialog sent by caller/callee, including periodic updates to database to store cseq numbers. So I went for a different approach, like stated in subject -- let's see your opinion if you think is going to work. Practically the keep alives will be OPTIONS with CSeq equal or lower than the last valid value (e.g., the cseq of the INVITE creating the call). If the caller/callee is gone, that is simply, the OPTION will be timed out and dialog module will send BYE. If the caller/callee are reachable, but for some reason the call was destroyed (e.g., a crash and restart meanwhile), since there is a To-tag, the OPTIONS should get a 481 Call/Leg transaction does not exists. Again, a case when kamailio will end the dialog. The one to be discussed here would be caller/callee are still on the call. Based on RFC (and the feedback from people on another thread here), a requests coming within a dialog with lower CSEq should be replies with 500. CSeq numbers are still ok in both sides (note that requests with lower CSeq can happen in reality, like two fast re-INVITEs sent over UDP, the second arriving first due to network transmission). I tested with snom phones and jitsi so far, seems to be a working solution (well, jitsi was replying 500 after 20secs to keep alive OPTIONS request, not sure for what reason, just reported back to the project). Is anyone here seeing any possible issues with the approach? Call are ended by dialog, with proper CSeq in the BYE, after 10seconds from the moment 408 or 481 is received, no other replies are taken in consideration for ending the dialog from server side. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 5

Can't load perl module
by Ladislav Jurák 02 May '12

02 May '12

Hello, I have problem with loading perl module on Kamailio v3.2 installed from repository on Debian sqeeze amd64. Perl modules was also installed from repository package kamailio-perl-modules. I have also installed all perl dependences listed in perl module documentary. Into Kamailio config file I added: loadmodule "perl.so" and modparam("perl", "filename", "/etc/kamailio/skript.pl") Now when I'm trying to start Kamailio I'll get following: May 1 23:13:46 server kamailio: DEBUG: <core> [cfg.y:1712]: loading module perl.so May 1 23:13:46 server kamailio: DEBUG: <core> [sr_module.c:557]: load_module: trying to load </usr/lib64/kamailio/modules_k/perl.so> May 1 23:13:46 server kamailio: WARNING: <core> [sr_module.c:620]: /usr/lib64/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead May 1 23:13:46 server kamailio: DEBUG: <core> [modparam.c:96]: set_mod_param_regex: 'perl' matches module 'perl' May 1 23:13:46 server kamailio: DEBUG: <core> [sr_module.c:763]: find_param_export: found <filename> in module perl [/usr/lib64/kamailio/modules_k/perl.so] May 1 23:13:46 server kamailio: DEBUG: <core> [modparam.c:113]: set_mod_param_regex: found <filename> in module perl [/usr/lib64/kamailio/modules_k/perl.so] May 1 23:13:46 server /usr/sbin/kamailio[31192]: DEBUG: <core> [sr_module.c:928]: DEBUG: init_mod: perl May 1 23:13:46 server /usr/sbin/kamailio[31192]: ERROR: perl [perl.c:238]: failed to load perl file "/etc/kamailio/skript.pl". May 1 23:13:46 server /usr/sbin/kamailio[31192]: ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module perl (/usr/lib64/kamailio/modules_k/perl.so) I am confused about the WARNIG too, can it have some influence to the errors? File skript.pl is in the specified directory. I also changed permissions to 777. It's content is from sample "branches.pl". I have similar problem with module app_python.so. Can you please give me some advice? It's important for me to get this module working. Thank you. -- Regards Ladislav Jurak

2 6

sdpops Siremis
by Grégoire Vandendeurpel 02 May '12

02 May '12

Hi, I'm using Siremis 3.2 ans Kamailio 3.2.3 for my final project of my bachloor degree I would like to know how Siremis and Kamailio work together? Like: If I decide to set a protocol in the acl trusted in Siremis like TLS, ok, it works well !! But I need to know the different links between siremis and kamailio to working well.. Why? Cause I need to implement in Siremis a choise form about the sdpops module, and it has to work well of course with the database of ser! So.. What are the different steps to do this? - Create new table sdpops in ser - create the form in siremis and view .. - what in kamailio.cfg?? I found this: http://kb.asipto.com/siremis:install32x:new-views OK, but in my case, I need to do that for SDPOPS, which is a module of kamailio, whithout any table and components about! I mean, there is no openser.sdpops in the database of kamailio, so create the table ok.. but what are the columns, datas, to set in ?? cause I tried php gen_meta.php Serdb sdpops ser.acl.sdpops of course it does not work --> no openser.sdpops ...... thx in advance for your help Best regards -- *Grégoire Vandendeurpel, * * * *IT Sector*

1 0

Problem with Route header
by Geoffrey Mina 02 May '12

02 May '12

Greetings, I am confused at some functionality I am seeing with Kamailio 1.5.4. I know this is an old version, but I don't have the time to go through a lengthy upgrade process right now. The issue I am seeing is that the server is inserting a Route header with it's own IP address for an unknown reason. Here is the initial invite (removed SDP for simplicity): INVITE sip:13@67.207.130.146:5060 SIP/2.0 Via: SIP/2.0/UDP 68.64.220.108:5060;branch=z9hG4bK78dd33c6;rport From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;tag=as1cad6370 To: <sip:13@67.207.130.146:5060> Contact: <sip:9546496707@68.64.220.108> Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 70 Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;privacy=off;screen=no Date: Tue, 01 May 2012 18:17:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Route: <sip:13@boulder-voip.mydomain.com> P-Account-ID: 99990023 P-Proxy-Route: Yes Content-Type: application/sdp Content-Length: 240 The basics of what happen next are: t_check_trans(); record_route(); remove_hf("P-Proxy-Route"); if(loose_route()){ route(3); } route[3]{ t_on_reply("1"); if(!t_relay()){ sl_reply_error(); } } The INVITE that goes out has the funky Route: header with the Kamailio IP in there. This is causing problems for some of the upstream proxy servers (obviously). INVITE sip:13@boulder-voip.mydomain.com SIP/2.0 Record-Route: <sip:67.207.130.146;lr;ftag=as1cad6370> Via: SIP/2.0/UDP 67.207.130.146;branch=z9hG4bKf183.456d51e1.0 Via: SIP/2.0/UDP 68.64.220.108:5060 ;received=68.64.220.108;branch=z9hG4bK78dd33c6;rport=5060 From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;tag=as1cad6370 To: <sip:13@67.207.130.146:5060> Contact: <sip:9546496707@68.64.220.108> Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 69 Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;privacy=off;screen=no Date: Tue, 01 May 2012 18:17:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces P-Account-ID: 99990023 Content-Type: application/sdp Content-Length: 240 Route: <sip:13@67.207.130.146:5060> Any idea what may be causing this to happen and how I could prevent it? I have tried removing the Route header using the remove_hf("Route") before doing the t_relay, but that doesn't seem to help. Thanks, Geoff

2 2

kamailio and openser
by Vineet Menon 01 May '12

01 May '12

Hi, I have a ubuntu 12.04 installation with me, and want to install kamailio in it... I have added repo from kamailio.ord site, but they are pretty old and require libmysqlclient16, which is no longer available. Ubunut 12.04 has a canonical repo for openser.... are they both interchangeble?? can i install openser and feel the same as kamailio?? Regards, Vineet Menon

3 3

LCR module from_uri module how to ?
by Sébastien Cramatte 01 May '12

01 May '12

Hello, Does anyone can give me a proper example how setup LCR to be able to use prefix and from_uri field. Does not a route should be preferred if prefix or from_uri is more accurate ? I mean that if you st from_uri value instead of default NULL route should be considered as better ? Thank you for you help

2 1

OpenXcap always returns full resource list
by Fabian Bernhard 01 May '12

01 May '12

Dear list, we are running Kamailio 3.2 together with the included OpenXcap server. I try to retrieve only part of a certain resource list using XPath, but the server always returns the full list, see example below. I was so far not able to retrieve only part of a resource list, independent of the XPath expression I'm using. Do I make something wrong or is this a bug? Thanks and regards, Fabian fubeh@orion:~$ curl --digest -u test4:XXXX http://192.168.1.134:5060/xcap-root/resource-lists/users/sip:test4@192.168.… <?xml version="1.0"?> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="Business"> <entry uri="sip:test1@192.168.1.134"> <display-name>Test 1</display-name> </entry> <entry uri="sip:test122@192.168.1.134"> <display-name>Test 1</display-name> </entry> </list> <list name="Family"> <entry uri="sip:test2@192.168.1.134"> <display-name>Test 2</display-name> </entry> <entry uri="sip:test222@192.168.1.134"> <display-name>Test 2</display-name> </entry> </list> <list name="Friends"> <entry uri="sip:test3@192.168.1.134"> <display-name>Test 3</display-name> </entry> <entry uri="sip:test322@192.168.1.134"> <display-name>Test 3</display-name> </entry> </list> <list name="Blocked" /> </resource-lists>

2 4

Re: [SR-Users] testing the value of $si
by Alex Balashov 27 Apr '12

27 Apr '12

It should work. What are your symptoms that it doesn't? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com x-kamailio(a)sidell.org wrote: >How do I compare $si to a particular IP address value? This doesn't >seem to work: > >if( $si == "123.123.123.123") > >_______________________________________________ >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

testing the value of $si
by x-kamailio＠sidell.org 27 Apr '12

27 Apr '12

How do I compare $si to a particular IP address value? This doesn't seem to work: if( $si == "123.123.123.123")

1 0

Re: [SR-Users] dispatcher and call transfer
by charles.chance＠sipcentric.com 27 Apr '12

27 Apr '12

1 0

Re: [SR-Users] dispatcher and call transfer
by charles.chance＠sipcentric.com 27 Apr '12

27 Apr '12

2 1

Re: [SR-Users] dispatcher and call transfer
by charles.chance＠sipcentric.com 27 Apr '12

27 Apr '12

1 0

Re: [SR-Users] dispatcher and call transfer
by Asgaroth 27 Apr '12

27 Apr '12

Hi All, Just resending this as my previous message was too big (bigger than 60k) ---. Thanks all for the suggestions. @Charles Yes that is correct, that is the scenario that is occurring, we are using asterisk, when a call is initiated via asterisk 1, the second call then routes through to asterisk 2, asterisk 1 then gets the REFER but doesn't know about the 2nd call. @Reda I like this option of using the hash table (I'd like to exclude a database lookup for each initial invite), however, would I be forced to use a database in the case of 2 loadbalancers, surely some shared state would need to exist between the loadbalancers to make this decision on where to route the call if it already exists. I suspect the hashtable suggestion may work perfectly in a single loadbalancer scenario, unless I'm missing something. For example, the initialy call may go via loadbalancer 1 however the second call may go through loadbalancer 2, that way loadbalancer 2 may not know of an existing call already been setup. In keeping with the hashtable train of thought, is it really neccessary to keep a reference to the callid, as the second call in the transfer scenario described by Charles, the callid of the second call is different to the original call. Just as an FYI, I am currently using dispatcher algorithm 0 which is a hash over callid. Thanks again all for the suggestions/tips

1 0

Re: [SR-Users] dispatcher and call transfer
by Asgaroth 27 Apr '12

27 Apr '12

Thanks all for the suggestions. @Charles Yes that is correct, that is the scenario that is occurring, we are using asterisk, when a call is initiated via asterisk 1, the second call then routes through to asterisk 2, asterisk 1 then gets the REFER but doesn't know about the 2nd call. @Reda I like this option of using the hash table (I'd like to exclude a database lookup for each initial invite), however, would I be forced to use a database in the case of 2 loadbalancers, surely some shared state would need to exist between the loadbalancers to make this decision on where to route the call if it already exists. I suspect the hashtable suggestion may work perfectly in a single loadbalancer scenario, unless I'm missing something. For example, the initialy call may go via loadbalancer 1 however the second call may go through loadbalancer 2, that way loadbalancer 2 may not know of an existing call already been setup. In keeping with the hashtable train of thought, is it really neccessary to keep a reference to the callid, as the second call in the transfer scenario described by Charles, the callid of the second call is different to the original call. Just as an FYI, I am currently using dispatcher algorithm 0 which is a hash over callid. Thanks again all for the suggestions/tips On 26/04/12 16:12, Charles Chance wrote: > > Yes, I realise Asterisk is not a media relay, but I don't think the OP > is using a pure media relay, otherwise the REFER message would not be > sent to it, would it? > > > > Thank you for the hash table suggestion by the way. > > > > Charles > > > > ------------------------------------------------------------------------ > > *From:*sr-users-bounces@lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *Reda Aouad > *Sent:* 26 April 2012 15:15 > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > *Subject:* Re: [SR-Users] dispatcher and call transfer > > > > Asterisk is not really a media relay. Asterisk establishes 2 legs for > each call, and I'm not sure what happens in this case. > > An improvement you can make is to use a hash table (in-memory) to > store the information you need (call-id, from, to) then lookup in > the htable for existing calls for same users. That should relieve your > database from a query on every invite and increase performance if you > have a large number of calls. > > > > Reda > > > > > > On Thu, Apr 26, 2012 at 15:52, Charles Chance > <charles.chance(a)sipcentric.com <mailto:charles.chance@sipcentric.com>> > wrote: > > Hi Reda, > > > > Sorry, I should have been more specific -- I am referring to instances > where the media server is for example Asterisk. If first call goes > through Asterisk 1, second call goes through Asterisk 2, when Asterisk > 2 receives the REFER it does not know of initial call on Asterisk 1. > The only way we have found for it to work is to ensure the second call > is dispatched to the same Asterisk box as the first. > > > > I would be pleased to hear of an alternative method. > > > > Regards, > > > > Charles > > ------------------------------------------------------------------------ > > *From:*sr-users-bounces@lists.sip-router.org > <mailto:sr-users-bounces@lists.sip-router.org> > [mailto:sr-users-bounces@lists.sip-router.org > <mailto:sr-users-bounces@lists.sip-router.org>] *On Behalf Of *Reda Aouad > *Sent:* 26 April 2012 14:34 > > > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > *Subject:* Re: [SR-Users] dispatcher and call transfer > > > > Hi, > > @Carsten > Dispatcher algorithm 0 based on call-id should do it in your case of > re-invite within dialog with same call-id. > > @Charles > In the case of attended transfers, shouldn't both media servers be > relaying media between them? I didn't understand why your are obliged > to dispatch to the same media server since they are 2 different calls > with different call-ids. > > > > Reda > > > > > > On Thu, Apr 26, 2012 at 14:30, Charles Chance > <charles.chance(a)sipcentric.com <mailto:charles.chance@sipcentric.com>> > wrote: > > Hi, > > Actually, this won't help for attended transfers where another call is > initiated first then the two are joined together by REFER. In this > case, the > second INVITE must be routed to the same media server as the existing call > for the transfer to work. > > What we do is store the dialogs in DB, then when a new call comes in, > prior > to doing ds_select_dst we query DB for existing call involving same > user. If > we find one, we simply replace destination host with that from the contact > (to/from depending on direction of call). > > It may not be the most elegant way but it works for us :) > > Charles > > > > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org > <mailto:sr-users-bounces@lists.sip-router.org> > [mailto:sr-users-bounces@lists.sip-router.org > <mailto:sr-users-bounces@lists.sip-router.org>] On Behalf Of Carsten Bock > Sent: 26 April 2012 13:25 > To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > Subject: Re: [SR-Users] dispatcher and call transfer > > Hi, > > if you look at the docs of the dispatcher module, you'll find this: > > alg - the algorithm used to select the destination address. The > parameter can be an integer or a variable holding an interger. > - "0" - hash over callid > (http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2498492) > > But probably you should look into record/loose_route for your setup. > Since REFER is normally an in-dialog request (belongs to another > voice-session), it should take the same route as the initial INVITE. > This is normally achieved by the record/loose-route mechanisms > described in RFC3261. > In the example config of Kamailio you find an configuration example > (below). > > It is not a bug in the dispatcher module, it's how you use it. > > So long, > Carsten > > 455 # handle requests within SIP dialogs > 456 route(WITHINDLG); > > [...] > > 473 # record routing for dialog forming requests (in case > they are routed) > 474 # - remove preloaded route headers > 475 remove_hf("Route"); > 476 if (is_method("INVITE|SUBSCRIBE")) > 477 record_route(); > > and the relevent parts in the "WITHINDLG" route: > > 566 # Handle requests within SIP dialogs > 567 route[WITHINDLG] { > 568 if (has_totag()) { > 569 # sequential request withing a dialog should > 570 # take the path determined by record-routing > 571 if (loose_route()) { > [...] > 580 route(RELAY); > 581 } else { > [...] > 587 if ( is_method("ACK") ) { > 588 if ( t_check_trans() ) { > 589 # no loose-route, but > stateful > ACK; > 590 # must be an ACK after a 487 > 591 # or e.g. 404 from upstream > server > 592 t_relay(); > 593 exit; > 594 } else { > 595 # ACK without matching > transaction ... ignore and discard > 596 exit; > 597 } > 598 } > 599 sl_send_reply("404","Not here"); > 600 } > 601 exit; > 602 } > 603 } > > 2012/4/26 Asgaroth <00asgaroth00(a)gmail.com > <mailto:00asgaroth00@gmail.com>>: > > Hi All, > > > > Currently we are running kamailio in a loadbalanced fashion whereby calls > > come in via the loadbalancers and distribute calls accross 2 media > servers. > > We have come accross and issue whereby call transfers may be distributed > > accross two media servers and when the REFER message comes along to > transfer > > the call, in some cases (if we're lucky) the message arrives at the wrong > > media server (transaction leg doesnt exist). > > > > Some googling later and it appears that dispatcher doesnt play nice when > it > > comes to this scenario. Some suggestions popped up in my previous > searches > > saying that a potential work around is to use the dialog module to check > if > > a call is eastablished and then to send all calls to the same media > server > > based on the dialog already being established. > > > > I'd appreciate some input from the guru's out there that have come > accross > > this same issue and, if possible, some suggestions on how to work around > the > > problem, does the dispatcher module have a hashing algorithm that can be > > suited for this particular scenario? > > > > Thanks in advance for any tips or sugestions. > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com <mailto:carsten@ng-voice.com> > > Mobile +49 179 2021244 <tel:%2B49%20179%202021244> > Office +49 40 34927219 <tel:%2B49%2040%2034927219> > Fax +49 40 34927220 <tel:%2B49%2040%2034927220> > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > -- > Meet ng-voice at LinuxTag 2012 in Berlin - May 23rd - 26th, 2012. Save the > date! > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ----- > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12 > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > ------------------------------------------------------------------------ > > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12 > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > ------------------------------------------------------------------------ > > No virus found in this message. > Checked by AVG - www.avg.com <http://www.avg.com> > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12 > > > > _______________________________________________ > 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

1 0

Copy reason field from 503 to 500.
by Vitaliy Aleksandrov 27 Apr '12

27 Apr '12

Hi all, I have Kamailio connected to PSTN gateway. Sometimes gateway replies with "503 temporary unavailable". PSTN gateway adds a "reason" header with ISUP code. But according to RFC 3261 kamailio doesn't forward 503 replies transparently. When 503 reply is received Kamailio replaces 503 with a 500 which is created from original request (build_res_buf_from_sip_req). I have tried append_hf from textops module, but it's not working with a locally generated replies. How can i copy Reason header from 503 to 500 ? I will appreciate your help with this situation. Best regards, Vitaliy Aleksandrov.

2 4

Gentoo ebuild v3.2.3
by caio 26 Apr '12

26 Apr '12

Good afternoon, It's ready the Gentoo ebuild for 3.2.3. You can get it from the Git repository or from the Gentoo Portage overlays: http://goo.gl/35uFX http://goo.gl/fmFIU Best regards, Caio

1 0

kamailio Crashes with rls module
by Gnaneshwar Gatla 26 Apr '12

26 Apr '12

Hello, I'm experimenting the usage of rls module with Kamailio. I have been able to successfully update xcap documents on Kamailio XCAP server. I have tried to use SIPp to get a NOTIFY for subscribing a rls-service list. Kamailio does get the xml from the xcap table and tries to parse it and crashes. I'm not sure if this is because the format of the XML, I derived the xml format from the RFC 4662 and 4826. I have attached the Kamailio log. I need help with this. Any help is appreciated, thank you very much. Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

1 0

Re: [SR-Users] dispatcher and call transfer
by Reda Aouad 26 Apr '12

26 Apr '12

Asterisk is not really a media relay. Asterisk establishes 2 legs for each call, and I'm not sure what happens in this case. An improvement you can make is to use a hash table (in-memory) to store the information you need (call-id, from, to) then lookup in the htable for existing calls for same users. That should relieve your database from a query on every invite and increase performance if you have a large number of calls. Reda On Thu, Apr 26, 2012 at 15:52, Charles Chance <charles.chance(a)sipcentric.com > wrote: > ** ** ** ** > > Hi Reda,**** > > ** ** > > Sorry, I should have been more specific – I am referring to instances > where the media server is for example Asterisk. If first call goes through > Asterisk 1, second call goes through Asterisk 2, when Asterisk 2 receives > the REFER it does not know of initial call on Asterisk 1. The only way we > have found for it to work is to ensure the second call is dispatched to the > same Asterisk box as the first.**** > > ** ** > > I would be pleased to hear of an alternative method.**** > > ** ** > > Regards,**** > > ** ** > > Charles**** > ------------------------------ > > *From:* sr-users-bounces(a)lists.sip-router.org [mailto: > sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Reda Aouad > *Sent:* 26 April 2012 14:34 > > *To:* **SIP Router - Kamailio** (OpenSER) and SIP Express Router (SER) - > UsersMailing List > *Subject:* Re: [SR-Users] dispatcher and call transfer > **** > > ** ** > > Hi, > > @Carsten > Dispatcher algorithm 0 based on call-id should do it in your case of > re-invite within dialog with same call-id. > > @Charles > In the case of attended transfers, shouldn't both media servers be > relaying media between them? I didn't understand why your are obliged to > dispatch to the same media server since they are 2 different calls with > different call-ids.**** > > ** ** > > Reda**** > > ** ** > > ** ** > > On Thu, Apr 26, 2012 at 14:30, Charles Chance < > charles.chance(a)sipcentric.com> wrote:**** > > Hi, > > Actually, this won't help for attended transfers where another call is > initiated first then the two are joined together by REFER. In this case, > the > second INVITE must be routed to the same media server as the existing call > for the transfer to work. > > What we do is store the dialogs in DB, then when a new call comes in, prior > to doing ds_select_dst we query DB for existing call involving same user. > If > we find one, we simply replace destination host with that from the contact > (to/from depending on direction of call). > > It may not be the most elegant way but it works for us :) > > Charles**** > > > > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Carsten Bock > Sent: 26 April 2012 13:25 > To: **SIP Router - Kamailio** (OpenSER) and SIP Express Router (SER) - > UsersMailing List > Subject: Re: [SR-Users] dispatcher and call transfer > > Hi, > > if you look at the docs of the dispatcher module, you'll find this: > > alg - the algorithm used to select the destination address. The > parameter can be an integer or a variable holding an interger. > - “0” - hash over callid > ( > http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2498492 > ) > > But probably you should look into record/loose_route for your setup. > Since REFER is normally an in-dialog request (belongs to another > voice-session), it should take the same route as the initial INVITE. > This is normally achieved by the record/loose-route mechanisms > described in RFC3261. > In the example config of Kamailio you find an configuration example > (below). > > It is not a bug in the dispatcher module, it's how you use it. > > So long, > Carsten > > 455 # handle requests within SIP dialogs > 456 route(WITHINDLG); > > [...] > > 473 # record routing for dialog forming requests (in case > they are routed) > 474 # - remove preloaded route headers > 475 remove_hf("Route"); > 476 if (is_method("INVITE|SUBSCRIBE")) > 477 record_route(); > > and the relevent parts in the "WITHINDLG" route: > > 566 # Handle requests within SIP dialogs > 567 route[WITHINDLG] { > 568 if (has_totag()) { > 569 # sequential request withing a dialog should > 570 # take the path determined by record-routing > 571 if (loose_route()) { > [...] > 580 route(RELAY); > 581 } else { > [...] > 587 if ( is_method("ACK") ) { > 588 if ( t_check_trans() ) { > 589 # no loose-route, but stateful > ACK; > 590 # must be an ACK after a 487 > 591 # or e.g. 404 from upstream > server > 592 t_relay(); > 593 exit; > 594 } else { > 595 # ACK without matching > transaction ... ignore and discard > 596 exit; > 597 } > 598 } > 599 sl_send_reply("404","Not here"); > 600 } > 601 exit; > 602 } > 603 } > > 2012/4/26 Asgaroth <00asgaroth00(a)gmail.com>: > > Hi All, > > > > Currently we are running kamailio in a loadbalanced fashion whereby calls > > come in via the loadbalancers and distribute calls accross 2 media > servers. > > We have come accross and issue whereby call transfers may be distributed > > accross two media servers and when the REFER message comes along to > transfer > > the call, in some cases (if we're lucky) the message arrives at the wrong > > media server (transaction leg doesnt exist). > > > > Some googling later and it appears that dispatcher doesnt play nice when > it > > comes to this scenario. Some suggestions popped up in my previous > searches > > saying that a potential work around is to use the dialog module to check > if > > a call is eastablished and then to send all calls to the same media > server > > based on the dialog already being established. > > > > I'd appreciate some input from the guru's out there that have come > accross > > this same issue and, if possible, some suggestions on how to work around > the > > problem, does the dispatcher module have a hashing algorithm that can be > > suited for this particular scenario? > > > > Thanks in advance for any tips or sugestions. > > > > _______________________________________________ > > 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 > > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 **Hamburg** / ****Germany**** > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Mobile +49 179 2021244 <%2B49%20179%202021244> > Office +49 40 34927219 <%2B49%2040%2034927219> > Fax +49 40 34927220 > > Sitz der Gesellschaft: ****Hamburg**** > Registergericht: Amtsgericht ****Hamburg****, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > -- > Meet ng-voice at LinuxTag 2012 in ****Berlin**** - May 23rd - 26th, 2012. > Save the > date! > > _______________________________________________ > 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**** > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12* > *** > > > > _______________________________________________ > 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**** > > ** ** > ------------------------------ > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12* > *** > > _______________________________________________ > 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 > >

2 1

Re: [SR-Users] About MSRP relay
by 罗俊明 26 Apr '12

26 Apr '12

3 2

Re: [SR-Users] dispatcher and call transfer
by Reda Aouad 26 Apr '12

26 Apr '12

Hi, @Carsten Dispatcher algorithm 0 based on call-id should do it in your case of re-invite within dialog with same call-id. @Charles In the case of attended transfers, shouldn't both media servers be relaying media between them? I didn't understand why your are obliged to dispatch to the same media server since they are 2 different calls with different call-ids. Reda On Thu, Apr 26, 2012 at 14:30, Charles Chance <charles.chance(a)sipcentric.com > wrote: > Hi, > > Actually, this won't help for attended transfers where another call is > initiated first then the two are joined together by REFER. In this case, > the > second INVITE must be routed to the same media server as the existing call > for the transfer to work. > > What we do is store the dialogs in DB, then when a new call comes in, prior > to doing ds_select_dst we query DB for existing call involving same user. > If > we find one, we simply replace destination host with that from the contact > (to/from depending on direction of call). > > It may not be the most elegant way but it works for us :) > > Charles > > > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Carsten Bock > Sent: 26 April 2012 13:25 > To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > Subject: Re: [SR-Users] dispatcher and call transfer > > Hi, > > if you look at the docs of the dispatcher module, you'll find this: > > alg - the algorithm used to select the destination address. The > parameter can be an integer or a variable holding an interger. > - “0” - hash over callid > ( > http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2498492 > ) > > But probably you should look into record/loose_route for your setup. > Since REFER is normally an in-dialog request (belongs to another > voice-session), it should take the same route as the initial INVITE. > This is normally achieved by the record/loose-route mechanisms > described in RFC3261. > In the example config of Kamailio you find an configuration example > (below). > > It is not a bug in the dispatcher module, it's how you use it. > > So long, > Carsten > > 455 # handle requests within SIP dialogs > 456 route(WITHINDLG); > > [...] > > 473 # record routing for dialog forming requests (in case > they are routed) > 474 # - remove preloaded route headers > 475 remove_hf("Route"); > 476 if (is_method("INVITE|SUBSCRIBE")) > 477 record_route(); > > and the relevent parts in the "WITHINDLG" route: > > 566 # Handle requests within SIP dialogs > 567 route[WITHINDLG] { > 568 if (has_totag()) { > 569 # sequential request withing a dialog should > 570 # take the path determined by record-routing > 571 if (loose_route()) { > [...] > 580 route(RELAY); > 581 } else { > [...] > 587 if ( is_method("ACK") ) { > 588 if ( t_check_trans() ) { > 589 # no loose-route, but stateful > ACK; > 590 # must be an ACK after a 487 > 591 # or e.g. 404 from upstream > server > 592 t_relay(); > 593 exit; > 594 } else { > 595 # ACK without matching > transaction ... ignore and discard > 596 exit; > 597 } > 598 } > 599 sl_send_reply("404","Not here"); > 600 } > 601 exit; > 602 } > 603 } > > 2012/4/26 Asgaroth <00asgaroth00(a)gmail.com>: > > Hi All, > > > > Currently we are running kamailio in a loadbalanced fashion whereby calls > > come in via the loadbalancers and distribute calls accross 2 media > servers. > > We have come accross and issue whereby call transfers may be distributed > > accross two media servers and when the REFER message comes along to > transfer > > the call, in some cases (if we're lucky) the message arrives at the wrong > > media server (transaction leg doesnt exist). > > > > Some googling later and it appears that dispatcher doesnt play nice when > it > > comes to this scenario. Some suggestions popped up in my previous > searches > > saying that a potential work around is to use the dialog module to check > if > > a call is eastablished and then to send all calls to the same media > server > > based on the dialog already being established. > > > > I'd appreciate some input from the guru's out there that have come > accross > > this same issue and, if possible, some suggestions on how to work around > the > > problem, does the dispatcher module have a hashing algorithm that can be > > suited for this particular scenario? > > > > Thanks in advance for any tips or sugestions. > > > > _______________________________________________ > > 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 > > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Mobile +49 179 2021244 > Office +49 40 34927219 > Fax +49 40 34927220 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > -- > Meet ng-voice at LinuxTag 2012 in Berlin - May 23rd - 26th, 2012. Save the > date! > > _______________________________________________ > 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 > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2411/4959 - Release Date: 04/25/12 > > > _______________________________________________ > 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 >

2 1

dispatcher and call transfer
by Asgaroth 26 Apr '12

26 Apr '12

Hi All, Currently we are running kamailio in a loadbalanced fashion whereby calls come in via the loadbalancers and distribute calls accross 2 media servers. We have come accross and issue whereby call transfers may be distributed accross two media servers and when the REFER message comes along to transfer the call, in some cases (if we're lucky) the message arrives at the wrong media server (transaction leg doesnt exist). Some googling later and it appears that dispatcher doesnt play nice when it comes to this scenario. Some suggestions popped up in my previous searches saying that a potential work around is to use the dialog module to check if a call is eastablished and then to send all calls to the same media server based on the dialog already being established. I'd appreciate some input from the guru's out there that have come accross this same issue and, if possible, some suggestions on how to work around the problem, does the dispatcher module have a hashing algorithm that can be suited for this particular scenario? Thanks in advance for any tips or sugestions.

3 2

generating cdrs
by Pirjo Ahvenainen 26 Apr '12

26 Apr '12

Hi all, I'm currently playing with accounting and cdrs, and meaning to test cdr record generation. Using Kamailio 3.2.3 on Ubuntu. My question is how to enable Kamailio to save cdr data into database (MySql in my case)? I have loaded the dialog module and enabled the cdr generation: modparam("acc", "cdr_enable", 1) ... but no cdrs in db, what else do I need to do? The script is mostly default config, any logic I need to add etc? I'll be fine with the most basic cdrs first and will expand from there. I do get entries in syslog for invite and bye of the transaction. I assume the cdr recodrs should be visible in cdrs table, but have nothing there. According to the acc module documentation: "In order to set up CDR generation, you must enable the CDR switch and load the dialog module", does cdr switch here mean the cdr_enable module parameter of the acc module? cheers, Pirjo -- Pirjo Ahvenainen +35844 559 7729

3 9

reply code for lower cseq
by Daniel-Constantin Mierla 25 Apr '12

25 Apr '12

Hello, I was wondering if someone here can point quickly where specs mention what is the right reply code to send when a request within dialog is received with lower cseq value than the previous request. I couldn't spot the part in the RFC yet, if any related exists. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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