- sr-users - kamailio.org

Kamailio NOTIFY
by Spinov Evgeniy 19 Feb '11

19 Feb '11

Hello all. I'm having problem with sending NOTIFY packet for event dialog. The problem is, that all NOTIFY packets sent with this event is empty. ( Content-length: 0 ) and in fact, not passing actual status to the phone. In my case this is SPA962. NOTIFY packets for event presence are sent fine and status is changing normally. The same phone, works fine with Asterisk. Asterisk is sending NOTIFY with non-zero content and changing status for the subscriber as well. I've turned on pua module to send PUBLISH requests at Kamailio, it's filling DB table with subscription, everything runs fine and NOTIFY packets are coming to subscribed phone, but they are empty. I also have presence_dialoginfo module on and pua_dialoginfo module on. I've tried on Kamailio 3.0.3 and 3.1.1. Content always empty. There is no sense of pasting here packet dumps, cause everything is like should be: (UAC) SUBSCRIBE, event: presence ---> K K ---> 202 OK (SPA) SUBSCRIBE, event: dialog ---> K K ---> 202 OK PUBLISH, event: presence ---> K K ---> 200 OK NOTIFY, event: presence, Content-Length: XXX ---> UAC NOTIFY, event: dialog, Content-Length: 0 ---> SPA962 As result, softphone changes status and hardphone ( SPA962 ) - no. May be it's cause of PUBLISH event value? Please advise.

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why function result -1 cannot be assigned to variable?
by Juha Heinanen 19 Feb '11

19 Feb '11

i have a local function x that returns a positive integer or -1. when it returns -1, assignment $var(tmp) = function_x("$rd"); produces to syslog an error: Feb 19 07:32:44 sip /usr/sbin/sip-proxy[27284]: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (1415,40-1415,40) i have had that function for years, but have not noticed these errors in earlier versions than 3.1. any ideas why the error is produced and how to get rid of it? -- juha

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Interfaces guestures
by Spinov Evgeniy 18 Feb '11

18 Feb '11

Hello, all. I'm having Kamailio 3.0.4. Server have 2 physical interfaces, one is private, second is public. INVITE request is coming to Kamailio from public interface and being routed by Kamailio to one of ther asterisk boxes, which is in private network. Packet is routed to correct interface, but SRC address of packet, containing public Kamailio IP. As of that, asterisk is unable to communicate properly with Kamailio. I'm not applying any packet changes in Kamailio config and cannot get why Kamailio is doing that? Kamailio is listening both networks and able to send packets with correct IP, but somewhy sending an invalid one. Am I doing something wrong? Thank you.

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Re: [SR-Users] pdb module timeouts
by Thomas Baumann 18 Feb '11

18 Feb '11

Hello Marius, thanks a lot for submitting the patch. It works like a charm now. But to be honest I had troubles to find this patch, I just checked the daily tarball [http://sip-router.org/tarballs/sr/] and found this change. How I can get patch e914f97b4 in a better way ? regards, Thomas >Hello >I have made a patch to 3.1(e914f97b4) . The poll() event list was not >cleared when a response was received so when 2 servers responded one >after the other it might block in an incorrect recv(). Can you please >test with this? >Marius ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

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Re: [SR-Users] ERROR: failover functions used, but AVPs paraamters required are NULL
by Gary Chen 18 Feb '11

18 Feb '11

Never mind. I used wrong kamailio.cfg file. From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Thursday, February 17, 2011 8:51 AM To: Gary Chen Cc: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] ERROR: failover functions used, but AVPs paraamters required are NULL Hello, can you send all log messages printed by: kamailio -E -ddd Thanks, Daniel On 2/17/11 2:42 PM, Gary Chen wrote: kamailio version 3.1.2 I am trying to setup dispatcher to use its failover feature. Here is dispatcher part of configure file: loadmodule "dispatcher.so" modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:lb1 <at> m-lab-ca805-sig.kd-lab.de") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "flags", 2) But it is still give the following error: 0(1917) ERROR: dispatcher [dispatcher.c:624]: failover functions used, but AVPs paraamters required are NULL -- feature disabled Does anybody know why? Gary _______________________________________________ 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

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Re: [SR-Users] Bad File Descriptor
by Daniel-Constantin Mierla 17 Feb '11

17 Feb '11

Please keep the mailing list cc-ed. My previous email had other comments/questions inline, can you answer them as well? They are relevant in troubleshooting. Thanks, Daniel On 2/17/11 7:57 PM, David J. wrote: > version: kamailio 3.1.1 (x86_64/linux) 88bda8 > 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, PKG_SIZE 4MB > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > id: 88bda8 > compiled on 04:39:27 Dec 9 2010 with gcc 4.3.2 > > > > On 2/17/11 1:46 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> what is the version you are using? >> >> On 2/17/11 7:35 PM, David J. wrote: >>> 5(20390) ERROR: <core> [udp_server.c:586]: ERROR: udp_send: >>> sendto(sock,0x7f67f7f94d14,4,0,69.117.34.101:2048,16): Bad file >>> descriptor(9) >>> 5(20390) ERROR: nathelper [nathelper.c:1722]: udp_send failed >>> >>> >>> I see this error when I try to restart kamailio after crash; >> >> What is the cause for the crash? Do you have a log or core for that? >> >>> I see the cause of this problem is 'stale' entries in the location >>> table; >>> >>> If I delete this entries kamailio starts fine; any suggestions to >>> prevent this from continuously happening. >> Is any change in the IP address of the server upon restart? >> >> Do the errors persist or they stop after a while? They are related to >> NAT keepalives sent by nathelper module, so they don't affect the sip >> traffic and if they are for stale contacts then they are fully >> harmless. The stale contacts should be automatically removed after a >> while by usrloc module. >> >> Cheers, >> Daniel >> > -- Daniel-Constantin Mierla http://www.asipto.com

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Bad File Descriptor
by David J. 17 Feb '11

17 Feb '11

5(20390) ERROR: <core> [udp_server.c:586]: ERROR: udp_send: sendto(sock,0x7f67f7f94d14,4,0,69.117.34.101:2048,16): Bad file descriptor(9) 5(20390) ERROR: nathelper [nathelper.c:1722]: udp_send failed I see this error when I try to restart kamailio after crash; I see the cause of this problem is 'stale' entries in the location table; If I delete this entries kamailio starts fine; any suggestions to prevent this from continuously happening. Thanks.

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Kamailio 1.5.5 No TLS Segmentation Fault
by Stagg Shelton 17 Feb '11

17 Feb '11

Hello, We have been having a problem with Kamilio faulting and dumping core files on occasion. I have not been able to reproduce the failure at will, but notice the back trace seems to point toward actions with the dialogue. Below is from a backtrace of a core file from just a few minutes ago. Can anyone determine what may have caused the system to error and stop processing? Thanks Stagg Core was generated by `/sbin/kamailio -m 512'. Program terminated with signal 11, Segmentation fault. #0 0x00007f8a11d55fa7 in unref_dlg (dlg=0x7f89f7e07470, cnt=1) at dlg_hash.c:474 474 d_entry = &(d_table->entries[dlg->h_entry]); Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.5-5.fc11.x86_64 db4-4.7.25-11.fc11.x86_64 e2fsprogs-libs-1.41.9-2.fc11.x86_64 elfutils-libelf-0.147-1.fc11.x86_64 glibc-2.10.2-1.x86_64 keyutils-libs-1.2-5.fc11.x86_64 krb5-libs-1.6.3-31.fc11.x86_64 libacl-2.2.49-3.fc11.x86_64 libattr-2.4.43-3.fc11.x86_64 libcap-2.16-4.fc11.1.x86_64 libconfuse-2.6-2.fc11.x86_64 libgcc-4.4.1-2.fc11.x86_64 libselinux-2.0.80-1.fc11.x86_64 lm_sensors-3.1.0-1.fc11.x86_64 lua-5.1.4-3.fc11.x86_64 mysql-libs-5.1.46-1.fc11.x86_64 net-snmp-libs-5.4.2.1-13.fc11.x86_64 nspr-devel-4.8.4-1.3.fc11.x86_64 nss-devel-3.12.6-1.2.fc11.x86_64 nss-softokn-freebl-3.12.6-1.2.fc11.x86_64 openssl-0.9.8n-1.fc11.x86_64 pcre-7.8-2.fc11.x86_64 perl-libs-5.10.0-82.fc11.x86_64 popt-1.13-5.fc11.x86_64 radiusclient-ng-0.5.6-4.fc11.x86_64 rpm-libs-4.7.2-1.fc11.x86_64 tcp_wrappers-libs-7.6-55.fc11.x86_64 xz-libs-4.999.9-0.1.beta.20091007git.fc11.x86_64 zlib-1.2.3-22.fc11.x86_64 (gdb) bt full #0 0x00007f8a11d55fa7 in unref_dlg (dlg=0x7f89f7e07470, cnt=1) at dlg_hash.c:474 d_entry = 0x0 __FUNCTION__ = "unref_dlg" #1 0x00007f8a11d5180f in unref_dlg_from_cb (t=0x7f89f7d9c660, type=4096, param=0x7f8a1836a6e0) at dlg_handlers.c:622 dlg = 0x7f89f7e07470 #2 0x00007f8a18138ea3 in run_trans_callbacks (type=4096, trans=0x7f89f7d9c660, req=0x0, rpl=0x0, code=0) at t_hooks.c:240 cbp = 0x7f89f7dc30e8 backup = 0x71a9d0 trans_backup = 0xffffffffffffffff __FUNCTION__ = "run_trans_callbacks" #3 0x00007f8a181273cc in free_cell (dead_cell=0x7f89f7d9c660) at h_table.c:132 b = 0x0 i = 1 rpl = 0x0 tt = 0x0 foo = 0x7fff4282f190 p = 0x7f89f7d3b068 #4 0x00007f8a18127bb6 in free_hash_table () at h_table.c:345 p_cell = 0x7f89f7d9c660 tmp_cell = 0x0 i = 4075 #5 0x00007f8a181342a4 in tm_shutdown () at t_funcs.c:109 __FUNCTION__ = "tm_shutdown" #6 0x00000000004529f6 in destroy_modules () at sr_module.c:321 t = 0x7349d0 foo = 0x734910 __FUNCTION__ = "destroy_modules" #7 0x000000000041f6b4 in cleanup (show_status=1) at main.c:331 No locals. #8 0x0000000000420597 in handle_sigs () at main.c:517 chld = 0 chld_status = 134 i = 12 do_exit = 1 ---Type <return> to continue, or q <return> to quit--- shutdown_time = 60 __FUNCTION__ = "handle_sigs" #9 0x00000000004217b5 in main_loop () at main.c:859 chd_rank = 12 i = 4 pid = 21442 si = 0x0 __FUNCTION__ = "main_loop" #10 0x0000000000423410 in main (argc=3, argv=0x7fff4282f498) at main.c:1321 cfg_log_stderr = 0 cfg_stream = 0x1fe1010 c = -1 r = 0 tmp_len = 0 port = 0 proto = 4910128 ret = -1 rfd = 4 tmp = 0x7fff4282ff8a "" options = 0x4b77e0 "f:cCm:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:" rand_source = 0x4b7d9c "/dev/urandom" seed = 3628387751 __FUNCTION__ = "main" (gdb) (gdb) quit

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Failover with UCARP and Monit
by Hugh Waite 17 Feb '11

17 Feb '11

Hello, I am setting up a high-availablilty kamailio system using UCARP to failover between active and standby instances. To detect failure, we intend to use Monit. Monit can monitor the kamailio PID and start the process when needed (Example on the wiki http://www.kamailio.org/dokuwiki/doku.php/install:configure-initd-script) and it can also do OPTIONS pings to verify it is working. If the pings fail we will initiate a ucarp swap. However if the server is currently in standby, it does not have the V-IP address, so I don't want to run the OPTIONS pings (I think). Does anyone use a similar system and can provide an example of how ucarp, monit and kamailio can work together? Many thanks, Hugh Waite

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Kamailio and Asterisk Realtime Integration using Asterisk Database
by Pavel Miskov 17 Feb '11

17 Feb '11

Hi all, I am trying to integrate Kamailio and Asterisk as explained in http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb document but I have some problems. 1. When Asterisk sends the call back to Kamailio, Asterisk generates new INVITE but with wrong From URI. In From header, "Display name" is OK (Phone A) but "From URI" is wrong (it is URI of Phone B). 2. When Phone B answers the call, Asterisk generates another INVITE to Phone B. When Phone B sends OK to this second INVITE Asterisk generates another INVITE to Phone A. 3. When Phone B sends BYE, Asterisk generates INVITE to Phone A and when it receives OK from Phone A it sends BYE to Phone A. I can fix the first problem with transformation in Kamailio but is this the way how it is supposed to work or I misconfigured something? I have tried this with kamailio 3.1 and asterisk 1.6.2.16.1 (and 1.8.2.3). Thanks Pavel

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Mediaproxy 2.4.4 sessions
by Ricardo Martinez 17 Feb '11

17 Feb '11

Hello. I’m using kamailio 3.1.2 and mediaproxy 2.4.4 . Is there a way to see the active media sessions like in the old mediaproxy? I was using the command ./sessions Thanks Ricardo-.

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Kamailio 3.0.4: Segfault at 4 [..] error 4 in dialog.so
by Klaus Feichtinger 17 Feb '11

17 Feb '11

Hello, does anybody have an idea, what this error messages could mean? Feb 16 11:23:27 SipSrvLstI1 kernel: [429767.707165] kamailio[10758]: segfault at 4 ip b78ba6e5 sp bfacece0 error 4 in dialog.so[b78a0000+32000] Feb 16 13:40:42 SipSrvLstI1 kernel: [438003.071797] kamailio[24880]: segfault at 4 ip b79416e5 sp bff7d460 error 4 in dialog.so[b7927000+32000] Feb 16 18:47:25 SipSrvLstI1 kernel: [456406.166885] kamailio[20939]: segfault at 4 ip b78d56e5 sp bfd821c0 error 4 in dialog.so[b78bb000+32000] I am setting the dialog flag for any incoming call..... Whenever a load test is running, this error occurs. I am not sure if it is reproduceable, or not. But I think it should be, because in worst case (during a load test over night) this error was indicated in a period of 1-2 minutes. But I've found this error only yesterday, because the server was down (as a result of this error....). The daemon (kamailio_3.0.4+lenny1_i386.deb) is running on a virtual machine (ESX) with OS Debian Lenny. How can I find / deliver additional info to this error / fault for analyzation? Thanks for any hints. Klaus

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ACC og ACC_RADIUS module
by Morten Isaksen 17 Feb '11

17 Feb '11

Hi, We have a OpenSER 1.1 platform running with radius accounting and I am in the progress of updating it to Kamailio 3.1. I am trying to decide if I should do accounting via Radius or directly to MySQL on the new platform. The only benefits a can see with Radius is that you can build some redundancy into your radius client. If one Radius server is failing then try the next and you can configure radius to log to a file if the DB is down. But i think you can get the same level of redundancy with a replicated DB setup with heartbeat/pacemaker. If I choose to do the accounting direct to MySQL I will skip the Radius layer (and one error source). Are there any other pros and cons? -- Morten Isaksen

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ERROR: failover functions used, but AVPs paraamters required are NULL
by Gary Chen 17 Feb '11

17 Feb '11

kamailio version 3.1.2 I am trying to setup dispatcher to use its failover feature. Here is dispatcher part of configure file: loadmodule "dispatcher.so" modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:lb1 <at> m-lab-ca805-sig.kd-lab.de") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "flags", 2) But it is still give the following error: 0(1917) ERROR: dispatcher [dispatcher.c:624]: failover functions used, but AVPs paraamters required are NULL -- feature disabled Does anybody know why? Gary

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Re: [SR-Users] pdb module timeouts
by Thomas Baumann 16 Feb '11

16 Feb '11

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Re: [SR-Users] Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 16 Feb '11

16 Feb '11

Hi Marius, Thanks for the reply. I found the patch in the 1.5 CVS branch. By the way, the 1.5.5 source tarball in /pub/kamailio/1.5.5/src was created before this patch was written, so the patch is not in there. The tarball I created from the 1.5 branch did have it. Thanks again, Sean O'Donnell Senior Engineer uReach Technologies, Inc. ---- On Mon, 14 Feb 2011, marius zbihlei (marius.zbihlei(a)1and1.ro) wrote: On 02/11/2011 09:14 PM, Sean O'Donnell wrote: > Hi all, > Hello Sean, > I checked the GIT repository and noticed that there was a patch in forward.c for > this issue. Looks like it was done 11/4/2010. Two questions: > > 1) Is there any reason that patch didn't make it into Kamailio 3.1.2 ? > Indeed, there is no reason. Actually I have cherry-picked the patches into the 3.1 branch (commit_notice 3.1 3ec1e9 and 9ef0a1e). Thanks for noticing. > 2) Any reason I shouldn't apply that patck to my 1.5.4 release? > The patch was already applied to the svn 1.5 branch (don't know if the latest 1.5.5 tar ball has it) but you can install it from svn and you can find it there (commit 6050) Cheers Marius > Thanks, > Sean O'Donnell > Senior Engineer > uReach Technologies, Inc. > > > > > > ---- On Thu, 10 Feb 2011, Sean O'Donnell (skodonnell(a)ureach.com) wrote: > > Hi all, > > I just started deploying Kamailio release 1.5.4, and I think there's an issue > with how Kamailio identifies an outgoing interface when mhomed is enabled under > Linux. > > I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a > voicemail platform. It > runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is > configured such that it listens on both interfaces. One interface (public > interface) handles traffic with the SBC, the other (private interface) handles > with the VM platform. The 'mhomed' option is enabled. > > After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP > packets coming out of the public interface. After looking at some ngrep > captures on that interface, I noticed that some packets had the source IP > address of the private interface and also had Record-Route and Via headers for > the private interface only - no headers for the public interface were there. > > Usually when I see the wrong source IP in a UDP packet, it's an issue with how > routes are set up on the host. However, I had our network engineer double check > them, and they seem fine (no ambiguous routes). The fact that I captured these > messages on the public interface also indicates to me that the kernel is routing > the message correctly. The missing Record-Route and Via for the public > interface, however, lead me to believe that the proxy didn't correctly identify > the outgoing interface in the first place. > > After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was > put in to improve performance when mhomed is enabled (r5971) in forward.c, and I > think this is the issue. > > As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined > the outgoing interface by creating a temporary UDP socket, invoking connect() on > the socket with the packet destination, then checking the source IP of the > socket that the kernel assigned using getsockname(). After the source address > was determined, the temp socket was closed closed. As of 1.5.4, this was > modified to reuse the temporary socket and just re-invoking connect() with a new > destination address. > > The problem with the enhancement is that Linux (again, at least in the 2.6 > kernels I'm using) > doesn't seem to rebind a new source address to the socket when connect() is > called more than once on > a UDP socket. Instead, it keeps the original one, and thus the wrong interface > is assumed. > > I wrote a small program to confirm this - basically creates a UDP socket, calls > connect()/getsockname() > multiple times using different destination addresses. I ran it on several 2.6 > kernels, including > Centos4.x and Centos5. The result was always that the source address of the > socket wasn't changed after the first connect(), regardless of the destination > address. The only way I could get it work as > required was to first do a connect() using a zero'd out AF_UNSPEC address before > doing the > connect() to the remote address. I also ran it on Solaris and it worked. Go > figure. > > I've downloaded the latest stable release (3.1.2) but I think the issue is still > there, and I don't see > anything in the user groups that addresses this. > > Any help would be appreciated. > > Thanks, > > Sean > > > _______________________________________________ > 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 > > > > > _______________________________________________ > 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 > > _______________________________________________ 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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Error when tring to load lcr module.
by Gary Chen 16 Feb '11

16 Feb '11

I am using Kamailio 3.1.2. I added lcr module in my kamailio.cfg file. After run this: ./kamctl -E I got this error: 0(23744) ERROR: lcr [lcr_mod.c:1235]: lcr_gw params at row <0> does not start with ';' 0 I have set up some gateway information in lcr_gw , lcr_rule and lcr_rule_target. From log it seems retrieve all the information but then got above error. Here is my lcr part of config: loadmodule "lcr.so" modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") Gary

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Re: [SR-Users] pdb module timeouts
by Thomas Baumann 15 Feb '11

15 Feb '11

Hello, I did a fallback to the original source, without sucess. I still have massive timeouts. So far I have no idea, how to go on. ps. what lenght of the carrier codes would be ok ? 3...10 digits, additional characters needed like A,B,C,D,E,F? regards, Thomas Hi Henning, this change was done at pdbt.c: $ diff pdbt.c pdbt_changed.c 245c245 < bufsize = slen + 1 + 1 + 3 + 1 + 1; // line buff er (telephone number + colon + white space + carrier ID + newline + \0) --- > bufsize = slen + 1 + 1 + 4 + 1 + 1; // line buff er (telephone number + colon + white space + carrier ID + newline + \0) 257,258c257,258 < ret = snprintf(p, 5, "%d\n", node->carrier); < if (ret < 1 || ret > 4) { --- > ret = snprintf(p, 6, "%d\n", node->carrier); > if (ret < 1 || ret > 5) { If I run querys in filemode against the server, where is no timeout. pdbt query -f /tmp/numbers3 -r 10.12.18.21:10001 -q 1 | grep answer regards, Thomas -----Ursprüngliche Nachricht----- Von: "Henning Westerholt" <henning.westerholt(a)1und1.de> Gesendet: 11.02.2011 09:33:05 An: sr-users(a)lists.sip-router.org Betreff: Re: [SR-Users] pdb module timeouts On Wednesday 09 February 2011, Thomas Baumann wrote: > I am using the PDB module and server components for number portability. 2 > instances of PDB Server runs on (10.12.19.51/10001/10002), Kamailio on > (10.12.19.21). With a small amount of traffic (-cmax 150 -cps 10 > -callduration 3), where are timeouts: WARNING: pdb [pdb.c:260]: exceeded > timeout while waiting for response. > > One requested number was 307111094, where the module prints out a timeout. > > The funny part is, that I can see the responses at least arriving at the > 10.12.19.21 interface on time. > > Request send: 0,200855 s > Answer received: 0,201027 s > > That are 0,172 ms and far away from a timeout. > > What could be the reason ? Hi Thomas, a bit difficult to say on a first sight.. Maybe some scheduling or load issues on the local machine? I've just checked two production server here, i don't see it so far in the available logs. How much load do you've on the machine? And what kind of timeout value do you specified? > ps. A small change on the server part was done: handle 4 digit carrier > codes. Ah, i see. I guess its not a big change? Regards, Henning ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

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CORE postgres module
by Bruno Bresciani 15 Feb '11

15 Feb '11

Hi, During my tests with kamailio 1.5.0 a core is generated when postgres is disconnected and I try register a user or make a call in kamailio. Analising the module postgres source code I notice that core is generated by the PQescapeStringConn located in db_postgres_val2str function. Someone know why this core is generated? Best Regards

2 2

Issue with avp
by Amit Nepal 14 Feb '11

14 Feb '11

Hi, I have been trying to load rpid while loading credentials. modparam("auth_db", "load_credentials", "$avp(i:123)=rpid") Now, I am trying to do a check in my routing logic. xlog("L_NOTICE","The avp is :$avp(i:123)"); (I dont get the value here either, i can't see the value when i do avp_print() if($avp(i:123)<5) { sl_send_reply("408","Message Here"); } I dont get the value of avp at that place. Any guidance please. -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

2 4

kamailio 1.5.5 crashes continue
by Andrew O. Zhukov 14 Feb '11

14 Feb '11

Core was generated by `/usr/local/sbin/kamailio -P /var/run/openser/openser.pid -m 32 -u openser -g op'. Program terminated with signal 6, Aborted. #0 0x00000039d8c30265 in raise () from /lib64/libc.so.6 (gdb) backtrace #0 0x00000039d8c30265 in raise () from /lib64/libc.so.6 #1 0x00000039d8c31d10 in abort () from /lib64/libc.so.6 #2 0x000000000046c397 in qm_debug_frag (qm=0x733c00, f=0x841420) at mem/q_malloc.c:137 #3 0x000000000046d99a in qm_free (qm=0x733c00, p=0x841450, file=0x4e4d30 "parser/digest/digest.c", func=0x4e4da0 "free_credentials", line=95) at mem/q_malloc.c:439 #4 0x0000000000495fac in free_credentials (_b=0x2b875404a130) at parser/digest/digest.c:95 #5 0x0000000000471a36 in clean_hdr_field (hf=0x2b875404a100) at parser/hf.c:116 #6 0x00002b8750ab78de in clean_msg_clone (msg=0x2b8754049130, min=0x2b8754049130, max=0x2b875404a748) at sip_msg.h:54 #7 0x00002b8750ab77b7 in run_trans_callbacks (type=2, trans=0x2b8754057ec0, req=0x2b8754049130, rpl=0x7c0eb8, code=200) at t_hooks.c:245 #8 0x00002b8750abe39d in t_reply_matching (p_msg=0x7c0eb8, p_branch=0x7fffbe5d11e8) at t_lookup.c:888 #9 0x00002b8750abe997 in t_check (p_msg=0x7c0eb8, param_branch=0x7fffbe5d11e8) at t_lookup.c:964 #10 0x00002b8750acd79b in reply_received (p_msg=0x7c0eb8) at t_reply.c:1395 #11 0x000000000041c6db in forward_reply (msg=0x7c0eb8) at forward.c:576 #12 0x000000000043ccf0 in receive_msg ( buf=0x712980 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP XXX.XX.XXX.13;branch=z9hG4bK8aa9.93e271f1.0;received=XXX.XX.XXX.13\r\nVia: SIP/2.0/UDP XXX.XX.XXX.236:5060;received=XXX.XX.XXX.236;branch=z9hG4bK37dacff3;rport=5060\r\nRec"..., len=576, rcv_info=0x7fffbe5d1340) at receive.c:212 #13 0x00000000004692e3 in udp_rcv_loop () at udp_server.c:449 #14 0x0000000000420ecb in main_loop () at main.c:774 #15 0x0000000000422e0f in main (argc=11, argv=0x7fffbe5d15c8) at main.c:1321 (gdb) q [root@ tmp]# openser -V version: kamailio 1.5.5-notls (x86_64/linux) flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 09:42:37 Feb 10 2011 with gcc 4.1.2 Andrew.

2 3

kamailio and MySQL licensing question
by Siddhardha Garige 14 Feb '11

14 Feb '11

hello all, I have a question regarding MySQL licensing. We configured kamailio to work in edge proxy mode with drive-by registration to support our network architecture. One of our clients is interested in conducting some trails with us and we are setting up a test network at client's facility. As a part of this process we will be installing our custom kamailio on their servers with MySQL. We are not charging our client for kamailio and mysql implementation since they both are open source implementations. But we will be charging them for our time at clients facility. Are we violating any MySQL license policies by installing our custom kamailio with MySQL for our client? thanks Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

2 1

Re: [SR-Users] Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 14 Feb '11

14 Feb '11

Hi all, I checked the GIT repository and noticed that there was a patch in forward.c for this issue. Looks like it was done 11/4/2010. Two questions: 1) Is there any reason that patch didn't make it into Kamailio 3.1.2 ? 2) Any reason I shouldn't apply that patck to my 1.5.4 release? Thanks, Sean O'Donnell Senior Engineer uReach Technologies, Inc. ---- On Thu, 10 Feb 2011, Sean O'Donnell (skodonnell(a)ureach.com) wrote: Hi all, I just started deploying Kamailio release 1.5.4, and I think there's an issue with how Kamailio identifies an outgoing interface when mhomed is enabled under Linux. I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a voicemail platform. It runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is configured such that it listens on both interfaces. One interface (public interface) handles traffic with the SBC, the other (private interface) handles with the VM platform. The 'mhomed' option is enabled. After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP packets coming out of the public interface. After looking at some ngrep captures on that interface, I noticed that some packets had the source IP address of the private interface and also had Record-Route and Via headers for the private interface only - no headers for the public interface were there. Usually when I see the wrong source IP in a UDP packet, it's an issue with how routes are set up on the host. However, I had our network engineer double check them, and they seem fine (no ambiguous routes). The fact that I captured these messages on the public interface also indicates to me that the kernel is routing the message correctly. The missing Record-Route and Via for the public interface, however, lead me to believe that the proxy didn't correctly identify the outgoing interface in the first place. After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was put in to improve performance when mhomed is enabled (r5971) in forward.c, and I think this is the issue. As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined the outgoing interface by creating a temporary UDP socket, invoking connect() on the socket with the packet destination, then checking the source IP of the socket that the kernel assigned using getsockname(). After the source address was determined, the temp socket was closed closed. As of 1.5.4, this was modified to reuse the temporary socket and just re-invoking connect() with a new destination address. The problem with the enhancement is that Linux (again, at least in the 2.6 kernels I'm using) doesn't seem to rebind a new source address to the socket when connect() is called more than once on a UDP socket. Instead, it keeps the original one, and thus the wrong interface is assumed. I wrote a small program to confirm this - basically creates a UDP socket, calls connect()/getsockname() multiple times using different destination addresses. I ran it on several 2.6 kernels, including Centos4.x and Centos5. The result was always that the source address of the socket wasn't changed after the first connect(), regardless of the destination address. The only way I could get it work as required was to first do a connect() using a zero'd out AF_UNSPEC address before doing the connect() to the remote address. I also ran it on Solaris and it worked. Go figure. I've downloaded the latest stable release (3.1.2) but I think the issue is still there, and I don't see anything in the user groups that addresses this. Any help would be appreciated. Thanks, Sean _______________________________________________ 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

Error running kamailio 3.1.0
by Lucas Alvarez 12 Feb '11

12 Feb '11

Hi, I have compiled kamailio 3.1.0 without any error and I having this error when running kamailio: ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/sl.so>: /usr/local/lib/kamailio/modules_k/sl.so: undefined symbol: fm_malloc Any will be appreciated. Regards, Lucas

2 1

After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set
by Andrew O. Zhukov 12 Feb '11

12 Feb '11

[root@ tmp]# /usr/local/sbin/kamailio -V version: kamailio 1.5.5-notls (x86_64/linux) flags: STATISTICS, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 12:38:36 Feb 2 2011 with gcc 4.1.2 ----------------------------- Core was generated by `/usr/local/sbin/kamailio -P /var/run/openser/openser.pid -m 32 -u openser -g op'. Program terminated with signal 11, Segmentation fault. #0 0x000000000046b0e3 in fm_malloc (qm=0x72dc00, size=32) at mem/f_malloc.c:354 354 if ((*f)->size>=size) goto found; (gdb) backtrace #0 0x000000000046b0e3 in fm_malloc (qm=0x72dc00, size=32) at mem/f_malloc.c:354 #1 0x00002b30f2803087 in build_rr (_l=0x76f110, _l2=0x76fe80, user=0x7fffe9c5a500, tag=0x777a58, params=0x0, _inbound=0) at record.c:176 #2 0x00002b30f2802b7a in record_route (_m=0x76e0e0, params=0x0) at record.c:322 #3 0x00002b30f28047db in w_record_route (msg=0x76e0e0, key=0x0, bar=0x0) at rr_mod.c:212 #4 0x000000000040ed9b in do_action (a=0x73f5a0, msg=0x76e0e0) at action.c:874 #5 0x000000000040c03a in run_action_list (a=0x73f5a0, msg=0x76e0e0) at action.c:145 #6 0x000000000040e6a7 in do_action (a=0x73f810, msg=0x76e0e0) at action.c:746 #7 0x000000000040c03a in run_action_list (a=0x73e418, msg=0x76e0e0) at action.c:145 #8 0x000000000040c2a9 in run_actions (a=0x73e418, msg=0x76e0e0) at action.c:120 #9 0x000000000040c357 in run_top_route (a=0x73e418, msg=0x76e0e0) at action.c:195 #10 0x000000000043bda4 in receive_msg ( buf=0x70c980 "NOTIFY sip:XXXXXX.com SIP/2.0\r\nVia: SIP/2.0/UDP XX.XXX.101.68:5060;branch=z9hG4bK-6ee3865\r\nFrom: VTHome <sip:101650@XXXXXX.com>;tag=129d73a13db8ec7fo0\r\nTo: <sip:XXXXX.com>\r\nCall-ID: e3fd1da9-142a0a17"..., len=373, rcv_info=0x7fffe9c5ae90) at receive.c:175 #11 0x0000000000467eeb in udp_rcv_loop () at udp_server.c:449 #12 0x000000000042097b in main_loop () at main.c:774 #13 0x00000000004228b0 in main (argc=11, argv=0x7fffe9c5b118) at main.c:1321 (gdb) print size $1 = 32 (gdb) quit -------------------------------------------- Core was generated by `/usr/local/sbin/kamailio -P /var/run/openser/openser.pid -m 32 -u openser -g op'. Program terminated with signal 11, Segmentation fault. #0 0x000000000046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609 609 size+=f->size,f=f->u.nxt_free,i++,j++){ (gdb) backtrace #0 0x000000000046bf7b in fm_status (qm=0x72dc00) at mem/f_malloc.c:609 #1 0x000000000041feb3 in sig_usr (signo=15) at main.c:563 #2 <signal handler called> #3 0x00000039d8cd4a51 in __recvfrom_nocancel () from /lib64/libc.so.6 #4 0x0000000000467bf4 in udp_rcv_loop () at udp_server.c:408 #5 0x000000000042097b in main_loop () at main.c:774 #6 0x00000000004228b0 in main (argc=11, argv=0x7fffe9c5b118) at main.c:1321 (gdb) print i $1 = 402 (gdb) print j $2 = 1 (gdb) print size $3 = 7234295468789601279 (gdb) print f $4 = (struct fm_frag *) 0x3738656435393838 (gdb) print f->size Cannot access memory at address 0x3738656435393838 ------------------------------------------------------------------- Andrew O. Zhukov

2 6

pdb module timeouts
by Thomas Baumann 11 Feb '11

11 Feb '11

Hello, I am using the PDB module and server components for number portability. 2 instances of PDB Server runs on (10.12.19.51/10001/10002), Kamailio on (10.12.19.21). With a small amount of traffic (-cmax 150 -cps 10 -callduration 3), where are timeouts: WARNING: pdb [pdb.c:260]: exceeded timeout while waiting for response. One requested number was 307111094, where the module prints out a timeout. The funny part is, that I can see the responses at least arriving at the 10.12.19.21 interface on time. Request send: 0,200855 s Answer received: 0,201027 s That are 0,172 ms and far away from a timeout. What could be the reason ? regards, Thomas ps. A small change on the server part was done: handle 4 digit carrier codes. level3_logs: Feb 9 17:19:42 node2 /openser/sbin/kamailio[21200]: DEBUG: pdb [pdb.c:201]: querying '307111094'... Feb 9 17:19:42 node2 /openser/sbin/kamailio[21200]: WARNING: pdb [pdb.c:260]: exceeded timeout while waiting for response. ___________________________________________________________ Schon gehört? WEB.DE hat einen genialen Phishing-Filter in die Toolbar eingebaut! http://produkte.web.de/go/toolbar

3 4

Change transport=tls within Contact Header
by Bernhard Suttner 11 Feb '11

11 Feb '11

Hi, is there a function to change the transport field within the Contact Header? t_relay does relay it to the UDP port of the PBX but the transport within the Contact Header is still set to TLS. Later, the PBX does try to establish a TLS connection. The idea is now, that kamailio does set the transport of the Contact Header to UDP. Another possible way would to be, that I use the re.subst method but maybe there is a special function for that. Thanks in advance. Best regards, Bernhard

2 1

NAT, TLS and location table
by Bernhard Suttner 11 Feb '11

11 Feb '11

Hi, I am using TLS and recognize the following problem: The TLS connection are build up successfully but the natping (natping_interval = 10) does not send small dummy packets to the phones. The phones are behind a firewall with NAT. Registered phones with NAT but UDP do work correctly. They are getting the natping every 10 seconds. After 120 seconds (should be the tcp_connection_timeout) kamailio does send a FIN to the TLS phone to close the TLS connection. Should I increase the tcp_connection_timeout to a value bigger than the registration timeout? I thought I do not need that, because of the natping_interval. Is it maybe better to use a SIP-Options Ping instead of the small dummy packets? I would prefer the dummy packets because they are much smaller. BTW: Kamailio 3.1.1 and snom360 with latest v8 firmware. Thanks in advance! Best regards, Bernhard

2 3

handle_new_connect ... Too many open files
by Efelin Novak 11 Feb '11

11 Feb '11

Hi, today during two hours kamailio made 2G of a syslog messages. This message was repeated for 10 000 000 times. /sbin/kamailio[24272]: ERROR: <core> [tcp_main.c:3959]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files It was preceded by following error. /sbin/kamailio[24265]: ERROR: <core> [tcp_read.c:269]: error reading: Connection reset by peer (104) /sbin/kamailio[24265]: ERROR: <core> [tcp_read.c:882]: ERROR: tcp_read_req: error reading Finally it stopped without any given reason. I'm using kamailio 3.1.0 (i386/linux) 1e204f, callcontrol 2.0.11 and clustered mysql. Was this some kind of attack, bug, error, misconfiguration or what? Thank for answer, Efelin

2 1

TLS Support
by Bernhard Suttner 11 Feb '11

11 Feb '11

Hi, how could I compile the TLS module and only the TLS module - not the other sources. BTW: The documentation within the http://www.kamailio.org/docs/tls.html is not up-to-date. Thanks in advance. Best regards, Bernhard

2 2

Kamailio as a sip client
by Amit Nepal 11 Feb '11

11 Feb '11

Hi, Is it possible to have kamailio register to a different proxy and route calls through that proxy ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 3

Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 10 Feb '11

10 Feb '11

Hi all, I just started deploying Kamailio release 1.5.4, and I think there's an issue with how Kamailio identifies an outgoing interface when mhomed is enabled under Linux. I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a voicemail platform. It runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is configured such that it listens on both interfaces. One interface (public interface) handles traffic with the SBC, the other (private interface) handles with the VM platform. The 'mhomed' option is enabled. After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP packets coming out of the public interface. After looking at some ngrep captures on that interface, I noticed that some packets had the source IP address of the private interface and also had Record-Route and Via headers for the private interface only - no headers for the public interface were there. Usually when I see the wrong source IP in a UDP packet, it's an issue with how routes are set up on the host. However, I had our network engineer double check them, and they seem fine (no ambiguous routes). The fact that I captured these messages on the public interface also indicates to me that the kernel is routing the message correctly. The missing Record-Route and Via for the public interface, however, lead me to believe that the proxy didn't correctly identify the outgoing interface in the first place. After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was put in to improve performance when mhomed is enabled (r5971) in forward.c, and I think this is the issue. As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined the outgoing interface by creating a temporary UDP socket, invoking connect() on the socket with the packet destination, then checking the source IP of the socket that the kernel assigned using getsockname(). After the source address was determined, the temp socket was closed closed. As of 1.5.4, this was modified to reuse the temporary socket and just re-invoking connect() with a new destination address. The problem with the enhancement is that Linux (again, at least in the 2.6 kernels I'm using) doesn't seem to rebind a new source address to the socket when connect() is called more than once on a UDP socket. Instead, it keeps the original one, and thus the wrong interface is assumed. I wrote a small program to confirm this - basically creates a UDP socket, calls connect()/getsockname() multiple times using different destination addresses. I ran it on several 2.6 kernels, including Centos4.x and Centos5. The result was always that the source address of the socket wasn't changed after the first connect(), regardless of the destination address. The only way I could get it work as required was to first do a connect() using a zero'd out AF_UNSPEC address before doing the connect() to the remote address. I also ran it on Solaris and it worked. Go figure. I've downloaded the latest stable release (3.1.2) but I think the issue is still there, and I don't see anything in the user groups that addresses this. Any help would be appreciated. Thanks, Sean

1 0

Merging nathelper modules
by Ovidiu Sas 10 Feb '11

10 Feb '11

Hello all, The nathelper module in modules_k was split in two: - nathelper (dealing with signaling); - rtpproxy (dealing with rtpproxy protocol). I would like to move the rtpproxy module from "modules_k" into "modules" and remove rtpproxy functionality from nathelper (s). This will give to (s) and (k) users: - rtpproxy: a single module for dealing with rtpproxy servers; - nathelper: two variants for dealing with NAT signaling. Next step, will be to merge the two nathelper modules into a single one. Thoughts? Regards, Ovidiu Sas

2 2

nathelper-module and sdp offer in 200 OK
by Emil Kroymann 10 Feb '11

10 Feb '11

Hi, We recently had a problem with the nathelper module and rtpproxy in a scenario where the SDP offer is sent only in the 200 OK. We use sip-router 3.1 and rtp-proxy from git master. The sip-router configuration uses the rtpproxy_offer() and rtpproxy_answer() functions in appropriate places. The problem is, that the arguments sent to the rtpproxy, when the ACK with the sdp answer arrives, seems to be not in the order, that rtpproxy expects. On the 200 OK, the nathelper module sends callid, to-tag, from-tag to rtpproxy. On the ACK, the nathelper module sends callid, from-tag, to-tag (with different command prefixes, of course, but I cannot remember them atm). The version of rtpproxy that we are using seems to expect, that the order of arguments sent on the ACK request is the same as on the 200 OK. My question: are there any module parameters, to correct this behaviour? Regards, Emil -- Emil Kroymann VoIP Services Engineer Email: emil.kroymann(a)isaco.de Tel: +49-30-203899885 ISACO GmbH Kurfürstenstraße 79 10787 Berlin Germany Amtsgericht Charlottenburg, HRB 112464B Geschäftsführer: Daniel Frommherz

3 12

Use different certificate for different client with TLS
by Daniel GROTTI 09 Feb '11

09 Feb '11

Hi all, I would like to use kamailio 3.1 with TLS and verified also a client certificate. My tls.cfg file is as follow: --- tls.cfg ---- ..... ..... [server:MY_IP:5061] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = default_key.pem certificate = default_cert.pem ca_list = default_ca.pem [client:default] verify_certificate = no require_certificate = no .... .... So I want to verify the client certificate, to do that I use "is_peer_verified()" function in kamailio.cfg. As tls.cfg shows, I have to send to my clients the CA certificate and the client certificate (default_cert.pem + default_key.pem - signing by the CAcert). This client certificate is unique for all clients. Everything works fine. But suppose I wanted to create a client certificate for client 1 (cert_1.crt), and a different client certificate for client 2 (cert_2.crt) and I want to configure kamailio to be able to verified this different certificates. Does it possible ? How can I configure the tls.cfg file to do that ? I try to do something like this: [server:MY_IP:5061] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = default_key_1.pem certificate = default_cert_1.pem private_key = default_key_2.pem certificate = default_cert_2.pem ca_list = default_ca.pem But when kamailio restart it seems that it read only the last couple of row certifcate/private_key. Regards, Daniel G

2 1

[OpenSER-Users] uaCSTA support in openser
by Hugo Koblmueller 09 Feb '11

09 Feb '11

high.all! i'm wondering if there is any support of uaCSTA in openser (planned)? i'm just working on the integration of asterisk (*) environment to OCS 2007 environment, having openSER in the middle (mainly for TCP/UDP translation and smoothing out the protocol deficienes on both sides). in this setup the * having the openSER in front is talking to the OCS (and vice versa) via the OCS mediation server, which is moreorless sending standard SIP messages, which enables normal softphone (integration to *) of the office communicator. this configuration is already working... now i'm planning to go for the CTI integration, where there is no OCS mediation server in between OCS and openSER, doing the translation of SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why i'm looking for a CSTA module or perl programm, which is capable of this functionality. afaik for the CTI communication there isn't the full complexity of CSTA needed, just a subset mainly for call setup and call clearing. anyone having experience on this topic? thx & cheers -hugo Great Ideas for Small Devices Hugo Koblmueller Senior Staff Engineer Software Development COMNEON electronic technology GmbH & Co. OHG Freistaedter Strasse 400 4040 Linz Austria hugo.koblmueller(a)comneon.com tel: fax: mobile: Skype ID: +43 (5) 1777 - 15730 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%285%2 9+1777+%2D+15730&Email=hugo(a)koblmueller.com> +43 (5) 1777 - 15810 +43 (676) 82051280 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%28676 %29+82051280&Email=hugo(a)koblmueller.com> drhookson Want to always have my latest info? <https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join &invite=1&lang=en> Want a signature like <http://www.plaxo.com/signature?src=client_sig_212_1_banner_sig&lang=en> this?

7 13

Re: [SR-Users] module postgres SER
by Daniel-Constantin Mierla 09 Feb '11

09 Feb '11

please keep the mailing list cc-ed, sending private messages is not in the spirit of public mailing lists. Others may want to follow up the discussion now or later. Thanks, Daniel On 2/9/11 5:01 PM, Bruno Bresciani wrote: > Daniel, > > thanks for your reply, really ser-0.8.1.4 is too old but i need to > solve this problem on that version. My great doubt is Why the aug_free > function corrupt the url of database after some attemps to reconnect. > Well, I'll try to understand this question... > > Best Regards > > > 2011/2/9 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 2/9/11 2:19 PM, Bruno Bresciani wrote: > > Hi, > > I've seen the problem in a postgres module (SER-0.8.1.4), if > the connection fails and module tries to reparse url it fails > as CON_SQLURL(_h) is corrupted by the function aug_free() > after some reconnect attempts . When the postgres database > back to work, some modules doesn't get reconnect because the > db_url is corrupted. Why this is happening? There are some > solution for this problem? > > ser 0.8.1.4 is soooo old and I cannot fully remember, but I think > postgres module had no reconnect functionality at all by that time. > > However, version 3.1.x of SER (as well as Kamailio flavour) has db > reconnect functionality for postgres. You can try it and report if > something is not working, it will be fixed in 3.x, but I think > nobody is still developing on 0.8.x to backport anything there. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

problem with bye using rtpproxy
by Amit Nepal 09 Feb '11

09 Feb '11

I have been trying to figure this out While using kamailio and rtpproxy, the caller is not receiving the bye when callee hangs up but audio is two way and everything seems to be working fine, any one had this issue ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

2 1

Multiple call accounting from an IP
by Amit Nepal 09 Feb '11

09 Feb '11

Hi everyone, I am sure someone have been working with this scenario, how about accounting multiple calls from same account or ip address while using ip auth ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

2 1

fosdem 2011 presentation about p_usrloc
by Henning Westerholt 09 Feb '11

09 Feb '11

Hello, this year the FOSDEM developer conference was a again a really nice event, the first time with an own room dedicated completely to open source telephony solutions! If you're interested in our presentation about the new p_usrloc module and how to scale location services with Kamailio, you can find it at the usual place on our webserver: http://kamailio.org/events/2011-fosdem/p_usrloc.pdf Best regards, Henning

3 3

module postgres SER
by Bruno Bresciani 09 Feb '11

09 Feb '11

Hi, I've seen the problem in a postgres module (SER-0.8.1.4), if the connection fails and module tries to reparse url it fails as CON_SQLURL(_h) is corrupted by the function aug_free() after some reconnect attempts . When the postgres database back to work, some modules doesn't get reconnect because the db_url is corrupted. Why this is happening? There are some solution for this problem? Best Regard

2 1

Fosdem2011 presentation: SIP Web20 Lua
by Daniel-Constantin Mierla 09 Feb '11

09 Feb '11

Hello, the presentation I did during Fosdem in Brussels last weekend is available at: http://www.kamailio.org/events/2011-fosdem/dcm-sip-web-lua.pdf The focus was to show how to interact with Kamailio via HTTP and how Kamailio can interact with Web services via HTTP, using Lua to make it easier. There are slides for a demo config of sending asynchronous notifications to Twitter on missed calls (using modules app_lua, mqueue, sqlops and rtimer). Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] problem with SER not proxying messages
by Asuncion Merino Rodríguez 09 Feb '11

09 Feb '11

Hola, no he visto respuesta a este foro, por favor alguien podría dar alguna respuesta?

1 0

Kamailio v3.1.2 Released
by Daniel-Constantin Mierla 08 Feb '11

08 Feb '11

Hello, Kamailio SIP Server v3.1.2 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.1. There is no change to database or configuration file required to upgrade to 3.1.2 from 3.1.0 or 3.1.1 versions, therefore it is strongly recommended to upgrade to v3.1.2. For more details about version 3.1.2, visit: http://www.kamailio.org/w/2011/02/kamailio-v3-1-2-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Problem with LCR!
by Ricardo Martinez 08 Feb '11

08 Feb '11

Hello. We’re having critical issues with our LCR, we have the Kamailio version 1.5.2, several calls are arriving to our LCR and from some reason the next message appears in the log : Feb 6 12:01:55 /usr/local/sbin/kamailio[14185]: NOTICE:lcr:gw_set_state: trunk "xx.xx.xx.135:5060" from group: <2099> is OFFLINE! We can see this message repeated for every single gateway in the “gw” table from the LCR, after this happened, the LCR rounting process is blocked and no more calls are processed. Can someone help us with this problem? Thanks in advance, *Regards,* *Ricardo Martinez.-* * *

3 3

planning release of v3.1.2
by Daniel-Constantin Mierla 08 Feb '11

08 Feb '11

Hello, I think it is time to release v3.1.2, first date that comes in my mind is next Thursday if everyone feels it is enough time to take care of backporting any fix he/she did and it is not yet there. That will provide us a fresh release for the FOSDEM event. If not, then maybe the other week, Tuesday, so the participants at the Kamailio Devel training in Barcelona can practice on it. Soon after we should plan also a release for previous stable, branch 3.0. Anyone having other options? Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 1

Problem with load_gws()
by Amit Nepal 08 Feb '11

08 Feb '11

Hi, Below is my config and i am getting error , missing load module modparam("lcr","db_url",DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") route[LCR] { if(!load_gws()){ xlog("yes"); }; This gives me error : loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio 0(1265) : <core> [cfg.y:3412]: parse error in config file routes/route-lcr.cfg unknown command, missing loadmodule? for load_gws() ERROR: bad config file (1 errors) Any guidance please ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 4

Handling timers from an Avaya PBX (throwing 404s)
by Graham Wooden 08 Feb '11

08 Feb '11

Hi there, I turned up a few new sip trunk customers this past week, from which one of them was an Avaya system. The customer wasn¹t able to do a username/password auth, so I set them up as a permanent entry in the location table. I am running Kamailio 1.5.5. The issue is that I am throwing back a 404 to them when they do a timer event, as they are passing sip:Unknown@my.ip.addr. I don¹t have a user name Unknown, hence the 404. I noticed that with the timer events from other customer¹s PBXs, that they are passing sip:my.addr, from which I am giving back 200 OK. What is the best way to handle this? Just let it go or add a ³Unknown² user (I don¹t like the latter, really)? Calls to and from their system is fine, so right now it¹s just entries in my logs but I hate spending CPU cycles on stuff that isn¹t going to benefit anyone. I sent an email back to the customer to see if maybe he can resolve it on his end, but I am doubtful on getting it resolved on his end. Thanks, -graham

3 5

Proposal to extend acc/dialog modules in order to log CDRs
by Timo Reimann 07 Feb '11

07 Feb '11

Hi all, I'd like to bring forward a proposal for creating CDRs (Call Data Records) from Kamailio by means of extending both acc and dialog modules. Feedback is very welcome. So far, CDR generation requires post-processing of log entries created by the (transaction-based) acc module or some other means. Looking at OpenSIPS (which is already capable of producing CDRs), my idea is to copy the basic notion of tracking dialogs for to-be-accounted calls and log a proper CDR on termination. Creating basic CDRs including start time, end time, and call duration can be accomplished quite easily this way. Apart from this very minimum CDR content, however, one can think of a number of CDR fields that cannot be filled easily. For instance, caller identity isn't always contained in the From header (think of CLIR calls) but need to be determined from other headers depending on the type of call, possibly even involving a database lookup. That's why I'd like to provide a way to pass additional, CDR-specific data from the configuration file to the dialog structure associated with the call. When the dialog is about to terminate, the extended acc module would make sure that the added CDR data associated to the dialog is included in the CDR. This way, users may define CDR fields to their personal needs from the configuration file in a flexible manner. In order to accomplish this, the dialog module would need to be extended such that dialog-specific data may be stored for the duration of a dialog (which is not possible to this day AFAICS). A new setter function attached to the dialog interface (struct dlg_binds) could accept a simple key/value pair for storage purposes. Exporting this function to the configuration file and accompanying it with a corresponding getter function will ensure that dialog-specific data may be written and read generically. Thus, improvements to the dialog module may be leveraged not just for CDR-accounting purposes. Regarding the acc module, a couple of new features need to be implemented: First, the introduction of another module parameter called something like "cdr_fields" that comprises the set of key names designated for CDR inclusion. Hence, a line like modparam("acc", "cdr_fields", "caller, callee, foo, bar") would require Kamailio configurators to store dialog-specific values for the keys "caller", "callee", "foo", and "bar", respectively, during call processing. The values will be fed to the dialog structure employing the new dialog store function explained above. Second, when the acc module is in CDR mode, dialogs shall be tracked from creation to termination using dialog callbacks. On termination of a dialog, values from keys referenced in the "cdr_fields" parameter will be fetched from the terminating dialog structure and associated with the new, fresh CDR. Third, the CDR is persisted to either log file, database, or both. Note that my approach would not touch the existing acc module functionality: CDR mode is supposed to be working in addition to transaction-based accounting as people will likely want to keep transaction logs for debugging or similar needs. A flag would enable and disable CDR generation at will. I have not fully investigated re-usability possibilities but believe that acc's persistence functions may be taken advantage of by the new CDR-producing parts as well. That's it with regards to the basic mode of operation. I am also thinking of having a few other features, such as using a switch to set the call start time to the moment the ACK or OK is seen instead of the INVITE (because users shouldn't be billed for call setup times). Looking forward to seeing your feedback on this proposal. Cheers, --Timo

3 6

topoh, angle brackets and Contact URI params interpretation
by Andrew Pogrebennyk 07 Feb '11

07 Feb '11

I'm having an issue with topoh module in Kamailio 3.1.0. When Contact header is formed like this: Contact: 0991 <sip:192.168.0.107;line=sr-N6IAzBysz.tyz.D4M.VLOBMfOBFuWxvfMxV4> The other party responds properly. But when there is no angle brackets in Contact: Contact: sip:192.168.0.107;line=sr-N6IAzB3AWxyfz.stM.quOBFZMJZfWxj7W.y-MljAWBy* Really, many parsers implement "prefer shift to reduce" principle, which means if something can be interpreted in more enclosed expression, it will be interpreted this way and no as part of less enclosed expression so ;line is interpreted as header parameter but not URI parameter. RFC 3261 section 20 suggests that any URI parameters be contained within angle brackets: === The Contact, From, and To header fields contain a URI. If the URI contains a comma, question mark or semicolon, the URI MUST be enclosed in angle brackets (< and >). Any URI parameters are contained within these brackets. If the URI is not enclosed in angle brackets, any semicolon-delimited parameters are header-parameters, not URI parameters. === I think the topoh module should force the angle brackets. BTW it seems that parameter needs to be urlencoded, see rule 'other-param' in RFC 3261 section 25.1: other-param = pname [ "=" pvalue ] pname = 1*paramchar pvalue = 1*paramchar paramchar = param-unreserved / unreserved / escaped param-unreserved = "[" / "]" / "/" / ":" / "&" / "+" / "$" No .-* characters are allowed in the paramchar. But at least that's not causing me any problems. -- Sincerely, Andrew Pogrebennyk

3 6

Compiling Just One Module
by Amit Nepal 07 Feb '11

07 Feb '11

Could someone guide me how to compile and install just one module with flavor set to kamailio ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 4

authentication is not working
by Danny Dias 06 Feb '11

06 Feb '11

Hello my friends, I'm trying to configure authentication on my Kamailio and is not working at all :( I've added the following to the script to make it work: (but it doesn't) ... loadmodule "auth.so" loadmodule "auth_db.so" ... modparam("usrloc", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("auth_db", "load_credentials", "") ... if (!(method=="REGISTER") && from_uri==myself) /*no multidomain version*/ { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!db_check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); } ... if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } ## if (!db_check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; } if (!save("location")) sl_reply_error(); exit; } But is not working at all...take a look: # U 2011/02/03 09:31:04.402891 172.30.140.22:48752 -> 172.30.140.8:5060 REGISTER sip:172.30.140.8 SIP/2.0 Via: SIP/2.0/UDP 172.30.140.22:48752 ;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport Max-Forwards: 70 Contact: <sip:1000@172.30.140.22:48752;rinstance=fcade2df86ce0ab8> To: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8>> From: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8>>;tag=cd3e2323 Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 # U 2011/02/03 09:31:04.404039 172.30.140.8:5060 -> 172.30.140.22:48752 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.140.22:48752 ;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport=48752 To: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8> >;tag=c97b4d1cb1f3d0da549e06a8d482ef63.34dc From: "1000"<sip:1000@172.30.140.8 <sip%3A1000(a)172.30.140.8>>;tag=cd3e2323 Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc. CSeq: 1 REGISTER Contact: <sip:1000@172.30.140.22:48752 ;rinstance=fcade2df86ce0ab8>;expires=3600 Content-Length: 0 Am i missing something in my configuration? Thanks in advance!!!

5 12

503 Server error response for REGISTER request (Asterisk v1.6.0.5)
by Vikram Ragukumar 04 Feb '11

04 Feb '11

Hello, I had posted this question earlier on asterisk-users, but didn't receive any reply, so I'm posting on sr-users in the hope that someone can provide guidance on how to debug this problem. Our setup involves a Sip softphone registering with a fresh install of Asterisk v1.6.0.5 through an outbound proxy (Kamailio v3.1.1). We observe that about 80% of the time, immediately after startup of the softphone, when the softphone sends a REGISTER message, Asterisk replies with a "503 Server error". However, subsequent REGISTER messages are replied to appropriately (with a 401 Unauthorized). There are very few devices (under 5) registered with Asterisk, so this is not an overload issue. Please find attached a SIP trace for device registration. - What might cause this behavior in Asterisk ? - Are there any config file parameters that require to be changed ? - Would it help to see any other specific logs, if so which ones ? Thanks and Regards, Vikram.

2 3

SIP Router 3.03 topoh
by dotnetdub 04 Feb '11

04 Feb '11

Hi All, Kamailio had just started to get SIP attacked and it looked like they were sending a bad header and Kamailio died.. Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [R-MAIN =>] Request method OPTIONS is to our domain (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3378]: INFO: <script>: [ROUTE-1] Received registration from sip:1368250498@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3378]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3381]: INFO: <script>: [ROUTE-1] Received registration from sip:100@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3381]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3374]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.401446] kamailio[3374]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3373]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.636774] kamailio[3373]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3388]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.680305] kamailio[3388]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3390]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.710444] kamailio[3390]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [ROUTE-1] Received registration from sip:2478223614@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3383]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3378]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.753334] kamailio[3378]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3379]: INFO: <script>: [ROUTE-1] Received registration from sip:noauth@sipproxy (212.124.123.170) Nov 20 19:48:51 sipproxy sip[3379]: INFO: <script>: [ROUTE-1 !] Credentials invalid; issuing challenge Nov 20 19:48:51 sipproxy sip[3375]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.768137] kamailio[3375]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3383]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.775628] kamailio[3383]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3386]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.790925] kamailio[3386]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3379]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.798926] kamailio[3379]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:51 sipproxy sip[3381]: INFO: <core> [parser/msg_parser.c:339]: ERROR: bad header field [CSeq: 1 REGISTER ACK] Nov 20 19:48:51 sipproxy kernel: [1033573.815493] kamailio[3381]: segfault at 18 ip b70070d9 sp bf916118 error 4 in topoh.so[b7004000+c000] Nov 20 19:48:54 sipproxy sip[3344]: INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD Nov 20 19:48:54 sipproxy sip[3399]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3396]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3400]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3392]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3401]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3394]: INFO: <core> [main.c:788]: INFO: signal 15 received Nov 20 19:48:54 sipproxy sip[3402]: INFO: <core> [main.c:788]: INFO: signal 15 received Thanks Brian

4 20

dispatcher - ds_mark_dst("i"); is inactive only few seconds
by Thomas Baumann 04 Feb '11

04 Feb '11

Dear sr-users, I use this straight forward dispatcher config. In the failure route I am calling this function: ds_mark_dst("i"); Also this action is logged: Feb 2 17:20:31 node5 /openser/sbin/kamailio[15945]: INFO: : FAIL_ONE:time [Wed Feb 2 17:20:31 2011] destination sip:10.12.19.31:5060 marked as inactive The gateway is inactive: [root@node5 ~]# kamctl 'fifo' ds_list URI:: sip:10.12.19.31:5060 flag=I priority=0 attrs=test1=123;test2=456 URI:: sip:10.12.19.21:5060 flag=A priority=0 attrs=test1=123;test2=456 Few seconds later the gateway is active again (there is still now response to SIP-OPTIONS) [root@node5 ~]# kamctl 'fifo' ds_list URI:: sip:10.12.19.31:5060 flag=A priority=0 attrs=test1=123;test2=456 URI:: sip:10.12.19.21:5060 flag=A priority=0 attrs=test1=123;test2=456 Does anybody have a idea why ? regards, Thomas DB: mysql> select * from dispatcher -> ; +----+-------+----------------------+-------+----------+---------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+----------------------+-------+----------+---------------------+-------------+ | 1 | 1 | sip:10.12.19.21:5060 | 0 | 0 | test1=123;test2=456 | AS 1 | | 2 | 1 | sip:10.12.19.31:5060 | 0 | 0 | test1=123;test2=456 | AS 2 | +----+-------+----------------------+-------+----------+---------------------+-------------+ CONFIG: #!KAMAILIO ####### Defined Values ######### #!define WITH_MYSQL #!define WITH_IPAUTH # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@10.12.18.51/openser" #!endif debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="m-lab-ca805-sig.kd-lab.de:5060" alias="m-lab-ca805-sig.kd-lab.de" alias="10.12.19.51" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:10.12.19.51:5060 listen=tcp:10.12.19.51:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 ####### Modules Section ######## # set paths to location of modules mpath="/usr/local/kamailio-3.1/lib/kamailio/modules_k/:/usr/local/kamailio-3.1/lib/kamailio/modules/" # ------------------ module loading ---------------------------------- loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dialplan.so" loadmodule "avpops.so" loadmodule "dispatcher.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- dispatcher params ----- #modparam("dispatcher", "list_file", "/openser/etc/kamailio/dispatcher.list") modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:lb1@m-lab-ca805-sig.kd-lab.de") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; #!ifdef WITH_IPAUTH if (!allow_source_address()) { sl_send_reply("403", "Forbidden"); exit; } #!endif # Send to application servers for number evaluation if ($rU=~"^0[1-9][0-9]{5,20}$") { ds_select_dst("1","0"); if (method=="INVITE") { t_on_failure("FAIL_ONE"); } t_relay(); exit(); } sl_send_reply("403", "Forbidden"); exit; } # Sample failure route failure_route[FAIL_ONE] { if (t_check_status("408")) { xlog("L_INFO", "FAIL_ONE:time [$Tf] destination $avp(dsdst) marked as inactive \n"); ds_mark_dst("i"); } exit; } ___________________________________________________________ NEU: FreePhone - kostenlos mobil telefonieren und surfen! Jetzt informieren: http://produkte.web.de/go/webdefreephone

2 5

Reloading a module without restart
by Stagg Shelton 04 Feb '11

04 Feb '11

Is it possible to reload a module such as siptrace without requiring a full restart of kamailio. I have changed my db_url modparam for the siptrace module and would like to avoid restarting kamailio if possible. Thanks Stagg

3 3

Problem with ser
by Asuncion Merino Rodriguez 02 Feb '11

02 Feb '11

Hello, i have insalled a ser server sucessfully. To try it, we was registered by a softphone, but when i am calling to the another registered user, he hang the phone but we cant ear anything. Can somebody help me? Thanks

1 0

Redundancy between 2 Kamailio servers
by Danny Dias 02 Feb '11

02 Feb '11

Hello my friends, I wonder if someone has done this before? i would like to implement 2 Kamailio servers with redundancy, something like the following: SERVER_A is working as the primary sip proxy (virtual IP as the sip signaling), if it fails, the other server (sleeping) should UP the virtual IP and takes all the traffic....i guess there is many Linux implementations thah can do this, but i would like to know if there is someone that has done this before and hear some recomendations... I've read some difficulty in the synchronisation of registrations because Kamailio works best when it stores registrations in memory and registrations are constantly changing - they expire and are renewed, as well as new ones joining and old ones leaving. To make the failover solution function seamlessly, it is necessary to synchronise the in-memory registrations between the primary and the backup server . This can be done by forking a copy of the registration request to the backup server, but there are some practical problems in doing this, has anyone do something with this? Thanks in advance!

6 21

Weird IPv6 Registration Issue
by Jon Farmer 02 Feb '11

02 Feb '11

Hi I am experimenting with a dual homed (IPv4 and IPv6) Openser 1.1.0-tls server. If I register using a IPv6 address then the initial registration works and the details are saved in the location table. However when the UA trys to re-register on expiry the SIP returns a 200 OK but the location data is not updated in the table. Looking at the syslog it is reporting that it didn't save the data as there was an existing entry on index with key 1. This is referring I presume to the primary index on "username, domain, contact) on the location table. So what happens is the first location entry expires and gets deleted from the table and thus there is no record for the subsequent register and thus openser thinks the UA has dropped off until the next re-register when because there is no longer a location record the data gets saved correctly. This doesn't happen if I register the same UA on IPv4 on the same openser server. Anyone got any ideas why this is happening? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 2

ACC error | failed to insert into database
by alex pappas 02 Feb '11

02 Feb '11

Hi all, Hve anyone seen before the following error? *Kamailio acc [acc.c:398]: failed to insert into database* After a Kamailio restart it is ok but it start again afetr x time. Thank you Alex

2 2

Re: [SR-Users] t_local_replied() ?
by Bernhard Suttner 01 Feb '11

01 Feb '11

argh. I just mixed up the true/false return values in the eMail but you still got it :-) Thanks a lot! ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Tue, 01 Feb 2011 20:31:17 +0100 Subject: Re: [SR-Users] t_local_replied() ? > Hello, > > not sure if you ask about the options, or you tried them and don't give > you the needed feature, since there are some improper true/false return > codes in your email. t_branch_replied() will return false if the 408 is > generated locally. > > Cheers, > Daniel > > On 2/1/11 8:25 PM, Bernhard Suttner wrote: > > Hi, > > > > just to be sure: > > > > - If the gateway does send back a 100 Trying and then a 408 is detected > within failure_route the method t_branch_replied does return false (means: > gateway is up and running) - dont go to next gateway (dispatcher) > > > > - If the gateway does not respond and a 408 is detected within > failure_route (= 408 was generated from kamailio) t_branch_replied does > return true (means: gateway is down) - go to the next gateway (dispatcher) > > > > Is that correct or am I wrong? > > > > Best regards, > > Bernhard > > > > > > > > ----- Original Message ----- > > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > > To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] > > Cc: sr-users(a)lists.sip-router.org > > Sent: Tue, 01 Feb 2011 13:30:54 +0100 > > Subject: Re: [SR-Users] t_local_replied() ? > > > > > >> Hello, > >> > >> On 2/1/11 12:08 PM, Bernhard Suttner wrote: > >>> Hi, > >>> > >>> I am using the dispatcher module and want to check within the > >> failure_route if the 408 was internally generated from kamailio or it was > >> received from the dispatcher gateway. > >>> There was previously a function called t_local_replied() in the > TM-module > >> but I could not find this function in the current documentation. Was it > >> removed? > >>> Is there a alternative to check if the 408 was local generated or if it > >> was received from the peer (= from the dispatcher gateway)? > >> see the example of: > >> http://kamailio.org/docs/modules/stable/modules/tm.html#t_branch_replied > >> > >> Cheers, > >> Daniel > >> > >> -- > >> Daniel-Constantin Mierla > >> http://www.asipto.com > >> > >> > > _______________________________________________ > > 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 > >

1 0

Re: [SR-Users] t_local_replied() ?
by Bernhard Suttner 01 Feb '11

01 Feb '11

Hi, just to be sure: - If the gateway does send back a 100 Trying and then a 408 is detected within failure_route the method t_branch_replied does return false (means: gateway is up and running) - dont go to next gateway (dispatcher) - If the gateway does not respond and a 408 is detected within failure_route (= 408 was generated from kamailio) t_branch_replied does return true (means: gateway is down) - go to the next gateway (dispatcher) Is that correct or am I wrong? Best regards, Bernhard ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Tue, 01 Feb 2011 13:30:54 +0100 Subject: Re: [SR-Users] t_local_replied() ? > Hello, > > On 2/1/11 12:08 PM, Bernhard Suttner wrote: > > Hi, > > > > I am using the dispatcher module and want to check within the > failure_route if the 408 was internally generated from kamailio or it was > received from the dispatcher gateway. > > > > There was previously a function called t_local_replied() in the TM-module > but I could not find this function in the current documentation. Was it > removed? > > > > Is there a alternative to check if the 408 was local generated or if it > was received from the peer (= from the dispatcher gateway)? > see the example of: > http://kamailio.org/docs/modules/stable/modules/tm.html#t_branch_replied > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

2 1

Radius and kamailio: avp unknown attribute
by Dominguez Jover, Ricardo 01 Feb '11

01 Feb '11

Hi everybody, I have a problem with Kamalio and Radius, when a try to authenticate a user I get the following: Feb 1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]: rc_avpair_new: unknown attribute 5 Feb 1 11:50:48 kamailiodes /usr/local/kamailio-3.1/sbin/kamailio[5941]: ERROR: auth_radius [sterman.c:412]: authorization failed In fact I can't see any packet sended to the Radius Server from Kamailio. I've read several posts and I've chmod of /var/run/radius.seq, include dictionary.sip in dictionary.radius and so on. Changing "if (!radius_www_authorize("$td"))" with " if (!radius_www_authorize(""))" I don't get any error but and can't neither see any packet sended to the Radius Server. I've checked the connection between radius client and radius server and it's OK, I get: Received response ID 91, code 2, length = 211 Any help would be appreciated. Kind regards, Ricardo Dominguez

2 2

Trunk connection asterisk-kamailio
by Mino Haluz 01 Feb '11

01 Feb '11

Hi, I have a question about kamailio-asterik interconnection. I'd like to connect 1000 numbers with a trunk, but it would be painful to add 1000 trunks on asterisk. Do you have some idea how could I group those numbers into one trunk connected to kamailio? asterisk and kamailio configuration settings. Thank you very much.

2 1

t_local_replied() ?
by Bernhard Suttner 01 Feb '11

01 Feb '11

Hi, I am using the dispatcher module and want to check within the failure_route if the 408 was internally generated from kamailio or it was received from the dispatcher gateway. There was previously a function called t_local_replied() in the TM-module but I could not find this function in the current documentation. Was it removed? Is there a alternative to check if the 408 was local generated or if it was received from the peer (= from the dispatcher gateway)? Thanks in advance! Best regards, Bernhard

2 1

kamailio and sip-router at FOSDEM 2011 - social event
by Henning Westerholt 31 Jan '11

31 Jan '11

Hello, in about two and a half week the annual open source developer conference FOSDEM (http://fosdem.org/2011/) takes place in Brussels. As in the last two years we would like to meet here for a social event, probably a dinner on Saturday evening, 6th February. This developers already confirmed their participance: - Daniel-Constantin Mierla - Marius Zbihlei - Henning Westerholt This year we'll also be present in the Open Source Telephony room on sunday with two talks, "Scaling location services in large SIP networks with Kamailio from Marius and myself, and "Unifying SIP and Web worlds with Lua (Kamailio)" from Daniel. Olivier Taylor agreed helping us again in choosing a nice place for the evening. Participation at the event is free of charge, everybody pays for its own expenses. In order to organize a restaurant and do a reservation please send me a short mail if you would like to attend. Looking forward to see you in Brussels! Best regards, Henning P.S.: Photos of the 2009 event can be found on the web site: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Karlsruhe, Germany

2 3

xlogl and pseudo variables
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, I am using xlogl for debugging and logging the most important actions within the kamailio script. Does it take much resources if I use the pseudo variables with the xlogl statements? Example: xlogl("L_NOTICE", "Route $rm to destination! F=$fu T=$tu IP=$si \n"); I´m planning to do this in all the xlogl statements to be able to debug / "see whats going on" on the system. Thanks in advance. Best regard, Bernhard

2 1

Re: [SR-Users] onreply_route and failure_route
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, that was _really_ fast. Thanks. :-) One more question: Could I somehow test with the route[NAT] for example, if the request was from the main route { ... } or from the onreply_route[] ? Best regards, Bernhard ----- Original Message ----- From: Alex Balashov [mailto:abalashov@evaristesys.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org [mailto:sr-users@lists.sip-router.org] Sent: Sat, 29 Jan 2011 21:37:34 +0100 Subject: Re: [SR-Users] onreply_route and failure_route > Yes. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Jan 29, 2011, at 3:35 PM, "Bernhard Suttner" <bernhard.suttner(a)winet.ch > > wrote: > > > Hi, > > > > is it possible to call a route within onreply_ and failure_route? I > > mean, within the main route { ... } its possible to call the route > > [NAT]. Is this also possible for this "special" routes? If this is > > possible, is it also allowed to call a route which will normally > > called from the main route { ... } > > > > Thanks in advance. > > > > Best regards, > > Bernhard > > > > _______________________________________________ > > 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

onreply_route and failure_route
by Bernhard Suttner 29 Jan '11

29 Jan '11

Hi, is it possible to call a route within onreply_ and failure_route? I mean, within the main route { ... } its possible to call the route[NAT]. Is this also possible for this "special" routes? If this is possible, is it also allowed to call a route which will normally called from the main route { ... } Thanks in advance. Best regards, Bernhard

2 1

Joining strings
by Lee Archer 29 Jan '11

29 Jan '11

Hi, is it possible to join strings? I'd like to prepend a number to a string prior to processing it to an int? Thanks Lee

2 2

Can you use a bigint with Kamailio?
by Lee Archer 29 Jan '11

29 Jan '11

Hi, I am trying to run dispatcher based on the inbound number called. This works fine for some numbers but not all. Is there a bigint string function as this would solve my problems? The numbers I am receiving are 14 digits long which s.int is returning as something completely different to what it should be. Thanks Lee

2 2

LCR next_gw() - certain GW, source from other IP address
by Graham Wooden 29 Jan '11

29 Jan '11

Hi there, I have a requirement now that for one of my GW¹s in my LCR, needs to be sourced from another IP address (two different rate-decks with the carrier). This ratedeck will always be at least 2nd inline ... I was thinking something along the sorts ... Some pseudo code: failure_route [2] { . . if (next_gw()) { if ( name = 'gw_name') { force_send_socket(udp:ip.address.of.certain.ip:5060); } else { # just dummy, as we will send out our default IP for everyone else. } t_on_failure("2"); route(19); } . . . } What variable will have the value that's in the gw_name column? I can't go by the IP address in $ru because the destination IP is the same between the two ratedecks. Overall, Is that the best way to handle this requirement? Thanks all, -graham

3 2

Re: [SR-Users] LCR next_gw() - certain GW, source from other IP address
by GP Wooden 29 Jan '11

29 Jan '11

Same destination IP for both rate decks, so I need to go by something else, like the gw_name. ----- Reply message ----- From: "Stagg Shelton" <stagg(a)vocalcloud.com> Date: Sat, Jan 29, 2011 8:27 am Subject: [SR-Users] LCR next_gw() - certain GW, source from other IP address To: <sr-users(a)lists.sip-router.org> Do the provider gateways have the same IP address for the different rate plans? If the gateways are different then the following may be relevant. if (next_gw()) { # prepare for lcr failover t_on_failure("2"); if($rd=="aaa.aaa.aaa.aaa") { xlog ("L_INFO", "This is a request for Some Rate Plan"); force_send_socket(udp:xxx.xxx.xxx.xxx:5060); } else { xlog ("L_INFO", "This is a request for another rate plan"); force_send_socket(udp:yyy.yyy.yyy.yyy:5060); }; route(2); } else { xlog("No gateways found!"); Stagg On 1/29/11 9:10 AM, Graham Wooden wrote: > Hi there, > > I have a requirement now that for one of my GW¹s in my LCR, needs to be > sourced from another IP address (two different rate-decks with the carrier). > This ratedeck will always be at least 2nd inline ... > > I was thinking something along the sorts ... Some pseudo code: > > failure_route [2] { > . > . > if (next_gw()) { > if ( name = 'gw_name') { > force_send_socket(udp:ip.address.of.certain.ip:5060); > } else { > # just dummy, as we will send out our default IP for everyone > else. > } > t_on_failure("2"); > route(19); > } > . > . > . > } > > What variable will have the value that's in the gw_name column? I can't go > by the IP address in $ru because the destination IP is the same between the > two ratedecks. > > Overall, Is that the best way to handle this requirement? > > Thanks all, > > -graham > > > > _______________________________________________ > 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 _______________________________________________ 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

3 4

Kamailio Devel Training, Barcelona - update
by Daniel-Constantin Mierla 28 Jan '11

28 Jan '11

Hello, a quick update about the next training in Barcelona, Feb 10-11, 2011, focusing on Kamailio development. We got a bigger room (for second time :-) ), so we increased the capacity to 28 seats. We are already 23 people in the class and several more joining the dinner Thursday evening, so there is going to be a great time. If you plan to come, register asap, there is no time to get a bigger room, so once we are full booked this time, we close the participants list. You can see more details at: http://www.kamailio.org/w/2011/01/kamailio-development-training-barcelona-f… Also, do not forget Fosdem next weekend, it is a bunch of us going there, two talks about Kamailio, come and say hi: http://www.kamailio.org/w/2011/01/social-networking-event-brussels-belgium/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Kamailio as a routing engine
by Gang Liu 28 Jan '11

28 Jan '11

Hello, I wrote a H.323/SIP IWF program before, it was based on B2BUA framework. Currently I am planning to use Kamailio as internal routing server and let IWF become a class 4 soft switch.That is Inbound ---- H.323/ SIP -- > ---- SIP routing request-----> Kamailio IWF/SBC | Outbound <--- H.323/SIP -- <--- SIP ----------------------------- Is it ok to store all voip provider's information at database of Kamailio and pass all to IWF at private SIP headers and let IWF as a outbound proxy? Is Kamailio as a SIP redirect server or it is better to stay at call signaling path until call session ended? It is great to share any information about this. regards, Gang

2 1

Load_gw failed
by Thiago Maluf 27 Jan '11

27 Jan '11

Hi Kamailio Users, I'm trying to use LCR module in Kamailio 3.1.1 and I', having some problems with this function (load_gws). I'm my kamailio.cfg , I have this route: # SEARCH DEST route[NUMBER] { xlog("L_ALERT", "NUMBER IS $rU \n"); if ($rU=~"^55211020[0-9]{4}$") { # IS MY USERS route(LDAP_NUM2USER); } else { # NOT IS MY USER if (!load_gws(1,$rU)) { xlog("L_ALERT", "LOAD LOCAL GW FAILED $rU \n"); route(DSER); return; } else { xlog("L_ALERT", "LOAD GW DEBUG $avp(i:709)\n"); } ; if (!next_gw()) { xlog("L_ALERT", "NEXT LOCAL GW FAILED\n"); sl_send_reply("503", "Service not available - No gateways"); return; }; setflag(29); route(RELAY); return; }; }; When my call arrive to load_gws, I have this problem: (2198) ERROR: <core> [action.c:1158]: runtime fixup failed for load_gws param 2 (2198) ERROR: <core> [action.c:1511]: run action error at: /etc/kamailio/kamailio.cfg:472 I already saw the Kamailio Code and this error doesn't make sense. Would somebody Help me with it? Very thanks, Thiago Maluf. -- ---------------------------------------------------------------- Thiago Maluf Resende Tel: +55 21 9700-9113 e-mail: malufrj(a)gmail.com

2 1

SIP Recorder
by Danny Dias 27 Jan '11

27 Jan '11

Hello my friends, I have a requeriment, which indicates that i have to record every SIP conversation between peers (also for callings to the PSTN); the recording server will be built for our company following this requeriments (also requested for the client): My doubt is: How can i handle sip conversations recording when all the calls are passing through a Proxy Server? I do understand that the media is always peer to peer and the signaling goes through the Proxy, but in this case the media not only has to pass between the peers because it must be recorded. How should i handle this?

7 19

Kamailio CFG file size ??
by Jijo 27 Jan '11

27 Jan '11

Hi All, is there any limitation on the size of kamailo.cfg file. When i tried to add few avp_subst, then the kamailio starts showing up the error out of package memory on receiving SIP message. If i remove those lines then everything works fine.. avp_subst is simple one. Our kamailio cfg file is quite big.. Please let me know how to debug this issue? Here is the error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_row.c:116]: no private memory left 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:293]: could not allocate row 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:258]: failed to convert row #1 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:75]: failed to convert rows 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_dbase.c:416]: error while converting result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_query.c:137]: error while storing result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: avpops [avpops_db.c:333]: cannot do the query 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: maxfwd [mf_funcs.c:62]: parsing MAX_FORWARD header failed! 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: rr [loose.c:98]: failed to parse headers 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: pv [pv_core.c:1761]: do action failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [msg_translator.c:2229]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: rr [loose.c:70]: failed to parse To header field 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1740]: do action failed) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1381]: error parsing headers 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [data_lump.c:318]: ERROR: del_lump: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: textops [textops.c:1081]: no memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [backLinkMode] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_shv.c:367]: no more pkg mem 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:1257]: osb_get_osv_current_binding(): no binding available 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <script>: R1 - binding IP=0 not available

2 3

NAT Problem in Asterisk Realtime + Kamailio config
by andrzej.ciupek＠asterisk.edu.pl 27 Jan '11

27 Jan '11

Hello I have started to use configuration like: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb I works fine, but I have problem with dealing with NAT. kamctl ul show gives me: AOR:: andrzej_104 Contact:: sip:andrzej_104@10.5.0.94:14368;rinstance=9c2b256a13210aa7 Q= Expires:: 46 Callid:: ZDUwYzIzMDNiZTI3MjkyNmFmOWEyOTI2ODcyNjVmNDg. Cseq:: 60 User-agent:: eyeBeam release 1102q stamp 51814 State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 5087 AOR:: andrzej_103 Contact:: sip:andrzej_103@10.5.0.165:5060 Q= Expires:: 926 Callid:: 11631436-5060-1(a)10.5.0.165 Cseq:: 2257 User-agent:: Grandstream HT-502 V1.2A 1.0.3.10 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 6879 So when I have incomming call from Asterisk to Kamailio, Kamailio start to send call to LAN IP. I have tried to use help from: http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-… But I have never get: Received:: and Cflag:: 192 it is always Cflag:: 0. field in my kamctl ul show. In my configuration Asterisk is Callweaver and is on the other server. I use rtpproxy-1.2.1 and kamailio-3.0.1 and mofified config from link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Please help.

2 2

RLS question
by "Andrés S. García Ruiz" 27 Jan '11

27 Jan '11

Hi, I'm trying to create a RLS "rls-services" document with a contacts-list inside it. Kamailio states that "No xcap document found for uri:x". When creating a subscribe I must set the uri to the service uri. What I want to know is the sql query Kamailio does in order to get the contacts list for that uri. For example: XCAP document: "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" + "<rls-services xmlns=\"urn:ietf:params:xml:ns:rls-services\"" + " xmlns:rl=\"urn:ietf:params:xml:ns:resource-lists\">" + "<service uri=\"sip:restricted_areas_presentities@open-ims.test\">" + "<list name=\"friends\">" + "<rl:entry uri=\"sip:testuser01@open-ims.test\"/>" + "<rl:entry uri=\"sip:testuser02@open-ims.test\"/>" + "</list>" + "<packages>" + "<package>presence</package>" + "</packages>" + "</service>" + "</rls-services>"; Is stored in /xcap-root/rls-services/users/sip:restricted_areas@open-ims.test/index The subscribe is from sip:restricted_areas@open-ims.test to sip:restricted_areas_presentities@open-ims.test What am I doing wrong? Thanks, Andrés.

3 5

[kamailio-users] kamailio on amazon
by Chandrakant Solanki 26 Jan '11

26 Jan '11

Hi I have installed Kamailio with MediaProxy and asterisk on Amazon Server.. While kamailio/MediaProxy and Asterisk both running on different amazon's instance. Kamailio : 1.5.0-notls MediaProxy : 2.3.8 Asterisk : 1.6.2.6 Firewall port is opened for mediaproxy from 10000-20000 (UDP), 5060 (UDP) etc on both amazon machine.. while I tried to play an audio... it plays sound file but unable to found audio on device. when I put tcpdump on asterisk machine it gives following error... # tcpdump -i eth0 udp portrange 10000-20000 -w test1.pcap 01:39:52.850279 IP (tos 0xb8, ttl 64, id 59, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 3cfd!] UDP, length 32 01:39:52.870279 IP (tos 0xb8, ttl 64, id 60, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 97fe!] UDP, length 32 01:39:52.890281 IP (tos 0xb8, ttl 64, id 61, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 75c0!] UDP, length 32 Any Idea..!! -- Regards, Chandrakant Solanki

3 2

Re: [SR-Users] fix_nated_sdp()
by Bernhard Suttner 25 Jan '11

25 Jan '11

Hi, thanks for your response. I think I understand the scenario very good - but im not so familiar with the fix_nated_sdp() functionality. Currently the SDP will be modified with the source IP of the message. This does work good but currently I only do that for some special devices (matching the User-Agent / Server string). I want to do that fix_nated_sdp() in all cases and not only for some special devices. The FreeSWITCH RTP-Auto Adjustment does work in nearly every case but in one (currently known) scenario it does not. Maybe I should just give it a try to do fix_nated_sdp() in combination with nat_uac_test (RFC1918 private address in SDP). Thanks a lot for this discussion. Best Regards, Bernhard ----- Original Message ----- From: Ovidiu Sas [mailto:osas@voipembedded.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: Carsten Bock [mailto:lists@bock.info], sr-users(a)lists.sip-router.org Sent: Tue, 25 Jan 2011 20:38:14 +0100 Subject: Re: [SR-Users] fix_nated_sdp() > You need to experiment with nathelper module and while doing that, you > need to inspect the SIP messages in order to understand how it works. > Only when you fully understand your scenarios, you will be able to > properly configure, debg and setup kamailio. > > Best thing to do here is to use rtpproxy and relay media if needed. > > Another option would be to let the SDP untouched and let freeswitch do > it's magic (if it can). Not sure if this will work in all scenarios. > > Calling fix_nated_sdp() on the SBC will definitely not work (you are > just messing with the IP on the SDP and you don't know if the real RTP > port on the NATed side will match the advertised port in your "fixed" > SDP). > > > Regards, > Ovidiu Sas > > On Tue, Jan 25, 2011 at 1:23 PM, Bernhard Suttner > <bernhard.suttner(a)winet.ch> wrote: > > Hi, > > > > thanks for your answer. > > > > The question was about, if the fix_nated_sdp() (in the example below) > could maybe break something. I am sure, that the fix_nated_sdp() would work > in "some" cases but I am not sure, if the function could break something: > > > > > > if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) > > > > and within the onreply route: > > > > if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) > fix_nated_sdp(3) > > > > > > Or is there a better solution to fix these damend NAT scenarios? > > > > Best regards, > > Bernhard > > > > > > -----Ursprüngliche Nachricht----- > > Von: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Im > Auftrag von Carsten Bock > > Gesendet: Dienstag, 25. Januar 2011 18:56 > > An: Bernhard Suttner > > Cc: sr-users(a)lists.sip-router.org > > Betreff: Re: [SR-Users] fix_nated_sdp() > > > > Hi Bernhard, > > > > have you checked the SDP which is going to and from the > > Freeswitch-Server? Is it modified? > > > > If yes: You may want to check your NAT-Firewall. A working (but not > > nice) solution might be to relay the RTP through an proxy in this > > case... > > If No: Verify the result of "nat_uac_test": Why is it returning "false"? > > > > Carsten > > > > 2011/1/25 Bernhard Suttner <bernhard.suttner(a)winet.ch>: > >> Hi, > >> > >> someone has an idea? > >> > >> Thanks in advance! > >> > >> Best regards, > >> Bernhard > >> > >> -----Ursprüngliche Nachricht----- > >> Von: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Bernhard > Suttner > >> Gesendet: Montag, 24. Januar 2011 20:38 > >> An: sr-users(a)lists.sip-router.org > >> Betreff: [SR-Users] fix_nated_sdp() > >> > >> Hi, > >> > >> I have a question about the fix_nated_sdp() function. Different devices > (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() > the message to a pool of FreeSWITCH Servers. > >> > >> The scenario does work quite well but: > >> > >> if a call is initiated from one of the freeswitch server, pass it through > kamailio to the PBX and the PBX does have a forwarding back to kamailio and > in the end to freeswitch I have no audio because the PBX is behind a NAT and > uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. > Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check > if IP in incoming packet is different to the IP in the SDP) to detect the > real IP address. > >> > >> Therefore I have to use fix_nated_sdp(). I am not sure, what this > function could break therefore I want to be very sure and use it only, if > really necessary. Test which User-Agent or Server the peer (in this example > the PBX) has, is not really possible because some devices does not send this > header. > >> > >> What do you think about the follwoing pseudo code: > >> > >> if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) > >> > >> and within the onreply route: > >> > >> if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) > fix_nated_sdp(3) > >> > >> Somebody has a better idea to handle this? What could go wrong? > >> > >> Thanks in advance. > >> > >> Best regards, > >> Bernhard Suttner > >> > >> _______________________________________________ > >> 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 > >> > >> > >> _______________________________________________ > >> 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 > > Schomburgstr. 80 > > 22767 Hamburg > > Germany > > > > Mobile +49 179 2021244 > > Home +49 40 34927217 > > Büro (Verl) +49 5246 801427 > > Fax +49 40 34927218 > > mailto:carsten@bock.info > > > > > > _______________________________________________ > > 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

fix_nated_sdp()
by Bernhard Suttner 25 Jan '11

25 Jan '11

Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner

3 4

accounting serial forked transactions with 302 from LCR
by thrillerbee 25 Jan '11

25 Jan '11

I'm converting my OpenSIPS routers to Kamailio & have run into a small complication. The proxy pushes all INVITEs to a least-cost router. This LCR responds with a list of routes as contact instances in a 302 Redirect. Calls are routing a serially forking normally. Connected & failed calls account normally. However, if the caller cancels the call, the acc module includes the 302 in the transaction record as the final response as opposed to the actual final response - the 487 Request Canceled. Is there something I could be missing that would cause this? Thanks, Ryan

2 4

Re: [SR-Users] $ua in on_reply route not set?
by Bernhard Suttner 25 Jan '11

25 Jan '11

Hi, found the problem. The device does sometimes use User-Agent and sometimes Server. Is it better to use $hdr() or the Search() function? I use this to check for the User-Agent and then to do a "fix_nated_sdp()" (in route[] and onreply[]) because I am not really sure, if the fix_nated_sdp() could break something. Or should kamailio break nothing here? Sometimes the User-Agent/Server is missing in Session-Progress 183. Therefore a "global" fix_nated_sdp() would be nice to have. Best regards, Bernhard ----- Original Message ----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Mon, 24 Jan 2011 17:09:12 +0100 Subject: Re: [SR-Users] $ua in on_reply route not set? > Hello, > > On 1/24/11 5:03 PM, Bernhard Suttner wrote: > > Hi, > > > > could it be, that the $ua pseudo variable is not set within in a onreply > route? (Version 3.1). > no, should be set, there was no change in this regard for quite long > time. Can you sent the sip reply plus log with debug=3? > > > > What is the best alternative for that? Search()? > > The alternative is $hdr(User-Agent) which is practically returning the > same value as $ua, using generic header search mechanism. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

2 1

Re: [SR-Users] pdb module best practice
by Henning Westerholt 25 Jan '11

25 Jan '11

On Monday 24 January 2011, Thomas Baumann wrote: > the downtime is not a problem, I already have 2 deamons running. > I want to reduce the time , how changes become active. > > 1. update xxx entry > 2. export table to *.csv (5s) > 3. generate *.mmap file (25 seconds) > > So changes will take at lease 30 seconds to become active. Only 20% of my > numbers are updated very often, I want to export them and the generation > of the *.mmap file takes only 5 seconds. > > But in this case I need to specify the daemons per request and not in the > general parameters. Because later on it might be necessary to ask the > daemons hosting the rest of my numbers. Or maybe 30 seconds are ok ? Hi Thomas, i understand. You could of course change/ extend the interface of the pdb, but at the moment this functionality is not available. During module startup the sockets and server data structures are initialized, so the structure needs probably a bit changed. But let me ask, what kind of changes are done to your routing? Is the routing data then completely invalid, or just a more expensive route is taken for this particular target? In the latter case a update period of 30 seconds sounds not like a big deal to me, after all you probably have also some manual steps involved before you do an update. Cheers, Henning

1 0

Re: [SR-Users] SIP between IPv4 and IPv6 agents
by Komáromi Péter 25 Jan '11

25 Jan '11

Hello! I went through your letters and start as following: ------------------------------------------------------- root@ubuntu:/home/yeti/rtpproxy# /etc/init.d/kamailio restart Restarting kamailio: kamailioloading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ Listening on udp: 192.168.124.158:5060 udp: FEC0:0:0:0:0:0:0:2:5060 tcp: 192.168.124.158:5060 tcp: FEC0:0:0:0:0:0:0:2:5060 Aliases: tcp: ubuntu:5060 udp: ubuntu:5060 root@ubuntu:/home/yeti/rtpproxy# kamctl restart INFO: Stopping Kamailio : INFO: stopped INFO: Starting Kamailio : INFO: started (pid: 7597) root@ubuntu:/home/yeti/rtpproxy# rtpproxy restart ------------------------------------------------------- So I guess everything is starting fine. The users can register and call each other. Their data are in the mysql database`s location table. I put this into the kamailio.cfg file: -------------------------------------------------------- # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { if (af == inet) { if ( $(rU{s.substr,0,1}) == "6") { force_rtp_proxy("ie"); } else { force_rtp_proxy("ee"); } } else { if ( $(rU{s.substr,0,1}) == "4") { force_rtp_proxy("ei"); } else { force_rtp_proxy("ii"); } } } else if (is_method("200OK")) { if (af == inet) { if ( $(rU{s.substr,0,1}) == "6") { force_rtp_proxy("ei"); } else { force_rtp_proxy("ee"); } } else { if ( $(rU{s.substr,0,1}) == "4") { force_rtp_proxy("ie"); } else { force_rtp_proxy("ii"); } } } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } ---------------------------------------------------------- But, unfortunately there is no RTP traffic yet. :( What do you thing guys, what can be the problem? Regards, Peter Am 02.12.2010 13:06, schrieb Komáromi Péter: > Hi! > > So if you say it is possible to solve the problem with the only > location table, the location_inet4 and location_inet6 is not > certainly necessary... do I _have to_ use the 4to6.cfg file from the > source of kamailio, or not? When you call force_rtp_proxy() then you have to provide the i/e flags to tell rtpproxy how it should bridge the call. Therefore you need to know for every call: - is the caller IPv4 or IPv6? - is the callee IPv4 or IPv6? Finding out the protocol version of the caller is rather easy: if (af == inet) { # caller ipv4 ... } else { # caller ipv6 ... } But finding out if the callee is v4 or v6 is difficult. Actually it may even be that the callee is registered with 2 clients at the same time - one using v4 and v6. As far as I know this multi-registration scenario can not be handled currently - only single-registrations or multi-registrations which use the same protocol version are supported. Anyway - there are several approaches how to detect the IP version of the callee. - fixed mapping: for example if you know that all usernames starting with 1 are IPv4 clients, and all usernames starting with 2 are IPv6 clients, then you can detect the callee's protocol version manually, e.g: if ( $(rU{s.substr,0,1}) == "1") { # callee ipv4 ... } else { # callee ipv6 ... } - check destination URI if it contains a IPv4 or IPv6 address, e.g. use http://www.kamailio.org/dokuwiki/doku.php/transformations:devel#resubst_exp… to test if $dd contains '.' which would indicate a ipv4 address. - use dedicated location tables > (Because it uses them, for instance here: if (method == "REGISTER") > { if (af == inet) { save("location-inet4"); } else if (af == inet6) > { save("location-inet6");...) > > If I do not have to use it, I do not need to merge it with my > kamailio.cfg file, the steps I have to do are: install rtpproxy, > start it with the appropriate parameters and set the forcing in the > kamailio.cfg file?! The more I think about it I prefer the approach with separated location tables. It should be rather easy to integrate the 2 location tables into your existing config. regards Klaus

3 4

Problem reloading carrierroute
by Morten Isaksen 25 Jan '11

25 Jan '11

Hi, Kamailio 3.0.3. A few days ago we got a problem when "kamctl cr reload" was executed. Jan 24 08:50:27 sip-core-2 /usr/local/sbin/kamailio[1639]: ERROR: db_mysql [km_dbase.c:346]: no memory left Jan 24 08:50:27 sip-core-2 /usr/local/sbin/kamailio[1639]: ERROR: carrierroute [cr_db.c:331]: Fetching rows failed Jan 24 08:50:27 sip-core-2 /usr/local/sbin/kamailio[1639]: ERROR: carrierroute [cr_data.c:181]: could not load routing data I tried to increase the shared memory from 32M to 64M but that did not seem to help. It failed again after a few hours. The last thing I tried was to insert modparam("carrierroute", "fetch_rows", 2000) into kamailio.cfg and is waiting to see if that helped. The carrierroute table is about 16K rows. Is it private og shared memory that is missing? Any other ideas how to fix this? -- Morten Isaksen

2 3

Building specific commit
by mayamatakeshi 25 Jan '11

25 Jan '11

Hello, I am using these commands to build and install kamailio: git clone http://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.1 origin/3.1 make FLAVOUR=kamailio cfg make group_include="mysql" all make group_include="mysql" install Now, If I want to build a specific commit like a0d376d85aae2f6e46e381a01fb85c5830250d4d, what should I do? regards, takeshi

2 2

kamctl fifo <command> not responding
by Anton Roman 24 Jan '11

24 Jan '11

Hi all, I'm having trouble trying to execute fifo commands with "kamctl fifo <command>". Just after restarting Kamailio it works fine, however, sometimes after some days running it doesn't respond. kamailio1:~#* kamctl fifo which* It doesn't respond so I input *Crtl+c* and I get: /usr/local/lib/kamailio//kamctl/kamctl.fifo: line 89: /tmp/kamailio_fifo: Interrupted system call If I delete and create the fifo file again (with "rm /tmp/kamailio_fifo" and "mkfifo /tmp/kamailio_fifo" and "chmod 660 /tmp/kamailio_fifo") it keeps not responding. Any help is welcome, what can be happening? Below you can find info about the pipe and the running kamailio. Thanks in advance, Best regards Antón kamailio1:~# *ls -hall /tmp/kamailio_fifo * prw-rw---- 1 root root 0 ene 17 12:01 /tmp/kamailio_fifo After deleting and creating the fifo file again: kamailio1:~# *ls -hall /tmp/kamailio_fifo * prw-rw-r-- 1 root root 0 ene 18 10:28 /tmp/kamailio_fifo kamailio1:~#* ps -ef | grep kama* root 17369 17245 0 10:12 pts/0 00:00:00 grep kama kamailio 23277 1 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23289 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23291 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23293 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23294 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23295 23277 0 Jan15 ? 00:01:14 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23299 23277 0 Jan15 ? 00:01:13 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23300 23277 0 Jan15 ? 00:01:13 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23303 23277 0 Jan15 ? 00:01:13 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23305 23277 0 Jan15 ? 00:05:29 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23306 23277 0 Jan15 ? 00:05:33 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23309 23277 0 Jan15 ? 00:05:30 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23311 23277 0 Jan15 ? 00:05:31 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23312 23277 0 Jan15 ? 00:00:02 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23313 23277 0 Jan15 ? 00:00:39 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23315 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23320 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23321 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23322 23277 0 Jan15 ? 00:00:05 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio kamailio 23323 23277 0 Jan15 ? 00:00:00 /usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 512 -u kamailio -g kamailio

2 5

$ua in on_reply route not set?
by Bernhard Suttner 24 Jan '11

24 Jan '11

Hi, could it be, that the $ua pseudo variable is not set within in a onreply route? (Version 3.1). What is the best alternative for that? Search()? Thanks in advance! Best regards, Bernhard

2 1

Multiple Sip
by aram amin 24 Jan '11

24 Jan '11

Dear Sir / Madam I registered for your services today and the main purpose was to register my other voip/sip providers with your services so that I can have more options in making out bound calls with my sip phone, however could you please explain to me that if for instance, I have three accounts with three different voip providers (i.e JustVoip, Smartvoip,and Voipcheap) ,then how do i register them with my account that you have opened for me today. could you also tell me how is it possible to make specific calls from one of the voip providers above . Thank you Regards Aram

2 1

pdb module best practice
by Thomas Baumann 24 Jan '11

24 Jan '11

Hello, I tested the pdb module and the pdb_server. It is a fast solution even with a huge amount of numbers. But how I can update the pdb_server in a fast and effective way ? I tried it in the following way: - number;carrier are stored in a mysql table (20 million entries) - 2 pdb_servers are running - the table is exported into a csv file (time: 5 seconds) - pdbt will create a new mmappable file (time: 25 seconds) - the pdb_servers will be restarted to activate the changes The time of 30 seconds is good but not fast enought. Some of my numbers are updated very often, to export them and create a new mmappable file will take only 3 seconds. I had the idea to split my numbers in a more or less static area and a dynamic area. What I want to do later is to have the pdb_server not static configured, I want to define the server within the request: 1. query in dynamic range: pdb_query("$rU", "$avp(i:1000)", "localhost:10001,localhost:10002"); if not found look into the static area: pdb_query("$rU", "$avp(i:1000)", "localhost:10003,localhost:10004"); Does it make sence, or has somebody a better idea how changes activated within 10s or less ? regards, Thomas ___________________________________________________________ Neu: WEB.DE De-Mail - Einfach wie E-Mail, sicher wie ein Brief! Jetzt De-Mail-Adresse reservieren: https://produkte.web.de/go/demail02

2 1

pdb module best practice
by Thomas Baumann 22 Jan '11

22 Jan '11

1 0

Problem with parallel forking
by Daniel Grotti 21 Jan '11

21 Jan '11

Hi all, I'm using kamailio 3.1 and I have some problems with parallel forking. I need to implement parallel forking to different users registered on kamailio. So, when call arrives with R-URI= sip:003912345678@<IP_server>, I need to fork te call to (for example) 2 users: 1001 and 1001. To do that, I've created my usr_preferences table like this: +----+------+------------------+--------+-----------+------+-------+---------------------+ |/ id | uuid | username | domain | attribute | type | value | /last_modified | +----+------+------------------+--------+-----------+------+-------+---------------------+ |/ 1 | | /003912345678/ | | fork | 0 | 1001 | | /|/ 2 | | /003912345678/ | | fork | 0 | 1002 | | /+----+------+------------------+--------+-----------+------+-------+---------------------+ and I've added a code to my kam.cfg like this: if (is_method("INVITE")) { xlog("L_INFO", "REQUEST Invite - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(avp_db_load("$ru/username","$avp(s:fork)")) { avp_pushto("$ru/username","$avp(s:fork)/g"); } if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); exit; } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); t_relay(); } } If 1001 and 1002 are registered everything works fine (1001 Ringing and 1002 ringing). Ifonly 1001 registered everything works fine (1001 ringing and 1002 is offline.). But when 1002 is registered and 1001 in offline, kamailio try to call 1001, find that it's offline ( and I get 404 User Offline") but no call to 1002 is attempted. What's wrong ? Regards, Daniel

2 2

Too many false positives?
by Jeremya 21 Jan '11

21 Jan '11

A large bunch of recent email on this list got consigned to the spam bin by my Thunderbird. This hasn't happened before. Is there some list change that triggered this? Regards Jeremy A

2 1

Proxying INVITE message without port information in R-URI
by Vikram Ragukumar 20 Jan '11

20 Jan '11

Hello, The diagram below shows the SIP message flow in our setup when Softphone A places a call to Softphone B. ----------- ----- ---------- ----- ----------- |Softphone A|-<->-|Proxy|-<->-|Sip Server|-<->-|Proxy|-<->-|Softphone B| ----------- ----- ---------- ----- ----------- IP: a.b.c.d IP: a.b.c.d Proxies shown in the diagram above are physically the same entity. For convenience of diagrammatic representation the figure above has been unwrapped. Both Softphone A and Softphone B (hereafter referred to as A,B) are behind a NAT and are registered to the Sip Server through the same proxy (IP: a.b.c.d). Proxy runs Kamailio v3.0.1. When A places a call to B, an INVITE is proxied to the Sip Server from A. The Sip Server now sends another INVITE to the proxy, however the R-URI of this INVITE does not contain the port number that the INVITE is supposed to be sent to. Hence the proxy sends the INVITE to the default SIP port 5060. This port is not open in the firewall hence Softphone B never receives the INVITE. My questions are 1) Is it mandatory for R-URI of the INVITE to contain the port number ? 2) If not, is there something that can be added to the config script in Kamailio so that the proxy forwards the INVITE to the appropriate port? (maybe by keeping track of the rcv port of the REGISTER message that Softphone B sends...) ? Thanks and Regards, Vikram. PS : The Sip Server in the above discussion is Voipswitch.

3 10

Re: [SR-Users] Refer Using UAC.
by Daniel-Constantin Mierla 20 Jan '11

20 Jan '11

private mails are simply ignored after first advise in this regard, please CC the mailing list always. If you read the config from the tutorial, you see how the invite is relayed to asterisk. Refer should go to asterisk in the same way if it is an out of dialog request, or follow record route/contact address for within dialog requests. Cheers, Daniel On 1/20/11 11:37 AM, David J. wrote: > could you point me to the docs? > > just use forward() or rewritehostport()? > > > > On 1/20/11 5:15 AM, Daniel-Constantin Mierla wrote: >> If asterisk is in the path of the call, then just forward the REFER >> to it, there is no need to generate a new one. >> >> Also, note that REFER is many times part of a dialog, uac_req_send() >> creates requests out of the dialog. >> >> Cheers, >> Daniel >> >> On 1/16/11 11:37 PM, David J. wrote: >>> I realize that kamailio is not a b2bua; >>> But because we are using Asterisk in the path; >>> >>> To extend the Asterisk Realtime Tutorial; >>> >>> I was wondering if I could do something like this... >>> >>> Kind of like how we use UAC to send a register to Asterisk; >>> Could we do the same and modify the method to use REFER instead? >>> >>> I know it is more complex; but I am not sure where to handle this case; >>> >>> Thanks for any pointers. >>> >>> if(is_method("REFER")){ >>> $var(rip) = $sel(cfg_get.asterisk.bindip); >>> $uac_req(method)="REFER"; >>> $uac_req(ruri)="sip:" + $var(rip) + ":" + >>> $sel(cfg_get.asterisk.bindport); >>> $uac_req(furi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $ft; >>> $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $tt; >>> $uac_req(hdrs)="Contact: <sip:" + $au + "@" >>> + $sel(cfg_get.kamailio.bindip) >>> + ":" + >>> $sel(cfg_get.kamailio.bindport) + ">\r\n"; >>> if($sel(contact.expires) != $null) >>> $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + >>> $sel(contact.expires) + "\r\n"; >>> else >>> $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + >>> $hdr(Expires) + "\r\n"; >>> >>> >>> uac_req_send(); >>> >>> >>> } >>> >>> _______________________________________________ >>> 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 Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

SL local ACK found -> dropping it! error ...
by Sébastien Cramatte 20 Jan '11

20 Jan '11

Not sure how todo this SIP capture. Can you give me an hand and explain how can I get this trace. I can post my configuration if it's necesary. But I'use the default NAT setup that comes with Debian package. 2011/1/19 Alex Balashov <abalashov(a)evaristesys.com> You'd need to post the SIP capture, but it sounds like Kamailio is not > associating the ACK with any known transaction, perhaps because the ACK has > an unexpected RURI, Via branch parameter, or CSeq number. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > _______________________________________________ > 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 5

Handling call transfer in the Asterisk Realtime setup.
by David J. 20 Jan '11

20 Jan '11

I am trying to add support for call transfer in the Asterisk realtime tutorial on Asipto; I am not sure what I would have to do to get this feature working; Perhaps I have to handle "refer" messages; but I am not sure how I send that to Asterisk; Any advice would be greatly appreciated.

2 1

Refer Using UAC.
by David J. 20 Jan '11

20 Jan '11

I realize that kamailio is not a b2bua; But because we are using Asterisk in the path; To extend the Asterisk Realtime Tutorial; I was wondering if I could do something like this... Kind of like how we use UAC to send a register to Asterisk; Could we do the same and modify the method to use REFER instead? I know it is more complex; but I am not sure where to handle this case; Thanks for any pointers. if(is_method("REFER")){ $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REFER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $ft; $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ";tag=" + $tt; $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); }

2 1

Reconnection to db
by Javier Gallart 20 Jan '11

20 Jan '11

Hi list we have been testing the mtree module. We do periodical reloads of the mtree table with the "mi mt_reload" command through xmlrpc. It works fine except when the db is restarted. If we manually restart the db and then try the mt_reload commands we get this error: Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: db_postgres [km_dbase.c:152]: 0xa6a490 PQsendQuery Error: server closed the connection unexpectedly#012#011This probably means the server terminated abnormally#012#011before or while processing the request.#012 Query: select partition,prefix,route from trees order by partition Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: <core> [db_query.c:101]: error while submitting query Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: mtree [mtree_mod.c:622]: Error while querying db Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: ERROR: mtree [mtree_mod.c:738]: cannot re-load info from database Is there a way to reconnect to the db before sending that command? Thanks in advance Regards Javi

3 15

avp_db_query() question
by 侯旭光 19 Jan '11

19 Jan '11

Hello avp_db_query(query[,dest]) can get a database query and store the results in the avps. But what if the results returns many rows,and how can I get all the results? How to set the [dest] parameter ? I've tried the method describered inhttp://www.kamailio.org/docs/avp_db_query.html,but it doesn't work. like below mysql<select mem_user from tgroup where grp_name='1234'; +----------+ | mem_user | +----------+ | 1013 | | 2013 | +----------+ 2 rows in set (0.00 sec) kamailio.cfg if(avp_db_query("select mem_user from tgroup where grp_name='1234'","$avp(name)")) { xlog("L_INFO","query results[1] :$avp(name[1])\n"); xlog("L_INFO","query results[2] :$avp(name[2])\n"); } syslog <INFO> query results[1] :<null> <INFO> query results[2] :<null> version: kamailio 3.0.2 MySQL 5.0 thank you very much!

2 1

SL local ACK found -> dropping it! error ...
by Sébastien Cramatte 19 Jan '11

19 Jan '11

Hello, I'm new with Kamailio. I've just setup on Debian Squeeze using mostly the example configuration. I've been able to route inbound / outbound call on local networks W/O NAT and over mobile 3G+NAT ... Unfortunateley I've tried over 3 wifi / local + NAT and Kamailio drop call and return me this message "SL local ACK found -> dropping it! error ..." My Kamailio server has public fixed IP and listen on 5060:udp It's look and issue with NAT setup ... How can we solve it ? Best regards

3 3

Return code after fr_inv_timer hit
by Mino Haluz 18 Jan '11

18 Jan '11

Hi, I would like to force kamailio to send another code as Request timeout when fr invite timeout is hit. Is there some nice way how to achieve it, or I have to edit the code ? :( Thank you!

3 7

ACK problem with FreeSwitch-Kamailio SBC implementation.
by Ricardo Martinez 18 Jan '11

18 Jan '11

Hello. I was wondering if someone can help me with some problem that I’m having with the combination of FreeSwitch and Kamailio as SBC. I’m having and ACK not routed by kamailio and dropped. Here is the debug. 100.100.100.211:5060 : Kamailio 100.100.100.211:5090 : FreeSwitch 100.100.100.108 : Caller 100.100.100.212 : Calle Request-Line: INVITE sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211>SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Bria release 2.2 stamp 45414 Content-Length: 388 Message Body Session Description Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Server: kamailio (3.1.1 (i386/linux)) Content-Length: 0 Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Message Body Session Description Protocol Request-Line: ACK sip:02131234@100.100.100.211:5090;transport=udp SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-182d537bcb534165-1---d8754z-;rport Max-Forwards: 70 Route: <sip:100.100.100.211;lr> Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 ACK User-Agent: Bria release 2.2 stamp 45414 Content-Length: 0 Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: True] [Suspected resend of frame: 7] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Message Body Session Description Protocol When I removed Freeswitch from the path the ACK is routed ok. The ACK is routed by loose_route, but then it fails because the if ( t_check_trans() ) { What could be happening? Regards, Ricardo.- *Ricardo Martínez **|* Ingeniero de Desarrollo RedVoiss *|* rmartinez(a)redvoiss.net Santiago - Chile *|* +56 2 2408196 www.redvoiss.net

2 1

ACK problem with FreeSwitch-Kamailio SBC implementation.
by Ricardo Martinez 17 Jan '11

17 Jan '11

Hello. I was wondering if someone can help me with some problem that I’m having with the combination of FreeSwitch and Kamailio as SBC. I’m having and ACK not routed by kamailio and dropped. Here is the debug. 100.100.100.211:5060 : Kamailio 100.100.100.211:5090 : FreeSwitch 100.100.100.108 : Caller 100.100.100.212 : Calle User Datagram Protocol, Src Port: 42772 (42772), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211>SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Bria release 2.2 stamp 45414 Content-Length: 388 Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Server: kamailio (3.1.1 (i386/linux)) Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:3031234@100.100.100.212:5060 SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: <sip:100.100.100.211;lr=on> Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Max-Forwards: 66 From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Contact: <sip:mod_sofia@100.100.100.211:5090> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 X-FS-Support: update_display Remote-Party-ID: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;party=calling;screen=yes;privacy=off Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Status-Code: 100 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0;received=100.100.100.211 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211>> Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Server: Thirdlane Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212> > Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 100.100.100.211;branch=z9hG4bK3916.41fc6fb3.0;received=100.100.100.211 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK1QH1XDH609cUm Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=as6727124a Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 INVITE Server: Thirdlane Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: <sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212> > Content-Type: application/sdp Content-Length: 239 Message Body Session Description Protocol User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:3031234@100.100.100.212<sip%3A3031234(a)100.100.100.212>SIP/2.0 Method: ACK [Resent Packet: False] Message Header Record-Route: <sip:100.100.100.211;lr=on> Via: SIP/2.0/UDP 100.100.100.211;branch=0 Via: SIP/2.0/UDP 100.100.100.211:5090 ;rport=5090;branch=z9hG4bK20atZ819Xj3Dg Max-Forwards: 69 From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=QmrB15eKrQ3Bj To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=as6727124a Call-ID: 54a74b6a-9cf6-122e-05a0-0050bad00883 CSeq: 7304076 ACK Contact: <sip:mod_sofia@100.100.100.211:5090> Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "Outbound Call" <sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211> >;party=calling;privacy=off;screen=no Message Body Session Description Protocol User Datagram Protocol, Src Port: 42772 (42772), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:02131234@100.100.100.211:5090;transport=udp SIP/2.0 Method: ACK [Resent Packet: False] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-182d537bcb534165-1---d8754z-;rport Max-Forwards: 70 Route: <sip:100.100.100.211;lr> Contact: <sip:1234@64.76.148.108:42772> To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp From: "camila"<sip:1234@100.100.100.211 <sip%3A1234(a)100.100.100.211> >;tag=8422566b Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 ACK User-Agent: Bria release 2.2 stamp 45414 Content-Length: 0 User Datagram Protocol, Src Port: sip (5060), Dst Port: 42772 (42772) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: True] [Suspected resend of frame: 7] Message Header Via: SIP/2.0/UDP 64.76.148.108:42772 ;branch=z9hG4bK-d8754z-da0b2e4b1b5d0843-1---d8754z-;rport=42772 Record-Route: <sip:100.100.100.211;lr=on> From: "camila" <sip:1234@100.100.100.211<sip%3A1234(a)100.100.100.211> >;tag=8422566b To: <sip:02131234@100.100.100.211 <sip%3A02131234(a)100.100.100.211> >;tag=pBZjZayFUeDSp Call-ID: MjY2NDY5NTE3OTk4MzdmMjE0YWQxNWQ4MWFmY2JhMWQ. CSeq: 1 INVITE Contact: <sip:02131234@100.100.100.211:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d41e649 2011-01-11 12-25-55 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "Outbound Call" <sip:02131234@100.100.100.211<sip%3A02131234(a)100.100.100.211> >;party=calling;privacy=off;screen=no Message Body Session Description Protocol When I removed Freeswitch from the path the ACK is routed ok. The ACK is routed by loose_route, but then it fails because the if ( t_check_trans() ) { What could be happening? * * *Regards,* *Ricardo.-*

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Re: [SR-Users] OpenIMSCore and Kamalio Integration
by Carsten Bock 17 Jan '11

17 Jan '11

Hi Andres, no problem ;-) See my answers inline. 2011/1/17 "Andrés S. García Ruiz" <asgarcia(a)um.es>: > > Hi Carsten, > > Thanks a lot for all the information and the access to your developments. > > I haven't been able to answer last week cause I got the flu. I'm fine right > now. > > In a previous mail you wrote: > > "The goal of the project is to create a stable, fast, open-source, > feature-rich IMS implementation based on Kamailio and the > OpenIMS-Core." > > > > I suppose that you've chosen Kamailio as a Sip Server for XCAP, Presence, > RLS, presence-rules, etc. and OpenIMSCore as the heart of the IMS. So your > work aims -I think- to make an easy integration of Kamailio with the > OpenIMSCore. Is that true? No, not exactly. I've replaced the SER-Core from the OpenIMS-Core with the Kamailio-Core and we are in the progress of creating a new OpenIMS-Core (based on the "old" one). We do not continue to use the existing OpenIMS-Core as such. The OpenIMS-Core implementation has some limitations and has some aspects, which i personally do not like and which i am trying to change (don't misunderstand me here: For a research project, there was really great work done). However, there are a few parts, which i want to change, in order to make the OpenIMS-Core more flexible (and in some issues more stable): - The current OpenIMS-Core borrowed some code from other modules, e.g. there was some code taken from the dialog module, from the Nathelper module and maybe others, in order to implement the CSCF-Components. As the intented show-case and prototype, that is totally acceptable. However, this results in improvements and fixes done to those modules not necessarily leading to fixes in the OpenIMS-Modules - The used Core is SER 2.1-dev1. I think, that was the version of SER when the OpenIMS-Core was released. I wanted to replace the core with a more current core (in my case Kamailio 3.1/3.2), in order to bring the OpenIMS-Core forward. - many aspects of the implementation can be replaced in the current version with Core-Functions (e.g. header manipulation), that will make the code easier to maintain - i want to add the missing documentation for the modules - i wanted to include the modules in the Kamailio/sip-router.org project, in order to keep them maintained My work has nothing to do with integrating a Kamailio-Server (e.g. Presence) into a system consisting of the OpenIMS-Core, but it has something to do with improving the OpenIMS-Core. First i want to improve the OpenIMS-Core, and then we will see, what's next. All i did so far, in order to make a Kamailio-based-Presence-Server work with the IMS-Core, was to modify the Trigger-Point on my HSS a little bit... (but more a dirty hack and nothing, were i am sure if it is the currect or even safe to do; i made it work rather than thinking of, if it was correct...). You need to reconfigure your trigger-points (check the Session Case) in order to make this work. I have attached my trigger-point for presence, may this will help you... > I'm trying to design and implement a similar architecture, but I've had some > issues, mainly with how the OpenIMSCore communicates with the external SIP > presence servers. For example, when I send a RLS SUBSCRIBE to the IMS core, > it does the following: > - If the subscribed URI is an user of the domain, it translates its SIP URI > to the sip contact uri. > - If the subscribed URI is not an user, it answers with "HSS User unknown". > > Could you tell me what kind of problems have you solved till now? > > Kind regards, > Andrés. > > PS: I'm using Ubuntu. Is Ubuntu compatible with your distribution? Since Ubuntu is based on Debian, i think that should work out of the box (but no warranty for that). Kind regards, Carsten P.S: Please always use the reply to all button, in order to keep the lists in CC.

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Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 17 Jan '11

17 Jan '11

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

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[OpenSER-Users] Strange ACK format gives OpenSER error?
by Nawfel Oujdi 17 Jan '11

17 Jan '11

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header.i m not really sure that cause teh problem. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

1 0

tm module question
by Prelle, Stefan 14 Jan '11

14 Jan '11

Hi folks, I have a problem with a scenario where an incoming INVITE is forwarded to a redirect server which responds with 302 and a new target in the Contact header (B). The call is being forwarded to the new contact correctly. All responses from B that are either provisional or a 200 OK are being routed back to the caller. If B responds with any code >=400 the 302 from the redirect server is being returned to the caller instead of the received code. A B C |--INVITE->| | | |--INVITE->| | |<---302---| D | |----INVITE--->| | |<-----404-----| |<---302---| Is this intended? And if so, is there any way to change this behaviour so the latest received response is returned? I tested using the version from repository from 13.01.2011 My reduced config looks like this: route{ if (@to.tag != "") { if (loose_route()) { record_route(); t_relay(); } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } record_route(); } exit; } else { record_route(); } t_on_failure("REDIRECT_REPLY"); if (!t_relay_to_udp("192.168.0.10","5060")) { sl_reply_error(); }; exit; } failure_route[REDIRECT_REPLY] { if (t_check_status("3[0-9][0-9]")) { get_redirects("*:1"); t_relay(); exit; } } Regards, Stefan

2 2

IP Address
by Bernhard Suttner 14 Jan '11

14 Jan '11

Hi, I have a very short question: Kamailio does listen to many IP Addresses and does forward messages to the peer with t_relay. Which IP address will kamailio use for the outgoing traffic as source-ip, if there are many IP addresses on the interface? Is it possible to specify this somehow? Thanks in advance. Best regards, Bernhard

2 2

manual dispatching
by Klaus Darilion 13 Jan '11

13 Jan '11

Hi! I would like to use probing implementation in dispatcher module to keep track of the state of the gateways, but I want to choosing a gateway manually in the config. For example I defined a dispatcher set with 6 entries. Then in .cfg I want to retrieve a certain entry from the set (by index) and its state (active, inactive, probing). AFAIS there is currently now way to do this. Does anybody knows a workaround or maybe another way for manual dispatching but automatic probing? Thanks Klaus

1 2

Looking for the RTP Proxy
by Mallik 13 Jan '11

13 Jan '11

Dear sir, we re looking for RTP proxy training or solution, can you please give us more details on this thanking you with regards Mallik

1 0

How to test xhttp with wget or curl
by mayamatakeshi 13 Jan '11

13 Jan '11

Hello, can someone give me a sample of calling to xhttp from curl or wget? I have this in my kamailio.cfg: listen=tcp:192.168.2.105:80 loadmodule "xhttp.so" event_route[xhttp:request] { xhttp_reply("200", "OK", "text/html", "<html><body>OK - [$si:$sp]</body></html>"); } But when I try wget or curl, kamailio closes the connection without sending a response: [root@centos55-02105 tmp]# curl http://192.168.2.105/test curl: (52) Empty reply from server [root@centos55-02105 tmp]# wget http://192.168.2.105/test --2011-01-14 00:11:49-- http://192.168.2.105/test Connecting to 192.168.2.105:80... connected. HTTP request sent, awaiting response... No data received. I see this in the log file: Jan 14 00:04:47 centos55-02105 /usr/local/sbin/kamailio[10479]: ERROR: <core> [tcp_read.c:904]: ERROR: tcp_read_req: bad request, state=7, error=4 buf: GET /test HTTP/1.0^M User-Agent: Wget/1.11.4 Red Hat modified^M Accept: */*^M Host: 192.168.2.105^M Connection: Keep-Alive^M ^M parsed: GET /test HTTP/1.0^M User-Agent: Wget/1.11.4 Red Hat modified^M Accept: */*^M Host: 192.168.2.105^M Connection: Keep-Alive^M ^M br, takeshi

2 3

Dispatcher Setting
by Bernhard Suttner 13 Jan '11

13 Jan '11

Hi, I am using the Dispatcher module to load-balance the calls to several freeswitch servers. On which responses within the failure_route should be a fallback to another server (dst_next_dst())? I am sure, that on a 408 response there should be a fallback but whats about other 4XX responses and 5XX or 6XX? Thanks in advance. Best regards, Bernhard Suttner

2 2

OpenIMSCore and Kamalio Integration
by "Andrés S. García Ruiz" 13 Jan '11

13 Jan '11

Hi everybody, I'm trying to deploy an IMS network with OpenIMSCore and Kamailio. Since OpenIMSCore has been already tested along with Mobicents, now I want substitute Mobicents for Kamailio. I've also successfully installed Kamailio. I can run it without any problem, but when SIP Publish messages arrive at Kamalio, it answers with "407 Proxy Authentication Required". The IMS presentity is already registered against the OpenIMSCore. How can I solve that problem? The publish message sent: PUBLISH sip:testuser01@open-ims.test SIP/2.0 Route: <sip:ciervo.inf.um.es:5060;lr>, <sip:iscmark@scscf.open-ims.test:6060;lr;s=1;h=0;d=0;a=7369703a74657374757365723031406f70656e2d696d732e74657374> Call-ID: K28QO9PYAE5vn3RNdYOkIEddMEd6yDiqeAW7DwSt6XV. CSeq: 1 PUBLISH From: "testuser01" <sip:testuser01@open-ims.test>;tag=ff123bda To: "testuser01" <sip:testuser01@open-ims.test> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKa31a.6cba1cd2.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKa31a.71481d13.0;i=1 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=41624;branch=z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z- Max-Forwards: 15 Content-Type: application/pidf+xml Expires: 30000 Event: presence Contact: <sip:testuser01@155.54.190.166:8060> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Mon_Nov_29_10-14-33_CET_2010 Content-Length: 451 P-Asserted-Identity: <sip:testuser01@open-ims.test> P-Charging-Vector: icid-value="P-CSCFabcd000000004cf3708400000002";icid-generated-at=155.54.210.134;orig-ioi="open-ims.test" <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' entity='sip:testuser01@open-ims.test'><tuple id='t6b9a6ab3'><status><basic>open</basic></status></tuple><dm:person id='p34b126e5'><rpid:activities><rpid:Online/></rpid:activities><dm:note>Online</dm:note></dm:person></presence> Kamailio answer: SIP/2.0 407 Proxy Authentication Required Call-ID: K28QO9PYAE5vn3RNdYOkIEddMEd6yDiqeAW7DwSt6XV. CSeq: 1 PUBLISH From: "testuser01" <sip:testuser01@open-ims.test>;tag=ff123bda To: "testuser01" <sip:testuser01@open-ims.test>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3d3a Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKa31a.6cba1cd2.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKa31a.71481d13.0;i=1 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=41624;branch=z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z- Proxy-Authenticate: Digest realm="open-ims.test", nonce="TPNwMkzzbwZRE1piAM8yHaXGsXvwS16v" Server: kamailio (3.1.0 (i386/linux)) Content-Length: 0 Kamailio log: 5(15391) DEBUG: <core> [parser/msg_parser.c:628]: SIP Request: 5(15391) DEBUG: <core> [parser/msg_parser.c:630]: method: <PUBLISH> 5(15391) DEBUG: <core> [parser/msg_parser.c:632]: uri: <sip:testuser01@open-ims.test> 5(15391) DEBUG: <core> [parser/msg_parser.c:634]: version: <SIP/2.0> 5(15391) DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <PUBLISH> 5(15391) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=10 5(15391) DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [45]; uri=[sip:testuser01@open-ims.test] 5(15391) DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: to body ["testuser01" <sip:testuser01@open-ims.test> ] 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bKa31a.6cba1cd2.0>; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 5(15391) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 5(15391) DEBUG: <core> [receive.c:145]: After parse_msg... 5(15391) DEBUG: <core> [receive.c:186]: preparing to run routing scripts... 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bKa31a.71481d13.0>; state=6 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 236, <i> = <1>; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=100 5(15391) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <41624>; state=6 5(15391) DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <z9hG4bK-d8754z-ce7487ff34be55f9-1---d8754z->; state=16 5(15391) DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 5(15391) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=100 5(15391) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 5(15391) DEBUG: <core> [parser/msg_parser.c:199]: DEBUG: get_hdr_body : content_length=451 5(15391) DEBUG: <core> [parser/msg_parser.c:101]: found end of header 5(15391) DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=ff123bda 5(15391) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 5(15391) DEBUG: sanity [mod_sanity.c:217]: all sanity checks passed 5(15391) DEBUG: siputils [checks.c:73]: no totag 5(15391) DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffff 5(15391) DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=41274, isACK=0 5(15391) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 5(15391) DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found 5(15391) DEBUG: tm [t_lookup.c:1150]: DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==9 && [open-ims.test] == [127.0.0.1] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==14 && [open-ims.test] == [155.54.190.245] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==9 && [open-ims.test] == [127.0.0.1] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 13==14 && [open-ims.test] == [155.54.190.245] 5(15391) DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 5(15391) DEBUG: auth_db [authorize.c:239]: realm value [open-ims.test] 5(15391) DEBUG: auth [api.c:85]: auth:pre_auth: Credentials with realm 'open-ims.test' not found 5(15391) DEBUG: auth_db [authorize.c:257]: not authenticated 5(15391) DEBUG: auth [challenge.c:102]: build_challenge_hf: realm='open-ims.test' 5(15391) DEBUG: auth [challenge.c:236]: auth: 'Proxy-Authenticate: Digest realm="open-ims.test", nonce="TPNwMkzzbwZRE1piAM8yHaXGsXvwS16v" ' 5(15391) DEBUG: sl [sl.c:278]: reply in stateless mode (sl) 5(15391) DEBUG: <core> [msg_translator.c:207]: check_via_address(155.54.210.135, 155.54.210.135, 0) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) 5(15391) DEBUG: <core> [receive.c:289]: receive_msg: cleaning up Thanks in advance, Andrés. -- ----------------------- Andrés S. García Ruiz ----------------------- e-mail: asgarcia(a)um.es Teléfono: (+34) 868 888258 Dpto. de Ingeniería de la Información y las Comunicaciones Facultad de Informática Universidad de Murcia 30100 Campus de Espinardo

5 13

set_dlg_profile not working?
by Ricardo Martinez 12 Jan '11

12 Jan '11

Hello. I’m trying to count the calls to a specific gateway. So I’m using the dialog module. This route is done before the final relay. route[CHANNELS] { if( is_method("INVITE") ) { if(!isflagset(31) ) { # user has max channel limit set as preference if(is_avp_set("$avp(s:channels)") && avp_check("$avp(s:channels)", "gt/s:0")) { # get current calls for uuid xlog("L_INFO","[$ci] : avp_channels existe y es mayor que 0 ?"); if (get_profile_size("gws","$rd","$var(calls)")) { xlog("L_INFO","Actualmente el gateway $rd tiene $var(calls) llamadas simultaneas\n"); } # check within limit if($avp(s:channels) > $var(calls)) { xlog("L_INFO", "CHANNELS : terminador '$rd' tiene '$var(calls)' de '$avp(s:channels)' active calls before this one\n"); $var(setprofile) = 1; } else { xlog("L_INFO", "Call crtl: user channel limit exceeded [$var(calls)/$avp(s:channels)]\n"); send_reply("487", "Request Terminated: Channel limit exceeded"); exit; } } else { xlog("L_INFO","[$ci] : the value if channels is $avp(s:channels)"); if( !is_avp_set("$avp(s:channels)") ) { xlog("L_INFO","[$ci] : avp_channel not set"); }; if(!avp_check("$avp(s:channels)", "gt/s:0") ) { xlog("L_INFO","[$ci] : avp_channel menor que 0?"); }; $var(setprofile) = 0; } } if($var(setprofile) > 0) { * if ( set_dlg_profile("gws","$rd") ) {* * xlog("L_INFO","set_dlg_profile exitoso");* } } When i made the call and try to see the profile with this command : [root@sipvoiss kamailio]# kamctl fifo profile_get_size gws 10.0.0.220 profile:: name=gws value=10.0.0.220 count=0 I’m not seeing the call added to the profile, despite the fact I using the * set_dlg_profile("gws","$rd")* What could be happening? This is part of the debug : Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] INVITE dominio:10.0.0.208 LOCAL DOMAIN Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:212]: success Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:133]: AVP 'tranum'/0=' sip:02444444444@10.0.0.208 <sip%3A02444444444(a)10.0.0.208>'/0 has been added Jan 11 18:07:03 .: DEBUG: misc_radius [functions.c:133]: AVP 'channels'/0='2'/0 has been added Jan 11 18:07:03 .: DEBUG: avpops [avpops_impl.c:1047]: 1 avps were processed Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] INVITE con dominio:10.0.0.208 checks de usuario via avp_radius OK Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] CHECKS SECTION :: INVITE:sip:02444444444@10.0.0.208 <INVITE%3Asip%3A02444444444(a)10.0.0.208>checks ok... Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1748]: added matched_gws[0]=[1, 4, 1, 8207736] Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1645]: added gw_uri_avp <1|1|4|550|3700771904||5060||1|0> with weight <8207736> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:1911]: r_uri < sip:5504444444@10.0.0.220:5060>, dst_uri <> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2015]: added ruri_user_avp <02444444444> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2050]: added flags_avp <0> Jan 11 18:07:03 .: DEBUG: lcr [lcr_mod.c:2057]: added defunct_gw_avp <3700771904> Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] Ruta LCR : cambiando sip:02444444444@10.0.0.208 <sip%3A02444444444(a)10.0.0.208> por sip:5504444444@10.0.0.220:5060 Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : entro en ruta 39 de call control Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : no esta seteado el flag 31 Jan 11 18:07:03 .: DEBUG: avpops [avpops_impl.c:1180]: check <2> against <0> as str /16 Jan 11 18:07:03 .: INFO: <script>: [ 71492292-34f2-a3ed-814e-0002a400d53d(a)10.0.0.214] : avp_channels existe y es mayor que 0 ? Jan 11 18:07:03 .: INFO: <script>: Actualmente el gateway 10.0.0.220 tiene 0 llamadas simultaneas Jan 11 18:07:03 .: INFO: <script>: CHANNELS : terminador '10.0.0.220' tiene '0' de '2' active calls before this one Jan 11 18:07:03 .: INFO: <script>: set_dlg_profile exitoso Thanks in advance, *Regards,* *Ricardo.-*

2 5

AFTER MAINTENANCE DIALOG MODULE NOT WORKING
by Phillman25 Kyriacou 12 Jan '11

12 Jan '11

Hi I performed the following maintenance actions to my server that is currently installed with kamailio v3.1 cd /usr/local/src/kamailio-3.1.0/kamailio git pull origin make all make install After this action the dialog module is not working, after investigation, i found out that the following entry is causing this issue: modparam("dialog", "db_mode", 1) In my syslog i find the following errors: Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: CRITICAL: dialog [dlg_db_handler.c:117]: BUG - db connection found already open Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 33 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 34 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: dialog [dialog.c:620]: failed to connect to database (rank=24) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 30, pid 7397, "tcp receiver child=7" Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7397]: ERROR: <core> [tcp_main.c:4811]: ERROR: tcp_main: fork failed: Success Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 37 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 43 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 30 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 41 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 5 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: WARNING: <core> [db_id.c:281]: identical DB URLs, but different DB connection pid [7396/7349] Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7391]: INFO: <core> [main.c:807]: INFO: signal 15 received Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 47 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: CRITICAL: dialog [dlg_db_handler.c:117]: BUG - db connection found already open Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: dialog [dialog.c:620]: failed to connect to database (rank=23) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so) Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 29, pid 7396, "tcp receiver child=6" Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 39 Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7396]: ERROR: <core> [tcp_main.c:4811]: ERROR: tcp_main: fork failed: Success Jan 11 16:45:54 SipProxy /usr/local/sbin/kamailio[7398]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 45 Only the dialog module is facing this problem Has anyone faced this before? Thanks Phillip

2 2

mi_fifo question
by Ricardo Martinez 11 Jan '11

11 Jan '11

Hello. I’m trying to use the kamctl command to get information about dialogs. I’m using the next line : kamctl fifo profile_get_size:kamailio_fifo and I’m getting the next output in the log file : Jan 11 12:34:38 /usr/local/sbin/kamailio[15051]: ERROR: mi_fifo [fifo_fnc.c:223]: open error (/tmp/kamailio_fifo:openser_receiver_15073): No such file or directory Jan 11 12:34:38 /usr/local/sbin/kamailio[15051]: ERROR: mi_fifo [fifo_fnc.c:484]: cannot open reply pipe /tmp/kamailio_fifo:openser_receiver_15073 What I’m doing wrong? This is part of the configuration file : # ----- mi_fifo params ------ modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_group", 0) modparam("mi_fifo", "fifo_group", "root") modparam("mi_fifo", "fifo_user", 0) modparam("mi_fifo", "fifo_user", "root") and this is the file kamailio_fifo in the /tmp directory : [root@kamailio]# ls -la /tmp total 52 drwxrwxrwt. 4 root root 4096 Jan 11 12:38 . dr-xr-xr-x. 25 root root 4096 Jan 7 09:25 .. drwxrwxrwt 2 root root 4096 Jan 7 09:25 .ICE-unix *prw-rw---- 1 root root 0 Jan 11 12:34 kamailio_fifo* drwx------ 2 root root 4096 Jan 11 09:54 ssh-gpTCQ12700 Hope someone could help me. Regards, Ricardo.-

2 3

Re: [SR-Users] Avp_write missing modules ?
by Alex Balashov 10 Jan '11

10 Jan '11

Please keep the discussion on list - cc: the list. I see no indication in the documentation that avp_write() is a function that exists, even going back to 1.4.x. On 01/10/2011 04:54 PM, Amit Nepal wrote: > Kamailio 3.1 > > Thank You > Amit Nepal > Systems Administrator > Phoenix Internet > Phone: 602-385-0731 > 602-234-0917#112 > http://www.phoenixinternet.net > > > On 1/10/2011 2:50 PM, Alex Balashov wrote: >> On 01/10/2011 04:48 PM, Amit Nepal wrote: >>> I am getting missing loadmodule issue with avp_write. I have the >>> following statement on line 11 and i am getting this error. Could you >>> please help me on this. But i can load from db using avp_db_load and i >>> can delete avp values but only avp_write is not working. >>> >>> >>> avp_write("$ruri", "s:fwd_blind"); >>> >>> loading modules under >>> /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ >>> 0(2361) : <core> [cfg.y:3412]: parse error in config file >>> /usr/local/etc/kamailio/cfgs/pstn.cfg, line 11, column 39: unknown >>> command, missing loadmodule? >>> >>> ERROR: bad config file (1 errors) >> >> Which version of Kamailio? >> -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 5

Avp_write missing modules ?
by Amit Nepal 10 Jan '11

10 Jan '11

I am getting missing loadmodule issue with avp_write. I have the following statement on line 11 and i am getting this error. Could you please help me on this. But i can load from db using avp_db_load and i can delete avp values but only avp_write is not working. avp_write("$ruri", "s:fwd_blind"); loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(2361) : <core> [cfg.y:3412]: parse error in config file /usr/local/etc/kamailio/cfgs/pstn.cfg, line 11, column 39: unknown command, missing loadmodule? ERROR: bad config file (1 errors) Thank you Amit

2 1

Problem with contact header parsing and get_redirects
by Stefan Prelle 10 Jan '11

10 Jan '11

Hi folks, Since upgrading vom Kamailio 1.5 to 3.1 I have a problem using get_redirects from the uac_redirect module. It seems that there is a problem with parsing the Contact header from the received 302. This is my script excerpt: if (t_check_status("3[0-9][0-9]")) { get_redirects("*"); if (!t_relay()) { xlog("L_ERR","t_relay failed\n"); } } Some 302 messages lead (as expected) to INVITEs send to targets in the Contact header, for others t_relay fails. A working example would be Contact:<sip:+49406686100@192.168.0.1:5060;user=phone>;q=0.5,<sip: +49406686100@192.168.0.2:5060;user=phone>;q=0.25 It fails on Contact: sip:+49406686100@192.168.0.1:5060 The following log messages appear when calling get_redirects (and failing t_relay): kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:74]: resume branch=0 kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:81]: checking branch=0 (added=0) kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:85]: branch=0 is a redirect (added=0) kamailio[7372]: WARNING: <core> [mem/q_malloc.c:366]: qm_malloc(0x82ba320, 32) called from <core>: parser/msg_parser.c: parse_headers(339) kamailio[7372]: WARNING: <core> [mem/q_malloc.c:406]: qm_malloc(0x82ba320, 32) returns address 0x8311d94 frag. 0x8311d7c (size=32) on 1 -th hit kamailio[7372]: DEBUG: uac_redirect [rd_funcs.c:233]: contact hdr not found in sh_rpl The Contact header that fails obviously has no parameters for the URI and contact itself. As far as I understand the RFC 3261 BNF this is valid. The 302 response is by the way created by kamailio when using the LCR module (like this) if (!load_gws("1")) {... exit;} if (!next_gw()) {... exit; } sl_send_reply("302", "Not here"); Did I miss anything when upgrading or is this a problem with the parser? Cheers, Stefan

2 5

Re: [SR-Users] ENUM changes?
by Juha Heinanen 09 Jan '11

09 Jan '11

klaus, could you try with master version if it fixes the problem. -- juha

2 2

Kamailio 3.1.1 + Sipp
by Rochdi ABID 09 Jan '11

09 Jan '11

Hello community, I want to test Kamailio 3.1.1 with SIPP. I followed many ideas that I found them on the internet but without no success. In particular, I followed this link http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling I can see traffic from the uac --> Kamailio --> uas ans in the other sens but no successful call. Can you give me any ideas please. Regards, Rochdi

2 1

Kamailio 3.1.1 + Sipp
by Rochdi ABID 08 Jan '11

08 Jan '11

Hello community, I want to test Kamailio 3.1.1 with SIPP. I followed many ideas that I found them on the internet but without no success. In particular, I followed this link http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling I can see traffic from the uac --> Kamailio --> uas ans in the other sens but no successful call. Can you any ideas please. Regards, Rochdi

1 0

Re: [SR-Users] Change the SIP-To-User
by Bernhard Suttner 07 Jan '11

07 Jan '11

Hi, thanks to all responses. I wonder if there is a function to modify the To/From Header stateful. I mean, change the TO header in the INVITE to A and then the 200 OK (for example) reverse the change back to the previous value to B? but the "magic" $tU is amazing... Best regards, Bernhard ----- Original Message ----- From: Olle E. Johansson [mailto:oej@edvina.net] To: Alex Balashov [mailto:abalashov@evaristesys.com] Cc: sr-users(a)lists.sip-router.org Sent: Fri, 07 Jan 2011 11:09:41 +0100 Subject: Re: [SR-Users] Change the SIP-To-User > > 7 jan 2011 kl. 10.55 skrev Alex Balashov: > > > On 01/07/2011 04:52 AM, Carsten Bock wrote: > > > >> the trick, when using "uac_replace_from" is, that it will > >> "automagically" change subsequest requests. This is required in order > >> to work with most SIP-Endpoints. > >> If you just want to change the "From" for the current request, that is > fine. > > > > I understand. Changing To/From by a proxy has long been a thorny issue, > in the sense that on the one hand those fields are purely cosmetic in > strict-3261 mode, but on the other hand, changing them breaks 2543 > compatibility, and it's just not something a proxy is supposed to do. > Well, that changed with a new rfc (put number here) about caller ID updates. > I do not remember the proxy role, but a UA can now change From/To identities > mid call, like after a call transfer in a pbx. > > /O > _______________________________________________ > 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 >

4 3

RLS server as SUBSCRIBE proxy?
by Jeremya 07 Jan '11

07 Jan '11

Hi, My situation is that I have a number of SIP end points (server) that support SUBSCRIBE method on various resources they hold. Primarily these are dialog resources, but include arbitrary other resources with custom MIME types. I also have a large number of SIP end-points (clients) that use SUBSCRIBE to resources held by these end-points. Some of these end-points are standard phones (SPA-962) monitoring dialog state. Other end points are custom clients that monitor a variety of custom resources and MIME types. I have got to the point where the number of SUBSCRIBE clients is making it impractical for the SUBSCRIBE sources to support them. I want to make a SUBSCRIBE proxy server that makes arbitrary SUBSCRIBE requests to my SIP end points (server) and in turn accepts SUBSCRIBE requests from my consumer SIP clients (Phones etc). In effect the server would aggregate SUBSCRIBE requests into a single SUBSCRIBE to a target resource. Ideally the SUBSCRIPTION would be dynamic, generated by the SUBSCRIBE requests from the SIP phones and other clients, but this can be static. I've read the documentation for the RLS module in Kamailio but I can't actually figure out what it does. It may do what I want (if only in some database mode) but I'd appreciate advice as to whether to pursue this as an option or go to plan-B Regards Jeremy

2 2

Re: [SR-Users] Change the SIP-To-User
by Bernhard Suttner 07 Jan '11

07 Jan '11

Hi, ah, thanks. Would be good to have a uac_replace_to. Sounds like a "30min" task to someone familiar with kamailio programming :-) BR, Bernhard ----- Original Message ----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Fri, 07 Jan 2011 21:59:55 +0100 Subject: Re: [SR-Users] Change the SIP-To-User > Bernhard Suttner wrote: > > Hi, > > > > thanks to all responses. I wonder if there is a function to modify the > To/From Header stateful. I mean, change the TO header in the INVITE to A and > then the 200 OK (for example) reverse the change back to the previous value > to B? > > > > but the "magic" $tU is amazing... > > If you use uac_replace_from(), the changes will stateful and fixed for > every in-dialog requests and responses. > > But there is no similar feature for To header manipulation. > > regards > klaus > > >

1 0

Using Dispatcher Module From Mysql
by Amit Nepal 07 Jan '11

07 Jan '11

Hi everyone, Again stuck at the dispatcher module.Do anyone happen to have a working config for kamailio with dispatcher list from mysql database ? if not can someone please guide me on this ? Do i need to pass module parameters for the column and the table names ? I have the following config : loadmodule "dispatcher.so" if(uri=~"^sip:1[2-9][0-9]{9}@") { if(is_user_in("credentials","longdistance")) { #route(PSTN); #exit; ds_select_dst("1","8"); forward(); } else{ sl_send_reply("403","No Permissions for Long Distance calls"); exit; }; }; -- Thank You Amit Nepal

3 5

Dispatcher Attributes field
by alex pappas 07 Jan '11

07 Jan '11

Dear all, I'm trying with the Dispatcher module to terminate a call to three different gateways serially. For each gw I use the attribute field to get the attribute and add it as a prefix to the RU. The problem is that the $avp(dsattrs) is returning always the first value of the first gw that is using. example: destination sip:100.12.12.12:5060 attrs 111 destination sip:100.12.12.11:5060 attrs 222 destination sip:100.12.12.10:5060 attrs 333 In this example the rU will always be 111+number. I tried also to initiate the avp in the Failure route by giving null value but still does not change the result. In Request route: if(ds_select_domain("$avp(s:disp_dstgrp)", "4")) { $avp(s:term_prefix) = $avp(dsattrs); # avp for CDR purpose $rU = $avp(dsattrs) + $rU; t_on_failure("FAIL_NATIONAL"); t_relay(); exit; } In Failure route: if(!ds_next_domain()) { t_reply("503", "Service Unavailable"); exit; } else { $rU = $avp(dsattrs) + $rU; t_on_failure("FAIL_THIS"); append_branch(); if(!t_relay()) { t_reply("503","Service Unavailable"); exit; }

2 5

Change the SIP-To-User
by Bernhard Suttner 07 Jan '11

07 Jan '11

Hi, what is the best way to replace the userpart of the SIP-To URI with the content of a pseudo variable? Thanks for every hint. Best regards, Bernhard

6 8

problem in MAKE command during installation Sip Express Router
by city spider 07 Jan '11

07 Jan '11

hello i want to install Sip Express Router on CENT OS 5.5. the problem is that after untar the ser-0.9.6.src.tar.gz package my system doesn't operate MAKE command ,and returns 127 errors containing message "the system doesn't find the folder...." is there any solution for this problem... waiting for your response.

2 1

Call Routing Not working
by Tim King 07 Jan '11

07 Jan '11

Below is my current code and my calls are just getting 100 trying -- your call is important to us even between to locally registered extension. Any guidance as to how to simplify troubleshooting this routing? ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE") ) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif #!ifdef WITH_CFGSAMPLES /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } #!endif if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route 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")) { 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; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); if($var(newbranch)==1) { append_branch(); $var(newbranch) = 0; } route(RELAY); exit; } #!endif #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); $var(newbranch) = 1; route(FSVBOX); } } #!endif # sample config file for dispatcher module #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", "mysql://openser:password@localhost /openser") route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } #!endif

2 1

Routing error
by Tim King 07 Jan '11

07 Jan '11

My calls are not making it to the dispatcher and I am not sure what I am doing wrong. Can anyone see what I need to modify so that my calls can be distributed to my free switch boxes via the dispatcher module? cat /usr/local/etc/kamailio/kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_FREESWITCH #!define WITH_ANTIFLOOD #!define WITH_DISPATCHER #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:password@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_FREESWITCH freeswitch.bindip = "127.0.0.1" desc "FreeSWITCH IP Address" freeswitch.bindport = "5090" desc "FreeSWITCH Port" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" # loadmodule "modules/tm/tm.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif #!ifdef WITH_CFGSAMPLES /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } #!endif if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route 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")) { 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; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); if($var(newbranch)==1) { append_branch(); $var(newbranch) = 0; } route(RELAY); exit; } #!endif #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); $var(newbranch) = 1; route(FSVBOX); } } #!endif # sample config file for dispatcher module #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", "mysql://openser:password@localhost /openser") route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } #!endif

3 2

Re: [SR-Users] avp_db_query returning wrong data
by Alex Balashov 07 Jan '11

07 Jan '11

On 11/23/2010 06:41 PM, Jijo wrote: > Yes it work with many UDP children. There were some concurrency issues with the Postgres module a while back, discovered after the 3.1.0 release. I remember something about it because we are extremely heavy users of the Postgres module as well. Unfortunately, I can't find the thread. Try the latest git master HEAD: git clone git://git.sip-router.org/sip-router cd sip-router make FLAVOUR=kamailio cfg make all And see if it changes. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street NE 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 4

SIREMIS logging problem.
by Ricardo Martinez 06 Jan '11

06 Jan '11

Hello. I have Siremis 2.0 installed from scratch, but I’m unable to logging. This is the error that I’m getting from the logs : '01/05/2011','12:41:36','ERR','DATAOBJ','Query Error: SQLSTATE[42S02]: Base table or view not found: 1146 Table 'siremis.contact' doesn't exist','n/a' I was checking the SIREMIS db and I can’t found the table “contact”. Is this a bug?.. can someone help me? Thanks Regards, * * *Ricardo Martinez.-*

2 3

Kamailio Development Training, Barcelona, Feb 10-11, 2011
by Daniel-Constantin Mierla 05 Jan '11

05 Jan '11

Hello, for those interested in learning how to write C code for Kamailio & SER, the next development training class takes place in Barcelona, Spain, during February 10-11, 2011. You can find more details at: http://asipto.com/u/r Number of seats is limited, the acceptance policy is first come first served. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

ENUM changes?
by Klaus Darilion 05 Jan '11

05 Jan '11

Hi! I just found out that my ENUM config didn't worked any more - and it worked for sure some time ago. Were there any recent changes, e.g. from 3.0 to 3.1? I used if enum_query("e164.arpa.","+sip+voice:sip") { which should accept normal E2U+sip and E2U+voice:sip NAPTRs. But it doesn't work with kamailio 3.1.0. But enum_query("e164.arpa.") works. thanks klaus

4 3

Re: [SR-Users] RLS issue
by Klaus Darilion 05 Jan '11

05 Jan '11

1. Kamailio will send the request to the address of the list member (normal DNS lookups). 2. If there is a problem with OpenIMS then you should ask the OpenIMS guys, we are not familiar with OpenIMS. regards klaus Am 22.12.2010 12:57, schrieb "Andrés S. García Ruiz": > > Thanks a lot for your response! > > I've been thinking about it and there's something abnormal. Firs, the > RLS is suppose to send the initial SUBSCRIBE to the S-CSCF instead of > sending it to the I-CSCF. Then, I don't know why the S-CSCF translates > the SUBSCRIBE URI (sip:testuser02@open-ims.test) to > "sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0". All the > routing is the default configuration of both cscfs and rls. > > How can I change the routing of the RLS in order it to send the > SUBSCRIBE messages to the S-CSCF? > > Here's the configuration file: > / > #!KAMAILIO > > #!ifdef ACCDB_COMMENT > ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL > DEFAULT ''; > #!endif > > ####### Defined Values ######### > > #!define WITH_DEBUG > #!define WITH_AUTH > #!define WITH_MYSQL > #!define WITH_USRLOCDB > #!define WITH_PRESENCE > #!define WITH_XCAPSRV > > > # *** Value defines - IDs used later in config > #!ifdef WITH_MYSQL > # - database URL - used to connect to database server by modules such > # as: auth_db, acc, usrloc, a.s.o. > #!define DBURL "mysql://openser:openserrw@localhost/openser" > #!endif > #!ifdef WITH_MULTIDOMAIN > # - the value for 'use_domain' parameters > #!define MULTIDOMAIN 1 > #!else > #!define MULTIDOMAIN 0 > #!endif > > # - flags > # FLT_ - per transaction (message) flags > # FLB_ - per branch flags > #!define FLT_ACC 1 > #!define FLT_ACCMISSED 2 > #!define FLT_ACCFAILED 3 > #!define FLT_NATS 5 > > #!define FLB_NATB 6 > #!define FLB_NATSIPPING 7 > > ####### Global Parameters ######### > > #!ifdef WITH_DEBUG > debug=4 > log_stderror=yes > #!else > debug=2 > log_stderror=no > #!endif > > memdbg=5 > memlog=5 > > log_facility=LOG_LOCAL0 > > fork=yes > children=4 > > /* uncomment the next line to disable TCP (default on) */ > #disable_tcp=yes > > > /* uncomment the next line to disable the auto discovery of local aliases > based on reverse DNS on IPs (default on) */ > #auto_aliases=no > > /* add local domain aliases */ > alias="open-ims.test" > > /* uncomment and configure the following line if you want Kamailio to > bind on a specific interface/port/proto (default bind on all available) */ > #listen=udp:10.0.0.10:5060 > > /* port to listen to > * - can be specified more than once if needed to listen on many ports */ > port=5060 > > #!ifdef WITH_TLS > enable_tls=yes > #!endif > > #!ifdef WITH_XCAPSRV > tcp_accept_no_cl=yes > #!endif > > ####### Custom Parameters ######### > > # These parameters can be modified runtime via RPC interface > # - see the documentation of 'cfg_rpc' module. > # > # Format: group.id = value 'desc' description > # Access: $sel(cfg_get.group.id) or @cfg_get.group.id > # > > #!ifdef WITH_PSTN > # PSTN GW Routing > # > # - pstn.gw_ip: valid IP or hostname as string value, example: > # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" > # > # - by default is empty to avoid misrouting > pstn.gw_ip = "" desc "PSTN GW Address" > #!endif > > > ####### Modules Section ######## > > # set paths to location of modules > #!ifdef LOCAL_TEST_RUN > mpath="modules_k:modules" > #!else > mpath="/usr/local/kamailio-3.1-xcap-rls/lib/kamailio/modules_k/:/usr/local/kamailio-3.1-xcap-rls/lib/kamailio/modules/" > #!endif > > #!ifdef WITH_MYSQL > loadmodule "db_mysql.so" > #!endif > > loadmodule "mi_fifo.so" > loadmodule "kex.so" > loadmodule "tm.so" > loadmodule "tmx.so" > loadmodule "sl.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "mi_rpc.so" > loadmodule "acc.so" > # loadmodule "pua_mi.so" > # loadmodule "pua_usrloc.so" > > > #!ifdef WITH_AUTH > loadmodule "auth.so" > loadmodule "auth_db.so" > #!ifdef WITH_IPAUTH > loadmodule "permissions.so" > #!endif > #!endif > > #!ifdef WITH_ALIASDB > loadmodule "alias_db.so" > #!endif > > #!ifdef WITH_MULTIDOMAIN > loadmodule "domain.so" > #!endif > > #!ifdef WITH_PRESENCE > loadmodule "presence.so" > loadmodule "presence_xml.so" > loadmodule "pua.so" > loadmodule "rls.so" > #!endif > > #!ifdef WITH_NAT > loadmodule "nathelper.so" > loadmodule "rtpproxy.so" > #!endif > > #!ifdef WITH_TLS > loadmodule "tls.so" > #!endif > > #!ifdef WITH_ANTIFLOOD > loadmodule "htable.so" > loadmodule "pike.so" > #!endif > > #!ifdef WITH_XMLRPC > loadmodule "xmlrpc.so" > #!endif > > #!ifdef WITH_XCAPSRV > loadmodule "xhttp.so" > loadmodule "xcap_server.so" > #!endif > > # ----------------- setting module-specific parameters --------------- > > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > > # ----- tm params ----- > # auto-discard branches from previous serial forking leg > modparam("tm", "failure_reply_mode", 3) > # default retransmission timeout: 30sec > modparam("tm", "fr_timer", 30000) > # default invite retransmission timeout after 1xx: 120sec > modparam("tm", "fr_inv_timer", 120000) > > > # ----- rr params ----- > # add value to ;lr param to cope with most of the UAs > modparam("rr", "enable_full_lr", 1) > # do not append from tag to the RR (no need for this script) > modparam("rr", "append_fromtag", 0) > > > # ----- registrar params ----- > modparam("registrar", "method_filtering", 1) > /* uncomment the next line to disable parallel forking via location */ > # modparam("registrar", "append_branches", 0) > /* uncomment the next line not to allow more than 10 contacts per AOR */ > #modparam("registrar", "max_contacts", 10) > > > # ----- acc params ----- > /* what special events should be accounted ? */ > modparam("acc", "early_media", 0) > modparam("acc", "report_ack", 0) > modparam("acc", "report_cancels", 0) > /* by default ww do not adjust the direct of the sequential requests. > if you enable this parameter, be sure the enable "append_fromtag" > in "rr" module */ > modparam("acc", "detect_direction", 0) > /* account triggers (flags) */ > modparam("acc", "log_flag", FLT_ACC) > modparam("acc", "log_missed_flag", FLT_ACCMISSED) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) > /* enhanced DB accounting */ > #!ifdef WITH_ACCDB > modparam("acc", "db_flag", FLT_ACC) > modparam("acc", "db_missed_flag", FLT_ACCMISSED) > modparam("acc", "db_url", DBURL) > modparam("acc", "db_extra", > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > #!endif > > > # ----- usrloc params ----- > /* enable DB persistency for location entries */ > #!ifdef WITH_USRLOCDB > modparam("usrloc", "db_url", DBURL) > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- auth_db params ----- > #!ifdef WITH_AUTH > modparam("auth_db", "db_url", DBURL) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "load_credentials", "") > modparam("auth_db", "use_domain", MULTIDOMAIN) > > # ----- permissions params ----- > #!ifdef WITH_IPAUTH > modparam("permissions", "db_url", DBURL) > modparam("permissions", "db_mode", 1) > #!endif > > #!endif > > > # ----- alias_db params ----- > #!ifdef WITH_ALIASDB > modparam("alias_db", "db_url", DBURL) > modparam("alias_db", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- domain params ----- > #!ifdef WITH_MULTIDOMAIN > modparam("domain", "db_url", DBURL) > # use caching > modparam("domain", "db_mode", 1) > # register callback to match myself condition with domains list > modparam("domain", "register_myself", 1) > #!endif > > > #!ifdef WITH_PRESENCE > # ----- presence params ----- > modparam("presence", "db_url", DBURL) > > # ----- presence_xml params ----- > modparam("presence_xml", "db_url", DBURL) > modparam("presence_xml", "force_active", 1) > modparam("presence_xml", "integrated_xcap_server", 1) > modparam("rls", "db_url", DBURL) > modparam("rls", "xcap_table", "xcap") > modparam("rls", "rlsubs_table", "rls_watchers") > modparam("rls", "rlpres_table", "rls_presentity") > modparam("rls", "clean_period", 100) > modparam("rls", "waitn_time", 50) > modparam("rls", "max_expires", 10800) > modparam("rls", "hash_size", 9) > modparam("rls", "xcap_root", "http://ciervo.inf.um.es/xcap-root:5060") > modparam("rls", "integrated_xcap_server", 1) > modparam("rls", "to_presence_code", 10) > modparam("rls", "rls_event", "presence") > #!endif > > > #!ifdef WITH_NAT > # ----- rtpproxy params ----- > modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") > > # ----- nathelper params ----- > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) > modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") > > # params needed for NAT traversal in other modules > modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") > modparam("usrloc", "nat_bflag", FLB_NATB) > #!endif > > > #!ifdef WITH_TLS > # ----- tls params ----- > modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") > #!endif > > #!ifdef WITH_ANTIFLOOD > # ----- pike params ----- > modparam("pike", "sampling_time_unit", 2) > modparam("pike", "reqs_density_per_unit", 16) > modparam("pike", "remove_latency", 4) > > # ----- htable params ----- > # ip ban htable with autoexpire after 5 minutes > modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") > #!endif > > #!ifdef WITH_XMLRPC > # ----- xmlrpc params ----- > modparam("xmlrpc", "route", "XMLRPC"); > modparam("xmlrpc", "url_match", "^/RPC") > #!endif > > #!ifdef WITH_XCAPSRV > # ----- xcap_server params ----- > modparam("xcap_server", "db_url", DBURL) > # modparam("rls", "integrated_xcap_server", 1) > #!endif > > ####### Routing Logic ######## > > > # Main SIP request routing logic > # - processing of any incoming SIP request starts with this route > route { > > # per request initial checks > route(REQINIT); > > # NAT detection > route(NAT); > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # CANCEL processing > if (is_method("CANCEL")) > { > if (t_check_trans()) > t_relay(); > exit; > } > > t_check_trans(); > > # authentication > route(AUTH); > > # record routing for dialog forming requests (in case they are routed) > # - remove preloaded route headers > remove_hf("Route"); > if (is_method("INVITE|SUBSCRIBE")) > record_route(); > > # account only INVITEs > if (is_method("INVITE")) > { > setflag(FLT_ACC); # do accounting > } > > # dispatch requests to foreign domains > route(SIPOUT); > > ### requests for my local domains > > # handle presence related requests > route(PRESENCE); > > # handle registrations > route(REGISTRAR); > > if ($rU==$null) > { > # request with no Username in RURI > sl_send_reply("484","Address Incomplete"); > exit; > } > > # dispatch destinations to PSTN > route(PSTN); > > # user location service > route(LOCATION); > > route(RELAY); > } > > > route[RELAY] { > #!ifdef WITH_NAT > if (check_route_param("nat=yes")) { > setbflag(FLB_NATB); > } > if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { > route(RTPPROXY); > } > #!endif > > /* example how to enable some additional event routes */ > if (is_method("INVITE")) { > #t_on_branch("BRANCH_ONE"); > t_on_reply("REPLY_ONE"); > t_on_failure("FAIL_ONE"); > } > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > > # Per SIP request initial checks > route[REQINIT] { > #!ifdef WITH_ANTIFLOOD > # flood dection from same IP and traffic ban for a while > # be sure you exclude checking trusted peers, such as pstn gateways > # - local host excluded (e.g., loop to self) > if(src_ip!=myself) > { > if($sht(ipban=>$si)!=$null) > { > # ip is already blocked > xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); > exit; > } > if (!pike_check_req()) > { > xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); > $sht(ipban=>$si) = 1; > exit; > } > } > #!endif > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > } > > if(!sanity_check("1511", "7")) > { > xlog("Malformed SIP message from $si:$sp\n"); > exit; > } > } > > # Handle requests within SIP dialogs > route[WITHINDLG] { > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > if (is_method("BYE")) { > setflag(FLT_ACC); # do accounting ... > setflag(FLT_ACCFAILED); # ... even if the transaction fails > } > route(RELAY); > } else { > if (is_method("SUBSCRIBE") && uri == myself) { > # in-dialog subscribe requests > route(PRESENCE); > exit; > } > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > t_relay(); > exit; > } else { > # ACK without matching transaction ... ignore and discard > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > } > > # Handle SIP registrations > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > if (!save("location")) > sl_reply_error(); > > exit; > } > } > > # USER location service > route[LOCATION] { > > #!ifdef WITH_ALIASDB > # search in DB-based aliases > alias_db_lookup("dbaliases"); > #!endif > > if (!lookup("location")) { > switch ($rc) { > case -1: > case -3: > t_newtran(); > t_reply("404", "Not Found"); > exit; > case -2: > sl_send_reply("405", "Method Not Allowed"); > exit; > } > } > > # when routing via usrloc, log the missed calls also > if (is_method("INVITE")) > { > setflag(FLT_ACCMISSED); > } > } > > # Presence server route > 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")) > { > $var(ret_code)= rls_handle_subscribe(); > > if($var(ret_code)== 10) { > handle_subscribe(); > } > t_release(); > > }else if(method=="NOTIFY") > { > rls_handle_notify(); > } > > 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; > } > > # Authentication route > route[AUTH] { > return; > #!ifdef WITH_AUTH > if (is_method("REGISTER")) > { > # authenticate the REGISTER requests (uncomment to enable auth) > if (!www_authorize("$td", "subscriber")) > { > www_challenge("$td", "0"); > exit; > } > > if ($au!=$tU) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > > #!ifdef WITH_IPAUTH > if(allow_source_address()) > { > # source IP allowed > return; > } > #!endif > # authenticate if from local subscriber > if (from_uri==myself) > { > if (!proxy_authorize("$fd", "subscriber")) { > proxy_challenge("$fd", "0"); > exit; > } > if (is_method("PUBLISH")) > { > if ($au!=$tU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > if ($au!=$fU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > > consume_credentials(); > # caller authenticated > } else { > # caller is not local subscriber, then check if it calls > # a local destination, otherwise deny, not an open relay here > if (!uri==myself) > { > sl_send_reply("403","Not relaying"); > exit; > } > } > } > #!endif > return; > } > > # Caller NAT detection route > route[NAT] { > #!ifdef WITH_NAT > force_rport(); > if (nat_uac_test("19")) { > if (method=="REGISTER") { > fix_nated_register(); > } else { > fix_nated_contact(); > } > setflag(FLT_NATS); > } > #!endif > return; > } > > # RTPProxy control > route[RTPPROXY] { > #!ifdef WITH_NAT > if (is_method("BYE")) { > unforce_rtp_proxy(); > } else if (is_method("INVITE")){ > force_rtp_proxy(); > } > if (!has_totag()) add_rr_param(";nat=yes"); > #!endif > return; > } > > # Routing to foreign domains > route[SIPOUT] { > if (!uri==myself) > { > append_hf("P-hint: outbound\r\n"); > route(RELAY); > } > } > > # PSTN GW routing > route[PSTN] { > #!ifdef WITH_PSTN > # check if PSTN GW IP is defined > if (strempty($sel(cfg_get.pstn.gw_ip))) { > xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); > return; > } > > # route to PSTN dialed numbers starting with '+' or '00' > # (international format) > # - update the condition to match your dialing rules for PSTN routing > if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) > return; > > # only local users allowed to call > if(from_uri!=myself) { > sl_send_reply("403", "Not Allowed"); > exit; > } > > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); > > route(RELAY); > exit; > #!endif > > return; > } > > # XMLRPC routing > #!ifdef WITH_XMLRPC > route[XMLRPC] > { > # allow XMLRPC from localhost > if ((method=="POST" || method=="GET") > && (src_ip==127.0.0.1)) { > # close connection only for xmlrpclib user agents (there is a bug in > # xmlrpclib: it waits for EOF before interpreting the response). > if ($hdr(User-Agent) =~ "xmlrpclib") > set_reply_close(); > set_reply_no_connect(); > dispatch_rpc(); > exit; > } > send_reply("403", "Forbidden"); > exit; > } > #!endif > > # Sample branch router > branch_route[BRANCH_ONE] { > xdbg("new branch at $ru\n"); > } > > # Sample onreply route > onreply_route[REPLY_ONE] { > xdbg("incoming reply\n"); > #!ifdef WITH_NAT > if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) > && status=~"(183)|(2[0-9][0-9])") { > force_rtp_proxy(); > } > if (isbflagset("6")) { > fix_nated_contact(); > } > #!endif > } > > # Sample failure route > failure_route[FAIL_ONE] { > #!ifdef WITH_NAT > if (is_method("INVITE") > && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { > unforce_rtp_proxy(); > } > #!endif > > if (t_is_canceled()) { > exit; > } > > # uncomment the following lines if you want to block client > # redirect based on 3xx replies. > ##if (t_check_status("3[0-9][0-9]")) { > ##t_reply("404","Not found"); > ## exit; > ##} > > # uncomment the following lines if you want to redirect the failed > # calls to a different new destination > ##if (t_check_status("486|408")) { > ## sethostport("192.168.2.100:5060"); > ## append_branch(); > ## # do not set the missed call flag again > ## t_relay(); > ##} > } > > #!ifdef WITH_XCAPSRV > # #!define WITH_XHTTPAUTH > event_route[xhttp:request] { > xdbg("===== xhttp: request [$rv] $rm => $hu\n"); > > #!ifdef WITH_XHTTPAUTH > if (!www_authorize("xcap", "subscriber")) > { > www_challenge("xcap", "0"); > exit; > } > #!endif > xdbg("===== xhttp: Antes del if de $hu\n"); > if($hu=~"^/xcap-root/") > { > xdbg("===== xhttp: Dentro del if de $hu\n"); > set_reply_close(); > set_reply_no_connect(); > # xcap ops > $xcapuri(u=>data) = $hu; > if($xcapuri(u=>xuid)=~"^sip:.+@.+") > $var(uri) = $xcapuri(u=>xuid); > else if($xcapuri(u=>xuid)=~".+@.+") > $var(uri) = "sip:" + $xcapuri(u=>xuid); > else > $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; > xlog("===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); > if($xcapuri(u=>auid)=="xcap-caps") > { > $var(xbody) = > "<?xml version='1.0' encoding='UTF-8'?> > <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> > <auids> > <auid>rls-services</auid> > <auid>pidf-manipulation</auid> > <auid>xcap-caps</auid> > <auid>resource-lists</auid> > <auid>pres-rules</auid> > <auid>org.openmobilealliance.pres-rules</auid> > </auids> > <extensions> > </extensions> > <namespaces> > <namespace>urn:ietf:params:xml:ns:rls-services</namespace> > <namespace>urn:ietf:params:xml:ns:pidf</namespace> > <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> > <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> > <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> > </namespaces> > </xcap-caps>"; > xhttp_reply("200", "ok", "application/xcap-caps+xml", > "$var(xbody)"); > exit; > } > > #!ifdef WITH_XHTTPAUTH > # be sure auth user access only its documents > if ($au!=$(var(uri){uri.user})) { > xhttp_reply("403", "Forbidden", "text/html", > "<html><body>$si:$sp</body></html>"); > exit; > } > > #!endif > xdbg("===== xhttp: Antes del switch $rm => $hu\n"); > switch($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); > } > exit; > break; > case "GET": > xlog("===== xhttp: get $var(uri) => $hu\n"); > xcaps_get("$var(uri)", "$hu"); > exit; > break; > case "DELETE": > xcaps_del("$var(uri)", "$hu"); > if($xcapuri(u=>auid)=~"pres-rules") > { > xlog("===== xhttp del: refreshing watchers for $var(uri)\n"); > pres_update_watchers("$var(uri)", "presence"); > pres_refresh_watchers("$var(uri)", "presence", 1); > } > exit; > break; > } > } > > # http ops > xhttp_reply("200", "ok", "text/html", > "<html><body>OK: $si:$sp</body></html>"); > exit; > } > #!endif/ > > > Regards, > Andrés. > > El 21/12/10 10:22, Klaus Darilion escribió: >> Am 16.12.2010 13:36, schrieb "Andrés S. García Ruiz": >>> >>> B.5) P-CSCF ---- SUBSCRIBE ----> Presentity??? >>> >>> SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0 >>> SIP/2.0 >>> Record-Route: <sip:mt@pcscf.open-ims.test:4060;lr> >>> Record-Route: <sip:mt@scscf.open-ims.test:6060;lr> >>> Via: SIP/2.0/TCP 155.54.210.134:4060;branch=z9hG4bK07df.baa7cf24.0 >>> Via: SIP/2.0/UDP >>> 155.54.210.135:6060;received=155.54.210.135;rport=6060;branch=z9hG4bK07df.8a35e4f3.0 >>> >>> >>> Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 >>> Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 >>> To: sip:testuser01@open-ims.test >>> From: >>> sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 >>> >>> >>> CSeq: 10 SUBSCRIBE >>> Call-ID: 7fd8dfdd-21694(a)127.0.0.1 >>> Content-Length: 0 >>> User-Agent: kamailio (3.2.0-dev1 (i386/linux)) >>> Max-Forwards: 14 >>> Event: presence >>> Contact: <sip:155.54.190.245:5060;transport=udp> >>> Expires: 10810 >>> Max-Forwards: 70 >>> Support: eventlist >>> P-Called-Party-ID: <sip:testuser01@open-ims.test> >>> >>> >>> The step B.5 is sent directly to the presentity testuser01. Instead of >>> that, I thought that message was suppose to be sent to the presence >>> server in the URI that is shown in Contact header. >> >> >> If the P-CSCF is not forwarding the request as supposed, I guess you >> have to fix the routing on the P-CSCF. >> >> klaus >

4 16

Re: [SR-Users] sr-users Digest, Vol 68, Issue 7
by Javier Gallart 05 Jan '11

05 Jan '11

Thanks Alex, it looks like we'll have to move to mysql in that case (we are much more experienced in postgres). Is this feature expected to be added for db_postgres? Regards Javi > > Message: 6 > Date: Tue, 04 Jan 2011 11:54:20 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] Reconnection to db > To: sr-users(a)lists.sip-router.org > Message-ID: <4D2350BC.8080207(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 01/04/2011 09:46 AM, Javier Gallart wrote: > > Hi list > > > > we have been testing the mtree module. We do periodical reloads of the > > mtree table with the "mi mt_reload" command through xmlrpc. It works > > fine except when the db is restarted. If we manually restart the db and > > then try the mt_reload commands we get this error: > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: db_postgres [km_dbase.c:152]: 0xa6a490 PQsendQuery Error: server > > closed the connection unexpectedly#012#011This probably means the server > > terminated abnormally#012#011before or while processing the request.#012 > > Query: select partition,prefix,route from trees order by partition > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: <core> [db_query.c:101]: error while submitting query > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: mtree [mtree_mod.c:622]: Error while querying db > > Jan 4 09:13:17 server /usr/local/kamailio-3.1.1/sbin/kamailio[28534]: > > ERROR: mtree [mtree_mod.c:738]: cannot re-load info from database > > > > Is there a way to reconnect to the db before sending that command? > > Failure to automatically reconnect to Postgres is one of the > distinguishing features of db_postgres in contrast to db_mysql for the > moment. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > > > ------------------------------ > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > End of sr-users Digest, Vol 68, Issue 7 > *************************************** >

2 1

Use of SIPP along with kamailio - REGISTER followed by INVITE not working
by Stephen McVarnock 05 Jan '11

05 Jan '11

Guys, I am not sure in which world to expect to get an answer to this so I am sending this to both SIPP and SR-User mailing lists. I am trying to get an SIPP end-point to register with a kamailio proxy. I do this by sending the REGISTER from the SIPP end-point, which is successfully responded to by kamailio with a 200 OK. No problems there. The problem is then with sending an INVITE from kamailio to the SIPP instance. This is where the problems arise. 1) If I include the expected INVITE in the same xml scenario file as the REGISTER/200, then SIPP will complain about the INVITE as it will have a different Call-ID from the REGISTER method. I have a hunt around the forums and this was acknowledged as an issue but as far as I can see this has not been fixed/implemented. The following link will take you to a proposed fix for this issue. http://sipp.sourceforge.net/wiki/index.php/Patches#Pre.2FPost_scenarios Does anyone know what the current state of play is for this proposed patch or if there is another way to get around this issue? 2) I tried to REGISTER the SIPP endpoint in a single xml scenario file with kamailio. This works as per usual. I then killed the SIPP instance and ran a new SIPP script listening on the same port before trying to send the INVITE to it. I expected this to work as the SIPP scenarios (both sending REGISTER and expecting INVITE) listened on the same port. However, the INVITE was not received by the SIPP endpoint. Can anyone think of a reason for this? If any further info is required then please let me know. I have spent quite a bit of time on this and its an itch that needs scratched. Thanks in advance, Steve. -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Stephen McVarnock, AePONA Ltd, Interpoint Building, 20-24 York Street, Belfast BT15 1AQ Tel: +44 (0)28 9026 9114 Fax: +44 (0)28 9026 9111 http://www.aepona.com ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

3 3

Faxing With Kamailio
by Amit Nepal 04 Jan '11

04 Jan '11

I was trying to use fax with kamailio / asterisk setup and sipce if possible. In asterisk we had to enable udptl for faxing to work. But i couldnot figure out where to start with for faxing in kamailio /asterisk or sipce. Could someone guide me to the right direction please ? Thank you Amit

2 1

Support for via ;maddr
by Olle E. Johansson 04 Jan '11

04 Jan '11

Friends, We've had an interesting discussion on the Asterisk-dev mailing list about supporting the ;maddr and ;ttl attributes in the via header when sending responses. We've agreed that it should be considered harmful and suggest making it configurable whether to support it in Asterisk. My question now is - Does kamailio support this automatically? - Can I disable it? Regards, /O

4 3

PATCH: parallel debian build support
by Noa Resare 04 Jan '11

04 Jan '11

Friends, The attached patch implements parallel building of the debian packages using the -j parameter to make. Predictably this gives package building a serious speed boost on an 8 way xeon with solid state drives :) To make things even simpler, I've pushed a branch debian-packaging to git:// git.resare.com/kamailio that you can pull from and merge with master using the following commands: git remote add resare git://git.resare.com/kamailio git fetch resare <check out the resare/debian-packaging branch, inspect> <merge it into master> A note on the patch: - I've replaced the DEB_BUILD_OPTIONS:="$(DEB_BUILD_OPTIONS) nostrip" (which puts literal quote characters in the make variable with a construct using the += operator. -It passes the basic test of building packages without failing with the DEB_BUILD_OPTIONS environment variable set to parallel=8 as well as unset on squeeze and lenny. /noa -- Everything is secret.

3 4

Re: [SR-Users] [Sipp-users] Use of SIPP along with kamailio - REGISTER followed by INVITE not working
by Stephen McVarnock 04 Jan '11

04 Jan '11

Hi, I have got the second scenario here to work i.e. REGISTER xml ran, kill sipp, run sipp with INVITE xml. There seems to be timing issue associated with this though - if I leave to long a delay between killing REGISTER xml and running INVITE xml then the INVITE will not be received by the sipp script. The Expires header for the REGISTER is set to 36000, so that is not the issue. Maybe this is a kamalio problem - anyway, for my purposes (as long as I am quick enough!), this works. Regards, Steve. > On 2011-01-04 08:59, Stephen McVarnock wrote: > >> http://sipp.sourceforge.net/wiki/index.php/Patches#Pre.2FPost_scenarios >> >> Does anyone know what the current state of play is for this proposed >> patch or if there is another way to get around this issue? >> > > Well, I developed this extension from june until december 2006 but we > never managed to merge this branch into the main-tree. > > >> 2) I tried to REGISTER the SIPP endpoint in a single xml scenario file >> with kamailio. This works as per usual. I then killed >> the SIPP instance and ran a new SIPP script listening on the same port >> before trying to send the INVITE to it. I expected this to work >> as the SIPP scenarios (both sending REGISTER and expecting INVITE) >> listened on the same port. However, the INVITE was not >> received by the SIPP endpoint. Can anyone think of a reason for this? >> > > Well, this is a standard task of sipp and usually works without any > limitations. Can you send both scnearios (Register and UAS) as well as > the used command line parameters? > > br > Michael > > -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Stephen McVarnock, AePONA Ltd, Interpoint Building, 20-24 York Street, Belfast BT15 1AQ Tel: +44 (0)28 9026 9114 Fax: +44 (0)28 9026 9111 http://www.aepona.com ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

2 1

Social Networking Event, Irvine, CA - Jan 25, 2011
by Daniel-Constantin Mierla 03 Jan '11

03 Jan '11

Hello, since I will spend some days in California this month, it is a good opportunity to host the first Social Networking Event for Kamailio & SER in 2011. I posted more details about it on Kamailio's project site, short link: * http://asipto.com/u/q Event is open for anyone interested to meet folks around the project or SIP and VoIP. Drop me an email if you want to participate. As for Europe, there will be most probably two similar events, one in Brussels (during Fosdem) and one in Barcelona -- more details to be posted very soon. Looking forward to meeting many of you during 2011. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

Re: [SR-Users] Change the SIP-To-User
by Bernhard Suttner 03 Jan '11

03 Jan '11

Hi, thank you very much for both answers. I did not know the pseudo variable magic, that it is possible to do a $tU = "newvalue" Amazing! Best regards, Bernhard Suttner ----- Original Message ----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Mon, 03 Jan 2011 18:42:47 +0100 Subject: Re: [SR-Users] Change the SIP-To-User > > > Am 03.01.2011 16:51, schrieb Bernhard Suttner: > > Hi, > > > > what is the best way to replace the userpart of the SIP-To URI with the > content of a pseudo variable? > > textops module: > > e.g. use > http://www.kamailio.org/docs/modules/3.1.x/modules_k/textops.html#id2910302 > > you could also remove_hf("to") and add a new header > append_hf("To: ..."); > > some devices do not like if the To gets changed. In this case you have > to reverse the change in the responses and keep track of cahnges for > in-dialog messages. > > Anyway - at the least the totag must not be changed (if present) > > regards > klaus >

1 0

Replace number in pseudo variable
by Bernhard Suttner 03 Jan '11

03 Jan '11

Hi, I try to replace the user-name of the Request URI and save that into a AVP later. I tried: $var(old_user) = $oU; $var(new_uri) = $(ou{re.subst,/$var(old_user)/004444444444/g}); and $var(new_uri) = $(ou{re.subst,/$oU/004444444444/g}); Both did not work: [lvalue.c:351]: non existing right pvar I am using latest git version of kamailio. Does someone has an idea? Best regards, Bernhard Bernhard Suttner Senior Entwickler Entwicklung _______________________ Winet Network Solutions AG Täfernstrasse 2a CH-5405 Baden-Dättwil E-Mail Winet: info(a)winet.ch E-Mail pers.: bernhard.suttner(a)winet.ch Voice: +41 56 470 4626 Fax: +41 56 470 4627

3 3

Segmentation fault when loading dialplan
by Javier Gallart 03 Jan '11

03 Jan '11

Hello I'd like to do some tests with the dialplan module, but Kamailio is crashing when it loads it. Actually the module is properly loaded if the table is empty, but it fails when a row is added; so I guess it's related with the way I've inserted the values. A core file is generated: Core was generated by `./kamailio -f ../etc/kamailio/kamailio.cfg'. Program terminated with signal 11, Segmentation fault. #0 build_rule (values=0x0) at dp_db.c:439 439 new_rule->dpid = VAL_INT(values); This is the relevant config: #----- dialplan params --- modparam("dialplan", "db_url", "mysql://xxxx:yyyyy@localhost/db") modparam("dialplan", "table_name", "dialplan") modparam("dialplan", "dpid_col", "dpid") modparam("dialplan", "attrs_pvar", "$avp(s:dp_attrs)") mysql> describe dialplan; +-----------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +-----------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | dpid | int(11) | NO | | NULL | | | pr | int(11) | NO | | NULL | | | match_op | int(11) | NO | | NULL | | | match_exp | varchar(64) | NO | | NULL | | | match_len | int(11) | NO | | NULL | | | subst_exp | varchar(64) | NO | | NULL | | | repl_exp | varchar(32) | NO | | NULL | | | attrs | varchar(32) | NO | | NULL | | +-----------+------------------+------+-----+---------+----------------+ 9 rows in set (0.00 sec) mysql> select * from dialplan; +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^00 | 0 | ^00 | 011 | dp1 | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ 1 row in set (0.00 sec) I'm probably making some stupid mistake...any clue? Thanks in advance Javier

3 21

MI FIFO command dlg_end_dlg in Kamailio 1.5.x
by Alex Balashov 03 Jan '11

03 Jan '11

Greetings, I have not tried in 3.0+, but in Kamailio 1.5.x, the documentation says that the MI FIFO command 'dlg_end_dlg' will only terminate confirmed dialogs. That's fair. However, I am concerned about what it does with unconfirmed dialogs (e.g. early dialogs). When I run 'kamctl fifo dlg_end_dlg SOME_HASH_ENTRY SOME_HASH_ID' it returns a positive value as if the dialog were deleted, but it is not actually deleted. It continues to show up in 'kamctl fifo profile_list_dlgs SOME_PROFILE SOME_VALUE', and continues to be linked to the profile, which is a problem if profiles are being used to enforce concurrent channel limits. Subsequent invocations of 'kamctl fifo dlg_end_dlg SOME_HASH_ENTRY SOME_HASH_ID' return: 404 Requested dialog not found. Is this a bug? A feature? What should one do with these permanent suspended dialogs other than to restart the proxy? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 1

kamailio logging
by Noa Resare 03 Jan '11

03 Jan '11

Is there a way to get a sensible amount of logging done from kamailio? Being new to the VoIP space (but with plenty of experience with i.e. web and email servers) I tried to get a running system by installing the kamailio package and starting up, trying to get it to behave by looking at log output. I use the blink SIP client in OSX, that has a SIP data dump log view and from that I can deduce that the server returns "SIP/2.0 483 Too Many Hops", however there is nothing helpful in the server logs. If I set debug=2 i get nothing useful, and if I set debug=3 I get insane amounts of logging, thousands of lines per connection attempt. Finding something meaningful in there seems like a herculean task. /noa ps. The actual problem with 483 Too Many Hops sounded a lot like 'Mail loops back to myself' that you get on smtp servers when the relevant virtual domain is not configured, so it seems I was able to handle that problem by adding WITH_MULTIDOMAIN and configuring my domain with kamctl domain add. Now I get "482 Loop Detected" instead, which seems like a non-fatal condition. The logging question still stands though. -- Everything is secret.

5 4

Happy New Year!
by Daniel-Constantin Mierla 03 Jan '11

03 Jan '11

Looking back to 2010, it was an amazing year - two major releases v3.0.x and v3.1.0, lot of new features, all in top of a better and more scalable core we have now after the integration of Kamailio with SER. Just to mention few here: asynchronous TCP and TLS, embedded Lua, XCAP server, HTTP server, pre-processor directives, full SCTP implementation with multi-homing, geoip API, asynchronous message queues, over 20 new modules -- for sake of completion, here are the links with the announcements: http://www.kamailio.org/w/kamailio-openser-v3.0.0-release-notes/ http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/ As for 2011, we will continue to deliver new features, but maintain as well our line of scalable, secure and rock solid SIP server. New projects related to Kamailio already announced their launch, participation to VoIP and Open Source events already booked, new thinks are baked and getting ready to launch soon -- 2011 is going to be for sure fascinating as well as challenging. Thanks for your support for the project! I wish you a great 2011 in personal life and business, enjoy tonight party! Happy New Year! Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

2 1

Deploy siremis with nginx
by Adieu 02 Jan '11

02 Jan '11

Hi everybody, I was trying to deploy siremis with nginx last night. Here are some problems I encountered during the installation process. Siremis has apache virtualhost settings. I just read the apache version and wrote nginx version as below: server { listen 80; server_name DOMAIN; index index.html index.htm index.php; root /path/to/siremis; location ~ .*\.(php|php5)?$ { fastcgi_pass 127.0.0.1:9000; fastcgi_index index.php; include fcgi.conf; } location ~^/siremis(.+)$ { try_files $uri $uri/ /siremis/index.php?$1; } } And I had to modify /siremis/bin/_forward.php a little bit to make siremis compatible with nginx. Here is the patch: *** _forward.php.old 2011-01-02 02:59:38.000000000 +0000 --- _forward.php 2011-01-01 16:19:26.000000000 +0000 *************** *** 16,25 **** --- 16,29 ---- if($_SERVER["REDIRECT_QUERY_STRING"]) { $url = $_SERVER["REDIRECT_QUERY_STRING"]; } + elseif($_SERVER["QUERY_STRING"]) + { + $url = $_SERVER["QUERY_STRING"]; + } elseif(preg_match("/\?\/?(.*?)(\.html)?$/si", $_SERVER['REQUEST_URI'],$match)) { //supports for http://localhost/?/user/login format //supports for http://localhost/index.php?/user/login format $url = $match[1]; The rest of the installation process is identical to the apache version from the document. Hope this can be helpful. Ivan Diao

1 0

some issues with database initialization (berkeley/txt)
by Noa Resare 31 Dec '10

31 Dec '10

Hello Friends, I thought I'd set up a SIP server during the holidays and kamailio seemed like a good choice. Since my setup won't be handling large amounts of users, a file based backend seemed like a good choice. Since I didn't find any "new users, please start with these steps" instruction on the web page I just installed the base package for Debian squeeze along with kamailio-berkeley-modules and proceded to try to configure the database. I set DBENGINE=DB_BERKELEY in /etc/kamailio/kamctlrc and ran 'kamdbctl create', which fails with this message: root@uma:/home/noa# kamdbctl create db4.6_load: /usr/share/kamailio//db_berkeley/kamailio/lcr_gw: reopen: No such file or directory ERROR: Creating standard tables failed! I tried DBENGINE=DBTEXT, and the command failed similarly: INFO: creating DBTEXT tables at: /usr/local/etc/kamailio/dbtext ... cp: cannot stat `/usr/share/kamailio//dbtext/kamailio/lcr_gw': No such file or directory ERROR: Creating core tables failed! Based on this I have a few questions: 1) Is there a text somewhere outlining the recommended steps to get a basic server going using the .deb packages? 2) What is the recommended backend for the least amount of install time trouble? 3) Is the issue I got into above a known problem? Is there a fix available? I asked my friend google, but she didn't seem to have an easily findable solution. /noa -- Everything is secret.

3 12

fifo dp_translate
by Borin 31 Dec '10

31 Dec '10

Hello everyone is there a possibility to check working dialplan rules in kamailio (ver 1.4) with the command kamctl fifo dp_translate ? "fifo which" shows that there is such command but I can't figure out how to use it. some rules are not working and I want to check if kamailio loads everything from dialplan table correctly. Mainly I use fifo commands to reload dialplan and dispatcher so fifo is configured right I suppose.

3 7

function get_user_group: parameter 1 is not constant
by Jijo 30 Dec '10

30 Dec '10

I'm getting following config error when i use the api from group module. CSBC-JJ:/srv/www/htdocs/core # /usr/sbin/kamailio -c -f /etc/kamailio/kamailio.cfg loading modules under /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ 0(5020) : <core> [cfg.y:3412]: parse error in config file /etc/kamailio/kamailio.cfg, line 455, column 20-36: function get_user_group: parameter 1 is not constant 0(5020) : <core> [cfg.y:3415]: parse error in config file /etc/kamailio/kamailio.cfg, line 455, column 48: bad command 0(5020) INFO: pv [pv_shv.c:61]: locks array size 16 ERROR: bad config file (2 errors) I have used the API get_user_group as shown below $avp(r_contact)=$ru; $avp(temp)=$null; get_user_group($avp(reg_contact),$avp(temp)); xlog("L_ERR", "request - M=$rm get_user_group=$avp(temp)\n\n"); Please let me know what could be reason for this error. Thanks Jijo

2 1

SIP CE vmware and vbox image
by Amit Nepal 30 Dec '10

30 Dec '10

I wanted to see how the sipce works and how it looks like , as Andreas suggested me to. But when i try to start the virtual machine i get the following error : This kernel requires an x86-64 CPU, but only detected an i686 CPU. Unable to boot - please use a kernel appropriate for your CPU I tried with both vmware and vbox image. I am trying to run the image on vmware infrastrcture running on dell dual xeon cpu. -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

3 2

Re: [SR-Users] Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration
by Kurt Mullen 29 Dec '10

29 Dec '10

Apparently the process of the installation did not setup even a default sip.conf file, and there were some other issues with the Kamailio.cfg file. Most of the issues were resolved by one of the other users in the group, and we were able to get the system running. Thanks everyone for your assistance. -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: Wednesday, December 29, 2010 11:49 AM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 67, Issue 75 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database (Klaus Darilion) 2. Load Balancing (Tim King) 3. Re: Load Balancing (Alex Balashov) 4. fifo dp_translate (Borin) 5. Re: some issues with database initialization (berkeley/txt) (Noa Resare) 6. Re: some issues with database initialization (berkeley/txt) (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Wed, 29 Dec 2010 15:17:41 +0100 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> Subject: Re: [SR-Users] Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database To: sr-users(a)lists.sip-router.org Message-ID: <4D1B4305.30504(a)pernau.at> Content-Type: text/plain; charset=windows-1252; format=flowed On 28.12.2010 22:07, Kurt Mullen wrote: > I am on my 20^th (I?m not kidding) attempt to successfully complete > this tutorial. > > I have installed on Ubuntu 10.10 x64 Server. I installed Kamailio & > Asterisk on the same server as in the tutorial. > > I have two SIP clients registered, but they are not able to call each other. > > No one answered my last two posts, so I hope someone can help this time. If you can describe the problem (why can't they call each other) we can help you. So, first you have to find out what is the problem. Use a packet sniffer (wireshark, ngrep) to capture the SIP traffic and analyze it (yes, if you want to use Kamailio you have to know at least the basics of SIP) for the problems. regards klaus ------------------------------ Message: 2 Date: Wed, 29 Dec 2010 09:31:28 -0500 From: Tim King <tim(a)compnetwork.net> Subject: [SR-Users] Load Balancing To: sr-users(a)lists.sip-router.org Message-ID: <AANLkTindx_r-RmTJSjTnpfdmLX8Nespzs_M9O0f6y385(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" I have read many examples and it seems I have found several pieces of how to do this but no real example of how to achieve this. I followed this setup guide: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc Performed this Kamailio install: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I have found the page for the dispatcher module here: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html I also am curious if this dispatcher setup can be driven solely from the dispatcher mysql table or if the dispatcher.list file is required. Can someone point me towards the proper documentation that truly show how to configure these features. *SIP Routing Capabilities* - - NAT traversal support for SIP and RTP traffic - load balancing with many distribution algorithms and failover support - flexible least cost routing - routing failover - replication for High Availability (HA) * Load Based Load Balancing ** ------------------------- ** ** +----------+ ** | Kamailio | ** +----------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 1 | [Calls 1,3,7] ** | +--------------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 2 | [Calls 2,6] ** +--------------+ *

2 1

Load Balancing
by Tim King 29 Dec '10

29 Dec '10

I have read many examples and it seems I have found several pieces of how to do this but no real example of how to achieve this. I followed this setup guide: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc Performed this Kamailio install: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I have found the page for the dispatcher module here: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html I also am curious if this dispatcher setup can be driven solely from the dispatcher mysql table or if the dispatcher.list file is required. Can someone point me towards the proper documentation that truly show how to configure these features. *SIP Routing Capabilities* - - NAT traversal support for SIP and RTP traffic - load balancing with many distribution algorithms and failover support - flexible least cost routing - routing failover - replication for High Availability (HA) * Load Based Load Balancing ** ------------------------- ** ** +----------+ ** | Kamailio | ** +----------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 1 | [Calls 1,3,7] ** | +--------------+ ** | ** | +--------------+ ** |-------| FreeSWITCH 2 | [Calls 2,6] ** +--------------+ *

3 3

Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database
by Kurt Mullen 29 Dec '10

29 Dec '10

I am on my 20th (I'm not kidding) attempt to successfully complete this tutorial. I have installed on Ubuntu 10.10 x64 Server. I installed Kamailio & Asterisk on the same server as in the tutorial. I have two SIP clients registered, but they are not able to call each other. No one answered my last two posts, so I hope someone can help this time. Kurt Mullen

3 2

LCR and gateways resource.
by Ricardo Martinez 29 Dec '10

29 Dec '10

Hello. I’m looking for a way to mix the features of LCR and a counter to keep track of the used resource of a gateway. Let me explain. Let’s suppose that I’m using LCR module to route the calls to all my gateways, in particular I have the calls with prefix 55 routed to the group “55” which are 3 gateways : GW1, GW2 and GW3. Let’s suppose also that each gateway is connected to digital E1 with 30 channels. At certain time of the day one of the gateways is at full capacity using the 30 channels. This is what I wanted to do : can I have a variable with the maximum channels available for a certain gateway, and for each call going to that gateway a counter?. When the counter reach the maximum, can I remove the gateway from the LCR choosing algorithm until it has capacity again? Can I have some thoughts about this idea? Is there a way to achieve this ? Thanks in advance. Regards, Ricardo.-

2 3

Swap of From and To - kamailio 3.1.x problem?
by Jeremya 28 Dec '10

28 Dec '10

Hi I have a problem which pops up occasionally. My scenario is a kamailio 3.0.x server coupled with SEMS 1.2.1 Calls go to SEMS and occasionally it will issue a refer to the caller (an SPA962 linksys) The problem is that the refer seems to get screwed up somewhere and the referred call changes the From field to the To field - in this case 711. The problems start below at 15:58:21 after the refer I'd like to know if this is a problem with Kamailio or a problem with the phone. The call fails on the SEMS_SEND section when the From field is validated. I suspect the phone is the culprit, but external advice appreciated. I'm not quite sure what the s_xxx call id means? Here is the log Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (New request) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 ( reply etc ) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (Loose route) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (RELAY, start of processing) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (New request) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (handle initial request) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (t_check_trans) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (Request For Me) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS, rewrites) Method:<INVITE> r-uri <sip:020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS_SEND) Method:<INVITE> r-uri <sip:020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[398]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS_SEND_REPLY) Method:<INVITE> Status:<603> Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[398]: CRITICAL: <script>: s_275259592d_358d20ea49bf (SEMS_SEND_FAIL) Method:<INVITE> Status:<<null>> Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (New request) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf ( reply etc ) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (Not a loose route) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (ACK in lr) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08

2 1

Crash
by michel freiha 28 Dec '10

28 Dec '10

Hi all, I have latest version of kamailio but it keep crashing...I know that the log could not be sufficient but maybe anyone faced the same issue before...Please find log below and let me know if you can find anything Nov 24 15:56:23 tunnel /usr/local/sbin/kamailio[19943]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 Nov 24 15:56:23 tunnel /usr/local/sbin/kamailio[19929]: ALERT: <core> [main.c:741]: child process 19935 exited by a signal 11 Nov 24 15:56:23 tunnel /usr/local/sbin/kamailio[19932]: INFO: carrierroute [cr_func.c:595]: uri 919437452006 was rewritten to sip:919437452006@x.x.x.x, carrier 5, domain 0 Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19930]: INFO: carrierroute [cr_func.c:595]: uri 008801823703200 was rewritten to sip:008801823703200@X.X.X.5, carrier 5, domain 0 Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19933]: ERROR: <script>: CARRIERROUTE_time_dbg:method=INVITE;time=1290614183;callid=14b6536a-d102-1910-9b23-9f3b3c3065ca - before cr Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19929]: ALERT: <core> [main.c:744]: core was generated Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19932]: INFO: <script>: time_dbg:method=SUBSCRIBE;time=1290614183;callid=YjVmODYzYjQxYWY0ZTU3MGRmMmZkYjRhOTg4ZGZmODQ. - before cr Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19930]: INFO: <script>: time_dbg:method=INVITE;time=1290614183;callid=cdc9556a-d102-1910-8b30-926b495dad44 - before cr Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19933]: INFO: carrierroute [cr_func.c:595]: uri 00919974265815 was rewritten to sip:00919974265815@X.X.X.X>, carrier 5, domain 0 Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19931]: ERROR: <script>: time_dbg:method=INVITE;time=1290614182;callid=14b6536a-d102-1910-9b23-9f3b3c3065ca - before t_relay Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19929]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19934]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19943]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19942]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19940]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19939]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19938]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19941]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19937]: INFO: <core> [main.c:807]: INFO: signal 15 received Nov 24 15:56:24 tunnel /usr/local/sbin/kamailio[19936]: INFO: <core> [main.c:807]: INFO: signal 15 received Regards

4 21

Merry Christmas!
by Daniel-Constantin Mierla 27 Dec '10

27 Dec '10

Hello, I could safety say that 2010 was the greatest year by achievements in the history of our SIP proxy server project so far. It will not be short to summarize it, so I will do a separate post for that. Now I just wast to thank everyone making that possible and wish you a Merry Christmas! Great winter holidays to all Kamailians and SERians!!! Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

3 3

Puzzling swap of To and From - Kamailio 3.0x
by Jeremy Ardley 24 Dec '10

24 Dec '10

Hi I have a problem which pops up occasionally. My scenario is a kamailio 3.0.x server coupled with SEMS 1.2.1 Calls go to SEMS and occasionally it will issue a refer to the caller (an SPA962 linksys) The problem is that the refer seems to get screwed up somewhere and the new call changes the From field to the To field - in this case 711. The problems start below at 15:58:21 after the refer I'd like to know if this is a problem with Kamailio or a problem with the phone. The call fails on the SEMS_SEND section when the From field is validated. I suspect the phone is the culprit, but external advice appreciated. Here is the log Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (New request) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 ( reply etc ) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (Loose route) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:20 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: f43507d4-e02f109b(a)10.1.39.192 (RELAY, start of processing) Method:<REFER> r-uri <sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08> To: sip:711@joondalup.fesa.sto From: sip:902@fesa.sto Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (New request) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (handle initial request) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (t_check_trans) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (Request For Me) Method:<INVITE> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS, rewrites) Method:<INVITE> r-uri <sip:020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[399]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS_SEND) Method:<INVITE> r-uri <sip:020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[398]: WARNING: <script>: s_275259592d_358d20ea49bf (SEMS_SEND_REPLY) Method:<INVITE> Status:<603> Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[398]: CRITICAL: <script>: s_275259592d_358d20ea49bf (SEMS_SEND_FAIL) Method:<INVITE> Status:<<null>> Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (New request) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf ( reply etc ) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (Not a loose route) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08 Dec 24 15:58:21 mc0 /usr/local/sbin/kamailio[404]: WARNING: <script>: s_275259592d_358d20ea49bf (ACK in lr) Method:<ACK> r-uri <sip:7020@fesa.sto> To: sip:7020@fesa.sto From: sip:711@10.50.20.10:5070;LINEID=fac9f6b7cb08

1 0

Billing
by Deon Vermeulen 24 Dec '10

24 Dec '10

Good Day list Is there someone that could guide me to installation guide for jBilling or recommend any other Free OpenSource Billing? Features Looking for: - Wholesale - Rating - Pre-Paid/Post-Paid Thank you very much Regards Deon

2 1

Source IP address in Failure_route
by Ricardo Martinez 24 Dec '10

24 Dec '10

Hello. I’m trying to obtain the source IP address from a reply message. This is part of the configuration file : failure_route[1] { if (t_was_cancelled()) { xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled in failure_route\n"); exit; } if ( ($T_rpl($si) != "10.0.0.4") || ($T_rpl($si) != "10.0.0.3") ) { if ( t_check_status("48[0-9]") ) { xlog("L_INFO","[$ci] 48X Received\n"); exit; } } … But it seems not to be working. Is there a way to do what I want? Thanks in advance, Regards, Ricardo.- * *

4 5

[dialog] No dialog stored if Via has no branch
by Iñaki Baz Castillo 23 Dec '10

23 Dec '10

Hi, using Kamailio 1.5.4. I receive llamadas from a Cisco pre-RFC 3261 (non symmetric SIP and no ;branch parameter). It seems that dialog module cannot account such calls. Maybe branch parameter is required? -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 4

kamailio restart and TLS ( relay_to_tls() )
by Juha Heinanen 22 Dec '10

22 Dec '10

Of course this won't work if you route to the other proxies also using lookup() with permanent entries in location table. If you really have such a setup you can try to set a certain flag for this entries and then do not call set_forward_no_connect(). klaus, i checked and i'm already calling set_forward_no_connect() when initial request is forwarded to a tcp contact. however, it seems more tricky to prevent useless tcp connection attempts for in-dialog requests. for example, here the UA has gone away when presence server sends a notify to it: Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: INFO: Routing in-dialog NOTIFY <sip:gmuodqxs@192.98.102.10:50279;transport=tcp> from <sip:jh@vm.test.fi> to <sip:192.98.102.10:60550;transport=tcp> Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: ERROR: <core> [tcp_main.c:2743]: connect 192.98.102.10:60550 failed (RST) Connection refused Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: ERROR: <core> [tcp_main.c:2753]: 192.98.102.10:60550: connect & send for 0xb2d1c490 failed: Connection refused (111) Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: ERROR: tm [../../forward.h:169]: msg_send: ERROR: tcp_send failed Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: ERROR: tm [t_fwd.c:1382]: ERROR: t_send_branch: sending request on branch 0 failed Dec 22 14:50:48 sip /usr/sbin/sip-proxy[3207]: INFO: Failed to relay in-dialog request NOTIFY <sip:gmuodqxs@192.98.102.10:50279;transport=tcp> it does not seem like a good idea to call set_forward_no_connect() for all in-dialog requests. for example, presence server might have been restarted when in-dialog subscribe arrives to it and it would make sense to for sr to try to setup a tcp session with presence server. same for pstn gateways and other proxies. -- juha

2 1

RTP to SRTP bridge
by Dominguez Jover, Ricardo 22 Dec '10

22 Dec '10

Hi everybody, I'm trying to deploy an scenario where all calls in the LAN are encrypted with SRTP, so I've forced my sip softphones to "mandatory SRTP". As in the Internet not all providers and clients have SRTP enabled, I couldn´t communicate them in this way, so I would need some kind of border element to bridge RTP and SRTP calls. Is there any bridge/proxy who can receive rtp packets and convert them into srtp and viceversa? I've been reading RTPproxy and Media proxy docs and also some forums, but I didn´t find anything useful. Regards, Ricardo Dominguez

2 1

Re: [SR-Users] 404 Not Found - If User Not Registered
by Klaus Darilion 22 Dec '10

22 Dec '10

Am 22.12.2010 13:29, schrieb David J.: > Klaus; > > The docs don't seem so clear as to which state is User Exists and Not > Registered vs No User Exists; You can not tell if a user exists by querying the location table - you can only tell if a user is registered. To find out if a user exists you would have to query the subscriber table. regards klaus

1 0

RLS issue
by "Andrés S. García Ruiz" 22 Dec '10

22 Dec '10

Hi everybody, I'm using the Kamailio RLS+Presence module integrated with OpenIMSCore. All the servers are running properly but when sending a RLS subscribe. The flow of SIP messages is the following: A.1) Watcher ---- SUBSCRIBE (rls-services uri) ---> RLS SUBSCRIBE sip:restricted_areas@open-ims.test SIP/2.0 Call-ID: dcKTUqpGi8lsPuQzqI0m922XEmSQpAxU36XScBNuu1A. CSeq: 1 SUBSCRIBE From: "restricted_areas" <sip:restricted_areas@open-ims.test>;tag=11afb48e To: <sip:restricted_areas@open-ims.test> Via: SIP/2.0/TCP 155.54.190.166:8060;rport;branch=z9hG4bK-d8754z-29095b2cee722e34-1---d8754z- Max-Forwards: 70 Event: presence Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/auth-policy+xml Expires: 30000 Contact: <sip:restricted_areas@155.54.190.166:8060> Supported: eventlist Route: <sip:orig@scscf.open-ims.test:6060;lr> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Thu_Dec_16_10-48-31_CET_2010 Content-Length: 0 A.2) RLS ---- 200 Ok ----> Watcher B.1) RLS ---- SUBSCRIBE (presentity in the rls-services user's list) ---> I-CSCF SUBSCRIBE sip:restricted_areas@open-ims.test SIP/2.0 Record-Route: <sip:mo@scscf.open-ims.test:6060;lr> Route: <sip:ciervo.inf.um.es:5060;lr>, <sip:iscmark@scscf.open-ims.test:6060;lr;s=1;h=0;d=0;a=7369703a726573747269637465645f6172656173406f70656e2d696d732e74657374> Record-Route: <sip:mo@pcscf.open-ims.test:4060;lr> Call-ID: dcKTUqpGi8lsPuQzqI0m922XEmSQpAxU36XScBNuu1A. CSeq: 1 SUBSCRIBE From: "restricted_areas" <sip:restricted_areas@open-ims.test>;tag=11afb48e To: <sip:restricted_areas@open-ims.test> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKfc46.88774fc7.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKfc46.8dfcbef7.0;i=2 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=42816;branch=z9hG4bK-d8754z-29095b2cee722e34-1---d8754z- Max-Forwards: 15 Event: presence Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/auth-policy+xml Expires: 30000 Contact: <sip:restricted_areas@155.54.190.166:8060> Supported: eventlist Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Thu_Dec_16_10-48-31_CET_2010 Content-Length: 0 P-Asserted-Identity: <sip:restricted_areas@open-ims.test> P-Charging-Vector: icid-value="P-CSCFabcd000000004d09e21300000005";icid-generated-at=155.54.210.134;orig-ioi="open-ims.test" B.2) I-CSCF ---- QUERY ----> HSS B.3) I-CSCF ---- SUBSCRIBE ----> S-CSCF SUBSCRIBE sip:testuser01@open-ims.test SIP/2.0 Route: <sip:scscf.open-ims.test:6060> Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 To: sip:testuser01@open-ims.test From: sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 CSeq: 10 SUBSCRIBE Call-ID: 7fd8dfdd-21694(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.2.0-dev1 (i386/linux)) Max-Forwards: 16 Event: presence Contact: <sip:155.54.190.245:5060;transport=udp> Expires: 10810 Max-Forwards: 70 Support: eventlist B.4) S-CSCF ---- SUBSCRIBE ----> P-CSCF SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0 SIP/2.0 Record-Route: <sip:mt@scscf.open-ims.test:6060;lr> Route: <sip:term@pcscf.open-ims.test:4060;lr> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bK07df.8a35e4f3.0 Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 To: sip:testuser01@open-ims.test From: sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 CSeq: 10 SUBSCRIBE Call-ID: 7fd8dfdd-21694(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.2.0-dev1 (i386/linux)) Max-Forwards: 15 Event: presence Contact: <sip:155.54.190.245:5060;transport=udp> Expires: 10810 Max-Forwards: 70 Support: eventlist P-Called-Party-ID: <sip:testuser01@open-ims.test> B.5) P-CSCF ---- SUBSCRIBE ----> Presentity??? SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0 SIP/2.0 Record-Route: <sip:mt@pcscf.open-ims.test:4060;lr> Record-Route: <sip:mt@scscf.open-ims.test:6060;lr> Via: SIP/2.0/TCP 155.54.210.134:4060;branch=z9hG4bK07df.baa7cf24.0 Via: SIP/2.0/UDP 155.54.210.135:6060;received=155.54.210.135;rport=6060;branch=z9hG4bK07df.8a35e4f3.0 Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 To: sip:testuser01@open-ims.test From: sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 CSeq: 10 SUBSCRIBE Call-ID: 7fd8dfdd-21694(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.2.0-dev1 (i386/linux)) Max-Forwards: 14 Event: presence Contact: <sip:155.54.190.245:5060;transport=udp> Expires: 10810 Max-Forwards: 70 Support: eventlist P-Called-Party-ID: <sip:testuser01@open-ims.test> The step B.5 is sent directly to the presentity testuser01. Instead of that, I thought that message was suppose to be sent to the presence server in the URI that is shown in Contact header. Regards, Andrés.

2 2

kamailio restart and TLS ( relay_to_tls() )
by Dominguez Jover, Ricardo 22 Dec '10

22 Dec '10

Hi everybody, Since I implemented Kamailio 3.1 with TLS I've found a strange behavior. That is, with some clients (Bria and Blink) registered, if I restart Kamailio, then when the clients re-register the strange behaivour happens. This behavior consist on receiving calls, it took about 15 seconds to receive the first tone since the call was made. I made the following modification in the "route[Relay]" config. The reason is I wanted my gateway and Kamailio to make signaling by TLS. Without this modification the signaling was unencrypted (SIP): route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } # Se comunica con el GWa traves de TLS if(!( ($od=~"mydomain.com") && ( ($rU=~"[a-z]{3,20}$") || ($rU=~"^xx[0-9][0-9]$") ) ) ) { ### If I'm calling a PBX extension do the signaling by TLS with the gateway (Cisco 2811) if (!t_relay_to_tls()) { sl_reply_error(); } } else if { if (!t_relay()) { sl_reply_error(); } } exit; } The rest of functionalities are working really fine. Any idea about what is happening? Cheers! Ricardo Domínguez

3 9

404 Not Found - If User Not Registered
by David J. 22 Dec '10

22 Dec '10

if I do lookup() What case does lookup return if entry exists but user not registered? lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; should I just wrap this like this: if(registered("location")){ lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; } }; The purpose I ask this rather simple question is because I want to redirect to voicemail if user is not registered. In the default script we have this case; failure_route[FAIL_ONE] { # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } But it seems this case is never met; Because this code lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; Seems to send a 404 response to the UA rather than go to failure route? Maybe I am wrong... Any advice would be appreciated.

2 1

Re: [SR-Users] SER0.9.7: How to prevent: 403 Max Registrations per IP exceeded
by cerieljacobs 22 Dec '10

22 Dec '10

Trying the other registrations takes 10 seconds extra for each registration above the first: registrations time before phone starts ringing --------------- -------------------------------- 1 3s 2 13s 3 23s 4 33s 5 43s --------------- -------------------------------- Does anyone see a way to limit the number of registrations to allow only 1 per User-Agent/IP-address combination? Op 22 dec 2010, om 02:08 heeft David J. het volgende geschreven: > Look at the registration module; > > There is a max_contacts param; > > Its fine if the UA re-registers; Its when SER sends out to the UA; Its only going to reach the one which is available; > > It may try the others but they obviously wont be reachable. >

2 1

Re: [SR-Users] SER0.9.7: How to prevent: 403 Max Registrations per IP exceeded
by cerieljacobs 22 Dec '10

22 Dec '10

Regarding the business rule, can you explain more about > YOu can specify max contacts on the registration of an endpoint For instance, how and where is the max contacts specified? There still might be an issue, as it is pointless to store multiple registrations for a single user-agent that re-regisers (without un-registering) using the same IP address, like below: bluesip/user,sip:pb@86.8.210.156:49569;rinstance=991CCEE9;transport=tcp2010-12-22 01:26:360.50 bluesip/user,sip:pb@86.8.210.156:49584;rinstance=991CCEE9;transport=tcp2010-12-22 01:33:250.50 bluesip/user,sip:pb@86.8.210.156:49585;rinstance=991CCEE9;transport=tcp2010-12-22 01:35:330.50 bluesip/user,sip:pb@86.8.210.156:49586;rinstance=991CCEE9;transport=tcp2010-12-22 01:37:430.50 bluesip/user,sip:pb@86.8.210.156:49587;rinstance=991CCEE9;transport=tcp2010-12-22 01:39:510.50 bluesip/user,sip:pb@86.8.210.156:49588;rinstance=991CCEE9;transport=tcp2010-12-22 01:42:000.50 Only the last line is the active one. The first five registrations are old and overridden by the newer registration. For instance: ports 49569, 49584, 49585, 49586 and 49587 are no longer in the NAT routing table, only 49588 is. Op 22 dec 2010, om 01:03 heeft David J. het volgende geschreven: > Its not an issue; > > AG Projects may allow multiple registrations from the SAME IP whereas the other company may not; > > Thats a business rule; Not a bug; YOu can specify max contacts on the registration of an endpoint. > > Its not a bug. > > > > On 12/21/10 6:58 PM, cerieljacobs wrote: >> Bluesip.net uses "Sip EXpress router (0.9.7 (i386/linux))" >> and on client registration it returns >> "SIP/2.0 403 Max Registrations per IP exceeded" >> due to some bug/error/failure in the version 1.4 Acrobits Business softphone user-agent >> to unregister correctly. >> >> Strange is that this never happens with accounts on AG Projects, "OpenSIPS (1.4.5-notls (x86_64/linux))" server. This server software seems to remove old registrations for the same user-agent from the same ip address. >> >> What should be changed in Sip EXpress router, >> to store only one single registration per user-agent/IP combination? >> >> Thanks,Ceriel >> _______________________________________________ >> 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

SER0.9.7: How to prevent: 403 Max Registrations per IP exceeded
by cerieljacobs 21 Dec '10

21 Dec '10

Bluesip.net uses "Sip EXpress router (0.9.7 (i386/linux))" and on client registration it returns "SIP/2.0 403 Max Registrations per IP exceeded" due to some bug/error/failure in the version 1.4 Acrobits Business softphone user-agent to unregister correctly. Strange is that this never happens with accounts on AG Projects, "OpenSIPS (1.4.5-notls (x86_64/linux))" server. This server software seems to remove old registrations for the same user-agent from the same ip address. What should be changed in Sip EXpress router, to store only one single registration per user-agent/IP combination? Thanks,Ceriel

1 0

SBC on kamailio
by Henry Dogger 21 Dec '10

21 Dec '10

Hi all, We have a kamailio installation for our voip system and would like to add a SBC. We found the freeswitch possibility: http://wiki.freeswitch.org/wiki/SBC_Setup But in this case, 302 Redirect SIP is used, we would like the SBC to act as a pass-through controller, just to be a gateway for SIP clients from outside. Is something like this possible? And will registrations be handled correctly since all SIP messages will be on port 5060. Thanks in advance. Kind regards, Henry Dogger Telecats BV

3 7

Recursive failure_route after CFNA
by Igor Potjevlesch 21 Dec '10

21 Dec '10

Hello all, I have setup a mechanism to forward outgoing calls to another gateway if the first one replies with 5xx. I have simply used ‘t_check_status’ in the failure_route. This works. But, with a CFNA it doesn’t work : the incoming call is treated by SER (0.9.9). After the timer has expired the call goes to ‘failure_route’ for the first time to setup the CFNA with, finally, an ‘append_branch’. At this moment, the first gateway replies with 5xx and the call never goes to ‘failure_route’. I can match the 5xx within the ‘onreply_route’, but that’s all. So, instead of trying another gateway, the initial incoming calls terminate on the voicemail. Can anyone help me? Regards, Igor.

2 2

SIP UA's behind same NAT
by Deon Vermeulen 20 Dec '10

20 Dec '10

Hi List Can someone please help me? I need to implement PBX Services for a customer that has 10 IP Phones at his Main site and then 20 IP Phones located over 5 Branches. My goal is to have RTP go end to end and not proxied. I had a look at the below link and tried it, but with no luck. http://kamailio.org/dokuwiki/doku.php/examples:caller-callee-behind-same-nat Here is a partial output of my kamailio.cfg file if someone would be so kind to help me. #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@x.x.x.x") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } Thank you in advance Kind Regards Deon

2 5

CRASH on qm_free for simultaneous calls (V3.1.0)
by Jijo 20 Dec '10

20 Dec '10

I'm observing a crash on qm_free, when we made two outgoing calls to the same number. One SIP phone receives ring back tone and the other busy tone. The back trace is shown below which shows that it is crashed in abort(). How can the condition if (f->u.is_free){ could be true? Please let me know if anybody observed this issue and let me know how to debug this isssue. f=(struct qm_frag*) ((char*)p-sizeof(struct qm_frag)); #ifdef DBG_QM_MALLOC qm_debug_frag(qm, f); if (f->u.is_free){ LOG(L_CRIT, "BUG: qm_free: freeing already freed pointer," " first free: %s: %s(%ld) - aborting\n", f->file, f->func, f->line);* abort();* } MDBG("qm_free: freeing frag. %p alloc'ed from %s: %s(%ld)\n", f, f->file, f->func, f->line); #endif BACKTRACE ------------------- Program received signal SIGABRT, Aborted. 0xb76fd7a5 in raise () from /lib/libc.so.6 (gdb) bt full #0 0xb76fd7a5 in raise () from /lib/libc.so.6 No symbol table info available. #1 0xb76ff070 in abort () from /lib/libc.so.6 No symbol table info available. #2 0x081824d5 in qm_free (qm=0xaf7ee000, p=0xaf9ed758, file=0xb7659aed "tm: h_table.c", func=0xb7659c9c "free_cell", line=141) at mem/q_malloc.c:447 f = <value optimized out> prev = <value optimized out> next = <value optimized out> size = <value optimized out> #3 0xb75ff6b9 in free_cell (dead_cell=0xaf9e7054) at h_table.c:141 b = <value optimized out> i = <value optimized out> rpl = <value optimized out> tt = <value optimized out> foo = <value optimized out> cbs = <value optimized out> cbs_tmp = <value optimized out> __FUNCTION__ = "free_cell" #4 0xb7625eb0 in t_unref (p_msg=0x847bb10) at t_lookup.c:1553 kr = <value optimized out> #5 0xb764b3ad in w_t_unref (foo=0x847bb10, flags=2147483649, bar=0x0) at tm.c:707 No locals. #6 0x0811543a in exec_post_script_cb (msg=0x847bb10, type=REQUEST_CB_TYPE) at script_cb.c:195 cb = 0x8867874 flags = 2147483649 #7 0x080e5a4e in receive_msg ( buf=0x82a89e0 "ACK sip:0845@10.80.13.54:5060;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 10.200.3.39:5060;branch=z9hG4bK0a992e8d3052e85e5\r\nRoute: <sip:0845@10.200.0.31:5060;lr;transport=udp>\r\nMax-Forwards: 69\r\nFrom: 554"..., len=<value optimized out>, rcv_info=0xbf95b7fc) at receive.c:221 msg = 0x847bb10 ctx = {rec_lev = 137434160, run_flags = 0, last_retcode = -1080707352, jmp_env = {{__jmpbuf = {-1224488143, 137089696, 137434092, 0, -1080707248, -1080707304}, __mask_was_saved = 0, __saved_mask = {__val = {3079133424, 134556827, 3079002384, 0, 3214260024, 3070479287, 137089696, 137434160, 3070572886, 3070573396, 228, 137347864, 3214260056, 3077419768, 3079002384, 137089664, 4294967295, 3079131124, 134556827, 134535424, 1, 3079058382, 3079133864, 3079003192, 1, 1, 0, 134555968, 0, 136548964, 3077419768, 142823652}}}}} ret = <value optimized out> inb = { s = 0x82a89e0 "ACK sip:0845@10.80.13.54:5060;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 10.200.3.39:5060;branch=z9hG4bK0a992e8d3052e85e5\r\nRoute: <sip:0845@10.200.0.31:5060;lr;transport=udp>\r\nMax-Forwards: 69\r\nFrom: 554"..., len = 457} __FUNCTION__ = "receive_msg" #8 0x08175cb6 in udp_rcv_loop () at udp_server.c:532 len = <value optimized out> buf = "ACK sip:0845@10.80.13.54:5060;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 10.200.3.39:5060;branch=z9hG4bK0a992e8d3052e85e5\r\nRoute: <sip:0845@10.200.0.31:5060;lr;transport=udp>\r\nMax-Forwards: 69\r\nFrom: 554"... from = 0x88350e4 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {654559242, 0, 3214260312, 3079002384}, addr32 = {654559242, 0, 3214260312, 3079002384}, addr16 = { 51210, 9987, 0, 0, 47192, 49045, 55568, 46981}, addr = "\n\310\003'\000\000\000\000X\270\225\277\020Ù \267"}}, dst_ip = {af = 2, len = 4, u = {addrl = {520144906, 0, 0, 0}, addr32 = {520144906, 0, 0, 0}, addr16 = {51210, 7936, 0, 0, 0, 0, 0, 0}, addr = "\n\310\000\037", '\000' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = { s = {sa_family = 2, sa_data = "\023\304\n\310\003'\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = { s_addr = 654559242}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 654559242, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x82f4140, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #9 0x080ac1d4 in main_loop () at main.c:1554 i = <value optimized out> ---Type <return> to continue, or q <return> to quit--- pid = <value optimized out> si = <value optimized out> si_desc = "udp receiver child=0 sock=10.200.0.31:5060\000\b A\206\b\002\000\000\000\004\000\000\000\n\310\000\037L \225\257H\271\225\277@ \030\207\267\020i\203\b\006\000\000\000\000\340~\257\001\000\000\000\001\000\000\000\000\000\000\000\000\340~\257\004\000\000\000?)\037\b\001\000\000\000\a\000\000\000\000\000\000\000H\271\225\277\350\353\017\b" #10 0x080af2ec in main (argc=9, argv=0xbf95ba84) at main.c:2398 cfg_stream = 0x8cbe008 c = <value optimized out> r = <value optimized out> tmp = 0xbf95cdc0 "" tmp_len = -1217394389 port = <value optimized out> proto = <value optimized out> options = 0x81ef340 ":f:cm:dVhEb:l:L:n:vrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 3337198521 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = <value optimized out> dont_fork_cnt = 0 p = <value optimized out> Thanks Jijo

2 5

new: pkg monitoring and troubleshooting enhancements
by Daniel-Constantin Mierla 20 Dec '10

20 Dec '10

Hello, I have committed some code to allow easier troubleshooting and monitoring of PKG usage per process. One new functions are exported by cfgutils module: - pkg_summary() - can be used to dump summary of PKG usage Btw, there was also added shm_summary() that dumps the summary of SHM usage. The kex module exports a new rpc command 'pkg.stats' that prints PKG usage metrics (used, free, real_used) per process. Here is a sample output: sercmd> pkg.stats { entry: 0 pid: 4793 rank: 0 used: 124788 free: 3813512 real_used: 380792 } { entry: 1 pid: 4794 rank: 1 used: 132740 free: 3805512 real_used: 388792 } { entry: 2 pid: 4795 rank: 2 used: 132740 free: 3805512 real_used: 388792 } The adding free to real_used values results in pkg memory size per process (4MB in this example). To get these stats, you have to compile with MALLOC_STATS enabled (this is default for Kamailio, so nothing actually to do for it). One benefit of this pkg.stats is ability to monitor usage of private memory and trigger alerts if the amount of available memory (free) for one process goes below a lower limit, useful when there is a memory leak (so you restart in time and report the problem) or the config operations for given traffic needs larger PKG size. In the near future there will be some new additions to be able to access these values also via PVs. Testing and feedback is appreciated. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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SER
by douglas ugalde 19 Dec '10

19 Dec '10

Can you tell me please what can I do to change the port on SER because I installed on Asterisk and SER listen on port 5060 right? so it blocked Asterisk when is running, please tell me how can I fix this problem. Regards.

2 1

Kamailio Compatability with Elastix
by Kurt Mullen 17 Dec '10

17 Dec '10

Well I finally made some progress. I have Kamailio 3.1.0 installed on one server and Elastix 2.0.3 installed on another server. The Kamailio server queries Elastix for authentication, and the SIP clients are able to register with Kamailio. However, the clients cannot call each other. Also, extensions added in the Elastix interface cannot authenticate with Kamailio. I have to manually add the users in the Asterisk database using mysql commands to get the authentication to work. I will need to use a GUI for staff to add / delete numbers. Am I way off base here? Kurt Mullen

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/mysql/#-create.sql: No such file or directory
by Kurt Mullen 17 Dec '10

17 Dec '10

When attempting to run : /usr/local/sbin/kamdbctl create I get the following results: /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 155: /usr/local/share/kamailio//mysql/#-create.sql: No such file or directory ERROR: Creating core tables failed! Contents of directory: root@Kamailio:/usr/local/share/kamailio/mysql# ls acc-create.sql drouting-create.sql purple-create.sql alias_db-create.sql group-create.sql registrar-create.sql auth_db-create.sql htable-create.sql rls-create.sql avpops-create.sql imc-create.sql siptrace-create.sql carrierroute-create.sql lcr-create.sql speeddial-create.sql cpl-create.sql matrix-create.sql standard-create.sql dialog-create.sql msilo-create.sql uac-create.sql dialplan-create.sql mtree-create.sql uri_db-create.sql dispatcher-create.sql pdt-create.sql userblacklist-create.sql domain-create.sql permissions-create.sql usrloc-create.sql domainpolicy-create.sql presence-create.sql Contents of kamdbctl.mysql # path to the database schemas DATA_DIR="/usr/local/share/kamailio/" if [ -d "$DATA_DIR/mysql" ] ; then DB_SCHEMA="$DATA_DIR/mysql" else DB_SCHEMA="./mysql" Fi for TABLE in $STANDARD_MODULES; do mdbg "Creating core table: $TABLE" sql_query $1 < $DB_SCHEMA/$TABLE-create.sql if [ $? -ne 0 ] ; then merr "Creating core tables failed!" exit 1 fi done Kurt Mullen

2 1

is_method with pv argument?
by Juha Heinanen 16 Dec '10

16 Dec '10

is there a means to test method of sip request against a pseudo variable, which would contain an integer bitmap of methods corresponding to enum request_method values? -- juha

2 5

Changing the text of reason header in CANCEL messages
by Klaus Feichtinger 16 Dec '10

16 Dec '10

Hello, I want to ask if you know any possibility in changing the text of the SIP reason header in CANCEL requests. Is the limitation still the same as described in the mail below from May 2010? For my scenario I have to use the TM module, because incoming requests from the PSTN gateway are parallelly forked to two destinations. Whenever the pending call is cancelled by the PSTN gateway the CANCEL request is still containing a reason header (Q.850;cause=16). But I would like to adapt the text of this header according RFC3326 (e.g. 'Reason: SIP ;cause=487 ;text="Request Terminated"'). The feature "e2e-cancel-reason" is working fine - the original text is copied, but no textops function can make any change..... Is there any trick possible to change the text of this header before it is handled by the proxy (e.g. in creating a spirale - change the text and forward it to the same proxy again)? Or in worst case if this is currently absolutely impossible - would this be an acceptable request for a new feature in any upcoming release of Kamailio 3.1.x? In the release 3.1.x it is still possible to store the text of an incoming reason header, so I guess it could be no problem in overwriting this text with any function...... Is anybody else interested in this feature? regards, Klaus ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Am 21.05.2010 09:26, schrieb François BERGANZ: When I use forward, it doesnt work! I don't know how to do It works fine for me: route{ if (is_method("CANCEL")) { if (!is_present_hf("Reason")) { xlog("L_ERR", "reason missing"); append_hf("Reason: Q.850; cause=31\r\n"); xlog("L_ERR", "Reason added"); } forward(); exit; } Note: above code snippet only works if the proxy does not do any parallel forking of the INVITE. It may also be necessary to change the request-URI to forward the request to the same destination as the INVITE. in: U 2010/05/21 10:29:38.372934 83.136.33.3:16534 -> 88.198.53.113:5060 CANCEL sip:01505641...@83.123.45.165 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.51:16534;branch=z9hG4bK-d8754z-7b591160b17cfc57-1---d8754z-;rport To: "01505641636"<sip:01505641...@83.123.45.165> From: <sip:kl...@83.123.45.165>;tag=7c681960 Call-ID: MzcxYTgzZmIwMGYyMTA4YzVkN2IzZDhjYjhjYjYwNTk. CSeq: 2 CANCEL User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 out: U 2010/05/21 10:29:38.373331 88.198.53.113:5060 -> 83.123.45.165:5060 CANCEL sip:01505641...@83.123.45.165 SIP/2.0 Via: SIP/2.0/UDP 88.198.53.113;branch=z9hG4bK-d8754z-7b591160b17cfc57-1---d8754z- Via: SIP/2.0/UDP 10.10.0.51:16534;received=83.136.33.3;branch=z9hG4bK-d8754z-7b591160b17cfc57-1---d8754z-;rport=16534 To: "01505641636"<sip:01505641...@83.123.45.165> From: <sip:kl...@83.123.45.165>;tag=7c681960 Call-ID: MzcxYTgzZmIwMGYyMTA4YzVkN2IzZDhjYjhjYjYwNTk. CSeq: 2 CANCEL User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 Reason: Q.850; cause=31 regards Klaus

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TCP Overload Congestion
by Jijo 16 Dec '10

16 Dec '10

Hi Andrei, I'm observing TCP Recv buffer getting full when we are doing a load run (30cps) on Proxy with TCP. Basically the congestion is happening on the TCP connection from SIP Server to Proxy. I beleive kamailio is not processing the message fast enough, due to this congestion is occuring. Did anyone observe this behavior? How do we resolve it? Thanks Jijo

3 3

Can't Auth against Kamailio 3.1.1 - fresh install
by Graham Wooden 16 Dec '10

16 Dec '10

Hi folks, Getting ready to stage my 1.4.2 upgrade to 3.1.1 - but apparently I am missing something pretty basic here as I can't get a test subscriber registered. Turned up debug and viewed the "401 Unauthorized" repeating messages. So I combed back through the INSTALL and the various references on kamailio.org. And for full disclosure, here are the relevant steps that I have took: Did a make clean on the source and re-did the make, which looks like: make FLAVOUR=kamailio group_include="standard mysql" include_modules="carrierroute acc mi_fifo kex tmx pv siputils mi_rpc" cfg all install Added my domain into kamctlrc. Created a fresh DB, started up kamailio and added a user: [root@qa-voip-01 kamailio-3.1.1]# kamctl add testnumber testpass new user 'testnumber' added Made the appropriate changes to the default kamailio.cfg by adding in the defines. Restarted kamailio - no errors and all looked good. At this point, all indications would think that I can get the subscriber registered and get the location table updated ... But no such luck: [root@qa-voip-01 kamailio]# kamctl ul show Domain:: location table=512 records=0 max_slot=0 Any thoughts on where to look? I have confirmed that the subscriber device is using the right credentials. Using plaintext and the password is in user's row in the subscriber table. I am sure it's something pretty basic; this is my first stab at 3.1 from my 1.4 production machines. Thanks, -graham

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couple of accounting questions
by Juha Heinanen 15 Dec '10

15 Dec '10

i have been using radius for accounting, but now looked at db accounting for another project. couple of issues: - why is sip reason (the comment after reply's reason code) a mandatory field in accounting table? if someone offers a good reason, my opinion is that it should be removed. if someone needs it, he/she can add it to db_extra list. - is there possibility to fall back to syslog accounting if database insert fails? -- juha

2 3

local_route
by Klaus Darilion 15 Dec '10

15 Dec '10

Hi! I wonder what is the status of local routes (to modify requests created by Kamailio)? Is it implemented at all? I found just one occurrence of USE_LOCAL_ROUTE in the code. (something obsolete)? thanks Klaus

2 1

Delaying 200 OK sent to originator
by Dan-Cristian Bogos 15 Dec '10

15 Dec '10

Hey Guys, I came in need to delay 200 OK sent to originator, I know that is not the job of a proxy (not a perfect world though), so if anyone think that should be in theory at least possible with kamailio/sr, I would appreciate any tip. Scenario: User A -> Proxy -> User B. User B will send 183 and then 200 to the User A. I would need to delay the 200 OK sent to User A with a number of seconds, but still send ACK to User B to "fake" the receiving of the message in time. When the ACK for the delayed 200 OK comes back from User A, I would simply absorb it. Would that be possible with kamailio/sr without using an external application? Ta for any kind of tip! DanB

3 3

modules_k/rls missing in Kamailio debs
by Klaus Darilion 15 Dec '10

15 Dec '10

rls is extra_excluded in debian rules file: # extra modules to skip, because they are not compilable now # - regardless if they go to the main kamailio package or to some module package, # they will be excluded from compile and install of all EXTRA_EXCLUDED_MODULES=bdb dbtext oracle pa rls iptrtpproxy I tested and modules_k/rls compiles fine. I think the above text is true only for ser's version. regards Klaus

2 2

404 failure route;
by David J. 14 Dec '10

14 Dec '10

I am using the default script; When I do a lookup for a user that is not registered; I get 404 back; I added in my failure route to the list of codes 404; ie: First on INVITE I have t_on_failure("FAIL_ONE"); failure_route[FAIL_ONE] { if (t_check_status("486|408|404")) { sethostport("mydomain.com:5060"); append_branch(); } } But for some reason I dont see it getting caught in the failure route and sending to the server specified; Thanks.

2 1

Kamailio 3.1 and Elastix 2.0 Integration
by Kurt Mullen 14 Dec '10

14 Dec '10

I have also tried the same using the following: one Kamailio server box and one Asterisk server box. Same results after following the tutorial, with exception of using two boxes. Kurt A. Mullen Practical PC, LLC (O) 830.542.4102 x204 (F) 210.767.3912

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Kamailio 3.1 and Elastix 2.0 Integration
by Kurt Mullen 14 Dec '10

14 Dec '10

I have two separate servers. A Kamailio box and an Elastix box. I have Kamailio all up and running. I installed Elastix 2.0 using an ISO. I followed all the steps to modify the Asterisk configuration from Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database to provide access to the asterisk database. 1) I am not able to register an X-lite softphone to Kamailio. a. Is this because Kamailio is not talking to the Asterisk DB or, would X-Lite connect to Kamailio anyway? b. kamailio.cfg : i. #!ifdef WITH_ASTERISK ii. asterisk.bindip = "192.168.15.211" desc "Asterisk IP Address" iii. asterisk.bindport = "5080" desc "Asterisk Port" iv. kamailio.bindip = "192.168.15.249" desc "Kamailio IP Address" v. kamailio.bindport = "5060" desc "Kamailio Port" vi. #!endif 2) From the Kamailio server I am able to make an MYSQL connection to the remote Elastix server. a. #!define DBURL "mysql://openser:openserrw@localhost/openser" b. #!define DBASTURL "mysql://asterisk:asterisk1017@'192.168.15.211'/asterisk" 3) From the Elastix server I am able to make a MYSQL connection to the remote Kamailio server (not sure this is necessary) Some items are unclear to me in the tutorial. 1) The first two steps, MYSQL INSTALLATION and INSTALL UnixODBC is this just on the Kamailio server, the Asterisk Server or both. 2) After creating the database on the Kamailio server "/usr/local/sbin/kamdbctl create" is the asterisk database configured on the Asterisk server? 3) After creation of the asterisk database, is the ODBC configuration on the asterisk server, Kamailio server, or both? 4) Note: "Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio". How do you do this? Kurt A. Mullen Practical PC, LLC (O) 830.542.4102 x204 (F) 210.767.3912

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. Re: Error opening Kamailio's FIFO
by Kurt Mullen 14 Dec '10

14 Dec '10

-----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: Monday, December 13, 2010 5:39 AM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 67, Issue 31 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org On 12/9/10 7:20 PM, Kurt Mullen wrote: > This issue has not been resolved. I am still getting the same error when > trying to use kamctl for anything. > > I have tried all of the suggestions posted to date with no resolution. Is > there anything more I can provide to help resolve this problem? can you post here the parameters of mi_fifo module in kamailio.cfg, content of kamctlrc related to fifo and output of 'ls -la /tmp | grep kamailio'? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com [[Kurt Mullen]] kamailio.cfg [[Kurt Mullen]] loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name","/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_mode",0666) kamctlrc CTLENGINE="FIFO" ## path to FIFO file OSER_FIFO="/tmp/kamailio_fifo" root@Kamailio:/# 'ls -la /tmp | grep kamailio' -bash: ls -la /tmp | grep kamailio: No such file or directory ------------------------------

2 2

Location of "kamailio.cfg"
by "Nicolas Rüger" 14 Dec '10

14 Dec '10

Hello, where can I change/configure the location of file "kamailio.cfg"? Reason: We like to use different "kamailio.cfg" files depending on the user that starts the kamailio. Thank you... Regards, Nicolas -- Neu: GMX De-Mail - Einfach wie E-Mail, sicher wie ein Brief! Jetzt De-Mail-Adresse reservieren: http://portal.gmx.net/de/go/demail

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Re: [SR-Users] sr-users Digest, Vol 67, Issue 33
by Nina Bersamina 13 Dec '10

13 Dec '10

remove --- On Mon, 12/13/10, sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> wrote: From: sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> Subject: sr-users Digest, Vol 67, Issue 33 To: sr-users(a)lists.sip-router.org Date: Monday, December 13, 2010, 6:48 AM Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Dialog module with 2 servers and 2 separate databases. (Daniel-Constantin Mierla) 2. First public release of sip:provider Community Edition (Andreas Granig) 3. First public release of sip:provider Community Edition (Andreas Granig) 4. Re: Crash (michel freiha) ---------------------------------------------------------------------- Message: 1 Date: Mon, 13 Dec 2010 15:12:13 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Dialog module with 2 servers and 2 separate databases. To: "Pan B. Christensen" <pan(a)ibidium.no> Cc: sr-users(a)lists.sip-router.org Message-ID: <4D0629BD.8050501(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, one option you can do is to write to db immediately when a call is active (see dialog module parameters) and do a query to the other server database in addition to counting the local instance active calls. Another one, different, is to use memcache for a communication system between two or more instances. Cheers, Daniel On 12/13/10 2:37 PM, Pan B. Christensen wrote: > Thanks for your reply, Daniel. > The purpose is to do busy forwarding without querying the client when > the user already has >= X active calls. X will normally be 1 (call > waiting inactive) or 2 (call waiting active). Advanced users may > possibly set a higher value. Counting the number of calls on > the server and doing busy forwarding based on that rather than waiting > for a "486 Busy here" from the client has several advantages. > Currently, I've written code to do this with the dispatcher module, > and it's working great with only one server. Here's a code snippet: > $var(dlg_busy) = 0; > get_profile_size("busy", "$avp(s:uid)", "$var(dlg_busy)"); > if ( $var(dlg_busy) >= $avp(s:busy_level) ) { > if ($avp(s:cfb_status) == "on") { > $rU = $avp(s:cfb_number); > xlog("L_INFO", "-------------------- $avp(s:uid) has > $var(dlg_busy) active calls. Treshold $avp(s:busy_level). Forwarding > on busy to $rU --------------------\n"); > route(10); > } > ... > } > Based on your reply, I guess one way to solve this would be to write > the get_profile_size function in sqlops, query the two dialog > databases and add the numbers. This would still require the customer > to change their database design. Is there an easier or better way to > do this? > I also wote code to do busy forwarding if the client replies with 486 > (do not disturb activated), 603 (call rejected) etc. > This code works for normal busy forwarding if Polycom is set to 1 call > per line key (default is 8). We'll then have to provision the > $avp(s:busy_level) variable to the clients instead of handling it > server-side. If a user now wants to change the setting, he'll have to > reboot his phone after doing so. Changing the > reg.x.callsPerLineKey setting in the phone also limits the number of > outgoing calls the user can make. We'll also have to make code for all > the other hardphones the customer is planning to use plus make guides > on how to change the setting for all kinds of softphones. We want to > avoid all this. > With kind regards, > Pan > > ----- Original Message ----- > *From:* Daniel-Constantin Mierla <mailto:miconda@gmail.com> > *To:* Pan B. Christensen <mailto:pan@ibidium.no> > *Cc:* sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > *Sent:* Monday, December 13, 2010 12:26 PM > *Subject:* Re: [SR-Users] Dialog module with 2 servers and 2 > separate databases. > > > > On 12/10/10 2:17 PM, Pan B. Christensen wrote: >> Hello, >> My customer has the following database design. >> Voip server 1 talks to SQL server 1. >> Voip server 2 talks to SQL server 2. >> Voip 1 and Voip 2 are load-balanced. >> Each SQL server has two databases. Database 1 contains >> semi-static data like call forwarding properties for users and is >> read-only. This is replicated from a third SQL server which the >> web interface writes to. Database 2 is read/write, is not >> replicated and contains data that is updated frequently like user >> location and now dialog info. >> Voip server 1 is not allowed to talk to SQL server 2 and vice versa. >> I'm using forward() to send authenticated REGISTERs to the other >> server so that it'll write this to RAM and its own SQL server. >> Thus, both servers are aware of clients authenticated and >> registered by the other server. >> How can I make both servers be aware of active calls on the other >> server? > what is the purpose? > > Practically, it is not possible to track a call in two instances, > because, unlike registration where is just a storage of mappings > between contact and aor, call states of dialog module involve more > processing logic, including timeouts and sending BYEs. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > Kamailio (OpenSER) Advanced Training > Jan 24-26, 2011, Irvine, CA, USA > http://www.asipto.com > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.5.449 / Virus Database: 271.1.1/3312 - Release Date: > 12/12/10 19:34:00 > > > _______________________________________________ > 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 Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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