- sr-users - kamailio.org

memory usage
by Uri Shacked 05 Sep '12

05 Sep '12

Hi, When I start kamailio it loads some data into memory. Now calls are starting to gather up. How can I check how mach memory does my avp's take? BR, Uri

2 1

event_route[dialog:failure]
by Uri Shacked 05 Sep '12

05 Sep '12

Hi, i seems to me that the "event_route[dialog:failure]" does not recognize dialog failures. i use dlg_manage when i recieve the INVITE. i get 301, 302, 402 or send_reply with some +300 responses. on the manage_failure i print the responses, on the event_route[dialog:start] and [dialog:end] i get good results as well. but, the dialog:failure] does not rise.... any ideas? BR, Uri

2 2

LUA Authentication.
by David | StyleFlare 05 Sep '12

05 Sep '12

I think I saw once an example from miconda using LUA for Auth? Does anyone else remember seeing that? I wanted to do a custom Auth using LUA. Thanks in advance for any pointers. David.

4 3

Re: [SR-Users] t_check_status not found in readmes
by Alex Balashov 05 Sep '12

05 Sep '12

It is true. It disappeared from the docs after the 3.0 sip-router convergence. It is present in 'tm' docs in <= 1.5.x, though. The current docs do have t_grep_status(), which seems in many respects identical. I wonder if one is an alias for another now. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Juha Heinanen <jh(a)tutpro.com> wrote: >looks like t_check_status is not documented in any readme file. > >-- juha > >_______________________________________________ >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

t_check_status not found in readmes
by Juha Heinanen 05 Sep '12

05 Sep '12

looks like t_check_status is not documented in any readme file. -- juha

1 0

Cassandra module - query question
by Boudewyn Ligthart 03 Sep '12

03 Sep '12

Hi, I would to query the Cassandra database for other information with either CQL or other commands. Could this be done through means of either avpops or sqlops, like an MySQL database. This could be done through an external script but I would like to utilize the connection that is already established. Thanks, Bou

3 4

Why append_branch should be called in a failure route ?
by Vitaliy Aleksandrov 03 Sep '12

03 Sep '12

Below is the example from http://kamailio.org/docs/modules/3.2.x/modules/tm.htm : failure_route[0]{ if (t_check_status("5[0-9][0-9]")){ # I do not like the 5xx responses, # so I give another chance to "foobar.com", # and I drop all the replies to make sure that # they are not forwarded to the caller. t_drop_replies(); rewritehostport("foobar.com"); append_branch(); t_relay(); } } Why do append_branch() is present before t_relay(). I'm asking because I didn't used it in failure routes and call rerouting on various conditions worked great.

1 0

Catch error "status rewrite by UAS" in on_reply
by Alex Hermann 03 Sep '12

03 Sep '12

Hello all, i sometimes get a very late reply from a broken client which triggers the following error 9because the call was already continued in another branch): ERROR: tm [t_reply.c:1193]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Not a problem perse, but i would like to catch this error in the script's on_reply route because I need to prevent some actions on the reply if it will not be forwarded to the UAC. Is there a way to do this? -- Greetings, Alex Hermann

1 0

Celebrating 11 Years
by Daniel-Constantin Mierla 03 Sep '12

03 Sep '12

Hello, eleven years ago, the first piece of code of the project was committed. Last year we organized a public event, for the 10 years celebration, presentations and more details about it at: * http://sip-router.org/10-years-ser/ For the 11th year, I made a news with the summary of the past year - definitively it was lot of work, adding many new features: * http://www.kamailio.org/w/2012/09/kamailio-at-11-years/ Looking forward to the next year celebration, surely it will list another great set of new features! Many thanks to everyone helping the project, from advocating to development! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

2 1

kamailio and clustered database
by Vieri 01 Sep '12

01 Sep '12

Hi, I'm new to Kamailio and would like to set it up with a "circular replication database" on several servers (master-to-master). Before running 'kamdbctl create' I edited kamctlrc and set DBENGINE to MYSQL (I can use either MySQL or PostgreSQL or both - I just need to use a cluster of servers). What would you suggest? Should I set DBENGINE=MYSQL and configure a cluster of databases as described here: http://dev.mysql.com/doc/refman/5.1/en/mysql-cluster-replication-multi-mast… Or does kamailio provide another way to do the same job? What does the db_cluster module do and where/how should I use it? Thanks, Vieri

1 0

weird dp_translate result
by Juha Heinanen 01 Sep '12

01 Sep '12

i have this piece of config code: modparam("dialplan", "attrs_pvar", "$var(plan)") ... xlog("L_INFO", "Translating <$var(calling_number)> based on dpid 4\n"); if (dp_translate("4", "$var(calling_number)")) { xlog("L_INFO", "Found attrs <$var(plan)>\n"); ... dpid 4 has three entries: 109 4 10 1 ^\+1 0 na 107 4 10 1 ^\+358 0 fi 108 4 10 1 ^\+46 0 se thus in each entry attrs field has some two letter value. when the above piece of code is executed, i get to syslog Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Translating <+35892345670> based on dpid 4 Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Found attrs <0> if i restart the proxy, attrs value may change to some other bogus value. any ideas? -- juha

2 10

OpenIMSCore icscf problem
by Siavash Safi 01 Sep '12

01 Sep '12

Hi, I fetched the latest source code from OpenIMSCore SVN repository and generated the debian packages on squeeze. Everything is working fine so far except the icscf module which fails to start. I think that there is something wrong with the configuration. Here is the output messages: root@spawn:~# icscf.sh Listening on udp: xxx.xxx.xxx.xxx [xxx.xxx.xxx.xxx]:5060 tcp: xxx.xxx.xxx.xxx [xxx.xxx.xxx.xxx]:5060 Aliases: tcp: spawn:5060 udp: spawn:5060 *: ims.mydomain.com:* *: icscf.ims.mydomain.com:* ------ Semaphore Arrays -------- key semid owner perms nsems root@spawn:~# 0(4597) ERROR: mk_proxy: could not resolve hostname: "" 0(4597) ERROR: fixup_hostport2proxy: bad host name in URI <> ERROR: error -478 while trying to fix configuration 0(4597) INFO:destroy_diameter_peer(): Bye Bye from C Diameter Peer test As you can see mk_proxy is called with a empty hostname! I have pasted my configuration and more verbose debug output: configuration: http://pastebin.com/nDQM1KLM debug output: http://pastebin.com/d6jh0CHF Any help would be appreciated. -- Regards, Siavash Safi Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

1 0

Kamailio - Nonce validity between Register and Invite
by patrice.bodeven＠orange.com 31 Aug '12

31 Aug '12

Hello, I am working on Kamailio 3.2.2. There is no traffic, only functional test done. Based on the SIP Client used until now (Xlite), the INVITE is systematically authenticated by 407 as there is no Proxy-Authorization in the initial INVITE. Expected/normal behavior. But when using an internal SIP client, the initial INVITE is generated with a Proxy-Authorization containing the nonce value used by the successful registration. In Authorization of Register (the answer is 200OK) Digest username="+33296488922(a)ims.v0.pftest.net", realm="sip:ims.v0.pftest.net", nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", uri="sip:ims.v0.pftest.net", response="4ed51b8e62cf769b19f8c4771e068ad3", cnonce="06440000000529330fe1", qop=auth, nc=00000011 In Proxy-Authorization of Invite Digest username="+33296488922(a)ims.v0.pftest.net", realm="sip:ims.v0.pftest.net", nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", uri="sip:0157361149@ims.v0.pftest.net", response="b247052d6d5b37e0b6677c816390cb77", cnonce="06440000000529330fe1", qop=auth, nc=00000012 As the nonce expire is high (see nonce_expire), I don't understand why Kamailio requests the Authentication of the INVITE by 407. 407 with Proxy-Authenticate: Digest realm="ims.v0.pftest.net", nonce="UDP6pVAzpkWDF+jSlP3zzVYcuNPhY8NQBj62WIA=", qop="auth", stale=true Does it mean that the REGISTER and INVITE are using different Nonce value ? ==> Information on my configuration file. modparam("auth_db", "db_url", DBURLAUTH) modparam("auth", "auth_checks_register", 6) # callid and From TAG modparam("auth", "qop", "auth") # enable qop=auth modparam("auth", "nonce_count", 1) # enable nonce_count support modparam("auth", "nonce_expire", 21600) # Set nonce_expire to 6 hours For register, I do if (!www_authorize("$ru", "subscriber")) { if($? == -2 || $? == -3) { sl_send_reply("403","Forbidden auth ID"); exit; } else { www_challenge("$ru", "17"); exit; } } For invite, I do if (!proxy_authorize("$fd", "subscriber")) { if($? == -2 || $? == -3) { sl_send_reply("403","Forbidden auth ID"); exit; } else { proxy_challenge("$fd", "17"); xlog("L_INFO","INVITE challenged by 407 to :$di:$dp \n"); exit; } } Followed by consume_credentials(); Thanks in advance for your support. Patrice B _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified. Thank you.

3 9

invite/ack avps/transactions
by Juha Heinanen 31 Aug '12

31 Aug '12

i tried to access an avp that was set during invite processing while processing ack to 200 ok, but i guess the avp is already gone by that time, because positive ack is its own transaction? what i tried to do is to save somewhere invite's $du and then t_relay ack to the same place without calling loose_route. should i go for htable as storage or is there a better choice? -- juha

2 1

LCR (next_gw) question
by Nord7 31 Aug '12

31 Aug '12

Hello everyone. Configures the module LCR. I can't get him to take the next on the list of priorities gateway if the first does not work. Function "next_gw" capable of doing that? Did anyone realties that? If possible show example. Kamailio 3.3 Regards, Kirill

2 1

spam messages breaking through filters
by Daniel-Constantin Mierla 31 Aug '12

31 Aug '12

Hello, there were some spam messages going through our mailing list filters. I was updating the filters on mailman, but there are new and new ideas coming from this guys. Sorry for the pollution and I wanted to say that these messages are marked as spam by spamassasin, you can filter them locally as well looking for the header X-Spam-Level: *** You can adjust the number of '*'. Some email services do that automatically, like gmail.com for example. Of course, if anyone has better ideas of protection, please write here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

1 1

db_cluster together with the registrar module = signal 11
by Øyvind Kolbu 30 Aug '12

30 Aug '12

Hi, I upgraded our system to Kamailio 3.3 to take advantage of the new db_cluster module. Currently we have two registration/location servers each with their own sql-server. Authenticated REGISTER messages are forwarded to the other server, which uses the 0x02 flag to save() in order to store it without generating an extra reply. This works fine. My goal was to use db_cluster to update both databases at once with save() and use the failover mode when reading with lookup(). Have anyone tested a similar setup? Anyhow, here is how I tried to do it: modparam("db_cluster","connection","voip1_data2=>postgres://user:pass@hostname1/voipt1_data2") modparam("db_cluster","connection","voip2_data2=>postgres://user:pass@hostname2/voipt2_data2") modparam("db_cluster", "cluster", "data2=>voip1_data2=8s7p;voip2_data2=9s7p") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "cluster://data2") But I get: : : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 33 : ALERT: <core> [main.c:785]: child process 28275 exited by a signal 11 : ALERT: <core> [main.c:788]: core was generated : INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Same with db_mode = 1 or 3, and either modparam("db_cluster", "cluster", "data2=>voip1_data2=8s9p") or modparam("db_cluster", "cluster", "data2=>voip2_data2=8s9p"). No difference between newly booted phones or phones reregistering. Backtrace from gdb: (gdb) bt #0 0x002fd7da in db_cluster_insert (_h=0xb7d2dc88, _k=0xbf9ac954, _v=0xbf9ac7ec, _n=18) at dbcl_api.c:334 #1 0x00fbb1cc in db_insert_ucontact (_c=0xb5d828d8) at ucontact.c:565 #2 0x00fcfc5d in insert_ucontact (_r=0xfd3a00, _contact=0xb7d3ce90, _ci=0xc5cc80, _c=0xbf9acb84) at urecord.c:478 #3 0x00c55bb0 in add_contacts (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:493 #4 save (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:864 #5 0x00c4dd43 in w_save2 (_m=0xb7d05e40, _d=0xb5d63060 "00ֵ", _cflags=0x0) at reg_mod.c:407 #6 0x080586ad in do_action (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1157 #7 0x0805fe5c in run_actions (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1644 #8 0x080604e1 in run_actions_safe (h=0xbf9adcb8, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1708 #9 0x080f96b7 in rval_get_int (h=0xbf9adcb8, msg=0xbf9ac954, i=0xbf9ad268, rv=0xb7c406cc, cache=0x0) at rvalue.c:920 #10 0x080fcf38 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad268, rve=0xb7c406c8) at rvalue.c:1914 #11 0x080fd009 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad544, rve=0xb7c40a70) at rvalue.c:1922 #12 0x080585fe in do_action (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1121 #13 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1644 #14 0x08058657 in do_action (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1136 #15 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1644 #16 0x0805853d in do_action (h=0xbf9adcb8, a=0xb7bd64d8, msg=0xb7d05e40) at action.c:761 #17 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7bc7128, msg=0xb7d05e40) at action.c:1644 #18 0x08060464 in run_top_route (a=0xb7bc7128, msg=0xb7d05e40, c=0x0) at action.c:1729 #19 0x080e11e4 in receive_msg (buf=0x82c4f20 "REGISTER sip:hometest.voip.uio.no SIP/2.0\r\nVia: SIP/2.0/UDP 193.157.149.56;branch=z9hG4bKb8e88f2eE34DCDDD\r\nFrom: \"kolbutest\" <sip:2549619@hometest.voip.uio.no>;tag=34CE9BA6-70FCC135\r\nTo: <sip:2549619@"..., len=809, rcv_info=0xbf9ade98) at receive.c:209 #20 0x08177f7b in udp_rcv_loop () at udp_server.c:544 #21 0x080b0426 in main_loop () at main.c:1633 #22 0x080b3d02 in main (argc=17, argv=0xbf9ae174) at main.c:2546 Core dumps and debug logs (debug=3) are available. -- Øyvind Kolbu University of Oslo

3 21

Job ad - see details! Sent through Search engine
by serusers＠iptel.org 30 Aug '12

30 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: I would like to take this time to welcome you to our hiring process and give you a brief synopsis of the position's benefits and requirements. If you are taking a career break, are on a maternity leave, recently retired or simply looking for some part-time job, this position is for you. [...] Content analysis details: (6.7 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 SPF_PASS SPF: sender matches SPF record 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [189.4.159.143 listed in bb.barracudacentral.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [189.4.159.143 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.3 RCVD_IN_RP_RNBL RBL: Relay in RNBL, https://senderscore.org/blacklistlookup/ [189.4.159.143 listed in bl.score.senderscore.com] 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?189.4.159.143>] -1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1% [score: 0.0000] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

Kamailio Crash using Cassandra 1.1.2 during SIP unregister
by Boudewyn Ligthart 29 Aug '12

29 Aug '12

Hi, Kamailio crash using Cassandra 1.1.2 during SIP unregister (registering works fine) The version of Kamailio is the latest GIT version: version: kamailio 3.4.0-dev3 (x86_64/linux) 10327c flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 10327c compiled on 14:53:11 Aug 22 2012 with gcc 4.6.3 BT of Core : Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `kamailio -ddddd'. Program terminated with signal 11, Segmentation fault. #0 fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at mem/f_malloc.c:527 527 if(unlikely(f->u.nxt_free!=NULL)) { (gdb) bt #0 fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at mem/f_malloc.c:527 #1 0x00007fa55d35ac53 in db_free_columns (_r=0x7fa55db38c30) at db_res.c:86 #2 0x00007fa55d83f87d in db_cassa_query (_h=<optimized out>, _k=0x7fa55adc6aac, _op=<optimized out>, _v=<optimized out>, _c=0x7fff72ef0e30, _n=<optimized out>, _nc=16, _o=0x7fa55afca990, _r=0x7fff72ef0f10) at dbcassa_base.cpp:659 #3 0x00007fa55adb58eb in db_load_urecord (_c=0x7fa55db218d0, _d=0x7fa5576eee38, _aor=0x7fff72ef11b0) at udomain.c:592 #4 0x00007fa55adb93cd in get_urecord (_d=<optimized out>, _aor=0x7fff72ef11b0, _r=0x7fff72ef1200) at udomain.c:1017 #5 0x00007fa55ab91698 in lookup (_m=0x7fa55db40c48, _d=0x7fa5576eee38, _uri=<optimized out>) at lookup.c:156 #6 0x00007fa55ab957f2 in w_lookup (_m=0x7fa55db40c48, _d=0x7fa5576eee38 "\350\355nW\245\177", _uri=<optimized out>) at reg_mod.c:437 #7 0x000000000041c624 in do_action (h=0x7fff72ef17f0, a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1153 #8 0x000000000041b493 in run_actions (h=0x7fff72ef17f0, a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1646 #9 0x0000000000423a24 in run_actions_safe (h=<optimized out>, a=<optimized out>, msg=<optimized out>) at action.c:1710 #10 0x00000000004d79a0 in rval_get_int (h=0x7fff72ef2790, msg=<optimized out>, i=0x7fff72ef1c30, rv=<optimized out>, cache=<optimized out>) at rvalue.c:920 #11 0x00000000004dc82c in rval_expr_eval_int (h=0x7fff72ef2790, msg=0x7fa55db40c48, res=0x7fff72ef1c30, rve=0x7fa55db23218) at rvalue.c:1914 #12 0x00000000004dc8f9 in rval_expr_eval_int (h=0x7fff72ef2790, msg=0x7fa55db40c48, res=0x7fff72ef20d0, rve=0x7fa55db238e8) at rvalue.c:1922 #13 0x000000000041cb66 in do_action (h=0x7fff72ef2790, a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1123 #14 0x000000000041b493 in run_actions (h=0x7fff72ef2790, a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1646 #15 0x000000000041c7e1 in do_action (h=0x7fff72ef2790, a=0x7fa55db15b38, msg=0x7fa55db40c48) at action.c:763 #16 0x000000000041b493 in run_actions (h=0x7fff72ef2790, a=0x7fa55db118c8, msg=0x7fa55db40c48) at action.c:1646 #17 0x0000000000423ab0 in run_top_route (a=0x7fa55db118c8, msg=0x7fa55db40c48, c=<optimized out>) at action.c:1731 #18 0x00000000004b39ee in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:209 #19 0x000000000054de1a in udp_rcv_loop () at udp_server.c:544 #20 0x0000000000478879 in main_loop () at main.c:1635 #21 0x000000000041b002 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2548 Thanks, Bou

2 2

[OT] sip:provider CE v2.6 is out!
by Andreas Granig 29 Aug '12

29 Aug '12

Hi all, I'm excited to announce the release of the Sipwise sip:provider Community Edition v2.6, a free and open source turn-key platform, which uses Kamailio, Sems and Asterisk in its core to provide a full-blown and feature-rich VoIP soft-switch. http://www.sipwise.com/news/announcements/spce-v2_6-release/ New core features are *full presence/xcap/im support* together with the awesome Jitsi client, allowing you to deploy your own Skype-like service for encryted voice/video communication, buddy lists, instant messaging, screen sharing and remote desktop control. We also support *media features for error announcements*, like busy, blocked, unknown number etc. Check our VM images to get started within a couple of minutes and learn what can be built with the power of Kamailio as a stateless load-balancer and a stateful proxy/registrar, and Sems as an SBC and application server. http://www.sipwise.com/products/spce/quickinstall/ Thanks a lot to all developers, contributors and members of this mailing list to make this possible and for helping us out! Andreas

1 0

Work offer inside
by serusers＠iptel.org 29 Aug '12

29 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: I would like to take this time to welcome you to our hiring process and give you a brief synopsis of the position's benefits and requirements. If you are taking a career break, are on a maternity leave, recently retired or simply looking for some part-time job, this position is for you. [...] Content analysis details: (7.4 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.4587] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [186.32.220.19 listed in bb.barracudacentral.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [186.32.220.19 listed in zen.spamhaus.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

[ot] virtualization systems
by Daniel-Constantin Mierla 29 Aug '12

29 Aug '12

Hello, just asking to see your experience deploying sip platforms on virtual systems. So far I was running Kamailio in virtual machines and no problems, but I insisted that media servers to be on physical machines. Lately is more pressure from the market to go everything virtual. So the question is more about having everything on virtual systems, proxy and media server, where the media server can deal with transcoding, conference rooms and IVRs. Any strong comments pro or against? What is your preferred virtualization system for such deployments? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

10 15

[OT] OverSIP SIP Proxy released
by Iñaki Baz Castillo 28 Aug '12

28 Aug '12

Hi all, I announce the first stable release of OverSIP SIP Proxy: http://www.oversip.net OverSIP features: ----------------------------------------------------------------------------- - SIP over UDP, TCP, TLS and WebSocket transports. - Full support for IPv4, IPv6 and DNS resolution (NAPTR, SRV, A, AAAA). - Fully programmable using Ruby programming language. - The perfect Outbound (RFC 5626) Edge Proxy. - Fully asynchronous event-based design. OverSIP is not a replacement for well-known proxies as Kamailio (sip-router). OverSIP does not implement basic proxy features such as Digest authentication, registration or parallel forking. OverSIP can behave as an Outbound Edge Proxy between clients and SIP registrars, or can be placed anywhere for building custom SIP / WebSocket applications. Cheers. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 1

Re: [SR-Users] [ot] virtualization systems
by JR Richardson 28 Aug '12

28 Aug '12

>> just asking to see your experience deploying sip platforms on virtual >> systems. So far I was running Kamailio in virtual machines and no problems, >> but I insisted that media servers to be on physical machines. Lately is more >> pressure from the market to go everything virtual. >> >> So the question is more about having everything on virtual systems, proxy >> and media server, where the media server can deal with transcoding, >> conference rooms and IVRs. >> >> Any strong comments pro or against? >> >> What is your preferred virtualization system for such deployments? >> I host ~120 asterisk virtual serves on OpneVZ for more than 3 years, runs great. These are business customers so they can be very demanding, allot of custom apps and such. I host up to 150 phones in virtual environment, if more is needed, I switch over to a physical server. This is just a personal preference, not limited virtually, but customers with that many or more phones usually want a dedicated server and can certainly afford the up charge. Ensure your resources for each VM are enough for the voice apps you want to run, be cautious about disk I/O like high volume recording, database dips per call and such. I just converted my physical sip-routers to virtual about a week now on OpenVZ, so far so good. Transcoding required higher proc and mem loads so partition resources accordingly. I do run Asterisk 10.0 conf bridge server virtual, SIP only, no need for timing interface like DAHDI. This is running well, no problem that I'm aware of, just works. I use VirtualBox and VMware for demo systems and lab work but OpenVZ for production due to higher density you can achive per physical server. Whatever virtualization you want to use or are comfortable with, ensure you know it backwards and forwards and what to do if it breaks and how to increase resource when needed and by all means do your lab work ahead of time. You will not be able to simulate every situation in the lab, but you can certainly stress test your environment with high call volume, disk I/O recordings, databases operations, live migration, upgrades, downgrades, backups, resource monitoring, alarm triggers, test, test, test and then test some more. Virtualizing voice systems has been around a long time, most virt platforms can be implemented to perform well for voice, just got to do your homework. Good luck!. JR -- JR Richardson Engineering for the Masses

1 0

AVP string or integer ID
by Vitaliy Aleksandrov 28 Aug '12

28 Aug '12

Hi all. I'm working with kamailio for about 1 year and still don't understand difference between integer and string IDs of AVPs. Is it just a question of usability ?

2 1

Trunk Utilization graphing using MI COMMANDS
by phillman25 28 Aug '12

28 Aug '12

Dear List I am trying to run the below command from a REMOTE server where i use Cacti to graph all my data. I am trying to graph on a per trunk basis. kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}' This command yields an output on the local server. Could someone perhaps point me in the right direction? Thanks!

2 6

DROUTING nub question
by Luis Fernando Urrea 28 Aug '12

28 Aug '12

I haven't been able to find the answer in the documentation. It seems DROUTING does not take regex in the dr_rules table to match routable numbers, are there any kind of wilcards that could be used. How does one specify different rules for 8 digit numbers and 10 digit numbers. TIA -- O: 4000-1020 C: 6040-3624

3 7

Re: [SR-Users] ADSL client-side problem
by Daniel-Constantin Mierla 27 Aug '12

27 Aug '12

Hello, please do not send private messages from mailing lists discussions, keep the mailing list cc-ed so other can participate or learn from archives. On 8/27/12 5:22 PM, Simon Hintermann wrote: > Hello, > > I will try with a smaller "Subscription Expiry (s):" in my SNOM. > > Just to be clear with the method of test: > > - firewall redirects SNOM to my first ISP > - telephone works perfectly, registered in Kamailio... > - firewall redirects SNOM to my second ISP > - telephone is unregistered in Kamailio, impossible to "re-register" > the identity1, says "network error" > - only way to recover is to change the SNOM's IP address what do you mean by 'firewall redirects Snom to Nth ISP'? Is the IP of snom static or taken via dhcp from adsl router? Cheers, Daniel > > Thanks for your support, greetings > > Simon > > Le 27.08.2012 09:04, Daniel-Constantin Mierla a écrit : >> Hello, >> >> what is the expire time of your registrations? 24 hours seems a lot >> to re-register. >> >> From kamailio point of view, the contact is expired based on >> un-registration or timeout. You can look with 'kamctl ul show' to see >> if the phone is still registered or not. >> >> Anyhow, in such cases is good to set registration time quite small, >> say 5 or 10 minutes, but if there is a problem in the client side, it >> is not much to do in server side. >> >> What is the adsl routed, does it do SIP ALG? I have snom on ADSL >> using tcp, does not seem to make big problems. >> >> Cheers, >> Daniel >> >> On 8/16/12 2:10 PM, Simon Hintermann wrote: >>> >>> Hello, >>> >>> I am experiencing some disconnections with my VOIP phones. >>> >>> Let me explain: >>> >>> My server is an Asterisk + Kamailio in a datacenter with a fixed IP, >>> and a bridge firewall in front of it, so no NAT here. >>> >>> My client (several offices) have usually no problem when they have a >>> fixed IP, but encounter fatal disconnection when they have a dynamic >>> IP. I had the case with Siemens, SNOM and Aastra phones, without any >>> difference. They are of course natted behind an ADSL router. I also >>> have clients doing load-balance between two ADSL lines, which is >>> also problematic. >>> >>> The only way for me to recover is to change the private IP of the >>> VOIP phones. Seems more like a TCP problem or a routing problem, but >>> I am no network guru. >>> >>> I have two ADSL lines to test it, and when I switch from one line to >>> another, every phone becomes "Not registered". Even falling back tot >>> he first line does not help recovering. I did not try to wait for >>> longer than 24 hours to recover. >>> >>> So my question is, more generally: is it possible to allow the >>> clients (VOIP phones) to re-connect with a new IP address without >>> having closed correctly the first connection (SIP session, I >>> presume)? It seems not by default, but I played with the timeouts >>> server-side and phone-side, without luck. >>> >>> I saw many posts with PBX having problems when behind a dynamic IP, >>> but nothing about the clients having dynamic addresses or >>> load-balancing. Normally, having a dynamic IP does not mean that >>> your IP will change « on-the-fly », but each time you shut down and >>> reboot later your ADSL router, but we came to the conclusion that >>> every client NOT having a fixed IP was experiencing fatal >>> disconnects. So I managed to test and validate this behavior with >>> our two ADSL lines. >>> >>> Hope someone can help me. >>> >>> Greetings >>> >>> Simon >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla -http://www.asipto.com >> http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

1 0

SMS module & IMS
by Raman Kumar 27 Aug '12

27 Aug '12

Hi, Can any one help me .. how to configure sms module? what changes required in scscf.cfg or any other cfg also?? Thanks and regards, Raman Kumar =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

Re: [SR-Users] centos (fedora) service startup file
by Daniel-Constantin Mierla 27 Aug '12

27 Aug '12

Do not send private messages from the mailing list discussions. Maybe others can jump in the thread and provide good information, if people from discussion cannot sort out things. Also, mailing list is archived online, so people can search later on topics they are interested in. Cheers, Daniel On 8/27/12 9:43 AM, andre wrote: > 27.08.2012 10:09, Daniel-Constantin Mierla пишет: >> Hello, >> >> On 8/14/12 11:31 AM, andre wrote: >>> 14.08.2012 12:21, Daniel-Constantin Mierla пишет: >>>> Hello, >>>> >>>> thanks, it was committed in git master branch, backports to follow >>>> before minor releases. >>>> >>>> Do you build your own rpms or you install from the opensuse build >>>> system? >>> From the opensuse build system. >> >> ok, I will update that system as well. is 'success' available in all >> distros using rpms (like opensuse)? > Not shure about Suse. I even have no someone to ask > > Better add to RH distros only. >> >> Cheers, >> Daniel >> >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 8/2/12 6:24 PM, andre wrote: >>>>> Somwhere losted since openser 1.0 :) >>>>> >>>>> >>>>> Please fix in: >>>>> >>>>> start() { >>>>> echo -n $"Starting $prog: " >>>>> ..... >>>>> [ $RETVAL = 0 ] && touch something >>>>> to >>>>> [ $RETVAL = 0 ] && touch someting && success >>>>> >>>>> I mean call function >>>>> success >>>>> in the end of the expression. >>>>> >>>>> Thanks in advance >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>> list >>>>> sr-users(a)lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >> > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

1 0

Advisement and promotion company seeking out local affiliate
by serusers＠iptel.org 27 Aug '12

27 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: As a company we deal with advising other business and end users on reusable materials, green advancements and nontypical forms of energy and also produce and promote projects that are web media focused. [...] Content analysis details: (7.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.6 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr 2) 0.0 TVD_RCVD_IP TVD_RCVD_IP 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [213.177.0.199 listed in zen.spamhaus.org] -0.0 SPF_PASS SPF: sender matches SPF record 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [213.177.0.199 listed in bb.barracudacentral.org] 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.4633] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

centos (fedora) service startup file
by andre 27 Aug '12

27 Aug '12

Somwhere losted since openser 1.0 :) Please fix in: start() { echo -n $"Starting $prog: " ..... [ $RETVAL = 0 ] && touch something to [ $RETVAL = 0 ] && touch someting && success I mean call function success in the end of the expression. Thanks in advance

2 3

ADSL client-side problem
by Simon Hintermann 27 Aug '12

27 Aug '12

Hello, I am experiencing some disconnections with my VOIP phones. Let me explain: My server is an Asterisk + Kamailio in a datacenter with a fixed IP, and a bridge firewall in front of it, so no NAT here. My client (several offices) have usually no problem when they have a fixed IP, but encounter fatal disconnection when they have a dynamic IP. I had the case with Siemens, SNOM and Aastra phones, without any difference. They are of course natted behind an ADSL router. I also have clients doing load-balance between two ADSL lines, which is also problematic. The only way for me to recover is to change the private IP of the VOIP phones. Seems more like a TCP problem or a routing problem, but I am no network guru. I have two ADSL lines to test it, and when I switch from one line to another, every phone becomes "Not registered". Even falling back tot he first line does not help recovering. I did not try to wait for longer than 24 hours to recover. So my question is, more generally: is it possible to allow the clients (VOIP phones) to re-connect with a new IP address without having closed correctly the first connection (SIP session, I presume)? It seems not by default, but I played with the timeouts server-side and phone-side, without luck. I saw many posts with PBX having problems when behind a dynamic IP, but nothing about the clients having dynamic addresses or load-balancing. Normally, having a dynamic IP does not mean that your IP will change << on-the-fly >>, but each time you shut down and reboot later your ADSL router, but we came to the conclusion that every client NOT having a fixed IP was experiencing fatal disconnects. So I managed to test and validate this behavior with our two ADSL lines. Hope someone can help me. Greetings Simon

2 1

Advisement and promotion company seeking out local affiliate
by serusers＠iptel.org 27 Aug '12

27 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: As a company we deal with advising other business and end users on reusable materials, green advancements and nontypical forms of energy and also produce and promote projects that are web media focused. [...] Content analysis details: (9.6 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [139.195.43.246 listed in zen.spamhaus.org] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] 0.6 URIBL_PH_SURBL Contains an URL listed in the PH SURBL blocklist [URIs: euroinjobs.com] 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.5000] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

Re: [SR-Users] Vacancy #120
by serusers＠iptel.org 27 Aug '12

27 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: As a company we deal with advising other business and end users on reusable materials, green advancements and nontypical forms of energy and also produce and promote projects that are web media focused. [...] Content analysis details: (15.1 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.6 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr 2) 0.0 TVD_RCVD_IP TVD_RCVD_IP 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [186.63.33.68 listed in zen.spamhaus.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [186.63.33.68 listed in bb.barracudacentral.org] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] 0.6 URIBL_PH_SURBL Contains an URL listed in the PH SURBL blocklist [URIs: euroinjobs.com] 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.4970] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

Re: [SR-Users] Vacancy #371
by serusers＠iptel.org 27 Aug '12

27 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: As a company we deal with advising other business and end users on reusable materials, green advancements and nontypical forms of energy and also produce and promote projects that are web media focused. [...] Content analysis details: (10.3 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.6 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr 2) 0.0 TVD_RCVD_IP TVD_RCVD_IP -0.0 SPF_PASS SPF: sender matches SPF record 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [184.77.159.149 listed in bb.barracudacentral.org] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] 0.6 URIBL_PH_SURBL Contains an URL listed in the PH SURBL blocklist [URIs: euroinjobs.com] -0.0 BAYES_40 BODY: Bayes spam probability is 20 to 40% [score: 0.2745] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [184.77.159.149 listed in zen.spamhaus.org] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

SEAS Module
by Fabian Borot 24 Aug '12

24 Aug '12

Hello, I would like to use the module and create our own version of the Application Server using the SEAS protocol but there are some pieces of information missing from the web site (links to images that contains information about headers etc ). The basic information on how it works is there, the AS registers to the OpenSER, receives information, sends information but the details of how to register, the format of the messages etc are not there. I checked for information in previous versions only to find the same situation.(3.1 thru 3.3) Is the information about the SEAS module specific to the WeSIP Application Server? Thank you fborot

2 4

Wrong index on expires column in pua module
by Andreas Granig 24 Aug '12

24 Aug '12

Hi guys, In the pua table, there is a unique key on the "expires" column, which seems wrong to me, because you'll sooner or later end up with different pua entries having the same expiry time, no? At least we ended up with messages like this in the logs, I think it's caused by pua_reginfo tests: ERROR: db_mysql [km_dbase.c:122]: driver error on query: Duplicate entry '1345474412' for key 'expires_idx' ERROR: <core> [db_query.c:210]: error while submitting query ERROR: pua [pua.c:1171]: while inserting in db table pua Any comments on that? Andreas

2 1

db_cassandra
by Boudewyn Ligthart 24 Aug '12

24 Aug '12

Hi, I am trying to get the db_cassandra working in kamailio 3.1.1 with thrift 0.6.1 API and however I get the following compile error. ol.h:33, from dbcassa_base.h:38, from dbcassa_base.cpp:47: ../../lib/srdb1/db.h:388:20: error: expected unqualified-id before âdeleteâ dbcassa_base.cpp: In function âColumnVecPtr cassa_translate_query(const db1_con_t*, str* const*, const db_val_t*, str* const*, int, int, int*)â: dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 7 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 5 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 6 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 4 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 4 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] ../../lib/srdb1/db_res.h: At global scope: ../../lib/srdb1/db_res.h:105:19: warning: inline function âdb1_res_t* db_new_result()â used but never defined [enabled by default] ../../lib/srdb1/db_res.h:121:12: warning: inline function âint db_allocate_columns(db1_res_t*, unsigned int)â used but never defined [enabled by default] ../../lib/srdb1/db_res.h:98:12: warning: inline function âint db_free_columns(db1_res_t*)â used but never defined [enabled by default] ../../lib/srdb1/db_res.h:129:12: warning: inline function âint db_allocate_rows(db1_res_t*)â used but never defined [enabled by default] ../../lib/srdb1/db_row.h:74:12: warning: inline function âint db_allocate_row(const db1_res_t*, db_row_t*)â used but never defined [enabled by default] ../../lib/srdb1/db_res.h:111:12: warning: inline function âint db_free_result(db1_res_t*)â used but never defined [enabled by default] make[1]: *** [dbcassa_base.o] Error 1 make: *** [install-modules] Error 1 Anybody has got any ideas? Thanks in advance, Bou

2 6

Продается коттедж в 42км от Москвы, у горнолыжного курорта
by jiri＠iptel.org 23 Aug '12

23 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: ðÒÏÄÁÅÔÓÑ ÚÄÅÓØ www.ËÕÐÉ-ÄÏÍ-ÚÄÅÓØ.ÒÆ [...] Content analysis details: (10.2 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.5 BAYES_99 BODY: Bayes spam probability is 99 to 100% [score: 1.0000] 0.8 RCVD_IN_SORBS_WEB RBL: SORBS: sender is an abusable web server [116.203.165.242 listed in dnsbl.sorbs.net] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [116.203.165.242 listed in bb.barracudacentral.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [116.203.165.242 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines

1 0

Kamailio with asterisk for outbound calls
by Vijay Thakur 23 Aug '12

23 Aug '12

Hi All Kamailio Experts, I have configured Kamailio (kamailio 3.1.5) as media server. All things are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound Calls. For this purpose i have followed the web page : http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. In this page, some points are not clear for me , as given below: (1) In case you use *sipregs* you have to create a record for each extension where to set the 'name' to value of 'name' from *sipusers*. The rest is populated by Asterisk from registrations. (2) Be sure you configure Asterisk *to not authenticate* SIP requests coming from Kamailio. I am not sure that my local users chat is working through kamailio or asterisk, who is used for authorization. Any specific Web page to correct the issue will highly appreciated according to my scenario. Kindly guide me. Thanks in advance. -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.thakur(a)loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

2 3

interfacing with kamailio
by Carlos Cruz 22 Aug '12

22 Aug '12

I'm just starting to evaluate Kamailio to be used as a front end to a couple of Asterisk Servers. Currently I have an Adobe Flex application that interfaces with Asterisk via AMI using a socket connection. I searched on how to interface with Kamailio but wasn't able to find anything that helped me. I would appreciate it if someone could point me to where I can find information on how this can be done using Kamailio. For example I would like my Flex Application to connect to Kamailio to know the status of the SIP channels. Thanks for any help!!! Carlos

2 1

new feature: option to remove contacts based on sip nat keepalives
by Juha Heinanen 22 Aug '12

22 Aug '12

daniel, you wrote: This feature should be useful in mobile networks, to avoid stacking lot of invalid contacts which could result in lot of branches causing many retransmissions. why would a sip ua that runs in a mobile network device be configured to use udp instead of tcp? poor battery life seems to be a big problem in so called smart phones. wouldn't this kind of active pinging make it even worse? -- juha

2 1

Websocket INVITE 407 error
by David Patiño 22 Aug '12

22 Aug '12

Hi there, I'm new on kamailio users list. Now I'm developing a client in javascript (using websockets) to SIP calls. I'm using last devel version of kamailio with websocket support. I've already get a correct REGISTER transaction but now I'm not able to get a correct INVITE transaction. Actually I'm not sure if this a kamailio error or a problem with my source code however I'm using the same function to get digest info from received nonce both REGISTER and INVITE transactions. The INVITE transaction always enters in an endless loop with 407 requests. Please, is there anybody who could help me? Those are the frames of my code (obtained from chrome browser console): wsUri: ws://10.1.20.40:80/ SOCKET CREATED SOCKET OPENED SENT: REGISTER sip:10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg From: sip:101@10.1.20.40;tag=Me0arqDG6K8o7l32oJgR To: sip:101@10.1.20.40 Call-ID: 0EQSBkoG9RGgfg3E CSeq: 1 REGISTER Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws> ;reg-id=98 ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>" RESPONSE: SIP/2.0 401 Unauthorized Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=Me0arqDG6K8o7l32oJgR To: sip:101@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c Call-ID: 0EQSBkoG9RGgfg3E CSeq: 1 REGISTER WWW-Authenticate: Digest realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re" Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 SENT: REGISTER sip:10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037 From: sip:101@10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612 To: sip:101@10.1.20.40 Call-ID: 0EQSBkoG9RGgfg3E CSeq: 2 REGISTER Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws> ;reg-id=98 ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>" Authorization: Digest username="101", realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="", uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da" RESPONSE: SIP/2.0 200 OK Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612 To: sip:101@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4 Call-ID: 0EQSBkoG9RGgfg3E CSeq: 2 REGISTER Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101@10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173@10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98 Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 SENT: INVITE sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7 From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40 Call-ID: 9KoKnFdh285k2jg4 CSeq: 1 INVITE Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws ;gr=urn:uuid:beb-68a9-6396e5f4;ob> Content-Type: application/sdp RESPONSE: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545 Call-ID: 9KoKnFdh285k2jg4 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re" Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 SENT: ACK sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545 Call-ID: 9KoKnFdh285k2jg4 CSeq: 1 ACK Max-Forwards: 70 Supported: path, outbound, gruu SENT: INVITE sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40 Call-ID: 9KoKnFdh285k2jg4 CSeq: 2 INVITE Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws ;gr=urn:uuid:beb-68a9-6396e5f4;ob> Proxy-Authorization: Digest username="101", realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="", uri="sip:102@10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798" Content-Type: application/sdp RESPONSE: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007 Call-ID: 9KoKnFdh285k2jg4 CSeq: 2 INVITE Proxy-Authenticate: Digest realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re" Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 SENT: ACK sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007 Call-ID: 9KoKnFdh285k2jg4 CSeq: 2 ACK Max-Forwards: 70 Supported: path, outbound, gruu SENT: INVITE sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40 Call-ID: 9KoKnFdh285k2jg4 CSeq: 3 INVITE Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws ;gr=urn:uuid:beb-68a9-6396e5f4;ob> Proxy-Authorization: Digest username="101", realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="", uri="sip:102@10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798" Content-Type: application/sdp RESPONSE: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7 Call-ID: 9KoKnFdh285k2jg4 CSeq: 3 INVITE Proxy-Authenticate: Digest realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re" Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 .......and so on in an endless loop with 407 requests Thanks a lot Kind Regards -- --DAVID--

4 9

SIP routing configuration howto
by Kaplan, Michael A 22 Aug '12

22 Aug '12

Folks, I'm working on a 5 node network in the following topology: Host1 - Edge Proxy 1 --- Core Proxy - Edge Proxy 2 - Host2 The proxy machines contain SIP Server installations, the hosts contain user agent software. I am able to make a call end to end between the hosts, but when I look at the traffic in wireshark, the core proxy is actually just forwarding the messages at the IP-layer. I would like actual SIP routing to occur, where a host would send the INVITE to the edge, and the edge has to forward it to the core etc, with these messages actually being processed at the SIP layer for all proxy nodes. I've been trying to use the dynamic routing module, thought I'm not sure this is really right for what I need, simple static routing would likely work as well. Is it necessary for me to rewrite the route{} functions of a proxy script? I'm not sure if I'm approaching this problem the right way. Eventually I would like to have multiple core servers and be able to failover between them, but trying to keep it simple for now. Thanks, Mike -------------------------------- Michael A. Kaplan Senior Research Scientist Applied Communication Sciences 732-699-3805 (office) 973-699-6775 (cell) mkaplan(a)appcomsci.com

2 1

Advertising and instructing company seeking region rep
by serusers＠iptel.org 22 Aug '12

22 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Collectively, we focus on different sources of power, new green tech, and reusable materials. We are advertising for someone from your region to be employed by us. We require: [...] Content analysis details: (11.3 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [189.58.67.119 listed in bb.barracudacentral.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [189.58.67.119 listed in zen.spamhaus.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL 1.3 RCVD_IN_RP_RNBL RBL: Relay in RNBL, https://senderscore.org/blacklistlookup/ [189.58.67.119 listed in bl.score.senderscore.com] 0.0 RCVD_IN_SORBS_DUL RBL: SORBS: sent directly from dynamic IP address [189.58.67.119 listed in dnsbl.sorbs.net] 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] 0.0 T_FRT_PROFILE2 BODY: ReplaceTags: Profile (2) -1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1% [score: 0.0001] 1.6 URIBL_SBL Contains an URL listed in the SBL blocklist [URIs: euroinjobs.com] 1.0 RDNS_DYNAMIC Delivered to internal network by host with dynamic-looking rDNS

1 0

New Job position
by serusers＠iptel.org 21 Aug '12

21 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Collectively, we focus on different sources of power, new green tech, and reusable materials. We are advertising for someone from your region to be employed by us. We require: [...] Content analysis details: (10.6 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [91.54.224.170 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL -0.0 SPF_PASS SPF: sender matches SPF record 0.0 T_FRT_PROFILE2 BODY: ReplaceTags: Profile (2) 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [91.54.224.170 listed in bb.barracudacentral.org] 1.3 RCVD_IN_RP_RNBL RBL: Relay in RNBL, https://senderscore.org/blacklistlookup/ [91.54.224.170 listed in bl.score.senderscore.com] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] -1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1% [score: 0.0000] 1.6 URIBL_SBL Contains an URL listed in the SBL blocklist [URIs: euroinjobs.com] 1.0 RDNS_DYNAMIC Delivered to internal network by host with dynamic-looking rDNS

1 0

Send/Receive IMs
by Marty Beyers 21 Aug '12

21 Aug '12

Hey guys. I am very new to Kamailio. Basically, at this time, I want to generate a SIP message (from an SMS) within Kannel, and send this message to my Kamailio server, which will forward the SIP message to another SIP server. Therefore my Kamailio Sip server would act as a single user agent connecting to only one other SIP server. So what I am unsure of is: 1. Is there a Kamailio interface (such as an HTTP server) which Kannel can connect to in order for Kannel to send multiple messages to my Kamailio Server, which in turn would be forwarded as SIP instant messages to another SIP server? 2. Which Kamailio api can I use to formulate and send the SIP instant messages? 3. Which Kamailio callback function could be called when I receive acknowledgement responses to the SIP IMs I have sent? 4. Which Kamailio callback function could be called when I receive new SIP instant messages sent to me? It would be great if someone could help point me in the right direction. Many thanks. Fred

2 1

vacancy #428
by serusers＠iptel.org 21 Aug '12

21 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Collectively, we focus on different sources of power, new green tech, and reusable materials. We are advertising for someone from your region to be employed by us. We require: [...] Content analysis details: (7.5 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 SPF_PASS SPF: sender matches SPF record 0.0 T_FRT_PROFILE2 BODY: ReplaceTags: Profile (2) 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [109.100.26.103 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [109.100.26.103 listed in bb.barracudacentral.org] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] -1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1% [score: 0.0000] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

SUBSCRIBE/NOTIFY record route
by Spencer Thomason 21 Aug '12

21 Aug '12

Hello all, I'm using Kamailio (on a public IP) as a border proxy that handles NAT traversal (detection and keep alives) for several Freeswitch servers which have public IPs as well. I'm having a problem record-routing SUBSCRIBEs and NOTIFYs for nated UACs. If the UAC is behind NAT, I am adding a Record-Route to the initial SUBSCRIBE and NOTIFY request. This works great until the UAC (in this case a Cisco SPA509G) refreshes the subcription and sends the SUBSCRIBE directly to the Freeswitch server. Since the request doesn't pass through the proxy, Freeswitch then sends NOTIFYs directly to the UAC which only recieves the first NOTIFY as the NAT pinhole is still open from the SUBSCRIBE. The UAC did receive a NOTIFY validating the subscription, and continues to refresh the subscription directly to the freeswitch server. The completely breaks presence. What is the correct way to ensure all SUBSCRIBE/NOTIFYs pass through the proxy? I have found that I can force this behavior by modifying the Contact header in the 202 Accepted to point back to the proxy but I'm not sure if this is what I "should" be doing. Thanks for any assistance and please forgive my ignorance when is comes to SIP presence! Spencer

2 1

Re: [SR-Users] Siremis login issue
by dave mccann 20 Aug '12

20 Aug '12

Hi All I managed to figure out that the table had not been created in the database. So i deleted the install.lock file in the Siremis directory to run the install again. This time I clicked on all the boxes that were on step 2: Create Siremis Database - this create database 'siremis' Import Default Data - this imports initial data into database 'siremis' Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules Replace DB Config - this updates Siremis configuration file with the details to access databases Then after it had finished I was able to login using the default admin/admin. Thanks for the reply Max greatly appreciated for the time taken to do so - Ivan - I didnt get chance to test what you had put however I will take a note of it for future reference.......hopefully I may be able to return the favor in the future if you guys need a hand. Cheers guys Dave > From: sr-users-request(a)lists.sip-router.org > Subject: sr-users Digest, Vol 87, Issue 64 > To: sr-users(a)lists.sip-router.org > Date: Mon, 20 Aug 2012 16:17:22 +0200 > > 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. Siremis login issue (dave mccann) > 2. Re: Siremis login issue (Max M?hlbronner) > 3. Re: [sr-dev] new feature: option to remove, contacts based on > sip nat keepalives (Yufei Tao) > 4. Re: Siremis login issue (Ivan Milivojevic) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 20 Aug 2012 12:51:57 +0100 > From: dave mccann <davemac2626(a)live.co.uk> > Subject: [SR-Users] Siremis login issue > To: <sr-users(a)lists.sip-router.org> > Message-ID: <DUB117-W1040D3073CA62A3AE627CF787B90(a)phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > > Hi All > > I have installed Kamailio with Siremis, Kamailio is working fine and starts and I have been through the initial setup for Siremis however when i go to login to the web management page it gives an error: > > "Error in query:: SELECT T0.id, T0.username, T0.password, T0.enctype, > T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, > T0.create_time, T0.update_by, T0.update_time FROM `user` T0 WHERE > T0.username='openser' and status='1' LIMIT 1. SQLSTATE[42S02]: Base > table or view not found: 1146 Table 'siremis.user' doesn't exist" > > I have tried the default "admin/admin" login as mentioned in the install guide and have also tried the default passwords for Kamailio "openser - (with default password 'openserrw') and openserro - (with default password 'openserro'). I have also tried root/*password* and also siremis/siremis which is contained in the /var/www/siremis-3.2.1/siremis/config.xml however none were successful and the same error appeared. I read that it was a permissions error however have set directory /var/www/siremis-3.2.1 to chmod 777 including all sub dirs and files. > > Can anyone please help as this is that last bit that I need to overcome :) > > Many thanks and best regards. > > Dave > >

1 0

UAC module
by Ron Hopson 20 Aug '12

20 Aug '12

Hello, I cannot create a configuration consistent with uac_reg_lookup. Does someone have a configuration with a simple example of uac_reg_lookup that works with db_text file uacreg? I have kamailio 3.3 running on linux. Based on suggestions in http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html and http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html I created this record in db_text file uacreg: 2:12345678:daniel:mysipserver.com:daniel123:sipprovider.com:sipprovider.com:daniel123:xxxxx:"sip:sipprovider.com" I assume that function uac_reg_lookup retrieves registration information from this record after it is loaded into a memory table. One version of the configuration file is attached. Here are three examples of failures to create a configuration. $rU = 12345678; if(uac_reg_lookup("$rU", "$ru")){ #line 73 0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 1 0(13856) DEBUG: <core> [rvalue.c:2861]: RV is int: 12345678 0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4 0(13856) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup() 0(13856) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:73 --------- $rU = "12345678"; if(uac_reg_lookup("$rU", "$ru")){ #line 73 0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2 0(13973) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678" 0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4 0(13973) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup() 0(13973) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:73 -------- $rU = "12345678"; $ru = "sip:mysipserver.com"; if(uac_reg_lookup("$rU", "$ru")){ #line 74 0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2 0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678" 0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2 0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "sip:mysipserver.com" 0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4 0(13976) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup() 0(13976) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:74 --------------------- regards, Ron Hopson

3 5

Re: [SR-Users] [sr-dev] new feature: option to remove, contacts based on sip nat keepalives
by Yufei Tao 20 Aug '12

20 Aug '12

That's a very useful feature for updating presence information on behalf of some clients that do not support presence, or simply crashed. I will be watching this space for TCP/TLS support :) Is there any other existing way to achieve this for clients over TCP/TLS? Cheers, Yufei On 20/08/12 09:13, sr-users-request(a)lists.sip-router.org wrote: > ------------------------------ Message: 5 Date: Mon, 20 Aug 2012 > 10:13:12 +0200 From: "Olle E. Johansson" <oej(a)edvina.net> Subject: Re: > [SR-Users] [sr-dev] new feature: option to remove contacts based on > sip nat keepalives To: miconda(a)gmail.com, Development mailing list of > the sip-router project <sr-dev(a)lists.sip-router.org> Cc: "SIP Router - > Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing > List" <sr-users(a)lists.sip-router.org> Message-ID: > <1CD23BC5-1002-4A6E-B42B-09C73530DB11(a)edvina.net> Content-Type: > text/plain; charset="us-ascii" 20 aug 2012 kl. 10:06 skrev > Daniel-Constantin Mierla: >> > Hello, >> > >> > over the weekend I committed the code that allows to remove contacts from location table if the device does not respond to several attempts of SIP nat keepalives (usually OPTIONS requests). >> > >> > The feature works only for SIP over UDP, being the only transport for which nathelper module sends keepalives (or sip ping requests as written in some docs). >> > >> > Also, the lack of keepalive reply is tracked only for location records stored in memory (so it does not work with db only mode for usrloc). This is because the keepalives are sent stateless, not using tm module, to avoid overloading of timers and shared memory when dealing with large subscriber base -- therefore the implementation is very lightweight in using resources, targeting of course scalability. Detection of replies for keepalives is done based on from tag which carries the internal unique id of the location contact (which is available from v3.3). >> > >> > In order to enable this feature you have to set the keepalive_timeout parameter of nathelper module -- that's the only addition you have to add to a config that sends SIP nat keepalives. Internally, the modules store the timestamp of the last SIP keepalive reply (nathelper detects the right reply and calls usrloc API for storing the timestamp in contact structure in memory). >> > >> > http://kamailio.org/docs/modules/devel/modules_k/nathelper.html#id2535244 >> > >> > Before sending a new keepalive to a contact, it is checking if the last keepalive reply is older than keepalive_timeout value and if yes, will lower the expires to 10 seconds (if the expires is higher). There is no special handling of keepalive reply codes, all of them will update the last keepalive timestamp. >> > >> > This feature should be useful in mobile networks, to avoid stacking lot of invalid contacts which could result in lot of branches causing many retransmissions. >> > >> > There is plant to go for tcp/tls as well and detect when the connection is down, but looks like no spare time in the near future. However, right now using set_forward_no_connect() will avoid trying to create new connections for natted uses whose registration connection is down. >> > > On my personal wish list is to have an event route when a registered device becomes unreachable - like a closed TCP connection for a device behind NAT or failure to response to NAT pings. > > /O -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

1 0

Siremis login issue
by dave mccann 20 Aug '12

20 Aug '12

Hi All I have installed Kamailio with Siremis, Kamailio is working fine and starts and I have been through the initial setup for Siremis however when i go to login to the web management page it gives an error: "Error in query:: SELECT T0.id, T0.username, T0.password, T0.enctype, T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM `user` T0 WHERE T0.username='openser' and status='1' LIMIT 1. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'siremis.user' doesn't exist" I have tried the default "admin/admin" login as mentioned in the install guide and have also tried the default passwords for Kamailio "openser - (with default password 'openserrw') and openserro - (with default password 'openserro'). I have also tried root/*password* and also siremis/siremis which is contained in the /var/www/siremis-3.2.1/siremis/config.xml however none were successful and the same error appeared. I read that it was a permissions error however have set directory /var/www/siremis-3.2.1 to chmod 777 including all sub dirs and files. Can anyone please help as this is that last bit that I need to overcome :) Many thanks and best regards. Dave

3 2

new feature: option to remove contacts based on sip nat keepalives
by Daniel-Constantin Mierla 20 Aug '12

20 Aug '12

Hello, over the weekend I committed the code that allows to remove contacts from location table if the device does not respond to several attempts of SIP nat keepalives (usually OPTIONS requests). The feature works only for SIP over UDP, being the only transport for which nathelper module sends keepalives (or sip ping requests as written in some docs). Also, the lack of keepalive reply is tracked only for location records stored in memory (so it does not work with db only mode for usrloc). This is because the keepalives are sent stateless, not using tm module, to avoid overloading of timers and shared memory when dealing with large subscriber base -- therefore the implementation is very lightweight in using resources, targeting of course scalability. Detection of replies for keepalives is done based on from tag which carries the internal unique id of the location contact (which is available from v3.3). In order to enable this feature you have to set the keepalive_timeout parameter of nathelper module -- that's the only addition you have to add to a config that sends SIP nat keepalives. Internally, the modules store the timestamp of the last SIP keepalive reply (nathelper detects the right reply and calls usrloc API for storing the timestamp in contact structure in memory). http://kamailio.org/docs/modules/devel/modules_k/nathelper.html#id2535244 Before sending a new keepalive to a contact, it is checking if the last keepalive reply is older than keepalive_timeout value and if yes, will lower the expires to 10 seconds (if the expires is higher). There is no special handling of keepalive reply codes, all of them will update the last keepalive timestamp. This feature should be useful in mobile networks, to avoid stacking lot of invalid contacts which could result in lot of branches causing many retransmissions. There is plant to go for tcp/tls as well and detect when the connection is down, but looks like no spare time in the near future. However, right now using set_forward_no_connect() will avoid trying to create new connections for natted uses whose registration connection is down. Testing and feedback will be welcome. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 1

does AVP's stay availeable after the transaction was answered?
by Uri Shacked 20 Aug '12

20 Aug '12

Hi, I am trying to use an AVP the I set on the INVITE when I recieve the BYE. for some strange reason, the AVP is NULL when I test it. Any ideas? Uri

3 2

avp usage limit
by Uri Shacked 20 Aug '12

20 Aug '12

Hi, I wonder, is there a limit for the amount of AVP's i can use? Let's say i store 6-9 characters (string) each time i set an AVP. Thanks, Uri

4 3

does AVP's stay availeable after the transaction was answered?
by Uri Shacked 20 Aug '12

20 Aug '12

OK, I understan the AVP's are per transaction. How can I save info from the invite and use it on the bye? $sht(...)? $dlg(flag), $dlg(set) ? Thanks, Uri > juha writes >it is not strange reason, since avps are transaction specific and bye >does not belong to the same transaction as invite. > juha >>Uri Shacked writes: >* >I am trying to use an AVP the I set on the INVITE when I recieve the BYE.* *> *>* for some strange reason, the AVP is NULL when I test it. *

2 1

Request for advice, running SIP-Router in OpenVZ Container
by JR Richardson 19 Aug '12

19 Aug '12

Hi All, I'm considering running Kamailio as a virtual machine, with such low utilization, it doesn't seem to make sense to keep running it on a physical host server. I've been virtualizing Asterisk PBX's for years and run a host of other virtualized servers with OpenVZ, VMware and MS V-Server without issues. I'd like any opinions or thoughts about pitfalls or caveats I should be aware of specific to virtualized SIP-Router/Kamailio. Thanks in advanced. JR

4 3

append_branch with pseudo variables
by Carlos Ruiz Díaz 17 Aug '12

17 Aug '12

Hello guys, Why I can only append_branch() a SIP URI as a literal constant string? Why not from a pseudovariable? It's the same for most of the core functions. My intention is to do something like this: append_branch("$cnx3a(route1)"); t_relay(); where cnx3a is a variable containing the SIP URIs. The only "fix" I have found for this is by doing it as follows: $ru = $cnx3a(route1); append_branch(); t_relay(); Is there any other (and more elegant) way? Regards. Carlos.

2 9

Kamailio packet sourcing
by Luis Fernando Urrea 17 Aug '12

17 Aug '12

Hi all, I am sort of a new comer to kamailio and I have a config in which Kamailio listens on different IPs that correspond to SSH VPN tunnels. The config handles a MULTIDOMAIN scenario with FreeSWITCH as multi tenant media server. The situation I am facing is that when a REGISTERED user calls in through one of the SSH tunnels to the corresponding listening IP, Kamailio does not seem to source the INVITE that goes to FreeSWITCH from that IP since on FS I use "destination_field=network_addr" to try to apply different dialplan rules per domain and I see the call on FS matching a non corresponding dial plan entry sourced with an IP of a different SSH tunnel. Do I have any other option to send the call to FS using some other field to match to my convenience? I hope the description is clear enough, otherwise I could elaborate if required. Your input is highly appreciated. Regards, Luis -- O: 4000-1020 C: 6040-3624

2 1

Kamailio NAT traversal
by Spencer Thomason 17 Aug '12

17 Aug '12

Hello, I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal. Is there a way I can allow freeswitch to handle all auth but store the registration in usrloc as I need to send an options ping to the endpoints? My thought is something like this: natted endpoint -> kamailio -> freeswitch If the response is 200OK, save the registration. Thanks, Spencer

4 5

Accessing received after lookup()
by DanB 17 Aug '12

17 Aug '12

Hey All, I was wondering if anyone could point me to the right direction on the following issue: I use registrar with path to store contacts after main proxy. There I have properly stored received db field (original address of the NAT router). When I query the registrar, I get the user contact details (natted ones) in 302 redirect. However at this point I cannot access received avp (or I did not find a way yet, except calling reg_fetch_contacts which I would prefer not to since it doubles the work lookup does). Is there any catch which could save me in this scenario? After loopkup() is there any pseudovariable which I can inspect to find out user's contact data? Any other way to extract received contact data out of contact info? Thanks in advance for any kind of tip! DanB

2 5

sms module & IMS
by Oleksandr Shchirskyi 16 Aug '12

16 Aug '12

Hi, I'm using OpenIMSCore based on the SER. At the moment I try to add SMS-GW to it. I've found, that SER has sms module, and I tried to use it. But, unfortunately, I couldn't make it work. It's not very clear for me how SIP MESSAGEs can be routed to the sms module. This module works not as application server (or I'm wrong?) At the moment I have such configuration in the scscf.cfg: ============================== modparam("sms","modems","Nokia [d=/dev/ttyACM0;b=9600]") modparam("sms","networks","D1[m=10]") modparam("sms","links","Nokia[D1]") modparam("sms","domain","open-ims.test") modparam("sms","max_sms_parts",3) modparam("sms","use_contact",1) # main routing logic route{ if(method=="MESSAGE"){ sms_send_msg_to_net("D1"); } ... } ============================== As a hardware GSM gateway my cell phone is using. As I see, nothing happens when s-cscf processes SIP MESSAGEs. I don't see that these messages are forwarded to the sms module, or smth like that. Final question - is it possible to use this module as IP-SMS-GW for the OpenIMSCore? And how? I would be very appreciated for any help. Thank you in advance. -- Oleksandr Shchirskyi | Software Testing Engineer GlobalLogic Inc. | Innovation by Design ARGENTINA | CHILE | CHINA | INDIA | ISRAEL | UKRAINE | UK | USA Office: +044.492.96.95, (ext. 4296) www.globallogic.com

2 1

Suggestions for time utils
by Andreas Granig 16 Aug '12

16 Aug '12

Hi, I'd like to implement a couple of helper functions for time handling, e.g. checking day of week, day of month etc. from within the kamailio config file. What would you prefer, a new module ("timeutils" maybe?) or adding it to cfgutils (there are already time-based functions there like sleep and usleep)? The idea is to implement time based call-forwards, and an approach could be to provision various time-related values in usr_preferences table, and then check it in the config. For example, when doing call-forwards from Monday to Friday only, I could put this into usr_preferences: attribute: cf_weekday value: [1, 2, 3, 4, 5] (each entry is a separate row in usr_preferences) And in kamailio config, I'd call this: avp_db_load(...); if(is_weekday("$avp(s:cf_weekday)")) { do CF } So is_weekday would iterate over the entries in the avp list and return true if the weekday at the time of routing matches an entry in the list. A module config param could control whether to use gmtime or localtime for matching. Does this make sense? Suggestions for other approaches? I'd rather prefer to do it directly in config instead of using some external interpreter like lua, python etc. Andreas

3 17

Re: [SR-Users] rtpproxy module issue
by phillman25 16 Aug '12

16 Aug '12

Dear SamyGo Thanks for your response. Yes you're right a folder called rtpproxy.so was left in the directory /usr/local/lib/kamailio/modules_k that had not been updated to the date kamailio was compiled and once i deleted it everything was then ok. I checked in directory /usr/local/lib/kamailio/modules and there i found a new rtpproxy.so folder that was updated on the installation date. Seems like rtpproxy.so in the new version is installed in /usr/local/lib/kamailio/modules and not in /usr/local/lib/kamailio/modules_k. Is that correct? Well everything is working ok now! Thanks for your assistance! Regards Phillip > Message: 4 > Date: Thu, 16 Aug 2012 13:29:35 +0500 > From: SamyGo <govoiper(a)gmail.com> > Subject: Re: [SR-Users] rtpproxy module issue > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAJUJwtjB7fmm1pa60wJMLPPte= > Qu_AeZ1z2aqysQaZch_Pb+dQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Though not an expert, I got few things in mind for your case. In the > upgrade process the modules folder remain the same ? right ! if that is so > is there any chances that the new installation skipped installation of > newer rtpproxy module and hence the old version rtpproxy module could've > been left in place. That could be a reason for not identifying the older > module. > This whole speculation could be wrong if the newer version supports the > older version modules. > > BR > Sammy > > > On Thu, Aug 16, 2012 at 10:56 AM, phillman25 <phillman25(a)gmail.com> wrote: > > > Dear List > > > > I have just upgraded from kamailio V3.2.2 to kamailio V 3.3.1 > > However, when trying to start kamailio its giving me the below error: > > > > > > ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open > module > > </usr/local/lib/kamailio/modules_k/rtpproxy.so>: > > /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: > qm_realloc > > : <core> [cfg.y:3591]: parse error in config file > > /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to > > load module > > > > > > Does anyone know why rtpproxy module cannot be loaded? > > > > > > This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so: > > > > > > > > shell-init: error retrieving current directory: getcwd: cannot access > > parent directories: No such file or directory > > linux-gate.so.1 => (0x00276000) > > libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000) > > libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 > (0x00768000) > > libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000) > > /lib/ld-linux.so.2 (0x00114000) > > > > > > > > Regards Phillip > > > > _______________________________________________ > > 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

Is Route Header in SIP Invite Mandatory?
by Apoorv Khare 16 Aug '12

16 Aug '12

Hi All, Does Kamailio has a mandatory requirement for SIP Invite to have Route header? Although RFC-3261 doesn't have. My requirement is to take out route header from the client if I do that I don't see PRACK being send out from Kamilio to Media GW which causes call drop eventually in 15sec. Any thoughts.......... BR, Apoorv

2 1

rtpproxy module issue
by phillman25 16 Aug '12

16 Aug '12

Dear List I have just upgraded from kamailio V3.2.2 to kamailio V 3.3.1 However, when trying to start kamailio its giving me the below error: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/rtpproxy.so>: /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc : <core> [cfg.y:3591]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to load module Does anyone know why rtpproxy module cannot be loaded? This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so: shell-init: error retrieving current directory: getcwd: cannot access parent directories: No such file or directory linux-gate.so.1 => (0x00276000) libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000) libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 (0x00768000) libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000) /lib/ld-linux.so.2 (0x00114000) Regards Phillip

2 1

Debian packages still installing 3.3.0
by DanB 16 Aug '12

16 Aug '12

Hey Guys, Not sure if important, but what I noticed is that the latest debian stable as well as latest 33 packages, they both install still kamailio 3.3.0. In order to get 3.3.1 I needed to use nightly build packages. Is this what is desired? Thanks, DanB

2 1

Kamailio and ACME
by Fatima Chahrour 15 Aug '12

15 Aug '12

Sent from iPhone

4 7

Dispatcher: ds_is_from_list does not longer populate grp_avp
by DanB 15 Aug '12

15 Aug '12

Hey Guys, Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1. Having the following code: """ if ds_is_from_list() { xlog("Request coming from SCloud"); xlog("Checking grp_internal, having: $avp(s:dsgrp)"); } """ will produce the following output to syslog (dsgrp null instead of group id): """ Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>: Request coming from SCloud Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>: Checking grp_internal, having: <null> """ I have in dispatcher set: modparam("dispatcher", "grp_avp", "$avp(dsgrp)") and I am loading the rules from file. I should mention that this block was working fine before upgrade. Am I doing something wrong? Thanks in advance! DanB

2 8

A much more modest initial goal: get Kamailio dev version up and running
by Juan Castro 14 Aug '12

14 Aug '12

1 1

Re: [SR-Users] rtpproxy module issue
by phillman25 14 Aug '12

14 Aug '12

Hello This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so: shell-init: error retrieving current directory: getcwd: cannot access parent directories: No such file or directory linux-gate.so.1 => (0x00276000) libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000) libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 (0x00768000) libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000) /lib/ld-linux.so.2 (0x00114000) Regards Phillip Message: 3 Date: Tue, 14 Aug 2012 08:49:13 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] rtpproxy module issue To: sr-users(a)lists.sip-router.org Message-ID: <502A4949.2000707(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Please post output of: ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so On 08/14/2012 08:47 AM, phillman25 wrote: > Dear List > > I have just installed kamailio V 3.3.1 > However, when trying to start kamailio its giving me the below error: > > > ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open > module </usr/local/lib/kamailio/modules_k/rtpproxy.so>: > /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc > : <core> [cfg.y:3591]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to > load module > > > Does anyone know why rtpproxy module cannot be loaded? > > > Thanks > Phillip > > > _______________________________________________ > 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 > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 0

rtpproxy module issue
by phillman25 14 Aug '12

14 Aug '12

Dear List I have just installed kamailio V 3.3.1 However, when trying to start kamailio its giving me the below error: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/rtpproxy.so>: /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc : <core> [cfg.y:3591]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to load module Does anyone know why rtpproxy module cannot be loaded? Thanks Phillip

2 1

Remove m=video from SDP header
by phillman25 14 Aug '12

14 Aug '12

Dear List I am trying to remove specific lines from the following original SDP body: Content-Type: application/sdp Content-Length: 406 v=0 o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx b=CT:384 t=0 0 m=audio 11904 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 11602 RTP/AVP 34 99 a=rtpmap:34 H263/90000 a=rtpmap:99 H264/90000 a=sendrecv I need to remove the following lines from the above SDP body as my International carrier does not support Video capabilities: m=video 15042 RTP/AVP 34 99 a=rtpmap:34 H263/90000 a=rtpmap:99 H264/90000 I used the following code in Kamailio config: if(has_body("application/sdp") && search_body("m=video")){ subst_body('#m=video ([0-9]+) RTP/AVP (.*)$# #'); subst_body('#a=rtpmap:34 (.*)$# #'); subst_body('#a=rtpmap:99 (.*)$# #'); subst_body('#a=sendrecv(.*)$# #'); } Content-Type: application/sdp Content-Length: 325 P-hint: outbound v=0 o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx b=CT:384 t=0 0 m=audio 11904 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv As you can see the m=video body has been removed, however, calls are still failing. Is there something i have missed? I am using Kamailio v3.2.2 Thanking you in advance! Phillip

3 3

How best to experiment with SIP over Websocket?
by Juan Castro 13 Aug '12

13 Aug '12

As per the instructions here... http://www.kamailio.org/w/2012/07/websockets ...I downloaded the latest source from Git... http://www.kamailio.org/w/download/ ...but I don't see any mention of Websocket anywhere in the source. I suspect I didn't actually download the development tree, but the 3.3 branch. Last time I messed with any VoIP code was in 2008 (OpenSER and Asterisk) and now I'm starting all over again. I apologize for the dumb newbie question, but how do I test the Websocket code? -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100

2 4

Possible bad lcr module, after upgrade to 3.3.0 from 3.2
by Alexey Mechanoshin 13 Aug '12

13 Aug '12

Hello. After upgrade kamailio and db, lcr module bad routing if entered *any value*to from_uri column in lcr_rule table: mysql> select * from lcr_rule where id=5; +----+--------+--------+-------------------------------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+--------+-------------------------------+-------------+---------+---------+ | 5 | 1 | yyyy | sip:x[xx]\d{2}@xxx.xxx.xx.xx | NULL | 0 | 1 | +----+--------+--------+-------------------------------+-------------+---------+---------+ and load_gw(...) skip this rule, in log: INFO: lcr [lcr_mod.c:1493]: skipping disabled <gw/rule> = <1/5> If remove value from 'from_ui' - rule enabled and all work. Why is turned off rule ? Linux Debian squeeze, upgraded from "deb http://deb.kamailio.org/kamailiosqueeze main" -- "В связи с отсутствием интереса общественности, конец света отменяется" (с) ? С Уважением, Механошин Алексей

3 11

Kamailio Presence with XCAP not working accordingly
by SamyGo 13 Aug '12

13 Aug '12

Hi, I've followed the tutorial on kab.asipto.com for presence using built-in xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple I'm using Kamailio version 3.3.1 and did minor changes in modparams and rtpproxy function calls and the kamailio accepted the configurations file posted on the page and started. But the problem is that I don't get the presence status of the contacts still. Please suggest what to look for and how to troubleshoot this. I get this error on Jitsi - image attached. ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource can not be read" Regards, Sammy

2 9

Best tool for crafting sip messages
by Anton Kvashenkin 13 Aug '12

13 Aug '12

Hello, guys. What is the best tool for crafting sip messages for testing purpose? For example, simple REGISTER or INVITE message. I'm new to kamailio (not sip) so it would be great for me to use debbuger module for step-by-step digging into cfg.

6 7

$avp reset to <null> with no reason?
by Uri Shacked 13 Aug '12

13 Aug '12

Hi,**** ** ** I encountered a problem. While doing some avp's settings I notice that one avp that I use is changing to <null> without me doing anything.**** ** ** Here is the part of the cfg file:**** ………….**** ………….**** Route[SET]{**** …………..**** $avp(Sfeaturetype)=$avp(next_feature_type);**** …………**** Route(DIDSRV);**** }**** ** ** route[DIDSRV] {**** xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV type = $avp(Sfeaturetype), index = $avp(SfeatureIndx) $C(xx)\n");**** if($var(srvcount)>3){**** xlog("L_NOTICE","$C(rg)Too many services loops $C(xx)\n");**** update_stat("nts_subs_srv_loop", "+1");**** $avp(TRMCS)="10";**** t_reply(403,"too many loops");**** exit;**** }**** $var(srvcount)=($var(srvcount)+1);**** xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV before switch to $avp(SfeatureType) or $avp(next_feature_type) $C(xx)\n");**** switch ($avp(SfeatureType)){**** case "0" :**** ……….**** ………**** ** ** On the route "SET" I put the value from the $avp(next_feature_type) in $avp(Sfeaturetype).**** After that I call the route "DIDSRV" and use switch($avp(Sfeaturetype)). Before the switch I xlog the value of the avp twice.**** As you can see in the log below, on the first xlog print, the value is 0 (as I expect). On the second print it resets to <null>**** WHY?**** ** ** Log:**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:413]: qm_malloc(0x7fe6f9aff000, 72) returns address 0x7fe6fcc4d330 frag. 0x7fe6fcc4d300 (size=72) on 1 -th hit**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>: #033[0;31;42m SCRIPT: before DIDSRV type = 0, index = 8888888889 #033[0;39;49m**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>: #033[0;31;42m SCRIPT*: in DIDSRV type = 0*, index = 8888888889 #033<8888888889%C2%A0%20%23033> [0;39;49m**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 848) called from <core>: rvalue.c: rval_new_empty(236)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 848) returns address 0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from <core>: rvalue.c: rval_destroy(141)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from <core>: rvalue.c: rval_new_empty(236)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>: #033[0;31;42m SCRIPT*: in DIDSRV before switch to <null>* or 0 #033[0;39;49m **** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 928) called from <core>: rvalue.c: rval_new_empty(236)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 928) returns address 0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from <core>: rvalue.c: rval_destroy(141)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from <core>: rvalue.c: rval_new_empty(236)

1 1

Kamailio LCR Module test scenario
by Fatima Chahrour~Vanrise Support 12 Aug '12

12 Aug '12

Dears, Am trying to apply Kamailio LCR feature lab test, after am finally able to run Kamailio with no errors, am not able to reach my target clarified in this following scenario: Call 961312345 - route the call to 192.x.x.15 in first place and if failed route the call to 192.x.x.10. I applied needed configuration in lcr tables (attached image) and the routing script is: route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (is_method("INVITE")) { if (!load_gws(1)) { sl_send_reply("503", "Unable to load gateways"); exit; } } if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using ourUSRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } BUT when attempting a call am getting the message 'Call failed: Not found .' Any help to achieve the successful call using above scenario is highly appreciated. Thanks in advance. F.Chahrour

2 2

postgres in kamailio 3.3.0-9.1
by Andrew O. Zhukov 09 Aug '12

09 Aug '12

The same trouble with: completely updated Centos 5 the last Kamailio RPM from http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_… and postgresql-libs.x86_64.8.1.23-5.el5_8 several month ago I try postgresql84-libs.x86_64 with the same result. It's not possible to use Kamailio RPM with postgres backend. Need to assemble it manually. Bruno Bresciani wrote: > Hi, > > I configure the Kamailio 3.1.2 with postgres but I cann't start. In the > log file is generated the following error: > > // > > ERROR: load_module: could not open module > </home2/local/kamailio/lib/kamailio/modules/db_ > postgres.so>: /home2/local/kamailio/lib/kamailio/modules/db_postgres.so: > undefined symbol: PQdescribePrepared Have you built Kamailio yourself? Looks like db_postgres.so can not find the postgresql libraries, or it was built with a different library version. Maybe ldd can give you some details: ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so regards Klaus

4 8

Re: [SR-Users] Problem Starting kamailio on Debian Squeeze [solved]
by JR Richardson 08 Aug '12

08 Aug '12

On Wed, Aug 8, 2012 at 12:13 PM, JR Richardson <jmr.richardson(a)gmail.com> wrote: > Hi All, > > I'm running redundant kamailio 3.0.4 servers in production, have been > for a long time with great success. They were installed on debian > Lenny. One of my servers crashed. I can't seem to do a debian lennyy > install because that version is archived now. I'm trying to install > on debian squeeze but still using kamailio 3.0.4 for compatibility > reasons. The install went pretty normal except during boot up, > kamailio starts before mysql and networking so it starts then exists. > > Once the server is fully booted, kamailio starts with the init script > without error. > > I have adjusted the init script to check for networking and mysql to > start first and performed an 'update-rc.d kamailio defaults' but still > no luck. > > Any help will be appreciated. > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses Yep, this is an http://wiki.debian.org/LSBInitScripts/DependencyBasedBoot issue. Thanks Alex. The old init scrip that ships with Kamailio 3.0.4 that I am using is not 100% compatible so I pulled a hack out of the hat and inserted 'sleep 20' before of 'check_fork ()' in the kamailio init script. Kamailio now waits long enough for all services to start before launching, simple but effective. Thanks. JR -- JR Richardson Engineering for the Masses

3 4

Porblem Starting kamailio on Debian Squeeze
by JR Richardson 08 Aug '12

08 Aug '12

Hi All, I'm running redundant kamailio 3.0.4 servers in production, have been for a long time with great success. They were installed on debian Lenny. One of my servers crashed. I can't seem to do a debian lennyy install because that version is archived now. I'm trying to install on debian squeeze but still using kamailio 3.0.4 for compatibility reasons. The install went pretty normal except during boot up, kamailio starts before mysql and networking so it starts then exists. Once the server is fully booted, kamailio starts with the init script without error. I have adjusted the init script to check for networking and mysql to start first and performed an 'update-rc.d kamailio defaults' but still no luck. Any help will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Kernel Module Issue
by Vijay Thakur 08 Aug '12

08 Aug '12

Thanks for the hint. What is the file name in /etc/modprob.d/ where i am supposed to enter this module name to disable it. Thanks Vijay Tha Message: 1 Date: Mon, 06 Aug 2012 09:01:03 -0400 From: Richard Fuchs<rfuchs(a)sipwise.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To:sr-users@lists.sip-router.org Message-ID:<501FC00F.6000604(a)sipwise.com> Content-Type: text/plain; charset="iso-8859-1" On 08/06/12 06:08, Vijay Thakur wrote: > Hi all, > > I have configure Kamailio 3.1.5 Server. All things are working fine. > When i make a call from Soft phone (X-Lite) to iphone, all is working > fine. But in other case call from iphone to Softphone is not working, > even not ringing. During checking the logs i am getting the error: > > Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= > MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 > DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF > PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 > RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) This is coming from nf_conntrack_sip, which is a netfilter connection tracking kernel module for SIP. I've never used it, but judging from what Google brings up, it seems to be very buggy. You should be able to just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work and/or if you want to keep it from auto-loading, you can blacklist it in /etc/modprobe.d/ and then reboot. HTH copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

2 1

AVPOPS / TM behavior
by Brandon Armstead 08 Aug '12

08 Aug '12

Hello, I am curious if there is any documentation on how AVP's processing works in the following scenario below. UAC 1 -> KAMAILIO -> KAMAILIO -> DEST It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I relay back to the same KAMAILIO proxy (self)? Is there any documentation on why or when this would occur? Is there a better way to handle such a scenario? i.e. more dynamic internal routing, vs relaying to self. Thanks as always in advance! Sincerely, Brandon Armstead

2 5

local_timer: add_timeout: 0 expire timer added
by Barthel Marco (CI/AFU1) 08 Aug '12

08 Aug '12

Hi, Occasionally I can see the following log output from Kamailio in our systems: WARNING: <core> [local_timer.c:97]: WARNING: local_timer: add_timeout: 0 expire timer added Can someone explain what it means? Many thanks. Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) www.bosch.com<http://www.bosch.com> Tel. +49 711 811-3602341 marco.barthel(a)de.bosch.com<mailto:marco.barthel@de.bosch.com> Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Franz Fehrenbach; Geschäftsführung: Volkmar Denner, Siegfried Dais; Stefan Asenkerschbaumer, Bernd Bohr, Rudolf Colm, Dirk Hoheisel, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Werner Struth, Peter Tyroller

1 0

Re: [SR-Users] Kamailio with APNS
by Duong Manh Truong 08 Aug '12

08 Aug '12

As your first solution with UAC module: "As a plan B we instead opted for sending via uac module the invite to an oversip instance (it's open-source since a week or two), which triggers an HTTP request towards APNS, and on kamailio check with inv-timers every few seconds whether the client came online in the meanwhile, then complete the call. The drawback is that you can wait only for so many loop interations due to the max branch limitation in kamailio (you'd need to recompile it to set it higher)." I read about the UAC module but feeling very hard to implement it as your suggestion, http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html Please give me and others more details about this? Which functions those we need ? Are they in: UAC module, Exec module, tm module ?? Thanks

1 0

set_advertised_port using pseudovariable
by Neven Boric 07 Aug '12

07 Aug '12

Hi, I'm trying to use Kamailio as an outbound proxy behind a NAT (all clients and kamailio itself are behind the same NAT): UACs -> Kamailio -> NAT router -> PBX (hosted, public server) I figured I could detect the external source port used by the router by periodically sending an OPTIONS request to the public server, then capture the rport value in the reply, and use that port as an input to set_advertised_port. Because I'm doing this periodically, it should have the added value of keeping the NAT mapping alive. Otherwise, I should be able to detect it, as the rport value will change. I managed to periodically send the OPTIONS request, read the rport value on the reply message and store it on a pseudovariable using: $var(rport) = $sel(v.rport); But when I try to use that value to set the advertised port, I get an error, as if set_advertised_port only accepted literal values: if( $var(rport) != 0) { set_advertised_port($var(rport)); } Not starting Kamailio: invalid configuration file! -e 0(4692) : <core> [cfg.y:3591]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 504, column 24-34: syntax error 0(4692) : <core> [cfg.y:3594]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 504, column 35: bad argument, string expected ERROR: bad config file (2 errors) So, am I missing something or does set_advertised_port actually only accept literal values? Thanks Neven Boric

2 2

Re: [SR-Users] sr-users Digest, Vol 87, Issue 18
by Vijay Thakur 07 Aug '12

07 Aug '12

Thanks all of your for reply. Here is the output of command: root@li496-23:~# lsmod | grep -i sip root@li496-23:~# This command is giving blank output. This is linnode server. So the zcat /proc/config.gz command reveals the kernel loaded modules. You can see complete output at pastebin: http://pastebin.com/a0iqpc4D. Thanks in advance Vijay Th. On Monday 06 August 2012 08:26 PM, sr-users-request(a)lists.sip-router.org wrote: > 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: Kernel Droping SIP packet (Richard Fuchs) > 2. Re: Kernel Droping SIP packet (Alex Balashov) > 3. Re: Kernel Droping SIP packet (Konstantin M.) > 4. Re: Kernel Droping SIP packet (Alex Balashov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 06 Aug 2012 09:01:03 -0400 > From: Richard Fuchs <rfuchs(a)sipwise.com> > Subject: Re: [SR-Users] Kernel Droping SIP packet > To: sr-users(a)lists.sip-router.org > Message-ID: <501FC00F.6000604(a)sipwise.com> > Content-Type: text/plain; charset="iso-8859-1" > > On 08/06/12 06:08, Vijay Thakur wrote: >> Hi all, >> >> I have configure Kamailio 3.1.5 Server. All things are working fine. >> When i make a call from Soft phone (X-Lite) to iphone, all is working >> fine. But in other case call from iphone to Softphone is not working, >> even not ringing. During checking the logs i am getting the error: >> >> Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= >> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 >> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF >> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 >> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) > This is coming from nf_conntrack_sip, which is a netfilter connection > tracking kernel module for SIP. I've never used it, but judging from > what Google brings up, it seems to be very buggy. You should be able to > just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work > and/or if you want to keep it from auto-loading, you can blacklist it in > /etc/modprobe.d/ and then reboot. > > HTH > >

1 0

LCR - inconsistent return results for gateway selection.
by Graham Wooden (personal) 06 Aug '12

06 Aug '12

Hello, I am noticing that some of the time (not always consistent) that load_gw() / next_gw() will not always return the gateway with the lowest priority number. For example, when I look up a route for 205595, I have three grp_ids (gateways) that services the destination; with them being separated with 1, 2, and 3 priorities: id prefix from_uri grp_id priority 7468 205595 NULL 1 1 7469 205595 NULL 2 2 7470 205595 NULL 7 3 About 1/2 of the time, it will chose the priority 1 gateway, as it should. But randomly, it will choose the other gateways on the first try. In this particular case, all three gateways are going to the same destination IP (just using different hostnames in the "gw" mySQL table) - so I know the gateways are up. Also, I am not noticing any failures and going to the next gateway, like it choosing #1, it failing and then going to #2. What kind of debugging can I do on this gateway selection process? I am running 1.5.5 notls, and have approximately 425K routes. Kamailio is started with "-m 512" for the additional memory. Thanks, -graham

2 3

Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

Can you paste this output too? lsmod | grep -i sip -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Thanks for prompt reply. Here is the out put if command iptables -L -n: Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server. Vijay TH -============================================================================== Date: Mon, 06 Aug 2012 06:35:04 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To: sr-users(a)lists.sip-router.org Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618(a)email.android.com> Content-Type: text/plain; charset="utf-8" You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH

2 1

Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

Ah, you short-circuited where I was going with this. Couldn't remember the name of the module. Yep, Vijay. What Richard said. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Richard Fuchs <rfuchs(a)sipwise.com> wrote:On 08/06/12 06:08, Vijay Thakur wrote: > Hi all, > > I have configure Kamailio 3.1.5 Server. All things are working fine. > When i make a call from Soft phone (X-Lite) to iphone, all is working > fine. But in other case call from iphone to Softphone is not working, > even not ringing. During checking the logs i am getting the error: > > Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= > MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 > DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF > PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 > RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) This is coming from nf_conntrack_sip, which is a netfilter connection tracking kernel module for SIP. I've never used it, but judging from what Google brings up, it seems to be very buggy. You should be able to just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work and/or if you want to keep it from auto-loading, you can blacklist it in /etc/modprobe.d/ and then reboot. HTH _______________________________________________ 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

Kernel Droping SIP packet
by Vijay Thakur 06 Aug '12

06 Aug '12

Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH

2 1

Re: [SR-Users] B2BUA issues
by Alex Balashov 06 Aug '12

06 Aug '12

For your relatively narrow, specific applications in your topology, no. You'd be better off installing SEMS per se. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/phillman25 <phillman25(a)gmail.com> wrote:Hello Alex Will try with SEMS first, found something called sip:provider CE v2.4 from http://www.sipwise.com/news/announcements/spce-v2_4-release/ if i'm not mistaken, this seems to combine Kamailio with SEMS? Do you think that this might be an easier installation rather than installing SEMS on its own as it seems to provide more documentation? Thanks again! On Mon, Aug 6, 2012 at 1:33 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported. Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ phillman25 <phillman25(a)gmail.com> wrote: Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip

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Re: [SR-Users] Kernel Droping SIP packet
by Vijay Thakur 06 Aug '12

06 Aug '12

Thanks for prompt reply. Here is the out put if command iptables -L -n: *Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server. Vijay TH -============================================================================== * Date: Mon, 06 Aug 2012 06:35:04 -0400 From: Alex Balashov<abalashov(a)evaristesys.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To:sr-users@lists.sip-router.org Message-ID:<17t1fd0fbly84yy2vuy8qlcp.1344249304618(a)email.android.com> Content-Type: text/plain; charset="utf-8" You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web:http://www.evaristesys.com/Vijay Thakur<vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables :http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH * *

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Re: [SR-Users] B2BUA issues
by Alex Balashov 06 Aug '12

06 Aug '12

The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported. Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/phillman25 <phillman25(a)gmail.com> wrote:Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip

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Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] B2BUA issues
by phillman25 06 Aug '12

06 Aug '12

Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? >From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b. 66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx; lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip

1 0

B2BUA issues
by phillman25 06 Aug '12

06 Aug '12

Dear List I am trying to accomplish the following: Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) When trying the above scenario, the call is silent and drops after a few seconds. In syslog i observe the following error: *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing name-addr (sip:22030305@192.168.10.189:5060>)* Looking at the sip trace i see that his might be caused by the ACK message received from the ASTERISK PABX? : ACK sip:22030305@192.168.10.189:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ yyy.yyy.yyy.yyy ;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> Max-Forwards: 70 From: "22498045" <sip:22498045@192.168.10.189>;tag=as166b1eea To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 CSeq: 102 ACK User-Agent: FPBX-2.8.1(1.8.12.0) Content-Length: 0 After contacting Cisco they informed us that issue is cause by B2BUA from our current version of Cisco PGW 2200 that doesn't support this feature. Is there a module, solution that i can implement on Kamailio that could temporarily resolve this issue? Thanking you in advance. Phillip

2 1

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