- sr-users - kamailio.org

Internet RTP filters
by Dominguez Jover, Ricardo 30 Jun '11

30 Jun '11

Hi everybody, First to say sorry if this is not the right list to send this post, anyway we hope somebody can help us. We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for NAT issues in the server side and STUN for NAT issues in the internet client side. Calls in Spain are working fine. This month one of our users has gone to Rwanda (our University has created a Hospital and a School in a region called Nemba). They are using an internet connection by satellite, with a low bandwith (128kb down, 16 up). In some countries we have trouble with internet calls. SIP signaling works fine, the call rings and is stablished, but there is a problem with RTP streams: - RTP streams sent by RTP proxy to client in internet never arrives to the client in the internet (although they RTP is sent to the right IP and port) - RTP streams sent by client in the internet never arrives to RTP proxy (although RTP is sent to the right IP and port) As I said this happens sometimes, and it's happening in Rwanda. We usually solve this problem making a VPN connection to the University, then the RTP stream works fine. However when our Rwanda's users connect to VPN, the Bandwith goes down (45 kbps down and 8kbps up), so they can hear as perfectly, but we can't hear them. Does anyone know if T companies are filtering RTP traffic? I've tried to change ports in RPT Proxy with no result. How could we solve this? Thanks in advance and best regards, Ricardo Dominguez Universidad Miguel Hernández de Elche

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Re: [SR-Users] MWI handling
by Ovidiu Sas 30 Jun '11

30 Jun '11

If you are using asterisk as a voicemail server, let asterisk to handle the SUBSCRIBE and don't involve the kamailio presence server in this process. If you really want to use kamailio as a mwi presence server, then you will need to build some external scripts for pooling the asterisk mwi db and send PUBLISH out. I would stick with asterisk handling the mwi subscription and configure kamailio as a relay only for those events. Regards, Ovidiu Sas On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy <spinov_evgeniy(a)intalisan.com> wrote: > This makes the main problem as in documents I've found on asterisk is > that it doesn't support PUBLISH, i.e. there is no way to force sending > them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme. > > Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio > about MWI? If yes, how? Cause right now, for me, it looks like a theory, > cause no mechanics to force asterisk send PUBLISHes. > > Thank you. > > Regards, > Evgeniy Spinov > > >> If you are using asterisk for voicemail, you should let asterisk >> notify the subscribers about mwi. >> If you want to use kamailio for notifications, then you need to >> configure asterisk to send PUBLISH requests to kamailio and kamailio >> will notify the subscribers. > > >> Regards, >> Ovidiu Sas > >

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Re: [SR-Users] MWI handling
by Spinov Evgeniy 30 Jun '11

30 Jun '11

I've done that in this fashion: Kamailio is only proxing SUBSCRIBEs to Asterisk and it's replying with NOTIFYs to Kamailio, Kamailio is just proxying reply to UAC. So Kamailio is transparent for MWI. But it looks like spike. I'd better send PUBLISHes from Asterisk to Kamailio for all the clients and Kamailio reply with NOTIFY to subscribed ones. This is like Kamailio MWI module was designed. Anyway, thanks for the help. Scripts for making PUBLISHES is spike even more. On Thu, 2011-06-30 at 09:53 -0400, Ovidiu Sas wrote: > If you are using asterisk as a voicemail server, let asterisk to > handle the SUBSCRIBE and don't involve the kamailio presence server in > this process. > If you really want to use kamailio as a mwi presence server, then you > will need to build some external scripts for pooling the asterisk mwi > db and send PUBLISH out. > > I would stick with asterisk handling the mwi subscription and > configure kamailio as a relay only for those events. > > > Regards, > Ovidiu Sas > > On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy > <spinov_evgeniy(a)intalisan.com> wrote: > > This makes the main problem as in documents I've found on asterisk is > > that it doesn't support PUBLISH, i.e. there is no way to force sending > > them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme. > > > > Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio > > about MWI? If yes, how? Cause right now, for me, it looks like a theory, > > cause no mechanics to force asterisk send PUBLISHes. > > > > Thank you. > > > > Regards, > > Evgeniy Spinov > > > > > >> If you are using asterisk for voicemail, you should let asterisk > >> notify the subscribers about mwi. > >> If you want to use kamailio for notifications, then you need to > >> configure asterisk to send PUBLISH requests to kamailio and kamailio > >> will notify the subscribers. > > > > > >> Regards, > >> Ovidiu Sas > > > >

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Re: [SR-Users] MWI handling
by Spinov Evgeniy 30 Jun '11

30 Jun '11

This makes the main problem as in documents I've found on asterisk is that it doesn't support PUBLISH, i.e. there is no way to force sending them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme. Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio about MWI? If yes, how? Cause right now, for me, it looks like a theory, cause no mechanics to force asterisk send PUBLISHes. Thank you. Regards, Evgeniy Spinov > If you are using asterisk for voicemail, you should let asterisk > notify the subscribers about mwi. > If you want to use kamailio for notifications, then you need to > configure asterisk to send PUBLISH requests to kamailio and kamailio > will notify the subscribers. > Regards, > Ovidiu Sas

1 0

mq_fetch and while loops
by Watkins, Bradley 30 Jun '11

30 Jun '11

I'm having an issue trying to use the construct show in the module documentation. The docs show: while(mq_fetch("myq")) { xlog("$mqk(myq) - $mqv(myq)\n"); } And I'm doing essentially the same thing (differently named mqueue, but otherwise identical), but invariably I get this in the logs: ERROR: <core> [action.c:1388]: ERROR: runaway while (559, 28): more then 100 loops Am I doing something wrong, or is mq_fetch not returning appropriate values for successful versus failed fetches? This is on Kamailio 3.1.4, FWIW. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

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Table location, methods field
by Andrew O. Zhukov 30 Jun '11

30 Jun '11

Can someone send me bitmasks for the Subj. Or just the reference to include file with comments in a source code. I can't find it in all set of availible document's since the SER router. :( I just have to know which UA except Grandstreams an CSipsample use RFC3311 :) Thanks in advance. Andrew O. Zhukov

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kamailio + rtpproxy behind nat possible?
by MingHon 30 Jun '11

30 Jun '11

will kamailio as proxy behind nat and UACs behind another nat work? port forward sip and rtp done in the router. UACs register successfully but no audio. advertised_address = public_ip advertised_port = sip both define after the line of listen=public_ip please advice. thanks. -- Regards, MingHon

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Proxy setup in a non-recursive mode
by Abdul Hakeem 29 Jun '11

29 Jun '11

Hello, Could anyone help with suggestions on how to setup a Proxy ( R-Proxy) that only does a route lookup and pass the parameter back to origination Proxy/Gateway A to forward the call based on the new parameter. On receipt of an invite from GW/Proxy A, R- Proxy will do a route lookup for the called number 447988870290, If it finds a route then it should append the network prefix to the new INVITE or rewrite the INVITE 23430442088870290 in the 302 Moved Temporarily message sent back to GW/Proxy A in the Contact field and there are two CC-Diversion header fields included; at GW/Proxy (IP address) who originated the call and already has a route or dial plan for the new prefix and also another one for same at GW/Proxy (IP address). Is there any way to do this without creating a loop in the network ? Previously this setup can be achieved with the originating network having 2 GW/Proxies i.e. GW/Proxy A and GW/Proxy B. In the 302 Temporarily moved message the Contact will show GW/Proxy B, while the 2 CC-Diversion header fields will add GW/Proxy B and GW Proxy A. Regards, Abdul Hakeem

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MWI handling
by Spinov Evgeniy 29 Jun '11

29 Jun '11

Hello. Trying to understand how to work with the presence_mwi module. It seems to me, that it's just for my purpose - inform subscribed peers about waiting messages. But it do not export any functions, noting. The scheme is simple: Peer->Kamailio->Asterisk. Asterisk can accept voicemail and when the message is saved I want to receive NOTIFY for subscribed peer. As I understand, peer should register on Kamailio, kamailio should accept this subscription and send subscription to asterisk. After this, need to proxy all NOTIFY to the peer. Anyway, cannot find any way to do that. How it's usually done, please, advice. Thanks in advance.

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Re: [SR-Users] Questions about LCR module
by Fabian Borot 29 Jun '11

29 Jun '11

Thanks Juha I populated the lcr tables by hand. This is how they look now, I have 2 lcr_ids [1 and 2], each instance has its own gw. And each instance has its own prefix to match for [4444 and 5555]. The idea is that a call comes in with R-URI 444413051234567, it should match the lcr_id= 200, then it should go out using gw 192.168.169.200 etc mysql> select * from lcr_gw; +----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | tag | flags | defunct | +----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+ | 1 | 200 | switchvox | 192.168.169.200 | pbx1.mydomain.com | 5060 | none | 1 | 1 | 0 | | 0 | 0 | | 2 | 201 | switchvox2 | 192.168.169.201 | pbx1.mydomain.net | 5060 | none | 1 | 1 | 0 | | 0 | 0 | +----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+ mysql> select * from lcr_rule; +----+--------+--------+----------+---------+---------+ | id | lcr_id | prefix | from_uri | stopper | enabled | +----+--------+--------+----------+---------+---------+ | 1 | 200 | 4444 | | 0 | 1 | | 2 | 201 | 5555 | | 0 | 1 | +----+--------+--------+----------+---------+---------+ mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 200 | 1 | 1 | 5 | 1 | | 2 | 201 | 2 | 2 | 5 | 1 | +----+--------+---------+-------+----------+--------+ Then I tried the sercmd commands to reload and dump and I do not see the contents o the screen. Is this how is supposed to work now?. with debug level set to 3 I do this "/usr/local/kamailio-3.1/sbin/sercmd lcr.dump_rules" and see these lines on the syslog: Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:390]: DBG: io_watch_add(0x2b772e494660, 12, 3, 0x9679e8), fd_no=1 Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:452]: handle_stream read: new connection (1) on /tmp/ser_ctl Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:505]: handle_stream read: eof on /tmp/ser_ctl Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:617]: DBG: io_watch_del (0x2b772e494660, 12, -1, 0x10) fd_no=2 called if I do this: "/usr/local/kamailio-3.1/sbin/sercmd lcr.reload" I see these: Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:390]: DBG: io_watch_add(0x2b772e494660, 12, 3, 0x9679e8), fd_no=1 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:452]: handle_stream read: new connection (1) on /tmp/ser_ctl Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db.c:294]: connection 0x951b78 found in pool Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 5 columns returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 40 bytes for result names at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 20 bytes for result types at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[0]=[id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952928)[1]=[prefix] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951a98)[2]=[from_uri] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951b08)[3]=[stopper] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[4] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952998)[4]=[enabled] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_dbase.c:326]: no rows returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: lcr [lcr_mod.c:985]: loading, cycle 0 with <0> rows Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 12 columns returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 96 bytes for result names at 0x951ef8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 48 bytes for result types at 0x951c38 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951b08)[0]=[gw_name] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951a98)[1]=[ip_addr] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952928)[2]=[port] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[3]=[uri_scheme] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[4] at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952610)[4]=[transport] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[5] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952998)[5]=[params] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[6] at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9527b8)[6]=[hostname] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[7] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951fb8)[7]=[strip] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[8] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952028)[8]=[tag] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[9] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952098)[9]=[flags] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[10] at 0x952108 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952108)[10]=[defunct] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[11] at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951288)[11]=[id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:169]: no rows returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 12 columns Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[5] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[6] at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[7] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[8] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[9] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[10] at 0x952108 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[11] at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x951ef8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x951c38 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 4 columns returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 32 bytes for result names at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 16 bytes for result types at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952098)[0]=[rule_id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952028)[1]=[gw_id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951fb8)[2]=[priority] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[3]=[weight] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_dbase.c:326]: no rows returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: lcr [lcr_mod.c:1344]: loading, cycle 0 with <0> rows Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 4 columns Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_pool.c:97]: connection still kept in the pool Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:505]: handle_stream read: eof on /tmp/ser_ctl Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:617]: DBG: io_watch_del (0x2b772e494660, 12, -1, 0x10) fd_no=2 called I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see > huge differences on the LCR module. there is a script somewhere (read wiki) that upgrades the tables. > I have these main questions: > > 1- does lcr_id replaces the grp_id? not really. lcr_ids denote independent lcr instances. > 2- what happened to the fifo commands to add gws etc: fifo commands to manipulate lcr tables are not supported by me. -- juha From: fborot(a)hotmail.com To: sr-users(a)lists.sip-router.org Subject: Questions about LCR module Date: Mon, 27 Jun 2011 17:23:05 -0400 I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module. I have these main questions: 1- does lcr_id replaces the grp_id? 2- what happened to the fifo commands to add gws etc: lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip> ............... add a gateway with flags, tag and strip ............ ................(flags, tag, and strip are optional arguments) ..... lcr rmgw <gw_name> .......................................................... ............... delete a gateway lcr addroute <prefix> <from> <grp_id> <prio> ................................. .............. add a route ( use '' to match anything in <from> ) lcr rmroute <prefix> <from> <grp_id> <prio> ................................. .............. delete a route 3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload" only the lcr tables What would be the best way to use this module on this version? Any help/hint would be greatly appreciated. thank you fborot

2 3

Kamailio+ Asterisk 491 error code ????
by Sébastien Cramatte 28 Jun '11

28 Jun '11

Hello, I've just setup Kamailio+ Asterisk following Daniel Constantin Mierla howto http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb Everything seemts to works and I'm able to place a call to local extensions. I'm able to establish a call but strangely I get and 491 error code on my SIP Phones In my Asterisk console I get this error : [Jun 28 18:14:42] WARNING[24432]: chan_sip.c:18087 handle_response_invite: just did sched_add waitid(23) for sip_reinvite_retry for dialog LIFuQLu1l3ZtKrNy0XDX7lEk.Y5JdFko in handle_response_invite == Spawn extension (from-local, 102, 1) exited non-zero on 'SIP/103-00000000' Anybody can give me an hand to solve this issue ? Best regards

2 1

Dialog module question
by Javier Gallart 28 Jun '11

28 Jun '11

Hello I have little experience with the dialog module -so sorry if the quesiton is silly-, and I was wondering if I could use it for doing accounting (acc works just fine, but I was looking for a unique stop record that includes all the call information). The basic idea would be to assign the initial INVITE to a profile; and then, when BYE is received, build a cdr based of the dialog attributes...but afaik , upon reception of the BYE the dialog is destroyed. is there a way to access the dlg attributes inside the script when the BYE transaction is processed? Thanks in advance Javi

2 2

Questions about LCR module
by Fabian Borot 28 Jun '11

28 Jun '11

I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module. I have these main questions: 1- does lcr_id replaces the grp_id? 2- what happened to the fifo commands to add gws etc: lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip> ............... add a gateway with flags, tag and strip ............ ................(flags, tag, and strip are optional arguments) ..... lcr rmgw <gw_name> .......................................................... ............... delete a gateway lcr addroute <prefix> <from> <grp_id> <prio> ................................. .............. add a route ( use '' to match anything in <from> ) lcr rmroute <prefix> <from> <grp_id> <prio> ................................. .............. delete a route 3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload" only the lcr tables What would be the best way to use this module on this version? Any help/hint would be greatly appreciated. thank you fborot

2 1

Re: [SR-Users] [sr-dev] new module: async - asynchronous SIP request processing in cfg
by Dan-Cristian Bogos 28 Jun '11

28 Jun '11

Hey Daniel, As said in the past, this module is good news, thanks for making it available. As a big fan of delayed billing, sleep() does a great job of delaying 200 OK from the upstream for me (useful for example with analogue/CDMA gateways sending 200 OK on ringback). Cheers, DanB

1 0

Kamailio and DTMF
by Phillman25 Kyriacou 28 Jun '11

28 Jun '11

Hi I was wondering what DTMF modes Kamailio supports? If i wanted to force a DTMF mode on Kamailio how could i go about doing it? thanking you in advance Phillip

3 3

rtpproxy and kamailio doesnt work out from the box.
by MingHon 28 Jun '11

28 Jun '11

hey list, i currently test on my asterisk 1.6 box on on pc with private ip and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc with public ip and private ip. everything installed successfully. i follow this tutorial realtime intergration with asterisk http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb my config. rtpproxy -l publicip -s udp:127.0.0.1:7722 -u user #!define WITH_NAT modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "sipping_from", "sip:pinger@publicip") nat_uac_test("19") # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); is my config correct? may i know the best version work out from the box? please advice.. thanks in adv. :) -- Regards, MingHon

4 9

new module: async - asynchronous SIP request processing in cfg
by Daniel-Constantin Mierla 27 Jun '11

27 Jun '11

Hello, a new module is now in GIT master branch, named async. Its purpose is to provide asynchronous SIP request processing using t_suspend()/t_continue() mechanism from tm module. While the tm module had it for quite some time, it was not available for usage in configuration file. There are two functions implemented by now: - async_route(routename, interval) - execute asynchronously a route block after a time interval - aync_sleep(interval) - sleep asynchronously for a time interval and then resume SIP request processing See the readme for more details: http://kamailio.org/docs/modules/devel/modules/async.html Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 4

LCR module
by Fabian Borot 27 Jun '11

27 Jun '11

After installing the latest stable version 3.1.4 I noticed that I can not run any 'lcr' commands from the 'kamctl fifo' interface. I noticed that the lcr.so is not on the lib64/modules_k or lib64/modules_s folders so I guess is was not included at installation time: [root@registrar kamailio]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_k/ [root@registrar modules_k]# ls acc.so db_text.so drouting.so mi_datagram.so permissions.so rtimer.so statistics.so usrloc.so alias_db.so db_unixodbc.so exec.so mi_fifo.so pike.so rtpproxy.so textops.so xlog.so auth_db.so dialog.so group.so msilo.so pua_mi.so siptrace.so tmx.so auth_diameter.so dispatcher.so htable.so nathelper.so pv.so siputils.so uac_redirect.so benchmark.so diversion.so imc.so nat_traversal.so qos.so speeddial.so uac.so call_control.so domainpolicy.so kex.so path.so registrar.so sqlops.so uri_db.so cfgutils.so domain.so maxfwd.so pdt.so rr.so sst.so userblacklist.so [root@registrar modules_k]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_s/ [root@registrar modules_s]# ls acc_db.so avp.so diversion.so fifo.so nathelper.so pike.so speeddial.so uri_db.so acc_syslog.so db_ops.so domain.so gflags.so options.so print.so textops.so uri.so auth_db.so dialog.so eval.so maxfwd.so pdt.so registrar.so timer.so usrloc.so avp_db.so dispatcher.so exec.so msilo.so permissions.so rr.so uac.so xlog.so [root@registrar modules_s]# I checked the modules.lst file and it is not on the exclude list, because I did removed it along with db_mysql and db_odbc, so I expected it to be included but it is not. Am I missing something? This is from the syslog Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command lcr is not available Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: entered consume Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: **** done consume and this is from the fifo interface: [root@registrar kamailio]# /usr/local/kamailio-3.1/sbin/kamctl fifo lcr reload 500 command 'lcr' not available [root@registrar kamailio]#

1 1

async processing in a module
by Matthew Williams 27 Jun '11

27 Jun '11

Hi, I'm working on a jsonrpc (client) module that we would like to make async, using t_suspend and t_continue from the tm module. I have a couple questions related to this. The way I understand these functions, t_suspend will freeze the transaction in shared memory and immediately return and t_continue will unfreeze the transaction. What I'm not clear on here is whether t_continue will continue processing of the transaction within the process that calls t_continue, or will it delegate to a worker process? I assume I should spawn an io process in child_init which will be responsible for sending/receiving on the socket(s), but what is the best way to communicate between processes? Is there a way to get a fd from fork_process (that can be written to by any child and read from using select() in the io thread)? This is my first attempt at a real module, so any pointers are greatly appreciated. Regards, Matthew WIlliams

3 4

Kamailio start failed
by Roman Dmytriv 27 Jun '11

27 Jun '11

Hello friends! Having trouble to start newly installed Kamailio This is what I got: root@debian:/home/user# /etc/init.d/kamailio status Status of kamailio: kamailio is running. root@debian:/home/user# kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed root@debian:/home/user# Any ideas will appreciated Regards Roman

3 3

changing script (perl, lua ...) without kamailio restart
by Klaus Darilion 27 Jun '11

27 Jun '11

Hi! Does any of the script interpreters (perl, lua, python ....) allow to use a modified shell script without restarting Kamailio? Thanks Klaus

2 1

The 15000th development commit
by Daniel-Constantin Mierla 27 Jun '11

27 Jun '11

Hello, I just wanted to mark the 15000th development commit in our project (counted only on GIT master branch). It happened several days ago and it was done by the same person that did the first commit, Andrei: git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -1 512dcd9 Andrei Pelinescu-Onciul Mon Sep 3 21:27:11 2001 +0000 Initial revision The 15000th one is: git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -15000 | tail -1 8a90dd3 Andrei Pelinescu-Onciul Sat Jun 11 11:24:05 2011 +0200 core: remove unused variables + coding style I published the news about it on project sites as well: * http://asipto.com/u/37 * http://asipto.com/u/38 Now it's the time to look forward to the 20000th commit! Enjoy the summer, autumn comes with v3.2.0, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 12

an idea - the SIP browser - elsipo
by Daniel-Constantin Mierla 24 Jun '11

24 Jun '11

Hello, I want to present an idea and see if it is something people find benefits with such approach. Some time ago, due to existence of only SIP client applications (aka softphones) that have static GUI, which were designed for generic purposes, do not present what servces the currently used provider offers nor are easy to customize, I played with the idea of a SIP application with the GUI in HTML/CSS/Javascript. Being busy with Kamailio, the whole thing was postponed, but during last days I took the time and updated the code to latest developing tools (QT/Webkit and pjsip) and made it public on github, see: * http://asipto.com/u/3c A screenshot and some details in the wiki at: * http://asipto.com/u/3b The UI can be pulled from a remote site via http/https for example. The approach was to have a framework to render the UI, used QT/Webkit for that (cross platform, Linux, Mac OS X, Windows) and exported SIP API to Javascript. I worked with pjsip, but the system is plugin based, so pjsip plugin can be replaced, plugins for other things (not only sip/voip) can be written easily. The code is just for demo/prototyping, with the goal of showing feasibility. At this moment can be used to make/receive calls and send/receive instant messaging. In addition, content of UI can be carried by SIP messages - the examples added so far is to request some server statistics when clicking on a link of UI and the SIP server pushing periodically some info to a section of UI. Personally I don't have any plan to go into client side development, still the idea was too appealing to try and I thought would give lot of value and enable new features on SIP services to make user experience more pleasant. Therefore the target is to see if any of you think is something that worth pursuing and if yes, get people on board to develop it further from prototype phase. In the wiki I listed a part of my wish list with current VoIP/SIP services that would be trivial to implement with this approach. The naming as SIP browser instead of softphone hopefully suggest also the purpose of the application - to be able to browse the services offered by provider at that time and the provider to be able to push content back to application (on demand or periodically). Development was done on Mac OS X, a tarball with binaries for this OS is available for download on github - compiling for other OSes should go pretty straightforward. At the moment the idea poped up there was no such approach that I could find (at least not in open source space), did I miss something, are you aware of any similar approach? What is your opinion overall? Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

Dispatcher module for INVITE
by Giuseppe Carella 24 Jun '11

24 Jun '11

Hi all, I'm using kamailio dispatcher module between the OpenIMSCore and sailfin. Normally, without the dispatcher, all works fine. From the IMS Bench I can see that there are many successful call, but when I use the dispatcher module I'm having a lot of retransmission in the first part of the INVITE protocol. I'm using the default configuration file: # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("dispatcher", "list_file", "../etc/dispatcher.list") # modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } I tried also the t_relay() function instead of the forward command, but the problem is the same. Thanks in advance. Bests, Giuseppe

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Don't publish Is not supposed the e-mail should be Hiden
by Omar 24 Jun '11

24 Jun '11

I really don't want to share my e-mail publicly, i remember there was an option before. O

3 2

Remote Party ID Header
by dotnetdub 23 Jun '11

23 Jun '11

Hi All, If an endpoint is sending us the remote party ID header, I know how to extract $(re{uri.user}) - how would I check if the Privacy Flag was true in Kamailio script? Version 1.41 Thanks, Brian

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Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hi Carsten, Thanks for the tip. All audio is going through RTPProxy on the Kamailio server, not directly to Asterisk. I will look into that patch. Thanks! Brett ----- Original Message ----- From: "Carsten Bock" <carsten(a)ng-voice.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Sent: Thursday, June 23, 2011 12:46:11 AM GMT -08:00 US/Canada Pacific Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint Hi, another solution might be, to either configure an RTP-Timeout on the Asterisk (if you send your calls through the asterisk anyway). You might also consider using the RTPProxy with the patch in the sip-router-repository. With the patch, the RTPProxy will trigger a teardown of calls (via XML-RPC) if the RTP-Session has a timeout. Carsten 2011/6/23 Brett Woollum <brett(a)woollum.com>: > Hi Alex, > > Thanks for this information. I've started researching the session-timer > capabilities in Asterisk, and I think that's my solution. I've already > implemented it on a test system and it works well, except that it's using > reINVITES to update as opposed to UPDATE messages, resulting in chops in the > audio every so often. I'll research this further though. > > Thanks again! > Brett > > ----- Original Message ----- > From: "Alex Balashov" <abalashov(a)evaristesys.com> > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users > Mailing List" <sr-users(a)lists.sip-router.org> > Cc: sr-users(a)lists.sip-router.org > Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific > Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to > SIP endpoint > > This is a complex topic. There is no way for a proxy like Kamailio to > detect this scenario per se. Kamailio reacts to and forwards signaling > events. If an endpoint disappears, it won't send any of those to indicate > that it has gone away. How would Kamailio know? Media stream timeout? > Kamailio doesn't relay media. > Your Kamailio-side solution is a dialog timeout, requiring use of > dialog-stateful tracking using the dialog module. But that will time out > calls indiscriminately, so you need to make it long enough to not anger your > users but short enough to be useful. > Your endpoint solution is SIP Session Timers. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > On Jun 23, 2011, at 1:10 AM, Brett Woollum <brett(a)woollum.com> wrote: > > Hello, > > We are running Kamailio as a registration point for our SIP phones, which > then interacts with Asterisk. SIP registrations are processed by Kamailio, > but everything else is passed to Asterisk. The Kamailio configuration is > close to the article at: > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. > Everything seems to be working well, until today. > > I found several calls today that were still connected to our provider, even > though our SIP phones were not active. There were three calls with timers at > 9 hours and counting. We had some IP connectivity issues earlier today, and > I'm wonder if it's related. > > If a SIP phone was connected and on a call (through kamailio), and the > kamailio/asterisk servers became unreachable, the SIP phones will drop the > call. But, it appears that kamailio/asterisk never drop the call in this > case, and the call stays live with the carrier. I had to manually kill the > calls by command prompt. > > What's the best way to handle this? Is there a way to have kamailio or > asterisk poll the phone to see if it's still on the call or something? How > can I give visibility to asterisk or kamailio so the calls are always > dropped properly? I don't want to run up a large bill because of calls that > didn't terminate when they should have. > > Thanks! > Brett > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ SIP Express Router (SER) and > Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 _______________________________________________ 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] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hi Alex, Thanks for this information. I've started researching the session-timer capabilities in Asterisk, and I think that's my solution. I've already implemented it on a test system and it works well, except that it's using reINVITES to update as opposed to UPDATE messages, resulting in chops in the audio every so often. I'll research this further though. Thanks again! Brett ----- Original Message ----- From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: sr-users(a)lists.sip-router.org Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint This is a complex topic. There is no way for a proxy like Kamailio to detect this scenario per se. Kamailio reacts to and forwards signaling events. If an endpoint disappears, it won't send any of those to indicate that it has gone away. How would Kamailio know? Media stream timeout? Kamailio doesn't relay media. Your Kamailio-side solution is a dialog timeout, requiring use of dialog-stateful tracking using the dialog module. But that will time out calls indiscriminately, so you need to make it long enough to not anger your users but short enough to be useful. Your endpoint solution is SIP Session Timers. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jun 23, 2011, at 1:10 AM, Brett Woollum < brett(a)woollum.com > wrote: Hello, We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb . Everything seems to be working well, until today. I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. Thanks! Brett _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hello, We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. Everything seems to be working well, until today. I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. Thanks! Brett

2 1

apt-get install doesn't work
by Roman Dmytriv 23 Jun '11

23 Jun '11

Hello, I was trying to install kamailio using your instruction from http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I run Debian Squeeze. Imported apt repository keys, created this file </etc/apt/sources.list.d/kamailio.list> with this information in it <deb http://deb.kamailio.org/kamailio31-nightly squeeze main> Now when I run apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules I got this response: root@s2:/etc/apt/sources.list.d# apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package kamailio E: Unable to locate package kamailio-mysql-modules E: Unable to locate package kamailio-tls-modules Can you give me a hint what could be wrong? Regards

2 5

OT: are there still asymmetric SIP clients?
by Klaus Darilion 22 Jun '11

22 Jun '11

Hi! Just a small survey: Are you still using NAT detection checks? I prefer to apply NAT traversal always (never trust the contact header) which of course only works if all the clients are symmetric. Thus - do you still see asymmetric SIP clients? Thanks Klaus

3 4

Kamailio/Asterisk/FreeSWitch dinner in Madrid - Thursday next week
by Olle E Johansson 22 Jun '11

22 Jun '11

Next week I'll be in the hot city of Madrid doing Asterisk/Kamailio training - The SIP master class. Maybe we can organize a voip nerd dinner on Thursday evening? If you're interested, please e-mail me off list and I'll send out more details later. Greetings /Olle PS. E-mail off list :-)

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Ukulele Kamailio Jingle
by Daniel-Constantin Mierla 22 Jun '11

22 Jun '11

Hello, a good friend, Javi Beltran, recorded a composition playing Ukulele (aka, ukelele - Hawaiian instrument - http://en.wikipedia.org/wiki/Ukulele). The jingle is available at: * http://asipto.com/u/39 If you like it, feel free to use it in relation with Kamailio project. Javi's compositions are open sourced, available at: * http://www.javibeltran.com/?lang=en I posted a news about it on project's site: * http://www.kamailio.org/w/2011/06/ukulele-kamailio-jingle/ If you or your friends have done any artistic work (music, graphics, a.s.o.) related to Kamailio, feel free to send it over and I will make them available on project's site. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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audio issue in same nat.
by †ஐû●чƃцίш† Goh 22 Jun '11

22 Jun '11

Hi List, below is my setup.. rtpproxy and kamailio in one PC with 2 nic. (ppp0 with public IP[60.49.119.XX] and eth1 with private IP[192.168.2.3])and asterisk is on another PC with private IP[192.168.2.23] i use realtime integration for kamailio and asterisk.http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6… i have two yealink hardphone ext 101(ip 192.168.1.200) and 102(ip 192.168.1.132) and a softphone ext 103 registered successful.both hardphone are behind same nat (175.136.221.XX)and softphone ext 103(ip 10.129.138.225) behind nat also (113.210.97.XX) ul show:database engine 'MYSQL' loadedControl engine 'FIFO' loadedentering fifo_cmd ul_dumpDomain:: location table=512 records=3 max_slot=1 AOR:: 102 Contact:: sip:102@175.136.221.XX:5062 Q= Expires:: 3109 Callid:: 2043273564(a)175.136.221.241 Cseq:: 4 User-agent:: T22 7.3.0.50 Received:: sip:175.136.221.241:1039 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.49.119.69:5060 Methods:: 16383 AOR:: 103 Contact:: sip:103@113.210.97.XX:58776;transport=UDP;ob Q= Expires:: 294 Callid:: oa8Pqx3mR.SVnzAVEYHTwVKZE8CbpY9l Cseq:: 27626 User-agent:: v1.0.0/iPhone State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:60.49.119.69:5060 Methods:: 8143 AOR:: 101 Contact:: sip:101@175.136.221.XX:5062 Q= Expires:: 1738 Callid:: 451417581(a)175.136.221.241 Cseq:: 2 User-agent:: T20 9.41.0.80 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:60.49.119.69:5060 Methods:: 16383FIFO command was::ul_dump:openser_receiver_17783 103 try to call 102 and 101 work fine. 101 and 102 try call 103 also fine.when 101 call 102 it work fine but when 102 call 101 there is no audio for both side.102 call 101 wireshark capture on 102 sidekeep send rtp but no receive.192.168.1.132 -> 60.49.119.XX RTP when capture on 101 side.keep send rtp but no receive.192.168.1.200 -> 60.49.119.XX RTP and also when 101 try to call into voicemail there is no audioit keep send rtp packet but to192.168.1.200 -> 192.168.2.23 RTP in kamailio.cfg#!WITH_NATlisten=60.49.119.XXlisten=192.168.2.3 # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); nat_uac_test("19")rtpproxy -l 60.49.119.XX -s udp:127.0.0.1 is running anyone can help me? how can i fix this?thanks in adv. Regards, minghon

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Re: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service when load the ldap.so module
by Daniel-Constantin Mierla 22 Jun '11

22 Jun '11

Thanks for reporting back, taking the chance to forward this email from sr-dev to sr-users since the thread was cross-posted to some point, for the benefit of future reads of the archive to show the solution of this case. Cheers, Daniel On 6/21/11 7:19 PM, laura testi wrote: > Found the problem after some investigation based on your suggestions. > In the openldap configuration file /etc/openldap/ldap.conf, the entry > TLS_CACERT is pointed to a certificate file in a directory which is > accessible only by root user. After I move the file to a directory > which all user can access it, now it works. I can start the kamailio > as a service with "service kamailio start" with the running user and > group is kamailio. > > Thank you all for your helps and suggestions! > > Best Regards, > Laura > > On Tue, Jun 21, 2011 at 5:44 PM, laura testi <lau.testi(a)gmail.com > <mailto:lau.testi@gmail.com>> wrote: > > The SELinux is disabled. > > > > > On Tue, Jun 21, 2011 at 5:26 PM, Andrew Pogrebennyk > <andrew.pogrebennyk(a)portaone.com > <mailto:andrew.pogrebennyk@portaone.com>> wrote: > > On 21.06.2011 17 <tel:21.06.2011%2017>:30, laura testi wrote: > > Should I need to set a some special environment variable > to load ldap.so > in the /etc/init.d/kamailio? > > > The env output for root does not confirm my theory :) I'm not > sure what is preventing kamailio from communication with LDAP, > could it be that SELinux is enabled and denying it in some way? > > -- > Sincerely, > Andrew Pogrebennyk > > > > > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

kamailio 3.1.3 fail to start as a service when load the ldap.so module
by laura testi 21 Jun '11

21 Jun '11

Hi, we try to use LDAP module in kamailio, first we try only load the ldap.so only to see if it can be loaded. When we start from the command line with either "kamctl start" or directly "kamilio", the kamailio is started normally with different children. When I try to start as service with "service kamailio start" or "/etc/init.d/kamailio start". It fails with the following error in the /var/log/messages: ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- Jun 21 10:28:39 kamailio3p1 kamailio: INFO: <core> [tcp_main.c:4716]: init_tcp: using epoll_lt as the io watch method (auto detected) Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: usrloc [hslot.c:53]: locks array size 512 Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: auth [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support disabled Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [parse_config.c:251]: file not found: //etc/kamailio/permissions.allow Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [permissions.c:606]: default allow file (//etc/kamailio/permissions.allow) not found => empty rule set Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [parse_config.c:251]: file not found: //etc/kamailio/permissions.deny Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [permissions.c:615]: default deny file (//etc/kamailio/permissions.deny) not found => empty rule set Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: ldap [ldap_mod.c:246]: OpenLDAP - 20343 Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_init.c:511]: tls: _init_tls_h: compiled with openssl version "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on, compression: on Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_init.c:519]: tls: init_tls_h: installed openssl library version "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on, zlib compression: on compiler: gcc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -I/usr/kerberos/include -DL_ENDIAN -DTERMIO -Wall -DMD32_REG_T=int -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wa,--noexecstack -DOPENSSL_USE_NEW_FUNCTIONS -fno-strict-aliasing -DOPENSSL_BN_ASM_MONT -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_init.c:373]: tls: init_tls: disabling compression... Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls [tls_init.c:564]: tls: init_tls_h: openssl kerberos malloc bug detected, kerberos support will be disabled... Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:176]: TLSs<default>: tls_method=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:188]: TLSs<default>: certificate='//etc/kamailio/FBCIDS01V01.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:195]: TLSs<default>: ca_list='//etc/kamailio/tiglobalca.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:202]: TLSs<default>: crl='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:206]: TLSs<default>: require_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:213]: TLSs<default>: cipher_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:220]: TLSs<default>: private_key='//etc/kamailio/FBCIDS01V01KEY.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:224]: TLSs<default>: verify_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:544]: TLSs<default>: No client certificate required and no checks performed Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f) has the SSL_OP_TLS_BLOCK_PADDING bug, but compression is disabled so no workaround is needed Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:176]: TLSc<default>: tls_method=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:188]: TLSc<default>: certificate='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:195]: TLSc<default>: ca_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:202]: TLSc<default>: crl='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:206]: TLSc<default>: require_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:213]: TLSc<default>: cipher_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:220]: TLSc<default>: private_key='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:224]: TLSc<default>: verify_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:227]: TLSc<default>: verify_depth=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:547]: TLSc<default>: Server MAY present invalid certificate Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f) has the SSL_OP_TLS_BLOCK_PADDING bug, but compression is disabled so no workaround is needed Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:176]: TLSs<10.10.5.214:5061>: tls_method=12 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:188]: TLSs<10.10.5.214:5061>: certificate='//etc/kamailio/FBCIDS01V01.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:195]: TLSs<10.10.5.214:5061>: ca_list='//etc/kamailio/tiglobalca.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:202]: TLSs<10.10.5.214:5061>: crl='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:206]: TLSs<10.10.5.214:5061>: require_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:213]: TLSs<10.10.5.214:5061>: cipher_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:220]: TLSs<10.10.5.214:5061>: private_key='//etc/kamailio/FBCIDS01V01KEY.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:224]: TLSs<10.10.5.214:5061>: verify_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:227]: TLSs<10.10.5.214:5061>: verify_depth=9 Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:544]: TLSs<10.10.5.214:5061>: No client certificate required and no checks performed Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f) has the SSL_OP_TLS_BLOCK_PADDING bug, but compression is disabled so no workaround is needed Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP server Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap (/usr/lib64/kamailio/modules_k/ldap.so) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core> [pt.c:337]: ERROR: fork_process(): init_child failed for process 1, pid 8786, "udp receiver child=0 sock=127.0.0.1:5060" Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: : <core> [main.c:1546]: main_loop: Cannot fork Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP server Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap (/usr/lib64/kamailio/modules_k/ldap.so) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core> [pt.c:337]: ERROR: fork_process(): init_child failed for process 2, pid 8788, "udp receiver child=1 sock=127.0.0.1:5060" Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: : <core> [main.c:1546]: main_loop: Cannot fork Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core> [main.c:738]: child process 8786 exited normally, status=255 Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core> [main.c:738]: child process 8788 exited normally, status=255 Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8801]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8799]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8797]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8795]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8793]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8790]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8812]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8808]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8810]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8814]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8816]: INFO: <core> [main.c:807]: INFO: signal 15 received ... -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- Without ldap.so loaded, we can start the kamailio as a service normally with "service kamailio start". The platform running kamailio is: RHEL5.6_X64 Can you help us please? Because we want to start the kamailio as a service from the boot Thanks a lot in advanced! Best Regards, Laura

4 10

an IDEA
by afshin afzali 21 Jun '11

21 Jun '11

Hi Guys, I need to integrate a SIP processing environment to H.248 and SS7/ ISUP links. For those communication links, OpenSS7 and Erlang/OTP MEGACO are good solutions. My idea is to write applications over OTP and SIP-Router to be able to bridge non-SIP protocols to SIP! On OTP the way is clear, but on SIP-Router I need to be able to generate / terminate SIP transactions. I'll appreciate every comment. BEST, -- afshin

3 3

Security in proxy
by Anto 20 Jun '11

20 Jun '11

Hi After reading the modules pike, pipelimit, etc. I wanted to know what measures can be used in the proxy, because like me, there will be more people interested ;-). We see that the module pike is a good security measure, but for users with many channels, we used the configuration of users with few channels (could we discriminate and make different rules for each?). If we have the proxy settings for normal users and have trunk configuration as well, I guess we also mark the traffic of the trunk to see it too aggressive about the user. As you might know if discriminate different types of users according to their traffic, one solution I can think of is to have different proxies for each user type (trunk, normal, etc). Is there any way to implement something that detects unusual behavior on the user (referred to countries that do not ever called, excessive traffic, etc)?. I ask not to try to implement something that exists or perhaps someone can advise me how to do so because it is very likely not get it or do it wrong :-P . Continue to seek information and if I find something to contribute, send him to the list. Can you think of some way more secure proxy (mainly user accounts)? Do some basic safety tips to take into consideration? Any guidance? My advice to avoid trying to schedule something with 100 lines of code when I can do with 5 lines? Does this indicate that I intend to do not I make it :-P ?. What I said, I will try paper on the subject and if I can find information on something (though it may seem a bad code :-( ), which publishes. Thanks ;-) Regards Anto

3 4

Keeping transactions in standby till future events
by Gustavo Garcia Bernardo 20 Jun '11

20 Jun '11

Is it possible to configure sip-router to keep a transaction for an incoming message (f.e. an INVITE) without sending a response and without forwarding it, and then reference and complete it in the future? Using pseudocode this is the logic I would like to implement: //When receiving an INVITE leave it pending if (request is INVITE) { store transactionID // This transactionID will be send somehow to the client return } //When receiving a PUBLISH referencing a pending transaction complete the forwarding if (request is PUBLISH) { transactionId = header("X-TransactionId") destination = header("X-Destination") transactions [transactionID]. forward(destination) } Thank you very much, G. Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx

4 7

Help find out where to modify request uri
by Simone Felici 20 Jun '11

20 Jun '11

Hi to all, I need to edit our ser installation (OLD ser-2.0.0-rc1) implementing a check to rewrite the INVITE URI address in case a flag on a DB is set or not. I'm not proof on this and as always the solution is to find out on the production system without the possibility (for now) to upgrade to most recent version. In fact I would load the flag with the db_ops module. The scenario should be the follow: 1. a call is coming from sip or from pstn to an internal sip account 2. the ser should check if a given flag for the called account is set (db_ops) 3. if it's set it should take the destination URI (taken from location db table) and rewrite the number to another known number. i.e. from sip:123456789@IP;uniqu=... to sip:new987654321@IP;uniq=... My first issue is to identify WHERE ser execute the INVITE to the sip account and how rewrite the URI. Could someone point me to the right place and give me some suggestion? The ser.cfg used could be read from a previous post of me: http://www.mail-archive.com/sr-users@lists.sip-router.org/msg04900.html Thank's a lot! Simon

2 4

Invite replies
by Lucas Alvarez 19 Jun '11

19 Jun '11

Hi, how may I catch the reply of an invite? I did this but it seems is not working: if (method=="INVITE") { $avp(sdp_ret) = fix_nated_sdp("11"); xlog("L_ALERT","#####********###### SDP return: $avp(sdp_ret)\n"); append_hf("P-hint: sdp rewritten\r\n"); t_on_reply("REPLY_TWO"); } onreply_route("REPLY_TWO") { xlog("L_ALERT","ALERT: ################ $rU - $rm from $fu (IP:$si:$sp--------$di)\n"); return; } Any help will be appreciate it, thanks in advance. Lucas

2 1

100rel without direct reachability
by Alex Balashov 19 Jun '11

19 Jun '11

(Sorry for cross-posting from Sip-Implementors; however, upon some reflection this is really more of a Kamailio mechanics question.) Greetings, I have a scenario where endpoint A is calling endpoint B through proxy P. Endpoints A and B are on different networks that are not directly reachable--A is on the public Internet, to which proxy P is connected, but NAT'd, and B is on a private RFC1918 network on which the proxy P has an interface as well. The purpose of proxy P is to make calling A <-> B possible, while providing the appropriate far-end NAT traversal functionality for A and so on and so on. It does this by inserting the correct Record-Route headers into the INVITE from A, and otherwise by staying in the signaling path; both A and B can reach the proxy, so all requests and replies go through it. The problem I'm having is prior to full dialog establishment. Upon receipt of INVITE from A by way of proxy P, B replies with 183 + SDP and also: Require: 100rel In the 183 it provides a Contact on its private network that A obviously cannot reach: Contact: <sip:xxxxx@192.168.0.10:5060> As a result, A attempts to send PRACKs to sip:xxxxx@192.168.0.10, which are obviously not going to go anywhere. After not receiving a few of these expected PRACKs, B replies to the call with a final negative failure. The question is, is there anything I can twiddle on the proxy to make this work? The problem with PRACKs is that although they are sequential requests, they arise (in this case) prior to the point at which RR can be used to guide them through the proxy. The most obvious answer is, "Make B not Require: 100rel", of course. However, that is not an option here. Thanks! -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 6

HTTP POST
by Matthew Williams 16 Jun '11

16 Jun '11

Hi, I'm working on an application that needs to create an HTTP POST from within Kamailio. The utils module has the http_query function, but that is only for GET. I think I'm looking for something more like the uac module's uac_req_send -- but for http. Any suggestions? Thanks in advance!

2 1

mi_datagram usage example
by Asgaroth 16 Jun '11

16 Jun '11

Hi All, I am trying to send a mi command over the mi_datagram udp socket. I dont seem to be getting a response back from the server. I am certain that it is something I am doing wrong. I have mi_datagram loaded and listening on udp:1.1.1.1:8080. I then try to send the following dispatcher mi command over to the mi_dg interface: echo -ne ":ds_list:\n\n" | nc -w 1 -u 1.1.1.1 8080 I dont see any error in the log file, so it appears that the mi_datagram is processing the request, but I am not getting any response back. The ds_list command works properly if I run it using "kamctl fifo ds_list", or if I go via "sercmd mi_dg ds_list" that works too. Would anyone have an example bash/ksh/perl snippet that would help me on my way to communicating with this mi interface. Any thoughts comments would be greatly appreciated. Thanks

3 3

Kamailio dispatcher
by Evgeniy Spinov 16 Jun '11

16 Jun '11

Hello. I had a Kamailio version of 3.0.3 and during this time configured a failover with simple routine. In short like this: if (ds_next_domain()) { xlog(.....); if (!t_relay()) { xlog(.....); return; } return; } else { t_reply("503"); } Then I've updated to 3.1.3 and was happy enough until I've discovered that my failover is not working anymore. What is observed: - It changes $rd variable, but not changing $du variable, is it ok? - Not any of the nodes receive packet on t_relay after ds_next_domain(). - In route decision section, where t_on_reply() and t_on_failure() are located I have the same t_relay() and it works fine, cause there is no any ds_next_domains(). Is it a bug or I'm doing something wrong? Thanks in advance.

3 11

How to check a bit (1/0) of a pseudo-variable?
by Iñaki Baz Castillo 16 Jun '11

16 Jun '11

Hi, summarizing I have two pseudo-variables: $var(allowed_zones) = 5; $var(id_zone) = 2; The binary representation of 5 (3+2), assuming 8 bits lenght, is: 00000101 so bit 0 = 1, bit 1 = 0, bit 2 = 1 ... Now I just need to know if bit number 2 ($var(id_zone)) of $var(allowed_zones) is 0 or 1. Is there any function like isflagset() but for pseudo-variables? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 6

Re: [SR-Users] Use of mtree module having multiple trees in the same table
by Javier Gallart 16 Jun '11

16 Jun '11

Hello not sure what you want to accomplish, but in case it might help this is how we use the mtree module. Our 3 columns are: partition, prefix, route Partition is just a name that defines a group, this is done inside the script before calling mt_match (via a table lookup for instance). Let's say if a call comes from ip x we assign it to partition 400; if it comes from ip y we assign it to partition 401. Each partition is a different tree. Consider these entries in the tree table: 400,123456,A 401,123456,A 401,1234567,B Let's say both x and y send a call with $rU 1234567; mt_match will return A for ip x, and B for ip y; we use that information for taking further routing decisions. A common situation might be: x has a unique route for a whole country while y has that entry but has more defined routes for more specific area codes inside that country. The column route could also be for instance the setid of a group of gw's for the dispatcher module, etc... Regards Javi > > Message: 3 > > Date: Tue, 14 Jun 2011 14:27:53 +0200 > > From: I?aki Baz Castillo <ibc(a)aliax.net> > > Subject: Re: [SR-Users] Use of mtree module having multiple trees in > > the same table > > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > > Users Mailing List" <sr-users(a)lists.sip-router.org> > > Message-ID: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww(a)mail.gmail.com> > > Content-Type: text/plain; charset=UTF-8 > > > > 2011/6/13 jose luis millan <doctore96(a)gmail.com>: > > > I wonder if anyone could help me understanding how can I use this > > > module in a scenario where multiple mtrees are defined behind the same > > > database table. > > > > > > I can't guess how to speficy to 'mt_match' the mtree I want to match > > > against, since the 'mtree' module parameter defines a specific memory > > > tree based on a specific table, but not based on a specific 'tname' > > > value. > > > > Indeed we have a table with these colums: > > > > - prefix: string to identify number destination. This is the regex > > column. > > - id_zone: integer, points to other table. This is the value column. > > - scope: integer, can be 0 or 1. > > > > So wee need two instances of mtree: > > 1) those rows with scope 0. > > 2) those rows with scope 1. > > > > Is it possible with current mtree module? or maybe we'd do better by > > creating two views of the table (one for entries with scope 0 and > > another for entries with scope 1)? > > > > Thanks a lot. > > > > -- > > I?aki Baz Castillo > > <ibc(a)aliax.net> > > > > > > > > >

2 1

Re: [SR-Users] sr-users Digest, Vol 73, Issue 40
by Javier Gallart 16 Jun '11

16 Jun '11

Hello not sure what you want to accomplish, but in case it might help this is how we use the mtree module. Our 3 columns are: partition, prefix, route Partition is just a name that defines a group, this is done inside the script before calling mt_match (via a table lookup for instance). Let's say if a call comes from ip x we assign it to partition 400; if it comes from ip y we assign it to partition 401. Each partition is a different tree. Consider these entries in the tree table: 400,123456,A 401,123456,A 401,1234567,B Let's say both x and y send a call with $rU 1234567; mt_match will return A for ip x, and B for ip y; we use that information for taking further routing decisions. A common situation might be: x has a unique route for a whole country while y has that entry but has more defined routes for more specific area codes inside that country. The column route could also be for instance the setid of a group of gw's for the dispatcher module, etc... Regards Javi Message: 3 > Date: Tue, 14 Jun 2011 14:27:53 +0200 > From: I?aki Baz Castillo <ibc(a)aliax.net> > Subject: Re: [SR-Users] Use of mtree module having multiple trees in > the same table > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww(a)mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > 2011/6/13 jose luis millan <doctore96(a)gmail.com>: > > I wonder if anyone could help me understanding how can I use this > > module in a scenario where multiple mtrees are defined behind the same > > database table. > > > > I can't guess how to speficy to 'mt_match' the mtree I want to match > > against, since the 'mtree' module parameter defines a specific memory > > tree based on a specific table, but not based on a specific 'tname' > > value. > > Indeed we have a table with these colums: > > - prefix: string to identify number destination. This is the regex > column. > - id_zone: integer, points to other table. This is the value column. > - scope: integer, can be 0 or 1. > > So wee need two instances of mtree: > 1) those rows with scope 0. > 2) those rows with scope 1. > > Is it possible with current mtree module? or maybe we'd do better by > creating two views of the table (one for entries with scope 0 and > another for entries with scope 1)? > > Thanks a lot. > > -- > I?aki Baz Castillo > <ibc(a)aliax.net> > > > >

1 0

t_relay_to("0x02") alternative with kamailio 3.1
by Herve Couplet 16 Jun '11

16 Jun '11

Hi, I had a setup with an old version where I used the function t_relay_to("0x02"). Following the 3.1 documentation, flag : * 0x02* - do not generate reply on internal error (NOTE: has no effect anymore) I used that function as a workaround with a pbx that stop to register when it receive an error response like 478 Unresolvable destination or 408 request timeout. I did upgrade to 3.1 to have dns srv caching. Do you know how this would be possible with kamailio 3.1. Thanks for your suggestions, hp

4 4

Operations With Variables $var- does it need to load a module
by Omar 15 Jun '11

15 Jun '11

In Kamailio 1.4.3 I have the following scenario: $var(a) = 0 $var(b) = 0 $var(a) = $(tU{s.len}); $var(b) = $var(a) - 5; xlog(" $var(b)"); the result $var(b) is always the same as $var(a) should i check if some specific module is loaded? the debugs does not show any error, and $var(a) and $var(b) are unique. Variables are working, but not the operations with the variables. regards O

3 4

200 OK Retransmission Error
by Jijo 15 Jun '11

15 Jun '11

Hi All, In an INVITE transaction, no ACK from A party and B party continue retransmission of 200 OK. I have observed that kamailio is not passing the retransmitted 200 OK message from B party to the Application(script) after 3 seconds. Instead kamailio directly sending to A party without passing to the script. I see in the logs that kamailio is cleaning up the transaction after 3 seconds..look for the string "DEBUG: t_lookup_request: no transaction found" before this log it is cleaning up the timer in the log "DEBUG: cleanup_uac_timers: RETR/FR timers reset" What does it mean by that?? How can we make the kamailio to pass the response to the application? Please let me know your comments.. Here is the flow. I cannot send the complete logs which is quite big. so i grep'd using the stirng "tcp_read_req: content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from the complete log.. A kamailo B ----INVITE-->- --------------------INVITE 180 Ringing--<---------------180 Ringing 200 OK-----------<---------------200 OK 0 200 OK-----------<---------------200 OK 500ms 200 OK-----------<---------------200 OK 1s 200 OK-----------<---------------200 OK 2s < From here it is not passing to the application- script> 200 OK-----------<---------------200 OK 4s 200 OK-----------<---------------200 OK 4s 28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[ sip:5616990102@192.168.243.151:5060] 36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 0 282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102 <sip: 5616990102@192.168.243.161:5060;transport=tcp> 331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction

2 2

NAPTR priorities doesn't seem to work properly
by Iñaki Baz Castillo 15 Jun '11

15 Jun '11

Hi, I'm testing Kamailio's NAPTR resolution: version: kamailio 3.2.0-dev4 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB DNS related configuration: dns_servers_no=2 dns_srv_lb=yes dns_try_naptr=yes dns_use_search_list=no use_dns_cache=yes use_dns_failover=yes dns_tls_preference=1 dns_tcp_preference=1 dns_udp_preference=1 As the doc says it should respect the priorities of the NAPTR records, see http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#dns_sctp_pref_dns…: "To use the remote site preferences set all dns_*_pref to the same positive value (e.g. dns_udp_pref=1, dns_tcp_pref=1, dns_tls_pref=1, dns_sctp_pref=1)" So my kamailio receives an INVITE for an external domain oversip.net (no transport param neither port in the RURI) and does a t_relay. Request RURI is sip:qwe@oversip.net. According to NAPTR: ~$ host -t naptr oversip.net oversip.net has NAPTR record 5 50 "S" "SIPS+D2T" "" _sips._tcp.oversip.net. oversip.net has NAPTR record 10 50 "S" "SIP+D2T" "" _sip._tcp.oversip.net. oversip.net has NAPTR record 20 50 "S" "SIP+D2U" "" _sip._udp.oversip.net. oversip.net has NAPTR record 40 50 "S" "SIP+D2S" "" _sip._sctp.oversip.net. oversip.net has NAPTR record 50 50 "S" "SIPS+D2S" "" _sips._sctp.oversip.net. So it should try TLS over TCP first, if it fails try TCP and if it fails try UDP. However it just uses UDP, why?? Even if I set a minor value to dns_tls_preference (so higher priority I expect) it still uses UDP. Am I doing something wrong? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

6 29

kamailio conference calls support
by Mino Haluz 15 Jun '11

15 Jun '11

Hi, does kamailio support conference calls or it has to be implemented in some B2BUA ? If it is supported, what are the requirements (modules etc.). Thanks, Mino

2 1

302 rewrite ruri based on contact
by Omar 15 Jun '11

15 Jun '11

I need to rewrite the destination after I receive 302 I have this logic Onreply Loads in avp the information needed I try sending to failure route and to a route in both cases the $ru become <null> In the route and failureroute I did try rewritehost, rewriteuri, $ru= number+ host And append the route But $ru is always null. Inside the route or failure route If there are no 302 nor the onreply involved everything works fine Any suggestion to rewrite the ruri after a 302 is welcome I am still using Kamailo1.4.3 Omar

3 6

Modifying sip packet
by Lucas Alvarez 14 Jun '11

14 Jun '11

Hi, I want to modify the following sip packet, I want to change the SDP IP address by $di, which rule could filter this packet? Something like: [ if method("OK") and $di is RFC1918 ] ??? And with which function could I change the Ip address in the SDP content? In this case I would like to change 172.16.99.2 by 190.244.125.41. Thanks in advance, Lucas T 172.16.99.2:5060 -> 190.244.125.41:9055 [AP] SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.99.2;branch=z9hG4bK7c5.bfc43456.0;i=1;received=172.16.99.2;rport=5060. Via: SIP/2.0/TCP 192.168.15.231:1952 ;received=190.244.125.41;branch=z9hG4bK-d8754z-be53247ec649be5b-1---d8754z-;rport=9055. Record-Route: <sip:172.16.99.2;r2=on;lr=on;ftag=b11c6a4e>. Record-Route: <sip:172.16.99.2;transport=tcp;r2=on;lr=on;ftag=b11c6a4e>. From: "Lucas"<sip:1104@67.152.18.231>;tag=b11c6a4e. To: <sip:1101@67.152.18.231>;tag=as5356a251. Call-ID: NzcwZWIxNGE1MDg5ZjVhZWRhYWNlM2RiNmMwM2M4MWM.. CSeq: 2 INVITE. Server: Asterisk PBX 1.8.4.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:1101@172.16.99.2:5080>. Remote-Party-ID: "Gus Office One" <sip:1101@67.152.18.231 >;party=called;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 305. . v=0. o=root 1124623741 1124623741 IN IP4 172.16.99.2. s=Asterisk PBX 1.8.4.1. c=IN IP4 172.16.99.2. t=0 0. m=audio 13640 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=ptime:20. a=sendrecv.

2 1

Use of mtree module having multiple trees in the same table
by jose luis millan 14 Jun '11

14 Jun '11

hello, I wonder if anyone could help me understanding how can I use this module in a scenario where multiple mtrees are defined behind the same database table. I can't guess how to speficy to 'mt_match' the mtree I want to match against, since the 'mtree' module parameter defines a specific memory tree based on a specific table, but not based on a specific 'tname' value. Any help is apreciated, Thanks in advance.

2 2

Fwd: 200OK Retransmission Error
by Jijo 14 Jun '11

14 Jun '11

Hi All, In an INVITE transaction, no ACK from A party and B party continue retransmission of 200 OK. I have observed that kamailio is not passing the retransmitted 200 OK message from B party to the Application(script) after 3 seconds. Instead kamailio directly sending to A party without passing to the script. I see in the logs that kamailio is cleaning up the transaction after 3 seconds..look for the string "DEBUG: t_lookup_request: no transaction found" before this log it is cleaning up the timer in the log "DEBUG: cleanup_uac_timers: RETR/FR timers reset" What does it mean by that?? How can we make the kamailio to pass the response to the application? Please let me know your comments.. Here is the flow. I cannot send the complete logs which is quite big. so i grep'd using the stirng "tcp_read_req: content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from the complete log.. A kamailo B ----INVITE-->- --------------------INVITE 180 Ringing--<---------------180 Ringing 200 OK-----------<---------------200 OK 0 200 OK-----------<---------------200 OK 500ms 200 OK-----------<---------------200 OK 1s 200 OK-----------<---------------200 OK 2s < From here it is not passing to the application- script> 200 OK-----------<---------------200 OK 4s 200 OK-----------<---------------200 OK 4s 28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[ sip:5616990102@192.168.243.151:5060] 36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 0 282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction

1 1

RTP proxy
by michel freiha 14 Jun '11

14 Jun '11

Dear All, I'm using the below config plan for routing my calls...The issue that forcing rtp proxy is not working well and the rtp proxy is never forced....Can you please provide me a guidance here? if($rU=~"^00.*" ) { if(!cr_route("default", "domain.com", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { setbflag(6); route(1); } exit; ## this was missing here -bogdan!!! } else { route(LOCATION); route(RELAY); } } ############# THIS IS THE GENERIC RELAY ROUTE THAT MUST BE USED ALL THE TIME FOR RELAY route[1] { # apply any NAT changes? if(isflagset(5)||isbflagset(6)) { route(4); } exit; } route[4] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ if (nat_uac_test("8")) { * force_rtp_proxy();* } else { force_rtp_proxy(); } xlog("L_ERR","66666666666666666666666666666666666666666"); t_on_failure("1"); t_on_reply("1"); }; The call is going through Route[1] then through Route[4], but rtp proxy is never forced...Any tips please? Regards

2 1

Meaning of empty body in NOTIFY
by Eugen Dedu 13 Jun '11

13 Jun '11

Hi, ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for users who are not registered an empty NOTIFY body is returned when asked by a SUBSCRIBE. What does this mean from SIP standard point of view, and from kamailio point of view (are they identical?) I see in RFC3265/3.1.6.2: .... If the resource has no meaningful state at the time that the SUBSCRIBE message is processed, this NOTIFY message MAY contain an empty or neutral body but is difficult for me to interpret what it means. Example: I ask the presence for a user xyz who registered and quit application long time ago: SUBSCRIBE sip:xyz@ekiga.net SIP/2.0 CSeq: 1 SUBSCRIBE Via: SIP/2.0/UDP 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport User-Agent: Ekiga/3.3.1 From: <sip:eugen.dedu@ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8 Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8@snoopy Supported: eventlist To: <sip:xyz@ekiga.net> Accept: application/pidf+xml Accept: multipart/related Accept: application/rlmi+xml Contact: <sip:eugen.dedu@82.238.108.175:5060> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK Expires: 300 Event: presence Content-Length: 0 Max-Forwards: 70 I receive the following answer: NOTIFY sip:eugen.dedu@82.238.108.175:5060 SIP/2.0 CSeq: 1 NOTIFY Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0 User-Agent: Kamailio (1.5.3-notls (i386/linux)) From: sip:xyz@ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0 Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8@snoopy To: sip:eugen.dedu@ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8 Contact: <sip:86.64.162.35:5060> Subscription-State: active;expires=370 Event: presence Content-Length: 0 Max-Forwards: 70 To resume: What does SIP standard say about this NOTIFY with empty body? Does this mean that the user xyz is offline? Or does this mean that user's status has not changed? In fact, the NOTIFY with empty body (as shown above) is the first one sent by kamailio, so there is no "previous" state of that user, hence "unchanged" status has no meaning. Thank you, -- Eugen

6 21

Proxy Address
by michel freiha 13 Jun '11

13 Jun '11

Dear All, I'm using kamailio as Proxy server for registration with RTP proxy as relay server for voice packets...I have an issue that when sending an Invite, the audio address in SDP (o header)is not RTP Proxy address while if I send another Invite to kamailio (second Invite), the RTP Proxy address will be present in SDP sent back by Kamailio... Anyone has a clue about what could be that matter? Regards

2 2

Double tuple in notify
by Eugen Dedu 12 Jun '11

12 Jun '11

Hi again, > Then it is the time for Eugen to add it in his sip client and send > the patch for enhancing kamailio's presence server with rfc4481, as > well :o) I start my application, and make my own status be shown. I change my status back and forth a few times between Away and DoNotDisturb (each time a publish is sent), the status returned (in notify) is good. When I change my status to Online/Available (and only to this), kamailio returns a double tuple. Is it normal to have a double tuple? If yes, which one to choose (the one with latest timestamp)? Here it is: NOTIFY sip:eugen.dedu@82.238.108.175:5060 SIP/2.0 CSeq: 7 NOTIFY Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK7634.4d863c14.0 User-Agent: Kamailio (1.5.3-notls (i386/linux)) From: sip:eugen.dedu@ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-29fc Call-ID: 8a0d723b-0f8d-e011-84d7-0024d693d8e8@snoopy To: sip:eugen.dedu@ekiga.net;tag=7c75723b-0f8d-e011-84d7-0024d693d8e8 Contact: <sip:86.64.162.35:5060> Subscription-State: active;expires=270 Event: presence Content-Length: 938 Content-Type: application/pidf+xml Max-Forwards: 70 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns: rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:eugen.dedu@ekiga.net"> <tuple id="T00000002"> <status> <basic>open</basic> </status> <contact priority="1">sip:eugen.dedu@ekiga.net</contact> <note>online - </note> <timestamp>2011-06-04T13:55:56+02:00+02:00</timestamp> </tuple> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="T00000001"> <status> <basic>open</basic> </status> <contact priority="1">sip:eugen.dedu@ekiga.net</contact> <note>away - jjk</note> <timestamp>2011-06-04T13:54:37+02:00+02:00</timestamp> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="pid1"> <rpid:activities> <rpid:away/> </rpid:activities> </dm:person> </presence> -- Eugen

3 2

Re: [SR-Users] Meaning of empty body in NOTIFY
by Daniel-Constantin Mierla 10 Jun '11

10 Jun '11

On 6/10/11 10:07 PM, Craig Southeren wrote: > On 10/06/2011 1:01 PM, Daniel-Constantin Mierla wrote:..deleted >>> >>> So, if a client uploads a presence document that contains a >>> <timed-status> element, that element could be sent by the PA when >>> the client goes offline, rather than the entire document being >>> discarded (which is what I suspect is happening now). >> >> we don't have this rfc implemented in kamailio at this time. Is any >> UA you know doing this kind of thing? > > No Then it is the time for Eugen to add it in his sip client and send the patch for enhancing kamailio's presence server with rfc4481, as well ;-) Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

Functions for DNS
by Nathan Angelacos 10 Jun '11

10 Jun '11

I'm looking to build a set of contacts with q values based on SRV records for serial/parallel forking. I want "enum_query" (loading the contact set with a q value based on the order,preference of the NAPTR record) but the ruri is not an e.164 number, and SRV records are used instead. Think in terms of "find all peer proxies based on SRV lookup, and sort them for use with t_load_contacts()/t_next_contacts()." So far I haven't found any api that exposes the DNS lookups in sip-router. Before rolling something custom with Lua, am I missing something that's already documented? Or are there plans to make such a module? Sure don't want to re-invent the wheel. tia

2 2

New developer: Peter Dunkley
by Daniel-Constantin Mierla 09 Jun '11

09 Jun '11

Hello, just to let everyone to know that starting with today Peter got developer access to GIT repository. He sent many patches in the past months to fix issues or add new features to presence modules (e.g., OMA extensions to embedded XCAP server) -- some are pending to be committed and expect more to come, as he is continuing to work with this part of SIP server. The user id is 'pd'. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Best way to handle OPTIONS message with Max-Forwards of 0
by Sean O'Donnell 09 Jun '11

09 Jun '11

Looks more comprehensive than mine. Thanks! Sean O'Donnell ---- On Wed, 8 Jun 2011, Klaus Darilion (klaus.mailinglists(a)pernau.at) wrote: Anyway - not standard conform but efficient (put somewhere in the beginning) force_rport(); if (is_method("OPTIONS")) { sl_send_reply("200","OK"); exit; } # sipura/linksys keepalive if (is_method("NOTIFY")) { if ( search("^Event: keep-alive") ) { sl_send_reply("200","Keep Alive"); exit; } } # ignore requests generated by sipvicious # User-Agent: friendly-scanner if ($ua == "friendly-scanner") { exit; } regards Klaus Am 08.06.2011 18:28, schrieb Sean O'Donnell: > Hi: > > We're running kamailio 1.5.5 as part of our VoIP infrastructure. One of our > service providers > has an SBC that sends an OPTIONS messages with a Max-Forwards of 0. > > Looking at RFC 3261 section 11, this seems to be a valid method of forcing the > proxy to respond to the OPTIONS. > However, the standard initial sanity checks in most proxy configs will reject > this message. I fixed our script > by changing to this: > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > if( method != "OPTIONS" ) { > sl_send_reply("483","Too Many Hops"); > return; > }; > }; > if ( msg:len > max_len ) { > sl_send_reply("513", "Message too big"); > return; > }; > > but does anyone have a better solution? > > Thanks, > > Sean O'Donnell > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] Best way to handle OPTIONS message with Max-Forwards of 0
by Sean O'Donnell 08 Jun '11

08 Jun '11

Thanks for the reply. In our case, the SBC treated the 483 as an error of some sort and wouldn't send traffic. It wanted a 200/OK. Anyway, I thought that when a proxy responds to an OPTIONS, its supposed to answer with a 200/OK (3261 11.2). Is that correct? Sean O'Donnell ---- On Wed, 8 Jun 2011, Klaus Darilion (klaus.mailinglists(a)pernau.at) wrote: What is the problem with the default config? The proxy will respond with 483 as the standard defines. I guess the SBC will doe this to check if the proxy is alive and will accept any response as "the proxy is alive". regards Klaus Am 08.06.2011 18:28, schrieb Sean O'Donnell: > Hi: > > We're running kamailio 1.5.5 as part of our VoIP infrastructure. One of our > service providers > has an SBC that sends an OPTIONS messages with a Max-Forwards of 0. > > Looking at RFC 3261 section 11, this seems to be a valid method of forcing the > proxy to respond to the OPTIONS. > However, the standard initial sanity checks in most proxy configs will reject > this message. I fixed our script > by changing to this: > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > if( method != "OPTIONS" ) { > sl_send_reply("483","Too Many Hops"); > return; > }; > }; > if ( msg:len > max_len ) { > sl_send_reply("513", "Message too big"); > return; > }; > > but does anyone have a better solution? > > Thanks, > > Sean O'Donnell > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Best way to handle OPTIONS message with Max-Forwards of 0
by Sean O'Donnell 08 Jun '11

08 Jun '11

Hi: We're running kamailio 1.5.5 as part of our VoIP infrastructure. One of our service providers has an SBC that sends an OPTIONS messages with a Max-Forwards of 0. Looking at RFC 3261 section 11, this seems to be a valid method of forcing the proxy to respond to the OPTIONS. However, the standard initial sanity checks in most proxy configs will reject this message. I fixed our script by changing to this: # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { if( method != "OPTIONS" ) { sl_send_reply("483","Too Many Hops"); return; }; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); return; }; but does anyone have a better solution? Thanks, Sean O'Donnell

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Incorrect registrar Expires handling
by Andreas Granig 08 Jun '11

08 Jun '11

Hi, Found an interesting call-flow today where a client (pjsip) is behind NAT and behaves like this during registration (kamailio-3.1.3): REGISTER sip:somedomain.com Contact: <sip:me@10.0.0.7:2001;ob> Expires: 900 SIP/2.0 200 OK Contact: <sip:me@10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001" REGISTER sip:somedomain.com Contact: <sip:me@1.2.3.4:3001;transport=UDP;ob> Contact: <sip:me@10.0.0.7:2001;ob>;expires=0 Expires: 900 SIP/2.0 200 OK Contact: <sip:me@10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001", <sip:me@1.2.3.4:3001;transport=UDP;ob>;expires=900 So obviously the client is trying to unregister its private contact (for whatever reason - probably to play nice with registrars not supporting far-end NAT traversal), and tries to register the Contact with the "received" address instead. Kamailio however ignores the ";expires=0" for the private Contact and just saves both of them with the expiry value given in the Expires header. Taking the client behavior aside for now: is this the correct behavior of the registrar module? In RFC3261 chapter 10.2.1.1 it says: #+ There are two ways in which a client can suggest an expiration interval for a binding: through an Expires header field or an "expires" Contact header parameter. The latter allows expiration intervals to be suggested on a per-binding basis when more than one binding is given in a single REGISTER request, whereas the former suggests an expiration interval for all Contact header field values that do not contain the "expires" parameter. #- So I'd say that in my specific case "expires" Contact header param (which is 0 for the binding having 10.0.0.7) should de-register this particular binding, whereas the Expires header should be applied to the second binding (having the 1.2.3.4) because it doesn't carry an "expires" param. Opinions on that one? Andreas

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incorrect port 0 in reply from rtp proxy
by Mokhtar Bengana 08 Jun '11

08 Jun '11

I have Kamailio and rtpproxy running behind NAT on the same server. Inbound calls to Kamailio from the outside work fine and I have two way audio but when I try an outbound echo test call the call connects but I have no audio. rtpproxy seems to be running and all rtp ports are poiting to Kamailio internal ip. The syslog is showing this error. Not sure what's missing in the config. Appreciate any inputs. Thanks. root@Kamailio:~# tail /var/log/syslog Jun 3 01:24:02 Kamailio kamailio[810]: INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled Jun 3 01:25:05 Kamailio kamailio[791]: INFO: <core> [forward.c:786]: broken reply to forward - no 2nd via Jun 3 01:25:29 Kamailio kamailio[791]: ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy Jun 3 01:25:30 Kamailio kamailio[793]: ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy

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t.38 fax
by Sanjeev Singh 07 Jun '11

07 Jun '11

does SER support t.38 fax? what other features related to fax does SER support. please expain in detail. thanks, Sanjeev Singh

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xhttp and dollar sign
by Andreas Granig 06 Jun '11

06 Jun '11

Hi, Is it possible to somehow quote the $ sign in xhttp_send_reply() in order not to be interpreted by kamailio-3.1.3? Tried with "\" and putting the whole body in single quotes, but I always get this error message during startup: ERROR: <core> [pvapi.c:720]: wrong char [$/36] in [${foo}] ERROR: <core> [sr_module.c:1156]: bad PVE format: "${foo}" ERROR: <core> [mod_fix.c:501]: Cannot convert function parameter 1 to spve So I actually just want to return the string "${foo}". Andreas

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Error bad syntax - DEBUG_FLAGS
by Skyler 06 Jun '11

06 Jun '11

Hi all, I'm new to Kamailio (3.1), learning the configuration language. I'm trying to take the advice from one of the docs to create a DEBUG block and place calls everywhere in my script. Right out of the gate I get these errors on loading. Can someone tell me where I went wrong and how I can fix this? It says flag not declared though it's there and I'm a bit lost. 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 6: syntax error 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 6: missing '(' or ')' ? 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 6: bad command: missing ';'? 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 18: '('')' expected (function call) 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 18: bad command: missing ';'? 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 20: bad command 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 802, column 24: flag not declared #------------------------------ # Flags #------------------------------ #!define FLAG_DEBUG 1 ... route { # Call debug flags route(ROUTE_DEBUG_FLAGS); ..... route[ROUTE_DEBUG_FLAGS] { resetflag(FLAG_DEBUG); #------------------------------------------------------------ # Methods that will have debugging turned on #----------------------------------------------------------- if (is_method("ACK")) { resetflag(FLAG_DEBUG); } else if (is_method("BYE")) { setflag(FLAG_DEBUG); } else if (is_method("CANCEL")) { setflag(FLAG_DEBUG); } else if (is_method("INVITE")) { setflag(FLAG_DEBUG); } else if (is_method("OPTIONS")) { setflag(FLAG_DEBUG); } else if (is_method("PUBLISH")) { setflag(FLAG_DEBUG); } else if (is_method("REGISTER")) { resetflag(FLAG_DEBUG); } else if (is_method("MESSAGE")) { setflag(FLAG_DEBUG); } else if (is_method("NOTIFY")) { setflag(FLAG_DEBUG); } else if (is_method("SUBSCRIBE")) { setflag(FLAG_DEBUG); } else if (is_method("INFO")) { setflag(FLAG_DEBUG); } else if (is_method("PRACK")) { setflag(FLAG_DEBUG); } else if (is_method("REFER")) { setflag(FLAG_DEBUG); } else { setflag(FLAG_DEBUG); } } }

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Using =~ to get the group identifier with longest match
by Iñaki Baz Castillo 06 Jun '11

06 Jun '11

Hi, I wonder how feasible is the folowing: I have a regular expresion like: REGEXP = "^((1|20)|(13|3|4)|(5|6))" There are 3 matching groups: 1) (1|20) 2) (13|3|4) 3) (5|6) I want to match the RURI user, let's suppose $rU = "138787686223". In this case the above regexp matches groups 1 and 2, but since group 2 matching is longest ("13" is longest than "1") I want to get value 2. In case $rU = "999999" I need to get "null", in case $rU = "1000022323" I need to get "1". AFAIK there is no way to do it with current regular expression utils in Kamailio, but maybe I miss something. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Questino about dialplan module
by Gary Chen 06 Jun '11

06 Jun '11

Kamailio 3.1.2 I am testing out dialplan module. Every time when I issue 'kamctl dialplan show', it shows what is in the database even if I have not yet issue "kamctl dialplan reload' after I changed data in database. I thought that dialplan is stored in the memory and you have to issue the reload before you can see the change. Why the ' kamctl dialplan show' display the data directly from mysql database? Does that mean that dialplan data is not stored in the memory? Gary

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callback function of tm module
by Bruno Bresciani 06 Jun '11

06 Jun '11

Hi All, I'm having doubts in the implementation of the callback function module tm. As she calls the function to record register_tmcb ()? Can someome help me? Regards

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Re: [SR-Users] Questino about dialplan module
by Gary Chen 03 Jun '11

03 Jun '11

I am talking about kamctl dialplan command not mi. When you type in kamctl, it lists a lot commands. Are these commands still valid? See below: kamctl ../../sbin/kamctl 3.1.0 Existing commands: -- command 'start|stop|restart' restart ............................ restart Kamailio start .............................. start Kamailio stop ............................... stop Kamailio -- command 'acl' - manage access control lists (acl) acl show [<username>] .............. show user membership acl grant <username> <group> ....... grant user membership (*) acl revoke <username> [<group>] .... grant user membership(s) (*) -- command 'lcr' - manage least cost routes (lcr) lcr show_gws....... show database gateways lcr show_routes.... show database routes lcr dump_gws....... show in memory gateways lcr dump_routes.... show in memory routes lcr reload ........ reload lcr gateways and routes lcr eval_weights .. evaluates probability for given GW's weights -- command 'cr' - manage carrierroute tables cr show ....................................................... show tables cr reload ..................................................... reload tables cr dump ....................................................... show in memory tables cr addcn <carrier id> <carrier name> .......................... add a carrier name cr rmcn <carrier id> ......................................... rm a carrier name cr adddn <domain id> <domain name> ............................ add a domain name cr rmdn <domain id> .......................................... rm a domain name cr addcarrier <carrier> <scan_prefix> <domain> <rewrite_host> ................ <prob> <strip> <rewrite_prefix> <rewrite_suffix> ............... <flags> <mask> <comment> .........................add a carrier (prob, strip, rewrite_prefix, rewrite_suffix,................... flags, mask and comment are optional arguments) ............... cr rmcarrier <carrier> <scan_prefix> <domain> ................ rm a carrier -- command 'rpid' - manage Remote-Party-ID (RPID) rpid add <username> <rpid> ......... add rpid for a user (*) rpid rm <username> ................. set rpid to NULL for a user (*) rpid show <username> ............... show rpid of a user -- command 'add|passwd|rm' - manage subscribers add <username> <password> .......... add a new subscriber (*) passwd <username> <passwd> ......... change user's password (*) rm <username> ...................... delete a user (*) -- command 'add|dump|reload|rm|show' - manage trusted trusted show ...................... show db content trusted dump ...................... show cache content trusted reload .................... reload db table into cache trusted add <src_ip> <proto> <from_pattern> <tag> ....................... add a new entry ....................... (from_pattern and tag are optional arguments) trusted rm <src_ip> ............... remove all entres for the given src_ip -- command 'dispatcher' - manage dispatcher * Examples: dispatcher addgw 1 sip:1.2.3.1:5050 1 'outbound gateway' * dispatcher addgw 2 sip:1.2.3.4:5050 3 '' * dispatcher rmgw 4 dispatcher show ..................... show dispatcher gateways dispatcher reload ................... reload dispatcher gateways dispatcher dump ..................... show in memory dispatcher gateways dispatcher addgw <setid> <destination> <flags> <description> .......................... add gateway dispatcher rmgw <id> ................ delete gateway -- command 'db' - database operations db exec <query> ..................... execute SQL query db roexec <roquery> ................. execute read-only SQL query db run <id> ......................... execute SQL query from $id variable db rorun <id> ....................... execute read-only SQL query from $id variable db show <table> ..................... display table content -- command 'speeddial' - manage speed dials (short numbers) speeddial show <speeddial-id> ....... show speeddial details speeddial list <sip-id> ............. list speeddial for uri speeddial add <sip-id> <sd-id> <new-uri> [<desc>] ... ........................... add a speedial (*) speeddial rm <sip-id> <sd-id> ....... remove a speeddial (*) speeddial help ...................... help message - <speeddial-id>, <sd-id> must be an AoR (username@domain) - <sip-id> must be an AoR (username@domain) - <new-uri> must be a SIP AoR (sip:username@domain) - <desc> a description for speeddial -- command 'avp' - manage AVPs avp list [-T table] [-u <sip-id|uuid>] [-a attribute] [-v value] [-t type] ... list AVPs avp add [-T table] <sip-id|uuid> <attribute> <type> <value> ............ add AVP (*) avp rm [-T table] [-u <sip-id|uuid>] [-a attribute] [-v value] [-t type] ... remove AVP (*) avp help .................................. help message - -T - table name - -u - SIP id or unique id - -a - AVP name - -v - AVP value - -t - AVP name and type (0 (str:str), 1 (str:int), 2 (int:str), 3 (int:int)) - <sip-id> must be an AoR (username@domain) - <uuid> must be a string but not AoR -- command 'alias_db' - manage database aliases alias_db show <alias> .............. show alias details alias_db list <sip-id> ............. list aliases for uri alias_db add <alias> <sip-id> ...... add an alias (*) alias_db rm <alias> ................ remove an alias (*) alias_db help ...................... help message - <alias> must be an AoR (username@domain)" - <sip-id> must be an AoR (username@domain)" -- command 'domain' - manage local domains domain reload ....................... reload domains from disk domain show ......................... show current domains in memory domain showdb ....................... show domains in the database domain add <domain> ................. add the domain to the database domain rm <domain> .................. delete the domain from the database -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry -- command 'fifo' fifo ............................... send raw FIFO command -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Juha Heinanen Sent: Friday, June 03, 2011 3:50 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List Subject: [SR-Users] Questino about dialplan module Gary Chen writes: > I am testing out dialplan module. Every time when I issue 'kamctl > dialplan show', it shows what is in the database even if I have not > yet issue "kamctl dialplan reload' after I changed data in database. I > thought that dialplan is stored in the memory and you have to issue > the reload before you can see the change. Why the ' kamctl dialplan > show' display the data directly from mysql database? Does that mean > that dialplan data is not stored in the memory? there is no mi function to show dialplan rules in memory. i have no idea, what 'show' does. please read readme of dialplan module before asking questions about the module. -- 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

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IRC meeting today, 15:00GMT
by Daniel-Constantin Mierla 03 Jun '11

03 Jun '11

Hello, everyone that wants to discuss about the next major release (v3.2.0), please join us on IRC: Channel: #sip-router Server: irc.freenode.net Time: 15:00GMT/London (16:00 Berlin/Paris/Madrid, 10:00am New York) Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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IMC module - #join to private room crashes kamailio
by Ozren Lapcevic 03 Jun '11

03 Jun '11

Hi, there is a bug in IMC module that crashes Kamailio. Here is the scenario: One user (pero(a)abc.hr) creates private chat room ( chat-999(a)abc.hr). Other user (oz(a)abc.hr) tries to join the chat-999(a)abc.hrby sending #join. Kamailio crashes at that point. Here is the log: Best regards, Ozren Jun 1 15:10:37 ser-test kernel: [5844312.456709] kamailio[20118]: segfault at 10 ip 7f28f89dae8e sp 7fffffffb750 error 4 in imc.so[7f28f89cf000+17000] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: INFO: <script>: going to imc: F [sip:oz@abc.hr] T [sip:chat-999@abc.hr] B [#join] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [receive.c:186]: preparing to run routing scripts... Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc [imc.c:525]: found command Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc [imc_cmd.c:146]: command: [join] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=223 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc [imc_cmd.c:292]: found room [chat-999] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: ERROR: imc [imc_cmd.c:317]: attept to join private room [chat-999] from user [oz] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core> [main.c:741]: child process 20118 exited by a signal 11 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core> [main.c:744]: core was not generated Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [msg_translator.c:204]: check_via_address(161.53.255.82, 161.53.255.82, 0) Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core> [parser/msg_parser.c:632]: method: <MESSAGE> Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20114]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20115]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20116]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20117]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20122]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20124]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20125]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20129]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20128]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20127]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20123]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20126]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20120]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20130]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20131]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc [imc.c:709]: room 3 chat-777 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc [imc.c:709]: room 11 chat-888 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc [imc.c:709]: room 13 chat-999

2 6

perlvdb error
by Roman Yeryomin 03 Jun '11

03 Jun '11

On 18 April 2011 15:47, Henning Westerholt <henning.westerholt(a)1und1.de> wrote: > On Monday 18 April 2011, Roman Yeryomin wrote: >> coredump? easy :) >> you can pick core file itself here (it's pretty heavy to send to the list): >> http://83.99.159.113/core.kamailio.20110418-1331 >> >> What do you mean by backtrace? log output or strace output? > > Hi Roman, > > he refers to a gdb backtrace, "gdb bt". More informations e.g. here: > http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:corefiles > ok, here is the backtrace: # gdb kamailio core GNU gdb 6.8-debian Copyright (C) 2008 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "i486-linux-gnu"... warning: Can't read pathname for load map: Input/output error. Reading symbols from /lib/tls/i686/cmov/libdl.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libdl.so.2 Reading symbols from /lib/tls/i686/cmov/libresolv.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libresolv.so.2 Reading symbols from /lib/tls/i686/cmov/libc.so.6...done. Loaded symbols for /lib/tls/i686/cmov/libc.so.6 Reading symbols from /lib/ld-linux.so.2...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/local/lib/kamailio/modules_k/mi_fifo.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/mi_fifo.so Reading symbols from /usr/local/lib/kamailio/libkmi.so.1...done. Loaded symbols for /usr/local/lib/kamailio/libkmi.so.1 Reading symbols from /usr/local/lib/kamailio/modules_k/kex.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/kex.so Reading symbols from /usr/local/lib/kamailio/libkcore.so.1...done. Loaded symbols for /usr/local/lib/kamailio/libkcore.so.1 Reading symbols from /usr/local/lib/kamailio/modules/tm.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/tm.so Reading symbols from /usr/local/lib/kamailio/modules_k/tmx.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/tmx.so Reading symbols from /usr/local/lib/kamailio/modules/sl.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/sl.so Reading symbols from /usr/local/lib/kamailio/modules_k/rr.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/rr.so Reading symbols from /usr/local/lib/kamailio/modules_k/pv.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/pv.so Reading symbols from /usr/local/lib/kamailio/modules_k/maxfwd.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/maxfwd.so Reading symbols from /usr/local/lib/kamailio/modules_k/usrloc.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/usrloc.so Reading symbols from /usr/local/lib/kamailio/libsrdb1.so.1...done. Loaded symbols for /usr/local/lib/kamailio/libsrdb1.so.1 Reading symbols from /usr/local/lib/kamailio/modules_k/registrar.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/registrar.so Reading symbols from /usr/local/lib/kamailio/modules_k/textops.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/textops.so Reading symbols from /usr/local/lib/kamailio/modules_k/siputils.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/siputils.so Reading symbols from /usr/local/lib/kamailio/modules_k/xlog.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/xlog.so Reading symbols from /usr/local/lib/kamailio/modules/sanity.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/sanity.so Reading symbols from /usr/local/lib/kamailio/modules/ctl.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/ctl.so Reading symbols from /usr/local/lib/kamailio/modules/mi_rpc.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/mi_rpc.so Reading symbols from /usr/local/lib/kamailio/modules_k/acc.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/acc.so Reading symbols from /lib/tls/i686/cmov/libpthread.so.0...done. Loaded symbols for /lib/tls/i686/cmov/libpthread.so.0 Reading symbols from /usr/local/lib/kamailio/modules_k/perl.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/perl.so Reading symbols from /usr/lib/libperl.so.5.8...done. Loaded symbols for /usr/lib/libperl.so.5.8 Reading symbols from /lib/tls/i686/cmov/libm.so.6...done. Loaded symbols for /lib/tls/i686/cmov/libm.so.6 Reading symbols from /lib/tls/i686/cmov/libcrypt.so.1...done. Loaded symbols for /lib/tls/i686/cmov/libcrypt.so.1 Reading symbols from /usr/local/lib/kamailio/modules_k/perlvdb.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/perlvdb.so Reading symbols from /usr/local/lib/kamailio/modules/auth.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/auth.so Reading symbols from /usr/local/lib/kamailio/modules_k/auth_db.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/auth_db.so Reading symbols from /usr/local/lib/kamailio/modules/app_lua.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/app_lua.so Reading symbols from /usr/lib/liblua5.1.so.0...done. Loaded symbols for /usr/lib/liblua5.1.so.0 Reading symbols from /lib/tls/i686/cmov/libnss_files.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libnss_files.so.2 Reading symbols from /lib/tls/i686/cmov/libnss_dns.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libnss_dns.so.2 Core was generated by `kamailio -dddd'. Program terminated with signal 11, Segmentation fault. [New process 13773] #0 0xb7228e3a in parseurl (url=0xb7213230 "T�0\b#") at perlvdbfunc.c:60 60 if (strlen(cn) > 0) If I understand this correctly error refers to row 60 in perlvdbfunc.c strlen expects const char* instead of char* ? Regards, Roman

2 17

Re: [SR-Users] Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

That works. Thanks. From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Thursday, June 02, 2011 4:28 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gary Chen Subject: Re: [SR-Users] Problem using dispatcher. On 6/2/11 9:41 PM, Gary Chen wrote: Thanks for the quick reply. How can I fix this? When you set t_on_failure for an INVITE, the respective failure route will be executed for any reply >=0. What you have to do is to not re-route to a new destination from failure route, simply exit for those reply codes, like: if(t_check_status("401|407")) exit; You don't handle as well the canceled transactions, when re-routing makes no sense -- you should do it (see default config file for example. Cheers, Daniel We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. Any suggestion? -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [ mailto:sr-users-bounces@lists.sip-router.org ] On Behalf Of Alex Balashov Sent: Thursday, June 02, 2011 3:30 PM To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Problem using dispatcher. On 06/02/2011 03:26 PM, Gary Chen wrote: > the dispatcher always get into FAILURE route although the log shows > that kamailio did send INVITE to 10.10.1.222 and received '407 - > Proxy Authentication Required' from Freexwitch1 and then dispatcher > send INVITE to 10.10.1.223 and also received '407 - Proxy > Authentication Required' from Freexwitch2 and eventually failed. That's because any reply >= 300 is considered a failure. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ 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://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

Thanks for the quick reply. How can I fix this? We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. Any suggestion? -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Alex Balashov Sent: Thursday, June 02, 2011 3:30 PM To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Problem using dispatcher. On 06/02/2011 03:26 PM, Gary Chen wrote: > the dispatcher always get into FAILURE route although the log shows > that kamailio did send INVITE to 10.10.1.222 and received '407 - > Proxy Authentication Required' from Freexwitch1 and then dispatcher > send INVITE to 10.10.1.223 and also received '407 - Proxy > Authentication Required' from Freexwitch2 and eventually failed. That's because any reply >= 300 is considered a failure. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

Kamailio 3.1.2 Here is my setup: Phone1 (10.10.1.114) | Kamailio (10.10.1.230) / \ / \ Freeswitch1 Freeswitch2 ----------------------> Other phones (10.10.1.222) (10.10.1.223) Both Freeswitch's share the same REGISTRATION information. Phone1 is already registered with Freeswitch. When I try to dial 'Other phones' using phone1, the dispatcher always get into FAILURE route although the log shows that kamailio did send INVITE to 10.10.1.222 and received '407 - Proxy Authentication Required' from Freexwitch1 and then dispatcher send INVITE to 10.10.1.223 and also received '407 - Proxy Authentication Required' from Freexwitch2 and eventually failed. Any idea? Here is my configuration: modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 10000) modparam("dispatcher", "ds_probing_threshhold", 3) #modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:proxy@SIP-router1.testnet.net") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "flags", 3) modparam("dispatcher", "force_dst", 1) route { .... if ( ds_select_dst("$var(group)", "9") ){ route(RELAY_DS); return; ... } route[[RELAY_DS] if (is_method("INVITE")) { t_on_branch("BRANCH_ONE"); t_on_reply("NAT_REPLY"); t_on_failure("FAIL_DS"); } if (!t_relay()) { sl_reply_error(); } } failure_route[FAIL_DS] { if (t_check_status("408")) { ds_mark_dst("i"); } if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } t_on_failure("FAIL_DS"); route(RELAY_DS); } onreply_route[NAT_REPLY] { xlog("L_INFO", "REPLY_ROUTE: S=$rs - $rr F=$fu T=$tu IP=$avp(s:ip):$avp(s:port) ($si:$sp) ID=$ci\n"); } And here is the SIP messages from ngrep: U 10.10.1.144:51029 -> 10.10.1.230:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..F rom: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet .net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 70..Date: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires: 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..C ontent-Length: 261..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.144..s=SIP Call..t=0 0..m=audio 20372 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.144..a=rtpmap: 0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv .. # U 10.10.1.230:5060 -> 10.10.1.144:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net; user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..Server: kamailio (3.1.2 (x8 6_64/linux))..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.222:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..Via: SIP/2.0/ UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "L ine1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;us er=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 2011 19:14: 14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires: 180..A ccept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..Content- Length: 277..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.231..s=SIP Call..t=0 0..m=audio 38426 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.231..a=rtpmap:0 PCMU/80 00..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..a=nortp proxy:yes.. # U 10.10.1.222:5060 -> 10.10.1.230:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060 ;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..From: "Line1" <sip:5022441011@fs2000.lightyea r.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>..Call-ID: 00082166-efcb0013 -692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 0 3-05-32 -0400..Content-Length: 0.... # U 10.10.1.222:5060 -> 10.10.1.230:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/U DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb0021 53bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9 -6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea lm="fs2000.testnet.net", nonce="8da1ddb1-e393-442c-8be0-17e439457363", algorithm=MD5, qop="auth"..Content-Length: 0... . # U 10.10.1.230:5060 -> 10.10.1.222:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Fro m: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.n et;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..CSeq: 101 ACK..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.223:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d.. From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.lightyea r.net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 201 1 19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires : 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces.. Content-Length: 328..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.23110.10.1.231..s=SIP Call..t=0 0..m=audio 3842638426 RTP/AVP 0 8 18 101..c=IN IP4 216.49. 129.23110.10.1.231..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8 000..a=fmtp:101 0-15..a=sendrecv..a=nortpproxy:yes..a=nortpproxy:yes.. # U 10.10.1.223:5060 -> 10.10.1.230:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060 ;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..From: "Line1" <sip:5022441011@fs2000. testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>..Call-ID: 00082166- efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011 -04-19 03-05-32 -0400..Content-Length: 0.... # U 10.10.1.223:5060 -> 10.10.1.230:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/U DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb0021 53bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9 -6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea lm="fs2000.testnet.net", nonce="8769298b-c52d-43b8-b69c-6ebf26f3184d", algorithm=MD5, qop="auth"..Content-Length: 0... . # U 10.10.1.230:5060 -> 10.10.1.223:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Fro m: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.n et;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..CSeq: 101 ACK..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.144:5060 SIP/2.0 503 Service unavailable: no more dst..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1 " <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=p hone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..Server: kamailio (3.1.2 (x86_64/linux))..Content-Length: 0.... # U 10.10.1.144:51030 -> 10.10.1.230:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From : "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.ne t;user=phone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..D ate: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 ACK..Content-Length: 0.... Gary Chen

2 1

discussing about next major release - v3.2.0
by Daniel-Constantin Mierla 02 Jun '11

02 Jun '11

Hello, we should decide the roadmap to next major release v3.2.0, so probably an IRC meeting for some real time chatting would be good. I am thinking of June 3 or 4, 15:00GMT. Anyone interested in the meeting and available in one of those dates? The date with most of devs available will be selected. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 7

RTP traffic
by Lucas Alvarez 01 Jun '11

01 Jun '11

Hi, I'm having a problem with RTP traffic, I have softphone connecting from the outside, I can register but Kamailio is sending all RTP traffic to the private IP address that the softphone has in it own local network. How can I fix this? Any help will be appreciated. Lucas

2 2

serweb & xSER 3.1
by Skyler 01 Jun '11

01 Jun '11

Hi all, Wondering if anyone has serweb working/functioning with Kamailio or SER 3.1 branch? If so which version of serweb are you using 2.0.0 or 2.1.0-dev1? Any issues or tips for first time install/setup of serweb? S.

1 0

has_tran_tmcbs function
by Bruno Bresciani 01 Jun '11

01 Jun '11

Hi, I can't understand the has_tran_tmcbs function... Can someone tell me how it works? Regards,

2 1

Kamailio Tables with index
by Klaus Darilion 01 Jun '11

01 Jun '11

Hi! With Openser 1.2 all the tables got an auto-increment-index column. Is this index column just for layout/performance purposes or is it also used somewhere insides the Kamailio code? If not (what I expect), then Kamailio 3.1 should still work with tables without those columns - correct? Thanks Klaus

2 1

ClueCon 2011 News and Notes
by marketing＠cluecon.com 31 May '11

31 May '11

1 0

Issue using pua_bla module
by Watkins, Bradley 31 May '11

31 May '11

I'm trying to get the pua_bla module to do its thing, and I'm running into an issue. I'm getting these errors when the first NOTIFY comes from the phone: ERROR: presence [publish.c:515]: Missing or unsupported event header field value ERROR: presence [publish.c:518]: event=[dialog;sla] I see the phone register and the subsequent SUBSCRIBE (both successful) from Kamailio, so this bit seems to be correct. I'm loading what I think are all the necessary modules: loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "pua.so" loadmodule "pua_bla.so" And I'm handling the NOTIFY as per the documentation (xlogs to see that in fact the script executes): if(is_method("NOTIFY") && to_uri=~"bla_aor(a)domain.com") { xlog("Matched bla_aor to_uri\n"); if(bla_handle_notify()) { xlog("Handled NOTIFY\n"); t_reply("200", "OK"); } } I see the log entries from the above xlogs, so it certainly is executing this part of the script. Any ideas why I'm getting the errors? All ideas and/or assistance are greatly appreciated. Thanks in advance, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

3 6

Siremis: changes to Kamailio DB
by Klaus Darilion 31 May '11

31 May '11

Hi! I just read the Siremis installation howto. One of the task of the installer is: "Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules" I wonder what siremis exactly does to the DB. Is greped over the siremis source file but couldn't find the changes. Are they somewhere documented? (e.g. .sql file with the statemens). Anyway, I wonder changes are needed at all. IMO, if possible, the Kamailio DB should not be changed by an external too - otherwise this may cause problems with updates to the Kamailio DB structure. Thanks Klaus

3 3

kamailio debian package bug report : 3.1.3+lenny2
by Philippe HENSEL 31 May '11

31 May '11

Hi developpers and packagers, The file "/usr/lib64/kamailio/libsrdb2.so.1.0" seems to be included in both "kamailio_3.1.3+lenny2_amd64.deb" and "kamailio-mysql-modules_3.1.3 +lenny2_amd64.deb". So it is not possible to install "kamailio-mysql-modules". Thanks for help

5 8

Multidomains - quick question
by Lucas Alvarez 31 May '11

31 May '11

Hi, if I want to register clients from outside of my local network, do I have to enable the domain module and add the external IP as a new domain? Thanks in advance, Lucas Alvarez

2 2

acc Locks
by Borin 31 May '11

31 May '11

Hi all, Recently I met the following problem with kamailio acc table. We are using kamailio for calls dispatching to a couple of rtp gateways and for call statistics acc is used (with mysql). I tried to delete some old entries from acc table and for this time mysql "show processes" showed a lot of entries like " kamailio | Query | 0 | System lock | insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,start_time,src_addr,dst_addr,src_ |". I think that meant that kamailio could not write to the acc table for the time the data were being deleted. And it also did not forward INVITE requests further. Does that mean that if kamailio does not manage to write acc record for any reason it does not forward that INVITE/200ok? Best Regards, Kate

2 4

Green VoIP - energy efficiency and performaces of v3.0
by Daniel-Constantin Mierla 31 May '11

31 May '11

Hello, Jan Janak conducted a very interesting research project regarding energy efficiency of VoIP systems during 2010, a collaboration between iptel.org and Columbia University. The team used the source code from sip-router.org GIT repository from January 2010, which corresponds to Kamailio (former OpenSER) and SER v3.0. The latest stable series v3.1 shares the same internal architecture with v3.0. As part of the research work, Jan could also gather some figures about capacity and performances of v3.0 with a quite complex configuration file: etc/sip-router-oob.cfg (involving authentication and NAT traversal as well). You can read the paper about energy efficiency at: - Green VoIP Article: http://asipto.com/u/2j The draft notes about capacity and performances of v3.0 are available at: - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k Some interesting results: - one instance of SIP server with 500 000 online users (mixed users – behind and not NAT routers) – consumed energy 210W - one instance of SIP server with 1 000 000 online users (no NAT involved) – consumed energy 190W - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for SIP server, the server could support 43 000 simultaneous TLS connections – consumed energy 203W - one SIP server instance with 80 000 permanent TCP connections, the SIP server could still handle at least 1000 requests per second and a connection arrival rate of 1000 new connections per second, done for 20 000 new connections. CPU load generated by the SIP server was from 6% to 8%. I added a new section to the draft notes to list the enhancements done for the latest stable release (v3.1.x) that contribute to performance improvements, like asynchronous TLS, fine tuning of memory for TLS connections and raw UDP sockets. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

5 9

Late parallel branch (without canceling previous pending branches)
by Iñaki Baz Castillo 31 May '11

31 May '11

Hi, just wondering whether the following scenario (which it's not possible in kamailio 1.5) is possible in 3.X: 1) Alice calls a PSTN number. 2) Proxy receives the call and routes it to a media server which plays an early-announcement (30 seconds long). 3) After N seconds (maybe 10 seconds) proxy starts a new parallel branch to the PSTN gateway *without* cancelling the previous branch with the media server. 4) Let's suppose the PSTN phone rings and proxy gets 183 with SDP (ringing in real audio). Proxy converts such 183 into a 180 with no SDP (already possible in Kamailio). 5) After 30 seconds media server rejects the call with 480. PSTN branch remains ringing as usual. I just would like to know if Kamailio 3.X allows point 3. I think it's hard but maybe I miss some feature. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

Dispatcher variables
by Lee Archer 31 May '11

31 May '11

Hi all, I am using dispatcher and on a failure I would like to log the server that has failed. I am using $dd but this only ever returns the address of the first server that was tried and not any subsequent servers that were tried and also failed. What should I be using to display the IP address of the server dispatcher is about to use? Thanks Lee

2 2

Fw: Load capacity of SER??
by Piyush.Bansal＠relianceada.com 31 May '11

31 May '11

Dear All, I'm using SER as a SIP proxy server for quite a long time. I'm not been able to find its load bearing capacity. If you can help me get a figure in terms of BHCA, that'll be quite helpful. Thanks and Regards, --Piyush Bansal The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

1 0

IMC module doesn't save data to mysql database
by Ozren Lapcevic 30 May '11

30 May '11

Hi all, I'm trying to configure IMC module and get it to work properly but I've run into problems with saving IMC module data to mysql database. I'm using following code provided in IMC module documentation: modparam("imc", "db_url", DBURL) ## route block ### if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@") || (uri=~ "sip:chat-manager@")) { xlog("L_INFO", "going to imc: F [$fu] T [$tu] B [$rb]\n"); if(imc_manager()) sl_send_reply("200", "ok - imc_manager()"); else sl_send_reply("500", "command error"); exit; } I am creating chat room with following message sent to chat-manager(a)abc.hr: "#create chat-111" DEBUG log returns following: May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: INFO: <script>: going to imc: F [sip:oz@abc.hr] T [sip:chat-manager@abc.hr] B [#create chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc.c:525]: found command May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:146]: command: [create] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:151]: parameter 0=[chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:171]: new room [chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:186]: added room uri= sip:chat-111@abc.hr May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_mng.c:322]: [uri]= sip:oz@abc.hr May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_mng.c:326]: [user]= oz May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:196]: added the owner as the first member [ sip:oz@abc.hr] "./kamctl fifo imc_list_rooms" returns following: ROOM:: URI=sip:chat-111@abc.hr MEMBERS=0 OWNER=sip:oz@abc.hr "./kamctl fifo imc_list_members sip:chat-111@abc.hr" returns following: ROOM:: sip:chat-111@abc.hr NR_OF_MEMBERS=1 MEMBER:: sip:oz@abc.hr However, that data is not persisted/saved in the imc_rooms and imc_members tables in the database. Any clues what might be the problem here? Any help is greatly appreciated. Best regards, Ozren Lapcevic PS I'm runing Kamailio 3.1.4 with default cfg file with enabled mysql, multidomain, usrlocdb, auth and NAT. I can register clients, establish calls, send IMs, etc. imc_rooms and imc_members tables were created when kamailio was installed. I've also checked imc-create.sql script and tables are correctly created. The same DBURL used for imc is also used for auth_db, domain and usrloc modules and I haven't experienced any problems with those modules and database connection/persistence. domain, location, version and subscriber tables are correctly populated. I'm not getting any errors from imc_manager(), it returns 200 OK. I can't trace any errors in DEBUG related to problems with mysql database.

1 0

Kamailio service not starting on reboot when using mysql database
by Rick Aldred 30 May '11

30 May '11

I have set up kamailio 3.1.2 on an install of Debian Squeeze, using the default config but with the following defines enabled: WITH_MYSQLWITH_AUTHWITH_USRLOCDBWITH_NAT Everything works as expected when starting Kamailio from a terminal, e.g.service kamailio start But after a reboot, the service fails to start and all SIP requests are dropped with a ICMP port unreachable error. The following is an extract of /var/log/syslog when this problem occurs. It looks like Kamailio is failing to access the mysql socket, as the mysql service has not yet started? Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2)Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [db.c:289]: could not add connection to the poolApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: usrloc [dlist.c:486]: failed to open database connectionApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: registrar [reg_mod.c:396]: failed to register domainApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:569Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:572Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:574Apr 12 10:08:34 SIPServer mysqld_safe: Starting mysqld daemon with databases from /var/lib/mysqlApr 12 10:08:34 SIPServer mysqld: 110412 10:08:34 [Note] Plugin 'FEDERATED' is disabled.Apr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 InnoDB: Started; log sequence number 0 44233Apr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 [Note] Event Scheduler: Loaded 0 eventsApr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 [Note] /usr/sbin/mysqld: ready for connections.Apr 12 10:08:36 SIPServer mysqld: Version: '5.1.49-3' socket: '/var/run/mysqld/mysqld.sock' port: 3306 (Debian) Both the Kamailio and mysql services start with priority 18 on run levels 2345. I tried adjusting the Kamailio service to priority 19, but the problem still occurs. I am relatively new to Kamailio and mysql as well as Linux / Debian administration, so I'm not really sure how to solve this problem. Any ideas would be greatly appreciated. Regards,Rick Aldred

5 10

kamailio and radius
by Asun 30 May '11

30 May '11

Hello I've installed kamailio 3.1 but It's doesn't Work with radius authentication. I've followed this manual <http://www.kamailio.org/docs/openser-radius-1.0.x.html> http://www.kamailio.org/docs/openser-radius-1.0.x.html but it does'nt work, because is for openser not kamailio. Is there another manual to do this? Anyone have radius authentication with kamailio? Can anyone help me? Thank you Asun

3 8

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