Hello I try to get some replies from redis. Time after time redis request
give me null result. But redis bs not disconnected.
This happens only with websocket endpoints. My queries is:
redis_cmd("srv1", "EXISTS $si", "s");
So at xLOG i see that $si correctly sended, but result is null. At db I
keep key IP with value TIMESTAMP
Intresting that route, that givesme this result idependend of WS or UDP
endpoint. IT does not know about it anything.
Good day. Is anyone able to let me in on what this means "WARNING: <core>
[local_timer.c:83]: _local_timer_dist_tl(): 0 expire timer added"? I just
started seeing this after pulling down 5c1a9df from git and it seems like
something that I probably don't want happening.
--
Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
Hello Daniel,
Here paste for gdb
http://fpaste.org/191338/25043949/
I got REGISTER and SUBSCRIBE start working correctly I see on asterisk correct record routes and sip traffic flow, but when asterisk or client ( soft phone) send OPTIONS or NOTIFY can't get properly relay it.
This is SUBSCRIBE route.
<--- Transmitting (NAT) to 10.18.130.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060
Via: SIP/2.0/TLS 192.168.88.252:5062;received=Client public ip;branch=z9hG4bK0bbe1f7d27257bba9;rport=5062
Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes>
Record-Route: <sip:PUBLIC_KAMAILIO_IP:5081;transport=tls;r2=on;lr=on;ftag=a185d974ec;nat=yes>
From: "Slava Bendersky" <sips:10101@networklab.ca>;tag=a185d974ec
To: <sips:10101@networklab.ca>;tag=as00757d3e
Call-ID: b08adb1ad1804a83
CSeq: 236711034 SUBSCRIBE
Server: FPBX-2.11.0(11.15.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:10101@10.18.130.50:5060>;expires=3600
Content-Length: 0
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "Slava Bendersky" <volga629(a)networklab.ca>
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Friday, February 27, 2015 6:42:33 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello,
I asked for the wrong command, as the bt full was already sent before -- I wanted the output from gdb for:
p *tcpconn
Daniel
On 27/02/15 04:10, Slava Bendersky wrote:
Hello Daniel,
Here bt full from back trace.
http://fpaste.org/191207/50064491/
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com , "Slava Bendersky" <volga629(a)networklab.ca>
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Thursday, February 26, 2015 9:56:48 PM
Subject: Re: [SR-Users] kamailio asterisk
Hello Daniel,
I tried $rz option on top of request route and that where I see wrong request uri like sip:sips : . And as far I can see it happenes only for SUBSCRIBE INVITE and NOTIFY.
if($rz=="sips") {
$ru = "sip" + $(ru{s.substr,4,0});
}
Slava.
Sent from mobile device typos are expected.
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Sent: Feb 25, 2015 1:04 PM
To: Slava Bendersky
Cc: sr-users
Subject: Re: [SR-Users] kamailio asterisk
Hello,
On 25/02/15 17:19, Slava Bendersky wrote:
BQ_BEGIN
Hello Daniel,
substr you suggested didn't worked.
See my previous email.
your previous email didn't say anything about the results. That's why I asked. Be sure you don't have those spaces that are in the email you wrote. Also, I had more parenthesis in the parameter of the subst_uri().
Or you can try the alternative with:
if($rz=="sips") {
$ru = "sip" + $(ru{s.substr,4,0});
}
I asked for more details from the backtrace to confirm that what I found is the cause for the crash in this case -- see one of my previous emails from today.
Cheers,
Daniel
BQ_END
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
I'm pretty new to SIP, RTP/SRTP, WebRTC and Websockets, so I hope this
question is coherent. I have a group of SIP Softphones that need to connect
to a WebRTC/SIP-over-Websockets server. Can Kamailio be configured to let
me do this?
Any examples, tutorials or documentation would be appreciated. I'm trying
to determine how feasible this task is. :)
Thanks!
I am getting errors when I run the following code
if (sql_xquery("mydb", "select * from account where account = 'demo'", "res") == 1) {
xlog("L_INFO", "my number: $xavp(res=>number)\n");
} else {
xlog("L_WARN", "Connection forbidden from $si\n");
sl_send_reply("403", "Forbidden");
exit;
}
sql_result_free("res");
}
10(4033) ERROR: <core> [db_ut.c:225]: db_str2time(): Error during time conversion
10(4033) ERROR: <core> [db_val.c:159]: db_str2val(): error while converting datetime value from string
10(4033) ERROR: db_mysql [km_row.c:68]: db_mysql_convert_row(): failed to convert value
10(4033) ERROR: db_mysql [km_res.c:190]: db_mysql_convert_rows(): error while converting row #0
10(4033) ERROR: db_mysql [km_res.c:219]: db_mysql_convert_result(): error while converting rows
10(4033) ERROR: db_mysql [km_dbase.c:252]: db_mysql_store_result(): error while converting result
10(4033) ERROR: <core> [db_query.c:188]: db_do_raw_query(): error while storing result10(4033) ERROR: sqlops [sql_api.c:454]: sql_exec_xquery(): cannot do the query
Any help please
Hi there,
I want to use kamctl to monitor the state of a sip-proxy.
I just have no idea which commands are available or should i use
"kamctl mi" or "kamctl fifo" commands.
Google dont give helpful links for that.
Im am using kamailio 4.2.3+wheezy on a Debian wheezy and kamctrl dont
work:
# kamctl fifo uptime
Control engine 'FIFO' loaded
entering fifo_cmd uptime
500 command 'uptime' not available
FIFO command was:
:uptime:kamailio_receiver_19959
# kamctl fifo which
Control engine 'FIFO' loaded
entering fifo_cmd which
500 command 'which' not available
FIFO command was:
:which:kamailio_receiver_19985
# kamctl mi which
Control engine 'FIFO' loaded
entering fifo_cmd which
500 command 'which' not available
FIFO command was:
:which:kamailio_receiver_20016
here is my config:
# grep -i fifo kamailio.cfg
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
modparam("mi_fifo", "fifo_user", "kamailio")
modparam("mi_fifo", "fifo_mode", 0640)
# ls -la /tmp/kam*
srw-r----- 1 kamailio root 0 Feb 27 12:39 /tmp/kamailio_ctl
prw-r----- 1 kamailio kamailio 0 Feb 27 14:25 /tmp/kamailio_fifo
The only known command what is working is to look into a hashtable:
# kamctl mi sht_dump foo
Control engine 'FIFO' loaded
entering fifo_cmd sht_dump foo
Entry:: 135
sip:0049xxxxxxxxxxx@sip.domain.tld:: 1
Entry:: 205
sip:0049yyyyyyyyyyy@sip.domain.tld:: 1
FIFO command was:
:sht_dump:kamailio_receiver_20144
foo
any idea what is wrong?
and is there a documentation how to use "kamcrl mi" and "kamctl fifo"?
Thanks
Thomas
Hello Everyone,
Trying resolve issue with NOTIFY and OPTIONS that it will be send to client through kamailio from asterisk.
1. SUBSCRIBE record routes (works). On asterisk I see two records routes.
2. NOTIFY from asterisk should re use Records routes from SUBSCRIBE and send it to the client (works). I see in asterisk logs Route record.
3. NOTIFY received, but kamailio should send to client (not working). I see To: private Ip of the client or #4.
4. NOTIFY|OPTIONS sips to sip translation (not working). I see To: <sip:sips: none substr_uri helps include $rz.
Here SDP conversation.
http://fpaste.org/191761/14251457/
Feb 28 12:54:58 canlvprx01 /usr/sbin/kamailio[12838]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 192.168.88.252:5064 failed (RST) Connection refused
Feb 28 12:54:58 canlvprx01 /usr/sbin/kamailio[12838]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 192.168.88.252:5064: connect & send for 0x7f00ed4a1228 failed: Connection refused (111)
Feb 28 12:54:58 canlvprx01 /usr/sbin/kamailio[12838]: ERROR: tm [../../forward.h:247]: msg_send(): tcp_send failed
Feb 28 12:54:58 canlvprx01 /usr/sbin/kamailio[12838]: WARNING: tm [t_fwd.c:1608]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed
Feb 28 12:54:58 canlvprx01 /usr/sbin/kamailio[12838]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL)
Slava.
From: "Slava Bendersky" <volga629(a)networklab.ca>
To: miconda(a)gmail.com
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Friday, February 27, 2015 8:40:10 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello Daniel,
Here paste for gdb
http://fpaste.org/191338/25043949/
I got REGISTER and SUBSCRIBE start working correctly I see on asterisk correct record routes and sip traffic flow, but when asterisk or client ( soft phone) send OPTIONS or NOTIFY can't get properly relay it.
This is SUBSCRIBE route.
<--- Transmitting (NAT) to 10.18.130.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060
Via: SIP/2.0/TLS 192.168.88.252:5062;received=Client public ip;branch=z9hG4bK0bbe1f7d27257bba9;rport=5062
Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes>
Record-Route: <sip:PUBLIC_KAMAILIO_IP:5081;transport=tls;r2=on;lr=on;ftag=a185d974ec;nat=yes>
From: "Slava Bendersky" <sips:10101@networklab.ca>;tag=a185d974ec
To: <sips:10101@networklab.ca>;tag=as00757d3e
Call-ID: b08adb1ad1804a83
CSeq: 236711034 SUBSCRIBE
Server: FPBX-2.11.0(11.15.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:10101@10.18.130.50:5060>;expires=3600
Content-Length: 0
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "Slava Bendersky" <volga629(a)networklab.ca>
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Friday, February 27, 2015 6:42:33 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello,
I asked for the wrong command, as the bt full was already sent before -- I wanted the output from gdb for:
p *tcpconn
Daniel
On 27/02/15 04:10, Slava Bendersky wrote:
Hello Daniel,
Here bt full from back trace.
http://fpaste.org/191207/50064491/
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com , "Slava Bendersky" <volga629(a)networklab.ca>
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Thursday, February 26, 2015 9:56:48 PM
Subject: Re: [SR-Users] kamailio asterisk
Hello Daniel,
I tried $rz option on top of request route and that where I see wrong request uri like sip:sips : . And as far I can see it happenes only for SUBSCRIBE INVITE and NOTIFY.
if($rz=="sips") {
$ru = "sip" + $(ru{s.substr,4,0});
}
Slava.
Sent from mobile device typos are expected.
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Sent: Feb 25, 2015 1:04 PM
To: Slava Bendersky
Cc: sr-users
Subject: Re: [SR-Users] kamailio asterisk
Hello,
On 25/02/15 17:19, Slava Bendersky wrote:
BQ_BEGIN
Hello Daniel,
substr you suggested didn't worked.
See my previous email.
your previous email didn't say anything about the results. That's why I asked. Be sure you don't have those spaces that are in the email you wrote. Also, I had more parenthesis in the parameter of the subst_uri().
Or you can try the alternative with:
if($rz=="sips") {
$ru = "sip" + $(ru{s.substr,4,0});
}
I asked for more details from the backtrace to confirm that what I found is the cause for the crash in this case -- see one of my previous emails from today.
Cheers,
Daniel
BQ_END
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.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
Hi,
while testing IPv6 with customers, we fell over quite a few cases, where
customers aren't reachable on inbound calls most of the time. And digging
into this, we found the home router firewall as the cause for those
problems.
Normally, you would think, all the NAT problems cease when switching to
IPv6. But actually, right now I don't know how to fix that problem.
In IPv4 NAT scenarios, we would flag the customer during the registration,
and Kamailio would send NAT pings (those 4 bytes of UDP junk) every few
seconds to keep the firewall in the NAT router open. And that worked pretty
great.
Now we have IPv6. We don't have NAT. But we still have a home router in
front of SIP devices, with a firewall. And this firewall will allow
outbound traffic. But after a few seconds it won't allow incoming
connections anymore. And the routers I have seen so far don't have a
configurable firewall where you could allow inbound traffic from our server.
Unfortunately, only our load balancer is IPv6, our registrar is still IPv4
only. And the loadbalancer doesn't know anything about registrations and
which customer needs an IPv6 keepalive.
Does anyone have a hint, how to keep the IPv6 registrations alive? Thanks
in advance.
Best Regards,
Sebastian
Hi Klaus, did you find a solution to this problem? I'm working with SIP-I and having the same issue with Binary encoding. I need to encode a 0x00 value and when the config script gets to that point, it breaks. If I encode any other hex value it's fine. This is the config part I'm using:
route[INSERT_ACM]{
if(has_body()){
# Save the SDP body for future reference
$var(sdp_body) = $rb;
replace_body_atonce("^.+$", "");
remove_hf("Content-Type");
append_hf("MIME-Version: 1.0\r\n", "Content-Length");
append_hf("Content-Type: multipart/mixed; boundary=unique-boundary-1\r\n", "Allow");
replace_body_atonce("^.+$", "--unique-boundary-1\r\nContent-Disposition: signal; handling=optional\r\nContent-Type: application/isup; version=itu-t92+\r\n\r\n'\x06''\x02''\x04''\x01''\x29''\x01''\x01''\x00'\r\n-unique-boundary-1\r\n");
msg_apply_changes();
replace_body_all("\047", "");
append_body_part("$var(sdp_body)","application/sdp");
}
return;
}
I tried with the escape character \0 which should be translated to ASCII hex value '00' but it's parsed as ASCII literal values \0...
Thanks!
Federico
Federico San Martín
e-mail : fsanmartin(a)telecentro.net.ar
On 9/17/13 4:34 PM, Daniel wrote:
> Hello,
>
> most of the functions for pseudo-variables work with string types, that
> because a script variable can have only integer or string values. Even
> the length can be higher, these functions truncate at first 0. But
> internally all should be kept, just not passed to config variables.
>
> Have you tried exec_msg(), that should pass entire msg?
>
> Otherwise, I presume there is need for some C coding to make binary
> values work with variables in config.
>
> Cheers,
> Daniel
On 9/17/13 4:03 PM, Klaus Feichtinger wrote:
> Hello,
>
> I have an update to the side effects of receiving an INVITE message that
> contains a MIME body with binary data:
> - pseudo variables are affected, too
> - the message buffer ($mb) does not include the whole message; it is
> ending with the nul character
> - as the message buffer does not include all data, the modification
> cannot be done with an external script / program (e.g. perl, bash script)
>
> The behaviour was tested with kamailio 4.0.3, too - no difference!
>
> In general, the whole message is stored in a buffer, but parts of it are
> inaccessible for parsing / text functions.
>
> Any idea, what could cause this problem?
>
> regards,
> Klaus
>
>> Hello,
>>
>> I have a problem with Kamailio version 3.2.4 (tested also with 3.3.5)
>> regarding "binary data" in message bodies. According RFC3261 it is
>> explicitly allowed using binary data within MIME bodies:
>>
>> RFC 3261 section 7.4.1: SIP messages MAY contain binary bodies or body
>> parts. When no explicit charset parameter is provided by the sender,
>> media subtypes of the "text" type are defined to have a default charset
>> value of "UTF-8".
>>
>> However, the Kamailio function "Regular Expression Transformation"
>> (re.subst), which is based on the transformation class (exported by the
>> textops module) is causing problems in our customer system. In regular
>> scenarios, Kamailio is receiving SIP INVITE messages with a MIME body,
>> which is containing following parts:
>> - application/sdp (standard conform)
>> - application/x-q931 (Cisco proprietary with BINARY data!)
>> - application/gtd (Cisco proprietary with ASCII strings)
>>
>> The content of the "x-q931" and "gtd" body parts is depending on (UUS1)
>> data that were received on the ISDN line. Kamailio has to forward
>> User-User specific information from the ISDN line to SIP user agents (in
>> form of the "User-to-User" header field). The content of these UUS1 data
>> may contain also byte values "00", which is legitimate. In general,
>> Kamailio is in every INVITE message searching specific content in the
>> last body (application/gtd) and copying this content to a config
>> variable. As soon as the x-q931 body contains nul values (0x00 in binary
>> format), the parser stops at this position and cannot parse the rest of
>> the message. Therefore, I am missing the information that should be
>> copied to the SIP header field, as the parser stops before the end of
>> the message body!
>>
>> As long as the message body does not contain 0x00, it is working fine!
>>
>> My question is:
>> - is this a bug in Kamailio parsing functions?
>> - is this a design issue of Kamailio text parsers (as binary data are
>> allowed acc. RFC3261) - does anybody know a solution for this problem?
>> This "bug" is causing big troubles....
>>
>> Thanks in advance,
>> Klaus Feichtinger
>>
>>
>> P.S. trace data (1)...(3) of my problem
>>
>> (1) exemplary content of the x-q931 body
>> ======================Sending: Binary Message Body
>> Sep 13 10:50:19: Content-Type: application/x-q931
>> 08 01 B4 05 04 03 80 90 A2 18 01 89 1E 02 82 88 6C 05 B1 35 30 30 34 70
>> 05 B1 35 30 30 31 7D 02 91 81 7E 18 04 1D 42 75 25 92 43 31 62 94 00 00
>> 2C 68 20 64 00 62 F2 10 41 B9 D7 BD 0D 0A
>>
>> (2) SIP INVITE message (debugger view)
>> =============================INVITE sip:115101@<ipv4>:5060 SIP/2.0 Via:
>> SIP/2.0/UDP <ipv4>:5060;branch=z9hG4bK171164E
>> From: <sip:1101015004@<ipv4>>;tag=29E16410-2541
>> To: <sip:115101@<ipv4>>
>> Call-ID: 1DBCDACB-1B9911E3-8992FF70-D2BED293@<ipv4>
>> Supported: timer,resource-priority,replaces,sdp-anat
>> Min-SE: 90
>> CSeq: 101 INVITE
>> Max-Forwards: 70
>> Contact: <sip:1101015004@<ipv4>:5060>
>> Expires: 1800
>> P-Asserted-Identity: <sip:1101015004@<ipv4>>
>> Content-Type: multipart/mixed;boundary=uniqueBoundary
>> Mime-Version: 1.0
>> Content-Length: 820
>>
>> --uniqueBoundary
>> Content-Type: application/sdp
>> Content-Disposition: session;handling=required
>>
>> v=0
>> o=CiscoSystemsSIP-GW-UserAgent 1770 5294 IN IP4 <ipv4>
>> s=SIP Call
>> c=IN IP4 <ipv4>
>> t=0 0
>> m=audio 16384 RTP/AVP 8 0
>> c=IN IP4 <ipv4>
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>>
>> --uniqueBoundary
>> Content-Type: application/x-q931
>> Content-Disposition: signal;handling=optional
>> Content-Length: 62
>> 4" l15004p15001}~Bu%C1b..,h d.brA9W
>> --uniqueBoundary
>> Content-Type: application/gtd
>> Content-Disposition: signal;handling=optional
>>
>> IAM,
>> PRN,isdn*,,,
>> USI,rate,c,s,c,1
>> USI,lay1,ulaw
>> TMR,00
>> CPN,08,,1,5001
>> CGN,08,,1,,,5004
>> UUS,3,1d427525924331629400002c6820640062f21041b9d7bd
>> CPC,09
>> FCI,,,,,,,y,
>> GCI,185ef1291b9911e381d500270dff3fa0
>>
>> --uniqueBoundary--
>>
>> (3) config excerpt
>> ============if (has_body("multipart/mixed")) {
>> if (filter_body("application/sdp")) {
>> remove_hf("Mime-Version");
>> remove_hf("Content-Type");
>> append_hf("Content-Type: application/sdp\r\n");
>> } else {
>> xlog("L_WARN", " <route> message body part 'application/sdp' not
>> found\n");
>> }
>> $var(UUS)=$(rb{re.subst,/^(.*)UUS,.,([a-z0-9,]*)..[A-Z][A-Z][A-Z],(.*)/\2/s});
>> append_hf("User-to-User: $var(UUS)\r\n", "CSeq");
>>
>> if
>> !(subst_body('/Content-Disposition..session.handling=required....//s'))
>> {
>> xlog("L_WARN", " <route> substituting Content-Disposition
>> FAILED!!! \n");
>> }
>> msg_apply_changes();
>> if (search_body("a=[a-z]+:.+[\r\n]{4}$")) {
>> #!ifdef WITH_XLOGDEBUG
>> xlog("L_INFO", " <route> double CRLF at the end of the message
>> body detected - is reduced to one now. \n"); #!endif
>> $var(sdp) = $(rb{s.striptail,2});
>> set_body("$var(sdp)", "application/sdp");
>> }
>> }
>>
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at 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 at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013
> - more details about Kamailio trainings at http://www.asipto.com -
Hello Daniel,
Here bt full from back trace.
http://fpaste.org/191207/50064491/
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com, "Slava Bendersky" <volga629(a)networklab.ca>
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Thursday, February 26, 2015 9:56:48 PM
Subject: Re: [SR-Users] kamailio asterisk
Hello Daniel,
I tried $rz option on top of request route and that where I see wrong request uri like sip:sips: . And as far I can see it happenes only for SUBSCRIBE INVITE and NOTIFY.
if($rz=="sips") {
$ru = "sip" + $(ru{s.substr,4,0});
}
Slava.
Sent from mobile device typos are expected.
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Sent: Feb 25, 2015 1:04 PM
To: Slava Bendersky
Cc: sr-users
Subject: Re: [SR-Users] kamailio asterisk
Hello,
On 25/02/15 17:19, Slava Bendersky wrote:
Hello Daniel,
substr you suggested didn't worked.
See my previous email.
your previous email didn't say anything about the results. That's why I asked. Be sure you don't have those spaces that are in the email you wrote. Also, I had more parenthesis in the parameter of the subst_uri().
Or you can try the alternative with:
if($rz=="sips") {
$ru = "sip" + $(ru{s.substr,4,0});
}
I asked for more details from the backtrace to confirm that what I found is the cause for the crash in this case -- see one of my previous emails from today.
Cheers,
Daniel
BQ_BEGIN
Feb 25 09:12:33 canlvprx01 kamailio: 11(4104) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=506 a=25 n=subst_uri
Feb 25 09:12:33 canlvprx01 kamailio: 11(4104) DEBUG: <core> [re.c:448]: subst_run(): subst_run: running. r=1
Feb 25 09:12:33 canlvprx01 kamailio: 11(4104) DEBUG: <core> [re.c:517]: subst_str(): subst_str: no match
but I changed to
subst("/sips:/ sip:/g ");
and log says that it match, but still ask for sips and then crash
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=507 a=25 n=subst
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: <core> [re.c:448]: subst_run(): subst_run: running. r=0
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: <core> [re.c:469]: subst_run(): subst_run: matched (229, 5): [sips:]
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: <core> [re.c:448]: subst_run(): subst_run: running. r=0
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: <core> [re.c:469]: subst_run(): subst_run: matched (299, 5): [sips:]
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: <core> [re.c:448]: subst_run(): subst_run: running. r=0
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: <core> [re.c:469]: subst_run(): subst_run: matched (364, 5): [sips:]
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: <core> [re.c:448]: subst_run(): subst_run: running. r=0
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: <core> [re.c:469]: subst_run(): subst_run: matched (538, 5): [sips:]
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: <core> [re.c:448]: subst_run(): subst_run: running. r=1
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: textops [textops.c:695]: subst_f(): textops: replacing at offset 266 [sips:] with [sip:]
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: textops [textops.c:695]: subst_f(): textops: replacing at offset 336 [sips:] with [sip:]
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: textops [textops.c:695]: subst_f(): textops: replacing at offset 401 [sips:] with [sip:]
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: textops [textops.c:695]: subst_f(): textops: replacing at offset 575 [sips:] with [sip:]
Feb 25 11:22:24 canlvprx01 kernel: [87678.490671] kamailio[4774]: segfault at 88 ip 00000000004bd27c sp 00007fff9a126ea0 error 4 in kamailio[400000+3b9000]
Feb 25 11:22:24 canlvprx01 kamailio: 11(4770) DEBUG: textops [textops.c:711]: subst_f(): lst was 0x7fd875e0dd98
That why I asked about this bug report, it close because of limitation in openser/kamailio.
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, February 25, 2015 10:57:29 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello,
that's from 2007, closed as invalid -- that was v1.2.
Again, as in my previous email, you haven't described what is the state after using the subst as I suggested previously.
Cheers,
Daniel
On 25/02/15 15:47, Slava Bendersky wrote:
BQ_BEGIN
Hello Daniel,
I found this bug report
http://sourceforge.net/p/openser/bugs/226/
Is this still valid ?
Slava.
From: "Slava Bendersky" <volga629(a)networklab.ca>
To: miconda(a)gmail.com
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, February 25, 2015 9:11:54 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello Everyone,
Feb 25 09:12:33 canlvprx01 kamailio: 11(4104) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=506 a=25 n=subst_uri
Feb 25 09:12:33 canlvprx01 kamailio: 11(4104) DEBUG: <core> [re.c:448]: subst_run(): subst_run: running. r=1
Feb 25 09:12:33 canlvprx01 kamailio: 11(4104) DEBUG: <core> [re.c:517]: subst_str(): subst_str: no match
I tried something like
if (proto==TLS)
{
#subst_uri('/^sips.+)$/ sip:\1/ ');
subst("/sips:/ sip:/g ");
}
but there some cases where client use ;transport=TLS which need remove too.
Slava
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, February 25, 2015 5:44:53 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello,
change the r-uri to use sip instead of sips.
You can try with:
subst_uri('/^sips:(.+)$/ sip:\1/ ');
Cheers,
Daniel!
On 25/02/15 01:14, Slava Bendersky wrote:
BQ_BEGIN
Hello Everyone,
I wonder in my case why kamailio is not bridging between TLS and UDP ? Is there additional configuration required ?
Slava.
From: "Slava Bendersky" <volga629(a)networklab.ca>
To: miconda(a)gmail.com
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Tuesday, February 24, 2015 11:42:02 AM
Subject: Re: [SR-Users] kamailio asterisk
Hello Everyone,
Here link to core dump bt full.
http://fpaste.org/189799/79603014/
password the same
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "volga629" <volga629(a)skillsearch.ca> , "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Monday, February 23, 2015 3:56:48 AM
Subject: Re: [SR-Users] kamailio asterisk
You don't need to run kamailio through gdb. If it crashes, then you get
a corefile -- in the logs it says the name of the file. It is usually
located in / or in the path you gave to -w parameter.
After you reproduced the crash, locate the corefile and run gdb like:
gdb /path/to/kamailio /path/to/corefile
bt full
The /path/to/kamailio should be /usr/sbin/kamailio if you installed from
rpms.
If you have improvements to init or sysconfig kamailio file, send a
patch and we will include the changes in kamailio repository.
Cheers,
Daniel
On 23/02/15 01:00, Slava Bendersky wrote:
> Hello Everyone,
> I upgraded to 4.2.3 version, but crash still there. What is my options to meet case like this.
>
> Client TLS -------> KAMAILIO Proxy ------ UDP/TCP ----> asterisk
> |
> |
> Public Private
>
> I will try try run gdb to get backtrace on crash.
> Also here link for Fedora 21 server rpms. There are couple small fixes for init file and sysconfig/kamailio
>
> http://ftpsrv01.networklab.ca/fedora/21/RPMS/x86_64/
>
> Slava.
>
> Sent from mobile device typos are expected.
>
> From: Daniel-Constantin Mierla <miconda(a)gmail.com>
> Sent: Feb 22, 2015 5:34 PM
> To: Slava Bendersky <volga629(a)networklab.ca> ;sr-users
> Subject: Re: [SR-Users] kamailio asterisk
>
>
>
> Hello,
>
> looking at the logs, the process routing the register is forwarding it,
> by opening a tls connection -- that is because the uri has sips as schema.
>
> The crash is reported in another process that doesn't print much logs
> messages. As Olle suggested, can you get the backtrace with gdb from the
> core file? That will help to see where the crash happened.
>
> gdb /path/to/kamailio /path/to/corefile
> bt full
>
> And again, it would be good to upgrade to 4.2.3 -- it is same config and
> database, just install new version and restart. In this way we rule out
> issues that were fixed already, avoiding to spend time on something fixed.
>
> Cheers,
> Daniel
>
> On 20/02/15 15:03, Slava Bendersky wrote:
>> Hello Everyone,
>> Thank you for reply,
>> On client I configured user @ domain.org and proxy point to kamailio
>> Here 1 debug where on client after doamin.org port is left configured
>> to 5061
>>
>> http://fpaste.org/188145/44047614/
>>
>> Second debug where port set to 0 and kamailio tries resolve and crashed
>>
>> http://fpaste.org/188148/24440702/
>>
>> Here config file
>>
>> http://fpaste.org/188149/24440841/
>>
>>
>> Thank you,
>> Slava.
>>
>>
>> ------------------------------------------------------------------------
>> *From: *"Olle E. Johansson" <oej(a)edvina.net>
>> *To: *"Daniel Constantin Mierla" <miconda(a)gmail.com> , "sr-users"
>> <sr-users(a)lists.sip-router.org>
>> *Sent: *Thursday, February 19, 2015 4:34:04 AM
>> *Subject: *Re: [SR-Users] kamailio asterisk
>>
>> We also need to check the core file from the crash.
>> /O
>> On 19 Feb 2015, at 09:30, Daniel-Constantin Mierla < miconda(a)gmail.com
>> <mailto:miconda@gmail.com> > wrote:
>>
>> Hello,
>>
>> can you send the REGISTER request received by kamailio and your
>> config to me?
>>
>> As you receive it over TLS, you can get the register by adding the
>> next line in kamailio.cfg at the beginning of request_route:
>>
>> xlog("received request: [[$mb]]\n");
>>
>> I will like to double check if the issue is still present.
>>
>> You should upgrade to 4.2.3, because it is the latest stable, you
>> have 4.2.1 and there were many fixes meanwhile.
>>
>> If you preserve sips as uri schema, then you force tls further for
>> forwarding. You should change that to sip:domain ...
>>
>> Cheers,
>> Daniel
>>
>> On 18/02/15 00:37, Slava Bendersky wrote:
>>
>> Hello Everyone,
>> I have standard case where kamailio play role of proxy for
>> asterisk servers.
>> Kamailio configured use TLS transport on public side and on
>> private side UDP 5060.
>> When client (SIP soft phone) connect to TLS socket everything
>> goes well until kamailio trying forward request. Kamailio
>> tries DNS resolve tls transport srv records instead of udp
>> then it just crashed when no tls configured on private side of
>> kamailio.
>>
>> Do I need manually fix sips in URI ? Or some different miss
>> configuration ?
>>
>>
>> [root@canlvprx01 kamailio]# rpm -qa | grep kamail
>> kamailio-carrierroute-4.2.1-4.2.fc21.x86_64
>> kamailio-mysql-4.2.1-4.2.fc21.x86_64
>> kamailio-outbound-4.2.1-4.2.fc21.x86_64
>> kamailio-4.2.1-4.2.fc21.x86_64
>> kamailio-tls-4.2.1-4.2.fc21.x86_64
>>
>>
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [parser/msg_parser.c:625]: parse_msg(): method: <REGISTER>
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [parser/msg_parser.c:627]: parse_msg(): uri:
>> <sips:domain.org> ---> Client come with TLS transport
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0>
>>
>>
>>
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [socket_info.c:583]: grep_sock_info(): grep_sock_info -
>> checking if host==us: 13==12 && [domain.org
>> <http://domain.org> ] == [10.18.130.46 <callto:10.18.130.46> ]
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [socket_info.c:587]: grep_sock_info(): grep_sock_info -
>> checking if port 5060 (advertise 0) matches port 5060
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [socket_info.c:583]: grep_sock_info(): grep_sock_info -
>> checking if host==us: 13==11 && [domain.org
>> <http://domain.org> ] == [67.34.12.56]
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [socket_info.c:587]: grep_sock_info(): grep_sock_info -
>> checking if port 5081 (advertise 0) matches port 5060
>> Feb 17 11:13:49 canlvprx01 kernel: [4130713.518667]
>> kamailio[22484]: segfault at 88 ip 00000000004bd30c sp
>> 00007fffa2f73a20 error 4 in kamailio[400000+3b8000]
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [forward.c:448]: check_self(): check_self: host != me
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: ***
>> cfgtrace:request_route=[SIPOUT]
>> c=[/etc/kamailio/kamailio-asterisk.cfg] l=850 a=25 n=append_hf
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: ***
>> cfgtrace:request_route=[SIPOUT]
>> c=[/etc/kamailio/kamailio-asterisk.cfg] l=851 a=5 n=route
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: ***
>> cfgtrace:request_route=[RELAY]
>> c=[/etc/kamailio/kamailio-asterisk.cfg] l=567 a=16 n=if
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: ***
>> cfgtrace:request_route=[RELAY]
>> c=[/etc/kamailio/kamailio-asterisk.cfg] l=563 a=25 n=is_method
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: ***
>> cfgtrace:request_route=[RELAY]
>> c=[/etc/kamailio/kamailio-asterisk.cfg] l=571 a=16 n=if
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: ***
>> cfgtrace:request_route=[RELAY]
>> c=[/etc/kamailio/kamailio-asterisk.cfg] l=567 a=25 n=is_method
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: ***
>> cfgtrace:request_route=[RELAY]
>> c=[/etc/kamailio/kamailio-asterisk.cfg] l=574 a=16 n=if
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) ERROR: ***
>> cfgtrace:request_route=[RELAY]
>> c=[/etc/kamailio/kamailio-asterisk.cfg] l=571 a=24 n=t_relay
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm
>> [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=1 ,
>> global msg id=1 , T on entrance=(nil)
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm
>> [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start
>> searching: hash=48550, isACK=0
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm
>> [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction
>> matching failed
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm
>> [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request:
>> no transaction found
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm
>> [t_hooks.c:380]: run_reqin_callbacks_internal(): DBG:
>> trans=0x7f598a9ced40, callback type 1, id 0 entered
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm
>> [t_hooks.c:380]: run_reqin_callbacks_internal(): DBG:
>> trans=0x7f598a9ced40, callback type 1, id 0 entered
>> Feb 17 11:13:49 canlvprx01 kernel: kamailio[22484]: segfault
>> at 88 ip 00000000004bd30c sp 00007fffa2f73a20 error 4 in
>> kamailio[400000+3b8000]
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 calculated:
>> 0475e0d0dd9778e889618cb724403b4d
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [dns_cache.c:566]: _dns_hash_find():
>> dns_hash_find(_sips._tcp.networklab.ca
>> <http://tcp.networklab.ca> (24), 33), h=646
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [resolve.c:967]: get_record(): get_record: skipping 1 NS
>> (p=0xa1f556, end=0xa1f588)
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [resolve.c:983]: get_record(): get_record: parsing 2 ARs
>> (p=0xa1f568, end=0xa1f588)
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [dns_cache.c:1772]: dns_get_related():
>> dns_get_related(0x7f598a9e89b0 (_sips._tcp.domain.org
>> <http://tcp.domain.org> , 33), 33, *0x7f5995bd55e0) (0)
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [dns_cache.c:869]: dns_cache_add_unsafe(): dns_cache_add:
>> adding _sips._tcp.domain.org <http://tcp.domain.org> (24) 33
>> (flags=0) at 646
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [dns_cache.c:869]: dns_cache_add_unsafe(): dns_cache_add:
>> adding camsgsrv02.domain.org
>> <http://camsgsrv02.domain.org> (24) 1 (flags=0) at 967
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [dns_cache.c:566]: _dns_hash_find():
>> dns_hash_find(camsgsrv02.domain.org
>> <http://camsgsrv02.domain.org> (24), 1), h=967
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [msg_translator.c:2871]: create_via_hf(): create_via_hf: id
>> added: <;i=1>, rcv proto=3
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [tcp_main.c:1818]: tcp_send(): tcp_send: no open tcp
>> connection found, opening new one
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection:
>> 10.18.130.50 <callto:10.18.130.50>
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 5061,
>> type 3
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 3263:0:0, 2
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) WARNING: <core>
>> [tcp_main.c:1221]: tcp_do_connect(): 10.18.130.50
>> <callto:10.18.130.50> :5061: could not find corresponding
>> listening socket for 10.18.130.46 <callto:10.18.130.46> , using
>> default...
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tls
>> [tls_server.c:184]: tls_complete_init(): Using TLS domain
>> TLSc<default>
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tls
>> [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake
>> started
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: <core>
>> [tcp_main.c:2697]: tcpconn_1st_send(): pending write on new
>> connection 0x7f598a9d4678 (-1/129 bytes written)
>> Feb 17 11:13:49 canlvprx01 kamailio: 15(22484) DEBUG: <core>
>> [tcp_main.c:3565]: handle_ser_child(): handle_ser_child: read
>> response= 7f598a9d4678, 5, fd 31 from 11 (22480)
>> Feb 17 11:13:49 canlvprx01 kamailio: 15(22484) DEBUG: <core>
>> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9daf00,
>> 31, 2, 0x7f598a9d4678), fd_no=19
>> Feb 17 11:13:49 canlvprx01 kamailio: 11(22480) DEBUG: tm
>> [t_funcs.c:394]: t_relay_to(): SER: new transaction fwd'ed
>>
>>
>>
>> Thank you Slava.
>>
>>
>> _______________________________________________
>> 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio World Conference, May 27-29, 2015
>> Berlin, Germany - http://www.kamailioworld.com
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>> list
>> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> _______________________________________________
>> 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.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
BQ_END
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.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
BQ_END
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
BQ_END
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Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com