Re: [SR-Users] Phone number formats per country

[Fred Posner]

Normally I run this against a carrier rate sheet, using the description.
Both Twilio and Flowroute have decent download-able sheets with prefix
<-> country/mobile description.

--fred

On 01/18/2017 09:21 AM, Daniel-Constantin Mierla wrote:
> Hello,
> 
> slightly off-topic, but related to voip -- does anyone have a link that
> can quickly share to a site with up to date details about phone
> numbering plans for as many country as possible?
> 
> Ideally to include the toll and premium number prefixes, the split
> between mobile and fixed lines if it is the case.
> 
> Cheers,
> Daniel
> 

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Re: [SR-Users] Kamailio not processing SIP TCP

[Fred Posner]

What happens when you try:

modparam("sipcapture", "hep_capture_on", 1)

On 01/13/2017 10:33 AM, JR Richardson wrote:
> Iptables is not blocking, but it was worth a check.
> 
> Thanks.
> 
> JR
> 
> 
> I assume you have ruled out firewall? It's something that can nab even
> experienced people:
> 
> # iptables -Ln
> 
> -- Alex
> 
> On Thu, Jan 12, 2017 at 03:25:27PM -0600, JR Richardson wrote:
> 
>> Hi All,
>>
>> Just enabled SIP TCP on a homer capture server, I can see the SIP TCP
>> Sessions on the server with ngrep, just like all the UDP traffic. I
>> have Kamailio listening on TCP ports but its not capturing any TCP
>> traffic.
>>
>> kamailio.cfg:
>>
>> #disable_tcp=yes
>> listen=tcp:10.99.99.99:5060#monitor port
>> listen=udp:10.99.99.99:5060   #monitor port
>>
>> loadmodule "pv.so"
>> loadmodule "db_mysql.so"
>> loadmodule "sipcapture.so"
>> loadmodule "textops.so"
>> loadmodule "rtimer.so"
>> loadmodule "xlog.so"
>> loadmodule "sqlops.so"
>> loadmodule "htable.so"
>> loadmodule "sl.so"
>> loadmodule "siputils.so"
>>
>>
>> modparam("sipcapture", "capture_on", 1)
>> modparam("sipcapture", "hep_capture_on", 0)
>> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070")
>> modparam("sipcapture", "raw_interface", "eth1")
>> modparam("sipcapture", "raw_ipip_capture_on", 0)
>> modparam("sipcapture", "table_name", "sip_capture")
>> modparam("sipcapture", "raw_sock_children", 4)
>> modparam("sipcapture", "db_insert_mode", 0)
>> modparam("sipcapture", "raw_moni_capture_on", 1)
>> modparam("sipcapture", "promiscious_on", 1)
>> modparam("sipcapture", "raw_moni_bpf_on", 1)
>> modparam("sipcapture", "capture_node", "homer02")
>> modparam("sipcapture", "authorization_column", "authorization")
>>
>>
>> ## logging all INVITES top of the [route] block
>> if (is_method("INVITE|REGISTER")) {
>> xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\"
>> from \"$si\"\n");
>>
>> Logging reports all SIP UDP traffic to logs fine, but no TCP traffic.
>>
>> root@homer02:~# netstat -al
>> Active Internet connections (servers and established)
>> Proto Recv-Q Send-Q Local Address   Foreign Address State
>> tcp0  0 homer02.me.com:sip*:* LISTEN
>>
>>
>> I don't think this is a homer issue because logging invites is prior
>> to any homer processing. I'm thinking this is something simple I'm
>> overlooking, any help is much appreciated.
>>
>> Thanks.
>>
>> JR
>> -- 
>> JR Richardson
>> Engineering for the Masses
>> Chasing the Azeotrope
>>


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Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.

[Fred Posner]

> listen=udp:MY_IP_ADDR:5060 advertise MY_PUBLICIP_ADDR:5060

That statement does not exist anywhere in the files you sent.

--fred

On 12/29/2016 11:19 AM, Pranathi Venkatayogi wrote:
> Yes. I defined advertised address and even used listen with advertise as 
> below. Still Kamailio does not send publicip in record route header.
> listen=udp:MY_IP_ADDR:5060 advertise MY_PUBLICIP_ADDR:5060
> 
> 
> -Original Message-
> From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
> Daniel Grotti
> Sent: Thursday, December 29, 2016 6:31 AM
> To: sr-users@lists.sip-router.org
> Subject: Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised 
> public IP.
> 
> Hi,
> not sure if I understood it right but, have you defined the 
> advertised_address ? That should be used in Via and RR as well:
> 
> https://www.kamailio.org/wiki/cookbooks/4.4.x/core#advertised_address
> 
> 
> Daniel
> 
> 
> On 12/29/2016 12:09 AM, Pranathi Venkatayogi wrote:
>> I implemented full NAT logic as per the sample config. Still unable to 
>> resolve the issue.
>>
>> How do I let Kamailio change record_route header to use public ip address?
>>
>>
>>
>> Please help!!!
>>
>>
>>
>> (attached are latest scripts)
>>
>>
>>
>> *From:* Pranathi Venkatayogi
>> *Sent:* Wednesday, December 28, 2016 12:39 PM
>> *To:* 'sr-users@lists.sip-router.org' 
>> *Subject:* Kamailio behind NAT, ACK to private IP not advertised public IP.
>>
>>
>>
>> Hi,
>>
>>   I am encountering the same problem described in google groups 
>> .
>> However I dint not find any resolution hence writing again.
>>
>>
>>
>>   200 OK sent from the server has private Ip in its record route. As 
>> you see below, though the message is received on public IP 
>> (63.149.103.72) , the record route is set to private IP 
>> (172.31.211.31)
>>
>>   I used listen with advertise of public IP, it did not work. Please 
>> find attached the config I am using.
>>
>>
>>
>>   How do I change it send public ip only when talking to external world.
>>
>>   Can someone point to me clear documentation how to configure 
>> Kamailio for NAT traversal.
>>
>>
>>
>> *The following message is sent from Kamailio behind NAT to the public
>> computer.*
>>
>> 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123,
>> +0:08:42.690309
>>
>> 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912
>>
>> SIP/2.0 200 OK
>>
>> Via: SIP/2.0/TLS
>> 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e38
>> 1a96979945bd969989ffe9dca3a9;alias
>>
>> Record-Route: > >
>>
>> Call-ID: eb8670eec4354acdb69fd26f5625b75c
>>
>> From: "cust1"
>> ;tag=2f25d2ae690747c48c874
>> e0b415ca03c
>>
>> To:
>> ;tag=1c33ad41f6f44cae8ae
>> 8e060f30fe119
>>
>> CSeq: 4665 INVITE
>>
>> Server: Blink 3.0.0 (Windows)
>>
>> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, 
>> MESSAGE, REFER
>>
>> Contact: 
>>
>> Supported: 100rel, replaces, norefersub, gruu
>>
>> Content-Type: application/sdp
>>
>> Content-Length:   355
>>
>> v=0
>>
>> o=- 3691844303 3691844304 IN IP4 10.0.27.108
>>
>> s=Blink 3.0.0 (Windows)
>>
>> t=0 0
>>
>> m=message 2855 TCP/TLS/MSRP *
>>
>> c=IN IP4 10.0.27.108
>>
>> a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp
>>
>> a=accept-types:message/cpim text/* image/* 
>> application/im-iscomposing+xml
>>
>> a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml
>>
>> a=setup:active
>>
>> --
>>
>>
>>
>> *The following is the ACK sent by public computer in reply to the 
>> above message. Note this message never reaches the Kamailio server as 
>> it is sent to private IP.*
>>
>> 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124,
>> +0:08:42.690309
>>
>> 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061
>>
>> ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0
>>
>> Via: SIP/2.0/TLS
>> 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;
>> alias
>>
>> Max-Forwards: 70
>>
>> From: "cust1"
>> ;tag=2f25d2ae690747c48c874
>> e0b415ca03c
>>
>> To:
>> ;tag=1c33ad41f6f44cae8ae
>> 8e060f30fe119
>>
>> Call-ID: eb8670eec4354acdb69fd26f5625b75c
>>
>> CSeq: 4665 ACK
>>
>> Route: 
>>
>> User-Agent: Blink 3.0.0 (Windows)
>>
>> Content-Length:  0
>>
>>
>>
>> Thanks,
>>
>> *Pranathi Venkatayogi*
>>
>> /System Developer II/
>>
>> (520) 745-9447 x4466
>>
>> www.cyracom.com 
>>
>>
>>
>> Join us:   cid:image001.gif@01D1C9C9.C4E8B6D0
>>  
>> cid:image002.gif@01D1C9C9.C4E8B6D0

Re: [SR-Users] Asterisk Security Advisory (AST-2016-009)

[Fred Posner]

Thank you for the post-- definitely appreciate you sharing it on this list.

--fred

On 12/8/16 6:02 PM, Matthew Jordan wrote:

Hey all -

The Asterisk project just released a security advisory for a security
vulnerability in which Asterisk using chan_sip with a proxy can allow for
unauthenticated calls. This affects all supported versions of Asterisk (11,
13, 14). Since that may be relevant to those on this mailing list who are
not also on the asterisk-users mailing list, I thought it prudent to
mention it here as well.

A description of the vulnerability follows:

Description  The chan_sip channel driver has a liberal definition for
 whitespace when attempting to strip the content between a
 SIP header name and a colon character. Rather than
 following RFC 3261 and stripping only spaces and horizontal
 tabs, Asterisk treats any non-printable ASCII character as
 if it were whitespace. This means that headers such as

 Contact\x01:

 will be seen as a valid Contact header.

 This mostly does not pose a problem until Asterisk is
 placed in tandem with an authenticating SIP proxy. In such
 a case, a crafty combination of valid and invalid To
 headers can cause a proxy to allow an INVITE request into
 Asterisk without authentication since it believes the
 request is an in-dialog request. However, because of the
 bug described above, the request will look like an
 out-of-dialog request to Asterisk. Asterisk will then
 process the request as a new call. The result is that
 Asterisk can process calls from unvetted sources without
 any authentication.

 If you do not use a proxy for authentication, then this
 issue does not affect you.

 If your proxy is dialog-aware (meaning that the proxy keeps
 track of what dialogs are currently valid), then this issue
 does not affect you.

 If you use chan_pjsip instead of chan_sip, then this issue
 does not affect you.


The announcement can be seen here:

http://lists.digium.com/pipermail/asterisk-announce/2016-December/000662.html

Thanks again to Walter Doekes for reporting the vulnerability and providing
the patch to fix it.

Matt



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Re: [SR-Users] Size of passwords fields in db

[Fred Posner]

On 10/06/2016 05:43 AM, Daniel-Constantin Mierla wrote:
> On 05/10/16 16:35, Fred Posner wrote:
>> On 10/05/2016 10:25 AM, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> writing here to decide on a topic opened by pull request 779:
>>>
>>>   - https://github.com/kamailio/kamailio/pull/779
>>>
>>> what would be a fair size for db column storing a password that one
>>> would like to have for proper security?
>>>
>>> I would like to make it consistent over all tables that have a password
>>> column by defining a xml entity for the size of these columns. The pull
>>> request suggests 64 chars, has anyone other opinions on making it larger
>>> or smaller?
>>>
>>> If they are defined varchar, then should not be a problem of allocated
>>> size, so we can go with 128 if someone feels it worth doing larger now
>>> so we don't have to change it again in the near future.
>>>
>>> This change is about db schema, the modules I expect to work with
>>> allocated strings (or have length checks) in this case and should not be
>>> affected.
>>>
>>> Cheers,
>>> Daniel
>>>
>> Although I can see why someone might consider the need for larger than
>> varchar 64, I really don't see a need for it. Assuming if you needed
>> more characters it would probably be time to use additional
>> authentication methods.
> That refreshed my mind that we have now support for sha 256, which means
> that ha1 fields need to be 64 (and they are now), but wondering if
> someone will want to have and add sha 512 any time soon, which means the
> ha1 fields need to be 128...

I guess there's really no cost to increasing it to 128 / sha 256 and
give ourselves some good time before we need to reconsider. The storage
cost will still be based on the actual value used and not the size
constraint; giving people options.

>>
>> I believe Polycom still max's out at 32.
> I haven't looked at phones restrictions on this, but I saw old devices
> accepting only digit based passwords... hopefully not many out there at
> this moment.
> 
> A stronger auth method would be highly desired, but I guess a lot of
> people are restricted by deployed devices.
> 
> Cheers,
> Daniel
> 

--fred

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Re: [SR-Users] Size of passwords fields in db

[Fred Posner]

On 10/05/2016 10:25 AM, Daniel-Constantin Mierla wrote:
> Hello,
> 
> writing here to decide on a topic opened by pull request 779:
> 
>   - https://github.com/kamailio/kamailio/pull/779
> 
> what would be a fair size for db column storing a password that one
> would like to have for proper security?
> 
> I would like to make it consistent over all tables that have a password
> column by defining a xml entity for the size of these columns. The pull
> request suggests 64 chars, has anyone other opinions on making it larger
> or smaller?
> 
> If they are defined varchar, then should not be a problem of allocated
> size, so we can go with 128 if someone feels it worth doing larger now
> so we don't have to change it again in the near future.
> 
> This change is about db schema, the modules I expect to work with
> allocated strings (or have length checks) in this case and should not be
> affected.
> 
> Cheers,
> Daniel
> 

Although I can see why someone might consider the need for larger than
varchar 64, I really don't see a need for it. Assuming if you needed
more characters it would probably be time to use additional
authentication methods.

I believe Polycom still max's out at 32.

--fred


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Re: [SR-Users] Commercial SBC or Kamailio

[Fred Posner]

On 09/15/2016 10:38 AM, Linux Vince wrote:
> We are trying to setup VOIP infrastructure, mainly wholesale and retail.
> 
> What is best option if money is not the problem?
> 
> Developing our own infrastructure using Kamailio and other open source
> packages to act as SBC and switch or buy a commercial solution like
> GenBand/Sonus/Sansay.
> 
> We are looking for high performance with scalability to handle thousands
> of call setups per second over the period of few years.
> 
> I have no idea on how/if can kamailio outperform commercial solutions or
> not.
> 
> Advantage of using kamailio is flexibility and possibility of
> customization as per requirement,
> 
> I am new to this list so please advise if this is not a valid question
> to be asked here.
> 

Normally, we try to keep commercial discussions on the business list:

http://lists.kamailio.org/cgi-bin/mailman/listinfo/business

You also may want to check out the business directory:

https://www.kamailio.org/w/business-directory/

There are some products that involve Kamailio at it's core, such as
Canonical SIP Routing Platform (CSRP), Enswitch, Sip:Wise, 2600hz, etc.

-- 
Fred Posner
@fredposner
The Palner Group, Inc.
http://www.palner.com
direct/sms: +1 (503) 914-0999
direct/sms: +1 (224) 334-FRED

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Re: [SR-Users] Video conferencing with Kamailio

[Fred Posner]

On 07/11/2016 11:40 AM, Jay Li wrote:
> Fred,
> 
> Thanks a lot your detailed explanation. About the media server addition
> to Kamailio, do you have any suggestions I should look into besides
> Jitsi and FreeSWITCH? Thanks.
> 
> Regards,
> Jay

You could look into Asterisk as well, but I've not used it for video
conferencing so cannot speak from experience.

-- 
Fred Posner
@fredposner
http://palner.com
direct/sms: +1 (503) 914-0999
direct/sms: +1 (224) 334-FRED


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Re: [SR-Users] Video conferencing with Kamailio

[Fred Posner]

On 07/08/2016 12:36 AM, Jay Li wrote:
> Dear All,
> 
> I'm curious if anybody has set up an infrastructure for video
> conferencing utilizing Kamailio as a proxy (like NAT support and so so).
> I found a kind of old tutorial "Run you own Skype-like service in less
> than one hour" kamailio:skype-like-service-in-less-than-one-hour [Asipto
> - SIP and VoIP Knowledge Base Site]
> <http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour> 
> written
> in 2013 on Kamailio 3.1. I wonder if this feature has been further
> developed in later releases. I know Kamailio is more often used as a SIP
> proxy, but somehow I got the impression that Kamailio supports WebRTC,
> so I wonder if it's possible to implement a many to many video
> conferencing infrastructure using Kamailio. If the answer is yes, how's
>  the performance or the bottleneck I should pay more attention to (maybe
> bandwidth for video traffic, especially with NAT? ) Anyway I'm a newbie
> to both Kamailio and video conference, so any suggestion/discussion is
> appreciated. Thanks.
> 
> Regards,
> Jay
> 

Jay,

The principles in the tutorial hold true today, and yes, kamailio does
support WebRTC. Assuming you want your video conference to have more
that two parties, you will also need a media server in addition to Kamailio.

Kamailio does not handle the mixing, timing, etc of media to enable
multi-person video conferencing. So, you will need this being done
either by a separate media server or endpoint capable of doing this.

There are some products like Jitsi Video Bridge and FreeSWITCH that
support video conferencing "out of the box." You can combine these with
Kamailio as well to handle additional security, authentication, etc.

-- 
Fred Posner
@fredposner
http://palner.com
direct/sms: +1 (503) 914-0999
direct/sms: +1 (224) 334-FRED


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Re: [SR-Users] Why SDPOPS does not remove attributes in SDP

[Fred Posner]

On 06/17/2016 09:25 AM, Richard Fuchs wrote:
> On 17/06/16 03:46 AM, Dmitry wrote:
>> Hi all
>> I have the following code:
>>
>>   if($T_reply_code=="200")
>>  {
>>  if(has_body("application/sdp"))
>>  {
>> xlog("L_INFO", "RTPENGINE received internal reply
>> $T_reply_code $rr SDP extra lines will be removed");
>>
>> set_rtpengine_set("0");
>> rtpengine_manage();
>>  sdp_remove_line_by_prefix("a=rtcp");
>>  sdp_remove_line_by_prefix("a=ssrc");
>>  sdp_remove_line_by_prefix("a=ice");
>>  sdp_remove_line_by_prefix("a=candidate");
>>
>>  xlog("L_INFO", "RTPENGINE received internal reply
>> $T_reply_code $rr SDP extra lines removed with SDPOPS");
>>
>>  }
>>
>>  }
>> When I look through traces  - I see that 200 ok(with SDP) has all these
>> attributes and they are not removed.
>>
>> Why SDPOPS does not remove these attributes?
> 
> Probably because there's a problem rewriting parts of the SDP body more
> than once. But if you don't want ICE attributes in the output SDP, you
> can use the rtpengine flags ICE=remove. You can influence rtcp-mux
> attributes in the same way. See docs.
> 
> Cheers
> 

Are your log messages triggered?

--fred



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Re: [SR-Users] High availability

[Fred Posner]

If it's just 2 servers, consider as Juha said, corosync/pacemaker with drbd. 

Fred Posner
direct: +1 (224) 334-FRED (3733)

> On Jun 5, 2016, at 5:26 PM, Moacir Ferreira <moacirferre...@hotmail.com> 
> wrote:
> 
> Hi,
> 
> Sorry... I should have mentioned before. You guys are thinking on the 
> standard Internet SIP calls' behavior while I am trying to use Kamailio on a 
> large "industrial" project. This said:
> 
> Assuming that the end-point is "smart", the DNS method is functional but it 
> would take quite a while before the UA (phone) recovers from the previous 
> name/IP binding it has in cache;
> SRV is good for a "smart" UA that, unfortunately, is not the case;
> Same for the phone units as they are industrial "Help Points" and so quite 
> "dummy".
> 
> While I never tested it, I thought I could use two Kamailio servers with a 
> mysql cluster like mariadb-galera where, for Kamailio functions, one server 
> would be "active" and another "passive" server. Then use keepalived for 
> monitoring the "active" Kamailio and starting the "passive" server if the 
> active Kamailio fails. Without any testing, tests that I think I should have 
> done before putting questions in here, my questions are:
> 
> Suppose that  I have two Kamailio servers, one "active" and another one 
> "passive" (not running) where the mysql databases are synchronized in between 
> two servers using MySQL Galera. Using keepalived I would monitor the active 
> Kamailio instance. Should it fails, start the "passive" Kamailio instance 
> using the same MySQL database that were supposed to be synchronized. Would 
> this new Kamailio instance be able to find a called number? Why this 
> question? As long as I understand, Kamailio will always challenge the UA for 
> authentication before making a call, so if this second server gets a call 
> request it would just challenge and authenticate the caller. The "key point" 
> would be having this new Kamailio instance aware about the called 
> destinations. So, delivering a MySQL database, with the latest data the 
> active Kamailio had, to this new Kamailio instance would be enough to allow 
> it find the called party?
> 
> Anyway, can you guys comment on my "thoughts"? Is it possible? Am I missing 
> something? Would you suggest another approach for such scenario?
> 
> Cheers!
> Moacir 
> 
> Date: Sun, 5 Jun 2016 21:07:41 +0200
> From: chabert.loic...@gmail.com
> To: sr-users@lists.sip-router.org
> Subject: Re: [SR-Users] High availability
> 
> Hello Bill,
> 
> I have made kamailio ha using exabgp with loopbacks.
> 
> Check https://github.com/Exa-Networks/exabgp
> 
> With bgp, kamailio cluster can be splited on severals datacenters.
> 
> Regards.
> 
> Le 5 juin 2016 20:53, "Bill" <b...@novatrope.us> a écrit :
> Hi Moacir
> 
> We have only found three ways to handle failover.
> 1. Change the DNS entry whenever a failure is detected.
> 2. Use SRV records to display an alternate route.
> 3. Use the failover mechanism in the phone itself
> 
> 1. works, but it may take some time for your ua's to become aware of the 
> change
> 2. never have been able to get this to work as advertised.
> 3. Works pretty well depending on the phone. (We use mostly Yealink's and 
> they seem to handle the failover pretty well.)
> 
> Hope this helps
> 
> On 06/05/2016 07:41 AM, Moacir Ferreira wrote:
> Hi,
> 
> I got two questions regarding high availability:
> 
> 1 - Should my Kamailio server fail, I would like another Kamailio 
> "box/server" to take over with minimum services disruption. What is the 
> "community" advice for such environment?
> 
> 2 - Should my main PSTN gateway fail, what would be the best mechanism to 
> redirect calls to a second PSTN gateway?
> 
> Cheers!
> Moacir
> 
> 
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Re: [SR-Users] username of Contact header

[Fred Posner]

On 05/27/2016 10:21 AM, Al S wrote:
> Hi,
> 
> I am trying to read username portion of Contact header:
> 
> My Contact header content has a URI value such as:
> sip:813111@10.10.10.10.:5060
> 
> and I am trying to read : 813111
> 
> I tried the following perl similar RE and it didn't work:
> 
> ($var(main_number2)) = $ct =~ /sip:(.*)@/;
> 
> Thanks,
> AS

Check out transformations:

$var(user) = $(ct{tobody.user});

http://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#uri_transformations

http://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#to-body_transformations

-- 
Fred Posner
@fredposner
The Palner Group, Inc.
http://www.palner.com
direct/sms: +1 (503) 914-0999
direct/sms: +1 (224) 334-FRED



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Re: [SR-Users] SRV and dialog stickiness

[Fred Posner]

On 05/25/2016 04:32 PM, Alex Balashov wrote:
> Hello,
> 
> Sorry if this is a tired, worn question, but I've not dealt much with
> Kamailio's SRV support before:
> 
> If a registrant has a contact binding whose domain component is subject
> to an SRV lookup with load-balanced or weighted entries, how does one
> solve the problem of ensuring that subsequent in-dialog requests go to
> the same host as the initial INVITE? Does Kamailio offer some facility
> for doing this? Is it somehow accommodated by SIP?
> 
> -- Alex
> 

Clarification:

Why would this be wanted?

I ask, as if you were using a srv record as the result of a load balance
lookup, wouldn't the point be to be able to quickly change location of
the domain in case of an outage/issue?

Otherwise, I'm not positive the benefit of doing a srv lookup for this
scenario.

-- 
Fred Posner
@fredposner
The Palner Group, Inc.
http://www.palner.com
direct/sms: +1 (503) 914-0999
direct/sms: +1 (224) 334-FRED



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Re: [SR-Users] LCR Module Question

[Fred Posner]

Have you considered either dispatcher or just using a failure route?

-- Fred

> On May 6, 2016, at 7:15 AM, Alberto Sagredo  
> wrote:
> 
> Hi
> 
> I have it working but i have re-read documentation and do not see how to do 
> what i need. 
> 
> I explain it :)
> 
> Now i have only one LCR provider and i need to add a backup one.
> 
> I do not know if its enough to add under same lcr_id or its better to add 
> with different one and add several lcr_rule and lcr_rule_target 
> 
> Acordding to next_gw() function, what is better to be used? In my case. a 
> primary providers is used and only a backup one for the momment.
> 
> Thanks for your help
> 
> Alberto
> 
> 
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Re: [SR-Users] set_advertised_address() / set_advertised_port() + PVs?

[Fred Posner]

On 03/23/2016 01:51 PM, Alex Balashov wrote:
> Hello,
> 
> I have a need to deploy Kamailio in AWS in a scenario of this sort:
> 
> 
>[Public Internet] <---> Kamailio <---> (Internal AWS servers)
> 
> In such a scenario, Kamailio would be multihomed.
> 
> 
> 1. Is this sane? Any unforeseen effects, e.g. vis-a-vis RR, provided
> enable_double_rr is enabled and that two genuinely different network
> interfaces are used?
> 

I'm not sure if it's sane, but I do this as well.

> 2. Do set_advertised_address()/set_advertised_port() accept PV
> arguments, or are they pre-PV "core function folk traditions" in the
> same way as rewritehostport() and force_send_socket()?
> 

I have a main listen=udp:192.168.25.31 advertise PUBLIC:5060

and then when needed...

set_advertised_address("192.168.25.31");

as in...

if ($rd=~"192.168.25") {
  set_advertised_address("192.168.25.31");
}


--fred

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Re: [SR-Users] KAMAILIO Installation Problem

[Fred Posner]

On 03/04/2016 01:47 PM, Ed Todd wrote:
> I have a local area network with:

[snip]

> I think the problem is that the reply to the PC is
> being sent to 192.168.1.12 instead of 92.13.147.87.

If it's on a LAN, I'd assume it would be ok to use the NAT address other
than the external address... but that being said, if you're using public
IP for everything on the LAN, look into advertised_address:

https://www.kamailio.org/wiki/cookbooks/4.3.x/core#advertised_address

This being said, I'm confused by your scenario.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)


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Re: [SR-Users] Kamailio 5.0 - B2BUA

[Fred Posner]

On 03/02/2016 12:45 PM, Alex Balashov wrote:
> Hi,
> 
> I wanted to raise the possibility of an inline signalling-only B2BUA
> component to Kamailio.
> 

+1 for this... for many reasons including seeing what happens when the
following threshold is met:

> I myself am philosophically opposed to a B2BUA in Kamailio
> to the threshold of physical violence.
> 
> -- Alex
> 



Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-224-334-FRED (3733) direct

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Re: [SR-Users] SQLOPS vs APP_LUA

[Fred Posner]

On 12/02/2015 06:22 AM, Luke Milbourne wrote:
> Hi All,
> 
> I need to run some SQL and then set the destination URI according to the
> results, if any.
> 
> I've got this working using APP_LUA, however, I've noticed SQLOPS.
> 
> Can anyone advise on which method is likely to produce better results.
> 
> I'm assuming my Lua function is connecting to the DB and then
> disconnecting each time.
> 
> If I use SQLOPS, does this mean Kamailio will keep a connection
> available to the DB?
> 
> -- 
> Luke Milbourne
> <mailto:luke.milbou...@gmail.com>
> 
> Tel: 07857154817
> Google Talk/Email: luke.milbou...@gmail.com
> <mailto:luke.milbou...@gmail.com>
> 

SQLOPS is a great tool and allows you to free when desired
(sql_result_free) as well as store the query results in xavps (or other
methods). You can run sql queries as well as call stored procedures.

You could easily run the query, store in xavp, free results and the use
the xavps later on (such as in failed route, etc).

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-224-334-3733 (direct)


0x6235BD69.asc
Description: application/pgp-keys
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Re: [SR-Users] Familiar topic about Kamailio with Asterisk behind NAT

[Fred Posner]

On 10/08/2015 09:09 AM, Ovidiu Sas wrote:
> If the SDP is correct, then you might have specific issues related to
> your specific deployment case. Snippets from others config files won't
> help. You really need to investigate and understand your particular
> issue that you are facing and fix it accordingly.
> 
> Regards,
> Ovidiu Sas
> 
> On Oct 8, 2015 09:04, "Dirk Teurlings - SIGNET B.V."
> <dteurli...@signet.nl <mailto:dteurli...@signet.nl>> wrote:
> 
> On 30-09-15 13:29, Fred Posner wrote:
> 
> 
> Without a version of rtpproxy using the -A flag, you'll need to
> either
> (1) update to a different version of rtpproxy or (2) skip
> rtpproxy and
> have your asterisk handle all the rtp.
> 
> 
> I tried rtpproxy v2, with the -A flag in bridge mode ( -A
> privateip/publicip ). This doesn't reflect anything in the SIP headers.
> 

A is advertise which should be different from bridging, and not certain
why you'd be in bridged mode. Are you using more than one nic?

Normally, for -A use something like:

rtpproxy -A PUBLICIP -F -l PRIVATEIP -m 1 -M 55000 -s udp:*:7722 -d INFO

(the -d INFO will increase your logging)

If you continue to have trouble, I'd recommend you post an ngrep of the
failure.

Fred Posner
http://www.palner.com (web)
+1-224-334-3733 (direct)

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Re: [SR-Users] Familiar topic about Kamailio with Asterisk behind NAT

[Fred Posner]

On 09/30/2015 07:15 AM, Dirk Teurlings - SIGNET B.V. wrote:
> On 30-09-15 12:23, Fred Posner wrote:
>>
>> Are you using -A flag in rttproxy?
>>
> 
> Unfortunately we're running a version of RTPPROXY at the moment that
> doesn't have this flag. I'm considering upgrading to rtproxy 2.0, but
> will need to test whether this doesn't affect anything else.
> 
> It's preferred to find a solution without major changes to
> software(version). But if anything else fails I might need to...
> 
> Cheers,
> Dirk
> 

Without a version of rtpproxy using the -A flag, you'll need to either
(1) update to a different version of rtpproxy or (2) skip rtpproxy and
have your asterisk handle all the rtp.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

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Re: [SR-Users] Familiar topic about Kamailio with Asterisk behind NAT

[Fred Posner]

On 09/30/2015 04:22 AM, Dirk Teurlings - SIGNET B.V. wrote:
> Hi,
> 
>  CLIENTS <-> (NAT) <-> INTERNET <-> KAMAILIO(4.2.5) with
> RTPPROXY(v1) <-> PRIVATE LAN <-> ASTERISK (v1.8)
> 
> 
> Any pointer or help would be greatly appriciated.
> 
> Cheers,
> Dirk
> 

Are you using -A flag in rttproxy?

So, to bridge to outside from kamailio you'll need to use advertised
address with public IP as well as -A flag in rtpproxy (I've been using
either the @miconda patched version or 2.0).

Then on inside communications, you'll need to set rtpproxy (override) to
use NAT/LAN IP, and use the public IP for the other side.

Something like...

if(dst_ip==ASTERISKIP) {
rtpproxy_manage("co","PRIVATEIP");
} else {
rtpproxy_manage("co");
}


Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

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Re: [SR-Users] OT: some github stuff

[Fred Posner]

On 08/25/2015 06:23 AM, Daniel-Constantin Mierla wrote:
 
 

 There's a special github markdown that shouldn't require a file:
 
 I don't understand what do you mean with the above, can you detail?
 
 I was looking to get README displayed as markdown file by github.com on
 the website of kamailio project repository, without renaming it to
 README.md.
 
 Cheers,
 Daniel
 

I was incorrect.


Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

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Re: [SR-Users] OT: some github stuff

[Fred Posner]

On 08/24/2015 02:34 PM, Daniel-Constantin Mierla wrote:
 2) is there any specific content/marker to add to README to have it
 interpreted as markdown (without adding .md extensions)? Something like
 a special first or last line, ... It will be displayed nicer, but
 renaming will result in changing lot of other places (makefiles, deb/rpm
 specs, etc...) and somehow README is the traditional approach for the
 kind of this file name for Linux/Unix apps.
 

There's a special github markdown that shouldn't require a file:
https://help.github.com/articles/github-flavored-markdown/

And of course a list of markups listed with
https://github.com/github/markup.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

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Re: [SR-Users] Mitel 5000 and 3000

[Fred Posner]

On 08/20/2015 04:10 AM, Daniel-Constantin Mierla wrote:
 
 By they try to speak directly with the media server vs the sip server
 -- do you mean they ignore Record-Route headers added by the sip server?
 
 Cheers,
 Daniel
 

Yes, that would be a much better way of expressing the issue.

Without topology hiding, the Mitel will ignore Record-Route headers. If
using topology hiding, they will send an in-dialog invite to the sdp c=
address...

Can hack around it by using both topology hiding and rtpproxy but would
rather see if I could get the Mitel to behave in a more respectful manner.

I've also thrown in either FreeSWITCH or Asterisk as a media server for
those end-points to just handle it; which may end up being a decent
solution for the trouble-making end-points.

--fred

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Re: [SR-Users] Mitel 5000 and 3000

[Fred Posner]

On 08/20/2015 08:25 AM, Fred Posner wrote:

[SNIP]
 
 Without topology hiding, the Mitel will ignore Record-Route headers. If
 using topology hiding, they will send an in-dialog invite to the sdp c=
 address...
 

EDIT: on the o= information... I need more sleep. =)

--fred

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Re: [SR-Users] Mitel 5000 and 3000

[Fred Posner]

On 08/20/2015 08:34 AM, Alex Balashov wrote:
 Fred,
 
 Throwing in B2BUAs is a common recipe for interop problems; one side
 is liberal in what it accepts, the other conservative in what it
 emits. In theory.
 
 ‎For lightweight, signalling-only duty, allow me to suggest SEMS +
 its 'sbc' module, though. It gives you a lot of low-level control to
 solve these kinds of interop problems, even if not quite as much as
 something programmable like Kamailio, and it's a lot less opaque and
 bureaucratic than Asterisk or Freeswitch. Email me offline if you
 want some sample configs, as the community documentation is a little
 light.
 
 -- Alex Balashov 

I've been meaning to test SEMS for a while... This may just push me to
it...

Thanks, and I'll get with you offline.

--fred

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Re: [SR-Users] Kamailio with NATted RTPproxy

[Fred Posner]

On 08/18/2015 08:36 AM, Jean-Marie Baran wrote:
 Hi,
 
 I am trying to have RTPproxy works with Kamailio behind a NAT. As it
 seems RTPproxy cannot handle NAT natively, I installed RTPproxy 1.2.1 by
 miconda: https://github.com/miconda/rtpproxy
 
 

This one is good or the new 2.0 version also supports the -A flag. You
may want to run in debug or info mode to see if you're passing packets.

Something like:

rtpproxy -A PUBLICIP -F -l PRIVATEIP -m 1 -M 2 -s
udp:127.0.0.1:7722 -d INFO

Making sure that the m (minimum) and M (maximum) ports match what you
have opened/forwarded on your firewall.

 
 My question here is what should be the addresses in the SDP when
 redirected from Kamailio to the client ?
 
 Cheer,
 -- 
 *Jean-Marie Baran*
 

The address should be your public ip (for outside connections), which
should also be in the advertised statement within kamailio.

--fred


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Re: [SR-Users] Kamailio with NATted RTPproxy

[Fred Posner]

On 08/18/2015 09:39 AM, Jean-Marie Baran wrote:
 Should I understand that the router should send the packet back to
 Kamailio which then send them to the SIP server ?

RTP packets will flow from client to client or with rtpproxy, from
client - rtpproxy - client.

--fred

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[SR-Users] Mitel 5000 and 3000

[Fred Posner]

I rarely integrate these with Kamailio and am having some resistance
in assisting. I was wondering if anyone would be willing to share any
configuration recommendations for Mitel when using Kamailio as a SIP
trunk (outbound/inbound).

Sincerely,

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

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Re: [SR-Users] Mitel 5000 and 3000

[Fred Posner]

On 08/18/2015 09:23 AM, Alex Balashov wrote:
 Are there any specific issues you're encountering?
 

Mostly hold issues where they try to speak directly with the media
server vs the sip server; as well as general audio issues in/out.

Most of these seem to be acl related with a set-up issue on the Mitel. I
don't work with Mitel enough to discuss in Mitel speak what to change.
Apparently RTP ports is a different language to the people I am working
with. =)

--fred

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[SR-Users] Handling Hold

[Fred Posner]

Hello all,

Always fun integrating with proprietary pbx's... is there a way that is
generally accepted as handling hold invites from devices such as Mitel
or Toshiba?

We're not proxying the media, so these devices are trying to send an
Invite to the endpoint when calls are placed on hold.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

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Re: [SR-Users] Handling Hold

[Fred Posner]

On 08/13/2015 02:30 PM, Alex Balashov wrote:
 I assume these are reinvites? If so, how would the strategy for
 handling them deviate from handling of all other in-dialog requests?
 
 -- Alex Balashov | Principal | Evariste Systems LLC 
 Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) 
 http://www.evaristesys.com/, http://www.csrpswitch.com/
 

Sadly, no. Straight up INVITE.

I was going to experiment with the is_audio_on_hold() function of textops...

http://www.kamailio.org/docs/modules/4.3.x/modules/textops.html#textops.f.is_audio_on_hold

however uncertain if my lack of proxied media would be a factor.

--fred

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Re: [SR-Users] Handling Hold

[Fred Posner]

On 08/13/2015 02:42 PM, Alex Balashov wrote:
 On 08/13/2015 02:34 PM, Fred Posner wrote:
 
 Sadly, no. Straight up INVITE.
 
 Wait, what? That doesn't make any sense. Can you provide a full
 signalling capture?
 

Here's an example exported from the pcap. This example has topology
hiding enabled with an address of 127.0.0.8.

--fred
Frame 1: 1441 bytes on wire (11528 bits), 1441 bytes captured (11528 bits)
Linux cooked capture
Internet Protocol Version 4, Src: [PUBLIC-KAMAILIO] ([PUBLIC-KAMAILIO]), Dst: 
[PUBLIC-MITEL] ([PUBLIC-MITEL])
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:pstn-did@[PUBLIC-MITEL] SIP/2.0
Message Header
Record-Route: 
sip:[PUBLIC-KAMAILIO];r2=on;lr;ftag=gK0e4a212a;vst=ABkJAw0IDQ4PAQF2AXIWFwYXGQEWHQc4OjUwNjA-;did=7.a51
Record-Route: 
sip:127.0.0.8;line=sr-XLSMq0lrx0hrq6lrx0lgX0dzPLkgP9dgTvDWTH2vZH1Se5CmxIuWqyTHF621E-R1E-uaZhRyxCSC-IDEEKR5xhRiZKF5F2SiD2RRKtWDiHD48SKyI.Q1fIwh8sEzemYWeIC*
Via: SIP/2.0/UDP 
[PUBLIC-KAMAILIO];branch=z9hG4bK05ee.871a25f6cea62efc9ab8a36e1198c199.0
Via: SIP/2.0/UDP 
127.0.0.8;branch=AX--tSEfHdrxbzKDRJlebk2eZknxmkYqIU2x6iMqyuMPyut4I-Me0JgKClzT5SLqLumisY08627qsW9e5ufx5Kbe5dte0eWe0lmT5Rhq6xyTE**
From: sip:+1callerid-did@[PSTN-IPADDRESS]:5060;tag=gK0e4a212a
To: sip:pstn-did@[PUBLIC-MITEL]
Call-ID: 1309573957_105226997@[PSTN-IPADDRESS]
CSeq: 23665 INVITE
Max-Forwards: 68
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Accept: application/sdp, application/isup, application/dtmf, 
application/dtmf-relay,  multipart/mixed
Contact: 
sip:127.0.0.8;line=sr-XLSMqGpnx0C2x0hMe6Xnq-Jtf0-2f0CHf0hYq0-Me0J*
P-Asserted-Identity: sip:+1callerid-did@[PSTN-IPADDRESS]:5060
Supported: 100rel
Content-Length:  255
Content-Disposition: session; handling=required
Content-Type: application/sdp
Message Body

Frame 2: 534 bytes on wire (4272 bits), 534 bytes captured (4272 bits)
Linux cooked capture
Internet Protocol Version 4, Src: [PUBLIC-MITEL] ([PUBLIC-MITEL]), Dst: 
[PUBLIC-KAMAILIO] ([PUBLIC-KAMAILIO])
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (100)
Status-Line: SIP/2.0 100 Trying
Message Header
Via: SIP/2.0/UDP 
[PUBLIC-KAMAILIO];branch=z9hG4bK05ee.871a25f6cea62efc9ab8a36e1198c199.0;received=[PUBLIC-KAMAILIO]
Via: SIP/2.0/UDP 
127.0.0.8;branch=AX--tSEfHdrxbzKDRJlebk2eZknxmkYqIU2x6iMqyuMPyut4I-Me0JgKClzT5SLqLumisY08627qsW9e5ufx5Kbe5dte0eWe0lmT5Rhq6xyTE**
To: sip:pstn-did@[PUBLIC-MITEL]
From: sip:+1callerid-did@[PSTN-IPADDRESS]:5060;tag=gK0e4a212a
Call-ID: 1309573957_105226997@[PSTN-IPADDRESS]
CSeq: 23665 INVITE
User-Agent: Mitel Border GW/4.4.0.80-01
Content-Length: 0

Frame 3: 588 bytes on wire (4704 bits), 588 bytes captured (4704 bits)
Linux cooked capture
Internet Protocol Version 4, Src: [PUBLIC-KAMAILIO] ([PUBLIC-KAMAILIO]), Dst: 
[PUBLIC-MITEL] ([PUBLIC-MITEL])
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (PRACK)
Request-Line: PRACK sip:pstn-did@[PUBLIC-MITEL]:5060;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 
[PUBLIC-KAMAILIO];branch=z9hG4bKd4ee.c8ccffc14692925a5f5b81e3b44b1889.0
Via: SIP/2.0/UDP 
127.0.0.8;branch=AX--tSEfHdrxbzKDRJlebk2eZknxmkYqIU2x6iMqyuMPyut4I-Me0JgKClzT5eLqLumisY08627qsW9e5ufx5Kbe5dtq6JMe6-nT0dHT6JtTl**
From: sip:+1callerid-did@[PSTN-IPADDRESS]:5060;tag=gK0e4a212a
To: sip:pstn-did@[PUBLIC-MITEL];tag=0_1970073968-102972850
Call-ID: 1309573957_105226997@[PSTN-IPADDRESS]
CSeq: 23666 PRACK
Max-Forwards: 69
RAck: 296987492 23665 INVITE
Content-Length: 0

Frame 4: 651 bytes on wire (5208 bits), 651 bytes captured (5208 bits)
Linux cooked capture
Internet Protocol Version 4, Src: [PUBLIC-MITEL] ([PUBLIC-MITEL]), Dst: 
[PUBLIC-KAMAILIO] ([PUBLIC-KAMAILIO])
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (200)
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 
[PUBLIC-KAMAILIO];branch=z9hG4bKd4ee.c8ccffc14692925a5f5b81e3b44b1889.0;received=[PUBLIC-KAMAILIO]
Via: SIP/2.0/UDP 
127.0.0.8;branch=AX--tSEfHdrxbzKDRJlebk2eZknxmkYqIU2x6iMqyuMPyut4I-Me0JgKClzT5eLqLumisY08627qsW9e5ufx5Kbe5dtq6JMe6-nT0dHT6JtTl**
Contact: sip:pstn-did@[PUBLIC-MITEL]:5060;transport=UDP
To: sip:pstn-did@[PUBLIC-MITEL];tag=0_1970073968-102972850
From: sip:+1callerid-did@[PSTN-IPADDRESS]:5060;tag=gK0e4a212a
Call-ID: 1309573957_105226997@[PSTN-IPADDRESS]
CSeq: 23666 PRACK
Date: Thu, 13 Aug 2015 00:31:40 GMT
User-Agent: Mitel Border GW/4.4.0.80-01
Content-Length: 0

Frame 5: 216 bytes on wire (1728 bits), 216 bytes captured (1728 bits)
Linux cooked capture

Re: [SR-Users] Handling Hold

[Fred Posner]

On 08/13/2015 03:30 PM, Alex Balashov wrote:
 I see a dialog established followed by a reinvite. Which frame # do you
 take to be the on-hold invite?
 

I was taking frame 8 as a new invite. I'll see what I'm doing wrong.

--fred

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Re: [SR-Users] Interpretation of Contact header

[Fred Posner]

Is there a reason why you cannot disable SIP ALG?

--fred

On 07/30/2015 03:12 PM, Joachim Büchse wrote:
 I’m coming back to this very old question as we have still not resolved
 this issue with Juniper.
 
 Are you aware of any RFC section that mandates, that the VIA headers
 IP+port should match the effective (transport) IP+port. Or how a UAS
 should interpret a mismatch?
 
 I’m seeing the following behavior of SER/OpenSER.
 
 REGISTER request is received from UAC via TCP.
 VIA header contains an IP + port.
 TCP source port of REGISTER request does NOT match port in VIA header:
 - UAS ignores CONTACT header and uses effective (TCP source) port + IP
 of REGISTER
 request as contact address for future INVITE messages it sends to
 the UAC
 
 I understand WHY this is done (i.e. to make UAC behind NAT work).
 However I wonder if this specific behavior is based on a particular RFC
 recommendation.
 
 The reason for my question is, that the above scenario happens with the
 SIP-Alg of our Juniper firewall. However the firewall rejects the
 INVITEs from the UAS. Juniper acknowledged, that the port in the
 rewritten VIA header of the REGISTER request does match the effective
 TCP port used to send it to the UAS, but they do not consider this being
 in contradiction to any RFC.
 
 Best regards,
 Joachim
 
 On 13/02/15 17:55, Joachim Büchse wrote:
 Good day,

 I’m experiencing some problems with our VoiP providers handling of
 REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a
 Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with
 UDP but our provider wants us to use TCP. With TCP enforced incoming
 calls don’t work. I’ve done some wire tracing and to me it seems
 that the providers configuration is to blame, but then - there are
 many RFCs out there and many NAT and UAC bug workarounds. Anyway, I
 wanted to get the opinion of “the experts about how the requests
 send to the UAS  SHOULD  be correctly interpreted.


 The REGISTER requests/responses look like this (outside of the
 firewall):

 Protocol TCP!
 client port 19091 - server port 5060

 REGISTER sip:pbx.peoplefone.ch SIP/2.0
 Via: SIP/2.0/TCP
 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec
 From: Michel sip:90780408...@pbx.peoplefone.ch;tag=2153084485
 To: Michel sip:90780408...@pbx.peoplefone.ch
 Call-ID: 2825358480@10_10_128_10
 CSeq: 1 REGISTER
 Contact: sip:90780408050@212.126.160.92:6717;transport=tcp
 Max-Forwards: 70
 User-Agent: N720-DM-PRO/70.089.00.000.000
 Expires: 180
 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
 Content-Length: 0

 SIP/2.0 401 Unauthorized
 Via: SIP/2.0/TCP
 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec
 From: Michel sip:90780408...@pbx.peoplefone.ch;tag=2153084485
 To: Michel
 sip:90780408...@pbx.peoplefone.ch;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc
 Call-ID: 2825358480@10_10_128_10
 CSeq: 1 REGISTER
 WWW-Authenticate: Digest realm=pbx.peoplefone.ch
 http://pbx.peoplefone.ch/, nonce=VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy
 Server: kamailio (3.2.1 (x86_64/linux))
 Content-Length: 0

 REGISTER sip:pbx.peoplefone.ch SIP/2.0
 Via: SIP/2.0/TCP
 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0
 From: Michel sip:90780408...@pbx.peoplefone.ch;tag=2153084485
 To: Michel sip:90780408...@pbx.peoplefone.ch
 Call-ID: 2825358480@10_10_128_10
 CSeq: 2 REGISTER
 Contact: sip:90780408050@212.126.160.92:6717;transport=tcp
 Authorization: Digest username=90780408050,
 realm=pbx.peoplefone.ch http://pbx.peoplefone.ch/,
 uri=sip:pbx.peoplefone.ch,
 nonce=VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy,
 response=764f371a08d258157a249f8d1b852514
 Max-Forwards: 70
 User-Agent: N720-DM-PRO/70.089.00.000.000
 Expires: 180
 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
 Content-Length: 0

 SIP/2.0 200 OK
 Via: SIP/2.0/TCP
 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0
 From: Michel sip:90780408...@pbx.peoplefone.ch;tag=2153084485
 To: Michel
 sip:90780408...@pbx.peoplefone.ch;tag=a0440f545f39b2694d387b475a5f6bc9.6bda
 Call-ID: 2825358480@10_10_128_10
 CSeq: 2 REGISTER
 Contact:
 sip:90780408050@212.126.160.92:6717;transport=tcp;q=0;expires=180;received=sip:212.126.160.92:19091;transport=TCP
 Server: kamailio (3.2.1 (x86_64/linux))
 Content-Length: 0


 The ip:port the firewall is sending those requests from is  ip
 212.126.160.92 port 19091. So this does NOT match the port from the
 Contact header. For TCP this seems rather logical to me, as one cant
 be listening on a TCP port and use it for sending at the same time.
 The UAC closes this “register connection” with TCP FIN after the
 register, and so does the firewall.

 However unfortunately subsequent requests from the provider (ie UAS)
 come in on port 19091 (not port 6717 from the Contact header) and
 the firewall simply drops them.

 Observations:
 - the server does NOT include received=212.126.160.92 in the Via of

Re: [SR-Users] Dispatcher module number of calls

[Fred Posner]

Have you tried with algorithm 10 (call load distribution)?

http://www.kamailio.org/docs/modules/4.3.x/modules/dispatcher.html

You will need to set a few parameters such as duid, hash_size, etc. and
then utilize the special attribute of maxload.

Then, using ds_load_update and ds_load_unset dispatcher will update the
load state for the node.

Fred Posner
f...@palner.com
http://palner.com



On 07/23/2015 03:48 AM, Alberto Sagredo wrote:
 Hi
 
 Hi have read documentation but it seems dispatcher does not keep how
 many calls has been dispatched or currently are in any of dispatcher
 destinations
 
 I have take a look to code on:
 
 http://lists.sip-router.org/pipermail/sr-users/2012-July/073919.html
 
 But it seems to use calls limit per user, 
 
 I would need to establish a limit per dispatcher destination i order to
 do not send more calls.
 
 Any idea where to look for?
 
 I have take a look to DMQ module, that maybe i could to use to spread
 information of how many calls to every destination but do not know how
 to extract calls from kamailio that have been dispatched to this
 destination using disptcher module.
 
 Best Regards
 
 Alberto
 
 
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0x6235BD69.asc
Description: application/pgp-keys
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Re: [SR-Users] HA+DRBD setup

[Fred Posner]

On 07/08/2015 12:31 AM, solution wrote:
 but while i manually killing that application fail over switching is not
 happening.

Are there any errors?

--fred

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[SR-Users] Pipelimit

[Fred Posner]

I know in ratelimit, you can define a method within queues... for
example, modparam(ratelimit, queue, 3:INVITE) to match on INVITE
method.

With pipelimit, is the method called from the config, such as:

if(is_method(INVITE)) {
  # Checking limit on INVITE
  if (!pl_check($au, traildrop, $var(limita))) {
pl_drop();
exit;
  }
}

if(is_method(REGISTER)) {
  # Checking limit on REGISTER
  if (!pl_check($au, traildrop, $var(limitb))) {
pl_drop();
exit;
  }
}

-- 
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

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Re: [SR-Users] Potential ip_addr_match_net bug

[Fred Posner]

On 06/29/2015 07:21 PM, Kyle Kurz wrote:
 I'm using allow_address function of the permissions module and I'm
 seeing strange behavior with network masks not divisible by 8. I have
 172.16.0.0/12 http://172.16.0.0/12 in my address table and Kamailio is
 marking packets from 172.56.16.232 as being part of that local subnet.
 If I change the subnet to 172.16.0.0/16 http://172.16.0.0/16, I no
 longer have an issue.
 
 I know there was a bug several years ago related to this
 (http://git.kamailio.org/gitlist/index.php/sip-router/commit/0ead7ab31bf0df6eb313dc3f9a4979c2a45cce8b),
 but that code is very different, so I'm wondering if a regression has
 crept in? I'm using Kam 4.1.6.
 
 Thanks,
 Kyle
 

Kyle,

How are you using allow_address?

--fred

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Re: [SR-Users] Kamailio with Blox SBC (opensips based) - Wrong CSeq

[Fred Posner]

On 06/22/2015 11:41 AM, kai.ohnac...@cbc.de wrote:
 In the txt file you can find the ngrep traffic.
 
 Cheers, 
 Kai
 
 -
 
 Perhaps you have an ngrep of the sip traffic?
 
 --fred
 

Is .3 the user, .1 the kamailio, and .33 the SBC?

U 2015/06/22 17:08:53.229016 192.168.0.1:5060 - 192.168.0.33:5060
INVITE sip:1001@192.168.0.33:5060 SIP/2.0
Record-Route: sip:192.168.0.1;lr=on
Via: SIP/2.0/UDP
192.168.0.1;branch=z9hG4bK3504.df39a8277f82eb6121d9c32883c70ead.0
Via: SIP/2.0/UDP 192.168.0.3:6060;branch=z9hG4bK0a9b.47eca853.0
From: PhonerLite sip:1000@192.168.1.3:8000;tag=3066030555
To: sip:1001@192.168.1.3:8000
Call-ID:
SBCbFhRRnpBd3ZEUWNUVF8gQWVZREoKMXhvWVMSVCUxUzIVWSpCCUJ4cEdWZV1NQ0tFYQ==
CSeq: 5 INVITE

U 2015/06/22 17:08:53.239973 192.168.0.33:5060 - 192.168.0.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.1;branch=z9hG4bK3504.df39a8277f82eb6121d9c32883c70ead.0
Via: SIP/2.0/UDP 192.168.0.3:6060;branch=z9hG4bK0a9b.47eca853.0
Max-Forwards: 70
To: sip:1001@192.168.1.3:8000;tag=krTfxNsCTuX
From: PhonerLite sip:1000@192.168.1.3:8000;tag=3066030555
Call-ID:
SBCbFhRRnpBd3ZEUWNUVF8gQWVZREoKMXhvWVMSVCUxUzIVWSpCCUJ4cEdWZV1NQ0tFYQ
CSeq: 1


It almost looks like the SBC is replying with a register request (which
is being responded with a correct CSeq increment).

On the invite initially sent to the SBC with a 5, the sbc responds with a 1.

--fred

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Re: [SR-Users] Kamailio with Blox SBC (opensips based) - Wrong CSeq

[Fred Posner]

On 06/22/2015 10:30 AM, kai.ohnac...@cbc.de wrote:
 Hello,
 
 my setup is a kamailio server as registrar and Blox as SBC.
 
 Blox is a freeware GUI with Opensips as framework.
 
 Kamailio is located in the private network and Blox operates with 2 NICs
 (private and public)
 
 Now the following problem:
 
 If I make a call from public (phonerlite) to private (mayah), the
 connection don’t reach the “framed” state and is “broken” after 30 sec.
 
 After evaluating the log files, I found that the CSeq from Kamailio is
 wrong.
 
 The invite comes from
 
 CSeq: 9 INVITE
 
 The reply is with CSeq 1
 
 SIP/2.0 180 Ringing
 Via: SIP/2.0/UDP 192.168.0.3:6060;branch=z9hG4bK38ae.cf317344.0
 Max-Forwards: 70
 To: sip:1001@192.168.1.3:8000;tag=qTKGZvr0ItW
 From: PhonerLite sip:1000@192.168.1.3:8000;tag=1005520878
 Call-ID:
 SBCbFgsQn5HcHpEUxJUVF8gQWVZREt6QwJvWVJkXCdAUzdkXl41CUJ4cEdWZV1NQ0tFYQ
 CSeq: 1
 Contact: sip:1001@192.168.0.33:5060
 Record-Route: sip:192.168.0.1;lr=on
 User-Agent: MAYAH 4.9.12.0-2.1.0.45
 Content-Length: 0
 
 And here the information from the opensips log files:
 
 Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]:
 ERROR:core:parse_cseq: no method found
 
 Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq:
 bad cseq
 
 Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]:
 ERROR:core:get_hdr_field: bad cseq
 
 Perhaps someone of you knows this failure or can tell me how to solve
 this problem?
 
 Cheers,
 
 Kai
 
 

Perhaps you have an ngrep of the sip traffic?

--fred

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Re: [SR-Users] Kamailio invite counter

[Fred Posner]

On 06/19/2015 09:54 AM, Ali Taher wrote:
 Hello,
  
 I want to define a counter  that increase in every invite.
 
 I tried the below :
 
 if (is_method(INVITE))
 
 { 
 
 $var(u)=$var(u)+1;
 
 xlog(L_INFO,Counter : $var(u)  );
 
 }
 
 
 But the counter is reset to 1 randomly (even before restarting kamailio
 service)
 
 
 How can have a proper counter that count invite messages.
 
 Thanks
 
 Ali
 

Have you considered using either Dialog module or HTABLE instead?

http://kamailio.org/docs/modules/stable/modules/dialog.html
http://kamailio.org/docs/modules/stable/modules/htable.html

I would start with htable...

--fred

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Re: [SR-Users] Kamailio license usage

[Fred Posner]

Kamailio  is released under GNU Public License v2 (GPLv2), which does
provide for commercial use. It's a very common licensing structure and
of course I am not a lawyer and this should not be taken as legal advise.

You can/should use the licensing information provided to have your
lawyer advise you as to your legal rights when conducting business use
specific to your unique application/service.


On 06/15/2015 12:09 PM, Alexandru Covalschi wrote:
 Maybe it may be an offtopic, but I'm not really into legal issues - so
 I'm sorry if this message is not fully related to this mailing list.
 
 Can I use Kamailio to provide VoIP backend for kind of CRM system in
 case of SaaS?
 
 ---
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/
 
 
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Re: [SR-Users] Removing parameters from user part of URI

[Fred Posner]

Even tel2sip shows it as a valid uri.

http://kamailio.org/docs/modules/4.2.x/modules/siputils.html#siputils.f.tel2sip

I again apologize.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

On 06/12/2015 02:52 PM, Alex Balashov wrote:
 
 On 06/12/2015 02:43 PM, Fred Posner wrote:
 
 I stand corrected.
 
 I'm not sure if these artefacts are an idiosyncrasy of conversion of
 tel: URIs to sip:, as 'cic' is also part of the ABNF layout in RFC 4694,
 which
 
The following syntax specification uses the Augmented Backus-Naur
Form (ABNF) as described in RFC 4234 [RFC4234] and defines the five
parameters, rn, npdi, cic, rn-context and cic-context, by extending
the parameter production rule of the tel URI defined in
[RFC3966].
 
 -- Alex
 

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Re: [SR-Users] Removing parameters from user part of URI

[Fred Posner]

I stand corrected.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

On 06/12/2015 02:41 PM, Alex Balashov wrote:
 Another example:
 
 http://www.dialogic.com/webhelp/IMG1010/10.3.3ER2/WebHelp/Description/Interworking/SIP_Carrier_Identification_Code.htm
 
 

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Re: [SR-Users] Removing parameters from user part of URI

[Fred Posner]

You could always to a search replace for the uri based on {uri.params}
(replacing with nothing).

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

On 06/12/2015 12:24 PM, Alex Balashov wrote:
 Sure, but that presumes that I know the position of the parameter and the 
 number of parameters. I'm looking for a generic approach to lob off all 
 parameters.
 
 --
 Alex Balashov | Principal | Evariste Systems LLC
 303 Perimeter Center North, Suite 300
 Atlanta, GA 30346
 United States
 
 Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
 Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
 
 Sent from my BlackBerry.
 
 
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Re: [SR-Users] Removing parameters from user part of URI

[Fred Posner]

 could, grammatically speaking, precede the user.

So now that I re-read your example... of a bad uri... I apologize for
not reading it well the first time.

 INVITE sip:4045551212;param=xyz@domain:port SIP/2.0

I think the only way to handle it would be with a regex; hopefully based
on parameters starting with a ; and ending with the last @.

The sed for it would be:

sed 's/^sip:\(.*\);.*@/\1@/'

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

On 06/12/2015 12:38 PM, Alex Balashov wrote:
 Could, but that would be a complicated regex since the parameters could, 
 grammatically speaking, precede the user. 
 
 --
 Alex Balashov | Principal | Evariste Systems LLC
 303 Perimeter Center North, Suite 300
 Atlanta, GA 30346
 United States
 
 Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
 Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
 
 Sent from my BlackBerry.
   Original Message  
 From: Juha Heinanen
 Sent: Friday, June 12, 2015 12:37
 To: Kamailio (SER) - Users Mailing List
 Reply To: Kamailio (SER) - Users Mailing List
 Subject: Re: [SR-Users] Removing parameters from user part of URI
 
 Alex Balashov writes:
 
 Sure, but that presumes that I know the position of the parameter and
 the number of parameters. I'm looking for a generic approach to lob
 off all parameters.
 
 ok, i just thought about your example. in that case perhaps you need to
 use one of regexp functions in textops module.
 
 -- juha
 
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Re: [SR-Users] Removing parameters from user part of URI

[Fred Posner]

My understanding of 4694 was that for SIP, it still references 3261,
which was :USERNAME:PASSWORD@HOSTPORT;PARAMS, no?

I thought the 4694 use of parameter was only applicable in a tel: or
similar field.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

On 06/12/2015 02:09 PM, Alex Balashov wrote:
 When you say bad URI... why? It's grammatically valid to put parameters in 
 the user part of the URI. If it weren't, it'd be hard to explain RFC 4694.
 
 --
 Alex Balashov | Principal | Evariste Systems LLC
 303 Perimeter Center North, Suite 300
 Atlanta, GA 30346
 United States
 
 Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
 Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
 
 Sent from my BlackBerry.
   Original Message  
 From: Fred Posner
 Sent: Friday, June 12, 2015 14:07
 To: sr-users@lists.sip-router.org
 Reply To: Kamailio (SER) - Users Mailing List
 Subject: Re: [SR-Users] Removing parameters from user part of URI
 
 could, grammatically speaking, precede the user.
 
 So now that I re-read your example... of a bad uri... I apologize for
 not reading it well the first time.
 
 INVITE sip:4045551212;param=xyz@domain:port SIP/2.0
 
 I think the only way to handle it would be with a regex; hopefully based
 on parameters starting with a ; and ending with the last @.
 
 The sed for it would be:
 
 sed 's/^sip:\(.*\);.*@/\1@/'
 
 Fred Posner
 The Palner Group, Inc.
 http://www.palner.com (web)
 +1-503-914-0999 (direct)
 
 On 06/12/2015 12:38 PM, Alex Balashov wrote:
 Could, but that would be a complicated regex since the parameters could, 
 grammatically speaking, precede the user. 

 --
 Alex Balashov | Principal | Evariste Systems LLC
 303 Perimeter Center North, Suite 300
 Atlanta, GA 30346
 United States

 Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
 Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

 Sent from my BlackBerry.
 Original Message 
 From: Juha Heinanen
 Sent: Friday, June 12, 2015 12:37
 To: Kamailio (SER) - Users Mailing List
 Reply To: Kamailio (SER) - Users Mailing List
 Subject: Re: [SR-Users] Removing parameters from user part of URI

 Alex Balashov writes:

 Sure, but that presumes that I know the position of the parameter and
 the number of parameters. I'm looking for a generic approach to lob
 off all parameters.

 ok, i just thought about your example. in that case perhaps you need to
 use one of regexp functions in textops module.

 -- juha

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Re: [SR-Users] Need help to configure rtpporyx

[Fred Posner]

On 06/03/2015 12:10 AM, Mir Jee wrote:
 Hi!
 Please help me to configure rtpproxy on centos + kamailio.
 

What type of trouble are you having?

I have a tutorial written for installing an older version of Kamailio
and RTPproxy on CentOS, but this is easily changed to the current version...

http://www.fredposner.com/1457/kamailio-behind-nat/

The only changes I would make would be to use the stable version of
Kamailio (or 4.3 at this point)... and install rtpproxy 2 from git.


Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)

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Re: [SR-Users] rtpproxy 2.0

[Fred Posner]

Have tested it on virtual and physical. Works well, no need to patch for VM or 
advertised address. Had no complaints from users with 1.2 and none since 2.0 

Installed from git for testing. 

-- Fred

 On May 22, 2015, at 4:37 AM, Klaus Darilion klaus.mailingli...@pernau.at 
 wrote:
 
 Hi!
 
 I just found out that there is a new rtpproxy release:
 http://www.rtpproxy.org/post/v2release/
 
 Has anybody tested it and want to share some experiences? Or have people
 turned to rtpengine meanwhile? (I am still using rtpproxy 1.2 as I do
 not need new features).
 
 regards
 Klaus
 
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Re: [SR-Users] Setup kamalio at AWS server

[Fred Posner]

I sent an answer to this same question from you yesterday.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

On 05/06/2015 10:44 AM, Yogendra Gupta wrote:
 Hello,
 
 I have setup kamailio at AWS server and it is connecting at private IP
 not public ip.
 
 I am getting following error:
 
  
 
 ERROR: core [udp_server.c:392]: udp_init(): bind(7, 0x7f9eca5f11bc,
 16) on 54.149.7.246: Cannot assign requested address
 
  
 
 Let me know where I need to fixed it
 
  
 
 Thanks
 
  
 
 *From:*Yogendra Gupta [mailto:yogen...@konstantinfosolutions.com]
 *Sent:* Tuesday, April 07, 2015 10:45 AM
 *To:* 'Kamailio (SER) - Users Mailing List'; 'Daniel-Constantin Mierla'
 *Subject:* RE: Kamailio audio call not working at some DNS network
 
  
 
 Hello,
 
  
 
 We have added domain and now call is connected but audio is not coming.
 
 What can be issue with audio call?
 
  
 
 Following is details:
 
  
 
 Username : user2
 
 Domain : mobileappplatform.com
 
 Password : 123456
 
  
 
 Let me know what can be issue?
 
  
 
 thanks
 
  
 
 *From:*Yogendra Gupta [mailto:yogen...@konstantinfosolutions.com]
 *Sent:* Wednesday, April 01, 2015 7:57 PM
 *To:* 'Kamailio (SER) - Users Mailing List'; 'Daniel-Constantin Mierla'
 *Subject:* Kamailio audio call not working at some DNS network
 
  
 
 Hello,
 
  
 
 We have setup Kamailio (SIP) at linux ubuntu server and it is working
 fine for call and text chat with users that are registered at server for
 some DNS networks.
 
 Audio call is not working at some DNS and chat is working at same DNS.
 
  
 
 We have already disabled firewall at server.
 
 What can be problem at server.
 
  
 
 I have also attached config file .. please check if there is any error.
 
  
 
 Let us know any communication option so we can sort out this issue at
 our server.
 
  
 
 Thanks
 
  
 
 *From:*sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
 Behalf Of *jay binks
 *Sent:* Wednesday, April 01, 2015 6:16 PM
 *To:* Daniel-Constantin Mierla; Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Make deb
 
  
 
 Sorry, this confused me a little..
 
 can you give an example of what you mean ?
 
  
 
 On 1 April 2015 at 22:40, Daniel-Constantin Mierla mico...@gmail.com
 mailto:mico...@gmail.com wrote:
 
 Hello,
 
 you have to edit spec files and remove the other modules in the folder
 matching your debian distribution inside 'pkg/kamailio/deb/'. This is
 the only way I know to change the packaging content for debs. Then make
 a symlink inside root folder of kamailio server named 'debian' to the
 respective folder inside 'pkg/kamailio/deb/' and run 'make deb'.
 
 Cheers,
 Daniel
 
  
 
 On 01/04/15 13:50, jay binks wrote:
 
 is there any way to make deb packages for only 1 of the modules ??
 
 currently it takes a long time to run make deb 
 
  
 
 im simply trying to build and package db_cassandra ( need the
 package for my environment )
 
  
 
 As a side note, is there any chance db_cassandra could be added to
 the kamailio deb packages ?
 
  
 
 -- 
 
 Sincerely
 
 Jay
 
  
 
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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
 
 
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 -- 
 
 Sincerely
 
 Jay
 
 
 
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Re: [SR-Users] Setup kamalio at AWS server

[Fred Posner]

On 05/06/2015 10:44 AM, Yogendra Gupta wrote:
 Hello,
 
 I have setup kamailio at AWS server and it is connecting at private IP
 not public ip.
 
 I am getting following error:
 
  
 
 ERROR: core [udp_server.c:392]: udp_init(): bind(7, 0x7f9eca5f11bc,
 16) on 54.149.7.246: Cannot assign requested address
 
  
 
 Let me know where I need to fixed it
 
  
 
 Thanks
 

In your LISTEN statement, you probably have an IP that is not on the system.

Try something like:

listen=udp:X.X.X.X:5060 advertise Y.Y.Y.Y:5060

where x is your private ip and y is your public ip.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)


 
 *From:*Yogendra Gupta [mailto:yogen...@konstantinfosolutions.com]
 *Sent:* Tuesday, April 07, 2015 10:45 AM
 *To:* 'Kamailio (SER) - Users Mailing List'; 'Daniel-Constantin Mierla'
 *Subject:* RE: Kamailio audio call not working at some DNS network
 
  
 
 Hello,
 
  
 
 We have added domain and now call is connected but audio is not coming.
 
 What can be issue with audio call?
 
  
 
 Following is details:
 
  
 
 Username : user2
 
 Domain : mobileappplatform.com
 
 Password : 123456
 
  
 
 Let me know what can be issue?
 
  
 
 thanks
 
  
 
 *From:*Yogendra Gupta [mailto:yogen...@konstantinfosolutions.com]
 *Sent:* Wednesday, April 01, 2015 7:57 PM
 *To:* 'Kamailio (SER) - Users Mailing List'; 'Daniel-Constantin Mierla'
 *Subject:* Kamailio audio call not working at some DNS network
 
  
 
 Hello,
 
  
 
 We have setup Kamailio (SIP) at linux ubuntu server and it is working
 fine for call and text chat with users that are registered at server for
 some DNS networks.
 
 Audio call is not working at some DNS and chat is working at same DNS.
 
  
 
 We have already disabled firewall at server.
 
 What can be problem at server.
 
  
 
 I have also attached config file .. please check if there is any error.
 
  
 
 Let us know any communication option so we can sort out this issue at
 our server.
 
  
 
 Thanks
 
  
 
 *From:*sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
 Behalf Of *jay binks
 *Sent:* Wednesday, April 01, 2015 6:16 PM
 *To:* Daniel-Constantin Mierla; Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Make deb
 
  
 
 Sorry, this confused me a little..
 
 can you give an example of what you mean ?
 
  
 
 On 1 April 2015 at 22:40, Daniel-Constantin Mierla mico...@gmail.com
 mailto:mico...@gmail.com wrote:
 
 Hello,
 
 you have to edit spec files and remove the other modules in the folder
 matching your debian distribution inside 'pkg/kamailio/deb/'. This is
 the only way I know to change the packaging content for debs. Then make
 a symlink inside root folder of kamailio server named 'debian' to the
 respective folder inside 'pkg/kamailio/deb/' and run 'make deb'.
 
 Cheers,
 Daniel
 
  
 
 On 01/04/15 13:50, jay binks wrote:
 
 is there any way to make deb packages for only 1 of the modules ??
 
 currently it takes a long time to run make deb 
 
  
 
 im simply trying to build and package db_cassandra ( need the
 package for my environment )
 
  
 
 As a side note, is there any chance db_cassandra could be added to
 the kamailio deb packages ?
 
  
 
 -- 
 
 Sincerely
 
 Jay
 
  
 
 ___
 
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 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
 
 
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 -- 
 
 Sincerely
 
 Jay
 
 
 
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Re: [SR-Users] Kamailio BLF Issue TCP vs UDP

[Fred Posner]

Yes, record route is being generated for all SUBSCRIBE, NOTIFY, and just
in case REFER, INFO, PUBLISH.

--fred

On 03/27/2015 06:38 AM, Daniel-Constantin Mierla wrote:
 202 is ok, so freeswitch has created the subscription dialog and should
 send notify requests with event dialog.
 
 As I can see in the traces, traffic is TCP to Kamailio and UDP in
 between Kamailio and FreeSwitch. Are you doing record_route() for all
 SUBSCRIBE and NOTIFY requests?
 
 Cheers,
 Daniel
 
 On 27/03/15 00:23, Fred Posner wrote:
 For both UDP and TCP I receive a 202.

 --fred

 On 03/26/2015 06:45 PM, Daniel-Constantin Mierla wrote:
 Hello,

 actually there is nothing wrong with the NOTIFY via TCP. But it is not
 for Event: dialog (which is for BLF), it is for Event: refer (which is
 for other purposes, having Content-Type: message/sipfrag -- see
 https://www.ietf.org/rfc/rfc3420.txt).

 You have to see why freeswitch is not sending the notify for Event:
 dialog. Does it send the 200ok for the SUBSCRIBE to event dialog?

 Cheers,
 Daniel

 On 26/03/15 23:33, Fred Posner wrote:
 Greetings all:

 Having a weird issue with BLF relay from Kamailio - FreeSWITCH on the
 EC2 network.

 Set-up:

 Natted Client - Natted Kamailio - Natted FreeSWITCH

 Kamailio has a public advertised IP (Amazon Cloud) and sends to
 FreeSWITCH's public IP (Amazon Cloud). Clients register to Kamailio and
 the registration sent to FreeSWITCH (via uac).

 What works:

 When a client registers to kamailio via UDP, all works beautifully.

 What doesn't work:

 When a client registers to kamailio via TCP, a NOTIFY is sent with
 almost no information.

 Example of UDP traffic:

 U 2015/03/26 22:12:30.483767 FREESWITCH_IPADDR:5060 -
 KAMAILIO_NAT_IPADDR:5060
 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0.
 Via: SIP/2.0/UDP FREESWITCH_IPADDR;rport;branch=z9hG4bKv7X88FmKg04UF.
 Route: sip:KAMAILIO_PUBLIC_IPADDR;lr=on;ftag=xew5nmpr9q;nat=yes.
 Record-Route: sip:KAMAILIO_PUBLIC_IPADDR;lr=on;ftag=xew5nmpr9q;nat=yes.
 Max-Forwards: 70.
 From: sip:3000@CLIENT_DOMAIN;user=phone;tag=YF4HMSQNlOOt.
 To: sip:3000@CLIENT_DOMAIN;tag=xew5nmpr9q.
 Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj.
 CSeq: 366877453 NOTIFY.
 Contact: sip:3000@FREESWITCH_IPADDR:5060.
 User-Agent:
 FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
 REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
 Supported: timer, path, replaces.
 Event: dialog.
 Allow-Events: talk, hold, conference, presence, as-feature-event,
 dialog, line-seize, call-info, sla, include-session-description,
 presence.winfo, message-summary, refer.
 Subscription-State: active;expires=2686.
 Content-Type: application/dialog-info+xml.
 Content-Length: 546.
 .
 ?xml version=1.0?
 dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=14
 state=full entity=sip:3000@CLIENT_DOMAIN
 dialog id=312b3bee-d405-11e4-a8b6-41b65d0bd045 direction=initiator
 stateconfirmed/state
 local
 identity display=3000sip:3000@CLIENT_DOMAIN/identity
 target uri=sip:3000@CLIENT_DOMAIN
 param pname=+sip.rendering pvalue=yes/
 /target
 /local
 remote
 identity display=*98sip:*98@CLIENT_DOMAIN/identity
 target uri=sip:**3000@CLIENT_DOMAIN/
 /remote
 /dialog
 /dialog-info


 U 2015/03/26 22:12:30.483877 KAMAILIO_NAT_IPADDR:5060 -
 CLIENT_PUBLIC_IPADDR:42477
 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0.
 Via: SIP/2.0/UDP
 KAMAILIO_PUBLIC_IPADDR:5060;branch=z9hG4bKfabf.4908255bf1623d250ddf7d51a3b49fb8.0.
 Via: SIP/2.0/UDP
 FREESWITCH_IPADDR;received=FREESWITCH_IPADDR;rport=5060;branch=z9hG4bKv7X88FmKg04UF.
 Record-Route: sip:KAMAILIO_PUBLIC_IPADDR;lr=on;ftag=xew5nmpr9q;nat=yes.
 Max-Forwards: 69.
 From: sip:3000@CLIENT_DOMAIN;user=phone;tag=YF4HMSQNlOOt.
 To: sip:3000@CLIENT_DOMAIN;tag=xew5nmpr9q.
 Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj.
 CSeq: 366877453 NOTIFY.
 Contact: sip:3000@FREESWITCH_IPADDR:5060.
 User-Agent:
 FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
 REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
 Supported: timer, path, replaces.
 Event: dialog.
 Allow-Events: talk, hold, conference, presence, as-feature-event,
 dialog, line-seize, call-info, sla, include-session-description,
 presence.winfo, message-summary, refer.
 Subscription-State: active;expires=2686.
 Content-Type: application/dialog-info+xml.
 Content-Length: 546.
 .
 ?xml version=1.0?
 dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=14
 state=full entity=sip:3000@CLIENT_DOMAIN
 dialog id=312b3bee-d405-11e4-a8b6-41b65d0bd045 direction=initiator
 stateconfirmed/state
 local
 identity display=3000sip:3000@CLIENT_DOMAIN/identity
 target uri=sip:3000@CLIENT_DOMAIN
 param pname=+sip.rendering pvalue=yes/
 /target
 /local
 remote
 identity display=*98sip:*98@CLIENT_DOMAIN/identity
 target uri=sip:**3000@CLIENT_DOMAIN/
 /remote
 /dialog
 /dialog-info


 Example

Re: [SR-Users] Kamailio BLF Issue TCP vs UDP

[Fred Posner]

I added the record route for SUBSCRIBE, UPDATE, INFO, PUBLISH, NOTIFY
when there were some issues with FreeSWITCH trying to bypass Kamailio...

That being said, when this was discussed on the FreeSWITCH channel, the
FreeSWITCH users flipped out over record-route. I removed it, had the
same results, channel went silent.

I've added it back, partly out of spite, but mostly for the proper
handling.

--fred

On 03/27/2015 06:47 AM, Alex Balashov wrote:
 I suppose this speaks to it:
 
 https://tools.ietf.org/html/rfc6665#appendix-B.19
 

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[SR-Users] Kamailio BLF Issue TCP vs UDP

[Fred Posner]

/2.0 200 OK.


T 2015/03/26 22:19:45.963407 KAMAILIO_NAT_IPADDR:5160 -
CLIENT_PUBLIC_IPADDR:5065 [AP]
NOTIFY sip:3001@CLIENT_PUBLIC_IPADDR:5065;transport=tcp SIP/2.0.
Via: SIP/2.0/TCP
KAMAILIO_PUBLIC_IPADDR:5160;branch=z9hG4bKe961.8d1f19f1acc0295e5e397afa76910115.0.
Via: SIP/2.0/UDP
FREESWITCH_IPADDR;received=FREESWITCH_IPADDR;rport=5060;branch=z9hG4bKjH5Nt8QKUU9Sp.
Max-Forwards: 69.
From: 3000 sip:3000@CLIENT_DOMAIN;tag=9jSFr02tgvHym.
To: sip:3001@CLIENT_DOMAIN;tag=1dba0f3ceb32436bi0.
Call-ID: 0882b1bc-4ea9-1233-f3a2-12e513dfd8f0.
CSeq: 73368127 NOTIFY.
Contact: sip:mod_sofia@FREESWITCH_IPADDR:5060.
User-Agent:
FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, path, replaces.
Event: refer;id=101.
Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer.
Subscription-State: terminated;reason=noresource.
Content-Type: message/sipfrag;version=2.0.
Content-Length: 16.
.
SIP/2.0 200 OK.


My thoughts is that this is on FreeSWITCH, just not sending a NOTIFY
with anything of value. That being said, I'd love to know if (a) anyone
else has had this issue or (b) if someone has a different theory.

Thanks!

-- 
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)


0x6235BD69.asc
Description: application/pgp-keys
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Re: [SR-Users] Kamailio BLF Issue TCP vs UDP

[Fred Posner]

For both UDP and TCP I receive a 202.

--fred

On 03/26/2015 06:45 PM, Daniel-Constantin Mierla wrote:
 Hello,
 
 actually there is nothing wrong with the NOTIFY via TCP. But it is not
 for Event: dialog (which is for BLF), it is for Event: refer (which is
 for other purposes, having Content-Type: message/sipfrag -- see
 https://www.ietf.org/rfc/rfc3420.txt).
 
 You have to see why freeswitch is not sending the notify for Event:
 dialog. Does it send the 200ok for the SUBSCRIBE to event dialog?
 
 Cheers,
 Daniel
 
 On 26/03/15 23:33, Fred Posner wrote:
 Greetings all:

 Having a weird issue with BLF relay from Kamailio - FreeSWITCH on the
 EC2 network.

 Set-up:

 Natted Client - Natted Kamailio - Natted FreeSWITCH

 Kamailio has a public advertised IP (Amazon Cloud) and sends to
 FreeSWITCH's public IP (Amazon Cloud). Clients register to Kamailio and
 the registration sent to FreeSWITCH (via uac).

 What works:

 When a client registers to kamailio via UDP, all works beautifully.

 What doesn't work:

 When a client registers to kamailio via TCP, a NOTIFY is sent with
 almost no information.

 Example of UDP traffic:

 U 2015/03/26 22:12:30.483767 FREESWITCH_IPADDR:5060 -
 KAMAILIO_NAT_IPADDR:5060
 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0.
 Via: SIP/2.0/UDP FREESWITCH_IPADDR;rport;branch=z9hG4bKv7X88FmKg04UF.
 Route: sip:KAMAILIO_PUBLIC_IPADDR;lr=on;ftag=xew5nmpr9q;nat=yes.
 Record-Route: sip:KAMAILIO_PUBLIC_IPADDR;lr=on;ftag=xew5nmpr9q;nat=yes.
 Max-Forwards: 70.
 From: sip:3000@CLIENT_DOMAIN;user=phone;tag=YF4HMSQNlOOt.
 To: sip:3000@CLIENT_DOMAIN;tag=xew5nmpr9q.
 Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj.
 CSeq: 366877453 NOTIFY.
 Contact: sip:3000@FREESWITCH_IPADDR:5060.
 User-Agent:
 FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
 REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
 Supported: timer, path, replaces.
 Event: dialog.
 Allow-Events: talk, hold, conference, presence, as-feature-event,
 dialog, line-seize, call-info, sla, include-session-description,
 presence.winfo, message-summary, refer.
 Subscription-State: active;expires=2686.
 Content-Type: application/dialog-info+xml.
 Content-Length: 546.
 .
 ?xml version=1.0?
 dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=14
 state=full entity=sip:3000@CLIENT_DOMAIN
 dialog id=312b3bee-d405-11e4-a8b6-41b65d0bd045 direction=initiator
 stateconfirmed/state
 local
 identity display=3000sip:3000@CLIENT_DOMAIN/identity
 target uri=sip:3000@CLIENT_DOMAIN
 param pname=+sip.rendering pvalue=yes/
 /target
 /local
 remote
 identity display=*98sip:*98@CLIENT_DOMAIN/identity
 target uri=sip:**3000@CLIENT_DOMAIN/
 /remote
 /dialog
 /dialog-info


 U 2015/03/26 22:12:30.483877 KAMAILIO_NAT_IPADDR:5060 -
 CLIENT_PUBLIC_IPADDR:42477
 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0.
 Via: SIP/2.0/UDP
 KAMAILIO_PUBLIC_IPADDR:5060;branch=z9hG4bKfabf.4908255bf1623d250ddf7d51a3b49fb8.0.
 Via: SIP/2.0/UDP
 FREESWITCH_IPADDR;received=FREESWITCH_IPADDR;rport=5060;branch=z9hG4bKv7X88FmKg04UF.
 Record-Route: sip:KAMAILIO_PUBLIC_IPADDR;lr=on;ftag=xew5nmpr9q;nat=yes.
 Max-Forwards: 69.
 From: sip:3000@CLIENT_DOMAIN;user=phone;tag=YF4HMSQNlOOt.
 To: sip:3000@CLIENT_DOMAIN;tag=xew5nmpr9q.
 Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj.
 CSeq: 366877453 NOTIFY.
 Contact: sip:3000@FREESWITCH_IPADDR:5060.
 User-Agent:
 FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit.
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
 REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
 Supported: timer, path, replaces.
 Event: dialog.
 Allow-Events: talk, hold, conference, presence, as-feature-event,
 dialog, line-seize, call-info, sla, include-session-description,
 presence.winfo, message-summary, refer.
 Subscription-State: active;expires=2686.
 Content-Type: application/dialog-info+xml.
 Content-Length: 546.
 .
 ?xml version=1.0?
 dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=14
 state=full entity=sip:3000@CLIENT_DOMAIN
 dialog id=312b3bee-d405-11e4-a8b6-41b65d0bd045 direction=initiator
 stateconfirmed/state
 local
 identity display=3000sip:3000@CLIENT_DOMAIN/identity
 target uri=sip:3000@CLIENT_DOMAIN
 param pname=+sip.rendering pvalue=yes/
 /target
 /local
 remote
 identity display=*98sip:*98@CLIENT_DOMAIN/identity
 target uri=sip:**3000@CLIENT_DOMAIN/
 /remote
 /dialog
 /dialog-info


 Example of TCP failure:

 U 2015/03/26 22:19:45.963196 FREESWITCH_IPADDR:5060 -
 KAMAILIO_NAT_IPADDR:5160
 NOTIFY sip:3001@CLIENT_PUBLIC_IPADDR:5065;transport=tcp SIP/2.0.
 Via: SIP/2.0/UDP FREESWITCH_IPADDR;rport;branch=z9hG4bKjH5Nt8QKUU9Sp.
 Route:
 sip:KAMAILIO_PUBLIC_IPADDR:5160;r2=on;lr=on;ftag=9jSFr02tgvHym;nat=yes.
 Route:
 sip:KAMAILIO_PUBLIC_IPADDR:5160;transport=tcp;r2=on;lr=on;ftag=9jSFr02tgvHym;nat=yes.
 Max-Forwards: 70.
 From: 3000 sip:3000@CLIENT_DOMAIN;tag=9jSFr02tgvHym.
 To: sip:3001@CLIENT_DOMAIN;tag

Re: [SR-Users] Kamailio Security Policy - How to handle vulnerability reports

[Fred Posner]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 02/25/2015 12:14 PM, Olle E. Johansson wrote:
 
 On 25 Feb 2015, at 17:24, Daniel Tryba d.tr...@pocos.nl wrote:
 
 On Wednesday 25 February 2015 16:14:43 Olle E. Johansson wrote:
 http://www.kamailio.org/wiki/securitypolicy


 We encourage your feedback!

 - Is this a good thing for the project?

 Yes

 - Do you have any changes to the policy to suggest?

 Yes:

 secur...@kamailio.org
 This address should have a PGP key associated, used by the security 
 officers.

 This is a security nightmare (a (for all purposes) shared private key).

 You might want to look at the Debian security announces, there the 
 individuals
 key is used for signing and the list filters on valid keys from individuals.
 https://www.debian.org/security/faq#signature
 This makes it a little more difficult to check if an announcement is actually
 from the list:
 -get key for fingerprint in mail
 -check key with currect securitylist member
 Thank you for the feedback!
 

 But I fail to see how a pgp key for security is really important. Is there a
 PKI for kamailio releases? http://www.kamailio.org/pub/kamailio/latest/src/
 contains the latest version, but there is no way to verify if this is really
 the latest release. No ssl, no dnssec, no signed checksums. These should be
 considered also.
 
 I would love seeing signatures on releases. I think there's a key for the RPM
 packages somewhere.
 
 /O

+1 on all points.


Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1

iQEcBAEBAgAGBQJU7geGAAoJEIvgPjxiNb1paTQH/iE2N47s4Iz44GgA8u+1RGsp
/OsUw80soI+u+Yu+m4Zp0qpn2ZZHbDgIqA7F79s2rwo7I6XfdT/ehITCjC9KZcTs
UpPymi8+JDT6EugbQPf7dBoI6Jwu9Hxq3OcRBQtRum0JWbuEXMy5YYLZwCPjmrt/
sOkxbJ4mZcMoaY0JtfbSk1U3KrCsHenngCaRnPhbKlw4vm7GNxeOpK+cNRSqYMPN
Xzss/Q8wd5f8OyjVOzydVBCUDKRP49/9YMfbfQhQVHi4V7xjuU6tVSteLcn0hUqc
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=imCz
-END PGP SIGNATURE-

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Re: [SR-Users] Kamailio

[Fred Posner]

Tom,

This list is for users, but we do have a list just for business requests:

http://lists.kamailio.org/cgi-bin/mailman/listinfo/business

We also have a business directory:

http://www.kamailio.org/w/business-directory/

You'll have great luck with both of the above.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

On 02/19/2015 03:30 AM, Tomasz Gass wrote:

Good morning,

I'm looking for someone who, according to my guidelines set kamailio and
connect the asterisk.
Kamailio to act as a proxy server to accept registrations sip and locate.
Pay for the correct configuration.
Please contact me with people interested.

Regards
Tom


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Re: [SR-Users] One Way RTP from destination SIP server using rtpengine

[Fred Posner]

Do you have any of the sip traffic?

Also, are you using FreeSWITCH for the media of WSS?

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

On 02/05/2015 09:38 PM, Don Fanning wrote:

Hello All,

I currently am running Kamailio in a WSS configuration with sipML5.  I
use rtpengine to convert a RTP/SAVPF packet to a RTP/AVP packet as the
destination server only supports AVP or SAVP.

RTPEngine has no issues rewriting the packet going out, the SIP session
comes up and handshakes correctly to start the session.  Then the remote
server sends the RTP stream back and I'm having issues getting Kamailio
or RTPEngine or something to take the RTP stream, process it back to
SAVPF and send it back out the WSS port.  I see no RTP data come from
Kam/RTPengine going towards the remote server.

On the server side, I do see STUN being called and the sipML5 bind
successfully but that's it.

Both the destination and server sit without firewalls in the way so the
problem has to be what I'm doing in Kamailio.

My config is located here:
http://pastebin.com/dWLdUz5j

Packet dumps available upon request.

Thanks for any assistance!



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Re: [SR-Users] kamailio passthrough user auth

[Fred Posner]

Hello Kai,

Many examples of this question exist on the mail list if you search 
through it.


Also, I believe this is an example of what you might want to look for:

https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg


Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

On 02/05/2015 06:43 AM, Kai Arif wrote:

Hi,

I have to implement a rather uncommon SIP setup and wanted to ask, if it is 
even possible with
Kamailio.
We are among other things a VoIP-provider for a couple of hundred customers. 
Right now we use a
rather old piece of software as a SIP-server, that was written in-house and is 
deeply integrated in
our user interfaces and the billing system and can't be replaced easily.
But now we need to make some modifications because our PSTN-gateway-provider is 
changing things.
The idea is to use Kamailio as proxy in front of our old SIP-server. But to do 
this we need to pass
through the REGISTER requests of our users to the old SIP-server.
So the question is if Kamailio supports this kind of passthrough and I would 
also be rely grateful
for a piece of example config, if somebody already has a similar setup running.
So far neither the documentation nor google have been very helpful.

Best regards
Kai



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Re: [SR-Users] How to install kamailio-4.2.2 in DEBIAN -7.1.0

[Fred Posner]

On 02/05/2015 06:18 PM, Yanko Marín Muro wrote:

Hello:

How to install kamailio-4.2.2 in DEBIAN -7.1.0-amd64?

This URL is the last version:
http://www.kamailio.org/pub/kamailio/4.2.2/src/kamailio-4.2.2_src.tar.gz

Best regards,

Yanko Antonio Marín Muro



Hi Yanko,

This url will give you step-by-step instructions for installing via git, 
which is the recommended method.


http://www.kamailio.org/wiki/install/4.2.x/git

You can copy / paste most of these commands to do a fairly quick install.

Regards,

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)

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Re: [SR-Users] kamailio for newbies!

[Fred Posner]

On 12/01/2014 08:35 AM, Aria Mill wrote:

Hello,

Kamailio is difficult. Many of us want to use it because is open source
and it's flexible. but to tell you the truth after 3-4 hours of playing
with it I am frustrated, and I am starting to hate it!


I hate to hear that you are starting to hate the software... many of us 
truly love it. I find one correction to your main statement.


SIP is difficult. Kamailio is flexible. The more you understand and know 
SIP, the easier using Kamailio becomes.




Why has nobody made a kamailio video series on youtube?
... snip ...
This is 2014, people don’t waste time reading
documentation, youtube is much more efficient...

... snip 



I for one have almost zero interest in making videos of configuring 
software. Plus, I have a face for radio. ;)


Some of us make tutorials... I have made two of them, and they get a lot 
of readers; so I do think it's not a waste of time.


If you're good at making videos, this would be a great way for you to 
contribute to the software.


The vast majority of asterisk videos I see aren't made by Digium or the 
Asterisk devs... but instead by either hardware companies selling 
asterisk based hardware, enthusiasts, or support vendors.


Again, this might be a great way for you to contribute.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)

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Re: [SR-Users] RFC: infrastructure upgrade - git, tracker, ...

[Fred Posner]

With no private repositories, gitlab is free which is nice. 

---Fred 


 On Nov 9, 2014, at 1:20 PM, Jan Janak j...@janakj.org wrote:
 
 On Thu, Nov 6, 2014 at 11:40 AM, Jan Janak j...@janakj.org wrote:
 If you prefer to keep a self-hosted git repository, I think we should at 
 least move it to gitolite:
 http://gitolite.com/
 
 To make it more manageable (doesn't require ssh access for individual 
 people).
 
 Gitlab is another potential candidate for a self-hosted repository:
 
 https://about.gitlab.com/
 
 (seems to provide more than gitolite).
 
 -Jan
  
 
 On Wed, Nov 5, 2014 at 9:25 AM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:
 Hello,
 
 as most of you know, we have a rather distributed infrastructure, with
 servers provided from different companies or persons.
 
 We came to the time when one of the servers is too old and considered to
 be decommisioned, so we have to decide how to move on. It is about
 sip-router.org, who was offered/sponsored by courtesy of Jan Janak. From
 project point of view, the server hosts:
 
 - git repository
 - bug tracker
 - website and wiki under domain sip-router.org
 
 Given that no matter what we like, there is work to do, I am looking to
 see what are the best options out there for everyone involved in the
 project.
 
 The sip-router.or wbesite and wiki, which are not really updated anyhow,
 will be relocated as virtual host in kamailio.org server and made static
 for historic purposes.
 
 For git and tracker, I thought of two variants:
 
 1) move to use an external hosting service - the first candidate and
 perhaps the only to be considered is GitHub, we have there already a
 real time mirror of git repository. Then we should get a read only
 mirror to kamailio.org. If the tracker on github is good enough for
 everyone, then we will use it. I could see quickly that lot of kamailio
 developers already have an account on github.
 
 2) get a new server and relocate those components there. It will need to
 be configured from scratch and the components eventually updated to use
 latest versions. In case of tracker, we have eventually to re-evaluate
 if flyspray worth keeping, as we had several discussions, due to the
 fact that the project doesn't seem to be very active.
 
 My personal preference at this moment is 1), given that offloads
 administration works from project.
 
 For 2) we will need someone to commit to a sysadmin job for long time.
 As probably you noticed lately, serious security vulnerabilities can
 appear and someone needs to take care of proper maintenance of the
 server. I don't want to get all services on kamailio.org, as it has
 other critical components (mailing lists, releases, website, ...) and
 messing it or overloading doesn't make sense.
 
 There is no real pressure to come to a decision, we can still rely on
 the server for a while, but I would rather not postpone it for long.
 
 While users are encouraged to give their opinion, I feel that existing
 developers should have the main role in decision, being something that
 impacts them directly.
 
 Your preference? Any other opinions?
 
 Cheers,
 Daniel
 
 --
 Daniel-Constantin Mierla
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
 Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com
 
 
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Re: [SR-Users] RFC: infrastructure upgrade - git, tracker, ...

[Fred Posner]

I'm certain that LOD would be willing to sponsor the server for git / 
tracker and I'd offer to handle the sysadmin of the server.


Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)

On 11/05/2014 09:25 AM, Daniel-Constantin Mierla wrote:

Hello,

as most of you know, we have a rather distributed infrastructure, with
servers provided from different companies or persons.

We came to the time when one of the servers is too old and considered to
be decommisioned, so we have to decide how to move on. It is about
sip-router.org, who was offered/sponsored by courtesy of Jan Janak. From
project point of view, the server hosts:

- git repository
- bug tracker
- website and wiki under domain sip-router.org

Given that no matter what we like, there is work to do, I am looking to
see what are the best options out there for everyone involved in the
project.

The sip-router.or wbesite and wiki, which are not really updated anyhow,
will be relocated as virtual host in kamailio.org server and made static
for historic purposes.

For git and tracker, I thought of two variants:

1) move to use an external hosting service - the first candidate and
perhaps the only to be considered is GitHub, we have there already a
real time mirror of git repository. Then we should get a read only
mirror to kamailio.org. If the tracker on github is good enough for
everyone, then we will use it. I could see quickly that lot of kamailio
developers already have an account on github.

2) get a new server and relocate those components there. It will need to
be configured from scratch and the components eventually updated to use
latest versions. In case of tracker, we have eventually to re-evaluate
if flyspray worth keeping, as we had several discussions, due to the
fact that the project doesn't seem to be very active.

My personal preference at this moment is 1), given that offloads
administration works from project.

For 2) we will need someone to commit to a sysadmin job for long time.
As probably you noticed lately, serious security vulnerabilities can
appear and someone needs to take care of proper maintenance of the
server. I don't want to get all services on kamailio.org, as it has
other critical components (mailing lists, releases, website, ...) and
messing it or overloading doesn't make sense.

There is no real pressure to come to a decision, we can still rely on
the server for a while, but I would rather not postpone it for long.

While users are encouraged to give their opinion, I feel that existing
developers should have the main role in decision, being something that
impacts them directly.

Your preference? Any other opinions?

Cheers,
Daniel



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Re: [SR-Users] Failed to install Kamailio database

[Fred Posner]

You will need to install mysql if you would like to use a mysql 
database. It is not required that you use mysql. Other databases are 
supported as well as a database not being a requirement for the software.


Fred Posner

On 10/24/2014 08:57 PM, Mahmoud Ramadan Ali wrote:

No ! i do not have mysql installed...does the script will install it
for me or i should install it previously ?

On 10/24/14, Fred Posner f...@palner.com wrote:

Do you have mysql installed?

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)

On 10/24/2014 08:52 PM, Mahmoud Ramadan Ali wrote:

Hiii everyone,

I can not create kamailio database and get this error message...

any ideas ?

Thanks in advance...


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Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX

[Fred Posner]

If you want to call a user on Kamailio from Asterisk...

example...

exten = s,1,Verbose(4,calling user on kamailio)
 same = n,Dial(SIP/USERNAME@KAMAILIO,time,options)
 same = n,--after dial logic --


Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)

On 10/23/2014 11:21 AM, Kenny Watson wrote:

Hi Fred,

Thanks for the quick response.  I already do use some Kamailio
features on our internal network for load balancing.

The use case that I'm interested in is to effectively replace an
asterisk server that I use for SIP trunking to remote phone systems
with a Kamailio registrar/proxy and a bank of asterisk servers
placing calls direct to extensions on the remote PBX.

I currently have this running on asterisk which I route to the
different remote PBX extensions using prefix based routing down to
the destination peer on asterisk which is essentially what I need to
replicate on Kamailio.

i.e.

2021 routes to   @remotepbx1

remotepbx1 maybe defined as either by IP address or via a normal
registered sip peer with a username/password combo.


I understand that I can dial a registered device directly but its how
to call a remote extension on a registered device via Kamailio.

Thanks Kenny Watson





-Original Message- From:
sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Fred
Posner Sent: 23 October 2014 16:00 To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX

Hi Kenny,

This depends on the carriers and scenarios that you may use. I know
depends is a horrible answer, but one of the great aspects of
Kamailio is the flexibility of the modules.

Some deployments may have a group of Asterisk servers all configured
similarly for handling calls. With this type of scenario, you would
benefit from using the dispatcher module.

Many people like to use Kamailio on the public side of their network
and keep their asterisk servers on the private. This would be an
example of when to use rtpproxy (in bridge mode).

Some carriers hate seeing the chain of systems on your network (ie
the asterisk boxes). Sometimes the use of TOPOH helps to integrate
with the carriers who have chosen their own interpretations of RFC
for security.

And there's more...

The bottom line, is that the devil is in the details.

Fred Posner The Palner Group, Inc. http://www.palner.com (web)
+1-503-914-0999 (direct) +1-954-472-2896 (fax)

On 10/23/2014 09:12 AM, Kenny Watson wrote:

Hi,


I have a few asterisk servers providing some basic SIP trunking and
routing.

We have remote PBXs trunked onto asterisk which calls come into
asterisk and are routing down to extensions on the remote PBX via
prefix routing.

I'm looking to have a central Kamailio
Registrar/Proxy/Loadbalancer which Invites come into and are routed
out to either SIP phones which are registered or to the remote
PBX.

I'm looking for some advice as to which modules would be best to
use to achieve this as the remote PBXs will be dynamically
registered rather than fixed gateways.

Please let me know what further information would be helpful.

Thanks

Kenny Watson



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Re: [SR-Users] paid support plans

[Fred Posner]

On Tue, 2014-09-16 at 19:46 +, Mike Hancock wrote:
 Hello,
 
 [SNIP] I would like to purchase support to help to get this up and
 running as well as configured to get us going. [/SNIP]  
 
  
 Mike Hancock

Hi Mike,

This is the link for the business directory:

http://www.kamailio.org/w/business-directory/

Disclaimer: whilst we do the best to select serious applications,
Kamailio and SER projects, along with their developers and management
groups, do not guarantee nor take responsibility for the quality of
services or products offered by the companies listed here. It is your
sole decision to do business with any of the entities listed here and
all commercial relations and liabilities are only between you and your
business partner, without any involvement of the two open source
projects.

On that page, you can find a great company to help you.

--fred


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Re: [SR-Users] Asterisk cluster behind kamailio natted to pubic IP, presenting internal ip addresses in From tag

[Fred Posner]

On 08/28/2014 12:44 PM, Tim Chubb wrote:
 Hi All
 
 ...snip... 
 
 So far everything is working fine, I can register via the public IP
 address, IM  presence is working, and as does audio, however when I
 dial an extension the caller id comes up like this
 12345@172.16.15.123:5080 mailto:12345@172.16.15.123:5080 which is the
 ip addresses of the asterisk server that the dispatcher has assigned to
 the call, I have tried setting the P-Asserted-Identity,
 P-Preferred-Identity  Remote-Party-ID headers to no effect.  What I
 would like to achieve is that the public IP or domain name comes up when
 I call an extension.  From examining the sip traffic traversing the
 kamailio box, it seems that the From and Contact headers sent by
 asterisk are the source of the internal implementation information
 reaching the end-user.
 
  .../snip...
  
 
 *Tim.*
 

I'm assuming with the 5080 that this call goes through the Asterisk box
before hitting the registered user on Kamailio... if that's correct,
have you also forced a CALLERID(name) on the call?

A grep of the sip traffic would show if you have something perhaps
removing this information before sending to the client.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)

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Re: [SR-Users] Kamailio behind NAT- alias issue

[Fred Posner]

Hello Yuriy,

 If I write at kamailio.cfg:
 alias=sip.myserver.com

 I see error at log - bad_uri sip.myserver.com

try adding the port...

alias=sip.myserver.com:5060

Also, since you're behind NAT make sure you also advertise the address
with advertised_address=sip.myserver.com.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)

On 08/02/2014 07:51 AM, Yuriy Gorlichenko wrote:
 Hello. I have Kamailio running behind NAT. It lesten eth0 with ip
 192.168.0.3 and
 I have external IP that have domain name (for example sip.myserver.com).
 
 Register packets from clients comes from external IP.
 
 If I write at kamailio.cfg:
 alias=sip.myserver.com
 
 I see error at log - bad_uri sip.myserver.com
 
 I think it happens because kamailio does not know atything about external
 ip pecause kamamiliol working with server interfaces (eth0, eth1, etc.)
 
 So my question - how to listen external IP with domain name on kamailio
 that running behind NAT?
 
 P.S. I do not have any access to router, that present NAT for me.

 

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Re: [SR-Users] Calls per second

[Fred Posner]

I've done this with dialog and a sql lookup/update.

The sql call updates the table with how many calls are in total for the
client and how many international calls. There's a max calls and a max
international.

If current  max, the call can go through.

The db allows me to combine the lookup over multiple media servers and
kamailio servers. The lookup checks the db so any modifications occur in
real-time.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)

On 07/03/2014 07:37 AM, Olle E. Johansson wrote:
 Hi!
 
 Have you implemented a per-customer rate limit in Calls per second? If so - 
 how?
 
 I've played with ratelimit/pipelimit and it seems like I can define a 
 database with one pipe per customer - but have to restart Kamailio to add 
 customers. There are warnings for low timer settings, like 1 second, but I 
 don't know how up-to-date those warnings are.
 
 I guess I could play with hash tables and implement something pike-like 
 there, but it seems like a workaround for something pretty common.
 
 So the question remains - how are you limiting on a cps per customer?
 
 /O
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Re: [SR-Users] Calls per second

[Fred Posner]

I think my head isn't fully woken up yet -- sorry about that.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)

On 07/03/2014 07:48 AM, Olle E. Johansson wrote:
 I am looking for calls setups per second - not concurrent calls.
 Sorry for not being exact.
 
 /O
 
 On 03 Jul 2014, at 13:41, Fred Posner f...@palner.com wrote:
 
 I've done this with dialog and a sql lookup/update.

 The sql call updates the table with how many calls are in total for the
 client and how many international calls. There's a max calls and a max
 international.

 If current  max, the call can go through.

 The db allows me to combine the lookup over multiple media servers and
 kamailio servers. The lookup checks the db so any modifications occur in
 real-time.

 Fred Posner
 The Palner Group, Inc.
 http://www.palner.com (web)
 +1-503-914-0999 (direct)
 +1-954-472-2896 (fax)

 On 07/03/2014 07:37 AM, Olle E. Johansson wrote:
 Hi!

 Have you implemented a per-customer rate limit in Calls per second? If so - 
 how?

 I've played with ratelimit/pipelimit and it seems like I can define a 
 database with one pipe per customer - but have to restart Kamailio to add 
 customers. There are warnings for low timer settings, like 1 second, but I 
 don't know how up-to-date those warnings are.

 I guess I could play with hash tables and implement something pike-like 
 there, but it seems like a workaround for something pretty common.

 So the question remains - how are you limiting on a cps per customer?

 /O
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Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls

[Fred Posner]

do you have an ngrep of the sip traffic? This can happen if the sip/rtp
cannot connect (perhaps blocked by the dsl router)

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)

On 07/01/2014 01:12 PM, Carlos Rangel wrote:
 Hello List
 
  
 
 Hopefully someone can help. This is the problem when the call is hug up 20-30 
 seconds after it initiates. The call is only hung on when the remote 
 extension initiates the call. If the remote extension receives the call there 
 is no problem the call is not hung on. I changed the remote cisco phone for a 
 yealink and it is the same behavior. It thought it was the phone.
 
  
 
 This is what I am using in kamailio.cfg
 
  
 
 #!define WITH_MYSQL
 
 #!define WITH_AUTH
 
 #!define WITH_ASTERISK
 
 #!define WITH_USRLOCDB
 
 #!define WITH_ANTIFLOOD
 
  
 
 Remote User Internet  
  Internal network
 
 Yealink IP TG28P DSL router ---|--Internet |-Cisco ASA 
 5500 FW--Kamailio/Freepbx (Same Box)--IAX 
 Trunk--Freepbx Production Server |-- PSTN
 
  
 
  
 
 Thanks
 
 Carlos Rangel
 
  
 
 De: Carlos Rangel [mailto:cran...@globaltelesourcing.com] 
 Enviado el: jueves, 26 de junio de 2014 01:27 p.m.
 Para: mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
 Asunto: RE: [SR-Users] Kamailio Freepbx Integration Dropping Calls
 
  
 
 Hi Daniel
 
  
 
 Thank you so much for your response. Here is the SIP trace of one of the 
 calls, I am not sure where the call initiates but you can see at the end of 
 the file in bold X-Asterisk-HangupCause: No user responding. I am not sure 
 why is it sending this message though.
 
  
 
 The variables are
 
  
 
 Extension/Username=X
 
 Ext_IP= Public IP
 
 Internal_IP= Asterisk/Kamailio internal IP
 
  
 
 Sorry for the long file but again I am not sure where the call initiates
 
  
 
 This is the part where that call is hung on.
 
  
 
 U 2014/06/26 13:36:11.831965 Kamailio_IP:5080 - Kamailio_IP:5060
 
 BYE sip:X@65.190.71.203:5060;user=phone;transport=udp SIP/2.0.
 
 Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47.
 
 Route: sip:Kamailio_IP;lr=on;ftag=000653dc394000970f227678-1fafb4e2.
 
 Max-Forwards: 70.
 
 From: sip:919707249077@Kamailio_IP:5060;tag=as2e670ea4.
 
 To: User sip:X@Kamailio_IP:5060;tag=000653dc394000970f227678-1fafb4e2.
 
 Call-ID: 000653dc-394b-33caf1b2-20ccd185@192.168.0.22.
 
 CSeq: 102 BYE.
 
 User-Agent: FPBX-2.11.0(11.10.2).
 
 X-Asterisk-HangupCause: No user responding.
 
 X-Asterisk-HangupCauseCode: 18.
 
 Content-Length: 0.
 
 .
 
 U 2014/06/26 13:36:11.832260 Kamailio_IP:5060 - 65.190.71.203:5060
 
 BYE sip:X@65.190.71.203:5060;user=phone;transport=udp SIP/2.0.
 
 Via: SIP/2.0/UDP 
 Kamailio_IP;branch=z9hG4bKcf68.d6ef5aa9cc5bd0fb0ab13a563b7cf284.0.
 
 Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47.
 
 Max-Forwards: 69.
 
 From: sip:919707249077@Kamailio_IP:5060;tag=as2e670ea4.
 
 To: User sip:X@Kamailio_IP:5060;tag=000653dc394000970f227678-1fafb4e2.
 
 Call-ID: 000653dc-394b-33caf1b2-20ccd185@192.168.0.22.
 
 CSeq: 102 BYE.
 
 User-Agent: FPBX-2.11.0(11.10.2).
 
 X-Asterisk-HangupCause: No user responding.
 
 X-Asterisk-HangupCauseCode: 18.
 
 Content-Length: 0.
 
  
 
  
 
  
 
  
 
 Descripción: Description: Description: DLR-Logo-No-TextCARLOS RANGEL | 
 INFORMATION TECHNOLOGY DIRECTOR
 
 Global Telesourcing México, S. de R.L. de C.V. | Aarón Sáenz #1891-1 | 
 Monterrey, N.L., México
 Direct 703 894 1667 | Mobile US 703 894 1667 | Mobile MX +52 1 812 000 7362 | 
 cran...@globaltelesourcing.com
 
 
 
 The information contained in this e-mail and any attached documents may 
 contain information that is confidential or otherwise protected from 
 disclosure. If you are not the intended recipient of this message, or if this 
 message has been sent to you in error, please immediately alert the sender by 
 reply e-mail and then delete this message, including any attachments. Any 
 dissemination, distribution or other use of the contents of this message by 
 anyone other than the intended recipient is strictly prohibited. 
 
  
 
  
 
 De: sr-users-boun...@lists.sip-router.org 
 [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Daniel-Constantin 
 Mierla
 Enviado el: jueves, 26 de junio de 2014 03:12 a.m.
 Para: Kamailio (SER) - Users Mailing List
 Asunto: Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls
 
  
 
 Hello,
 
 can you gran the SIP trace on kamailio server for such case?
 
 You can use ngrep, like:
 
 ngrep -d any -qt -W byline port 5060
 
 and send the output to the mailing list. You can replace any sensitive 
 information (e.g., ip address) before sending to mailing list.
 
 The typical call drop after 30-40 secs is when ACK is not routed properly

Re: [SR-Users] avpops issue

[Fred Posner]

For the is_user_in... are you loading the group module?

For avp_write, that function hasn't existed in some time. You can use
logic such as:

$avp(s:fwd_blind) = $ru;

Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

Good. Fast. Cheap. - Pick two.

On 05/22/2014 04:49 PM, Gilbert T. Gutierrez, Jr. wrote:
 I am trying to setup call forwarding but I am getting a failure when I
 attempt to implement it.
 
 Kamailio Version 4.1.1
 Centos 6 x64
 Using precompiled RPMs from telephony.repo
 
 I am following guidance from the following urls...
 http://www.kamailio.org/wiki/tutorials/mini-howto-admin/call_forwarding
 http://www.kamailio.org/dokuwiki/doku.php/tutorials:avpops
 http://www.kamailio.org/dokuwiki/doku.php/examples:set-blind-call-forwarding
 
 
 
 0(16018) ERROR: core [cfg.y:3272]: yyparse(): cfg. parser: failed to
 find command is_user_in
  0(16018) : core [cfg.y:3411]: yyerror_at(): parse error in config
 file //etc/kamailio/kamailio.cfg, line 1046, column 41: unknown command,
 missing loadmodule?
 
  0(16018) ERROR: core [cfg.y:3272]: yyparse(): cfg. parser: failed to
 find command avp_write
  0(16018) : core [cfg.y:3411]: yyerror_at(): parse error in config
 file //etc/kamailio/kamailio.cfg, line 1048, column 39: unknown command,
 missing loadmodule?
 
 My understanding is that when I load module avpops that it should
 include those 2 procedures (avp_write and is_user_in). I am loading the
 module avpops.so
 
 Can someone please set me straight.  Thank you.
 
 Gilbert T. Gutierrez, Jr.
 Phoenix Internet
 
 
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Re: [SR-Users] Adding dialogs to a profile in a failure_route

[Fred Posner]

On 04/30/2014 04:16 AM, Alex Balashov wrote:


So, do you suppose I could achieve my objective by tracking every
dialog--that is, by calling dlg_manage() in the initial request route
for every call, and then calling set_dlg_profile() out of a failure
route conditionally?



Wouldn't you need to set it within the original invite and then do 
something with it in the event_route[dialog:failed]?


Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

Good. Fast. Cheap. - Pick two.

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Re: [SR-Users] Adding dialogs to a profile in a failure_route

[Fred Posner]

On 04/28/2014 01:49 PM, Alex Balashov wrote:

Hi,

[SNIP] So, the question is, am I doing something wrong? What's the best way to
accommodate this scenario? I don't know if I want to track the dialog or
add it to a profile until after I get the 302.



Out of curiosity, if you call set_dlg_profile() at the initial invite 
and then so something with this in event_route[dialog:failed], does it 
still error?


Fred Posner
The Palner Group, Inc.
f...@palner.com
@fredposner

Good. Fast. Cheap. - pick two

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Re: [SR-Users] Adding dialogs to a profile in a failure_route

[Fred Posner]

On 04/28/2014 06:54 PM, Alex Balashov wrote:

I don't think that will work, because no dialog is created by the 302 redirect.



11.3. event_route[dialog:failed]

Executed when dialog is not completed (+300 reply to INVITE).

--fred

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Re: [SR-Users] sql_xquery() and xavp checks

[Fred Posner]

Have you tried something like...

if (sql_xquery(ca, SELECT * FROM gateways, gateways) == 1) {
#do stuff
} else {
#dang nabbit
}

Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

On 04/06/2014 02:37 PM, Alex Balashov wrote:

On 04/06/2014 12:37 AM, Kelvin Chua wrote:


$dbr for SELECTs


Unless it's not, because you're using sql_xquery().  :-)



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Re: [SR-Users] sql_xquery() and xavp checks

[Fred Posner]

When I use xquery, I generally use it with a stored procedure that 
returns a value if not matched... so I always return at least one row 
with a variable of 'fail', -1, etc. to evaluate.


I like this for a variety of reasons (ie changing sql without changing 
the config)... but that being said...


Wouldn't this still work for you:

if($dbr(gateways=rows)0) { }

Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

On 04/05/2014 11:32 AM, Alex Balashov wrote:

Hi,

When using sql_xquery() like this:

sql_xquery(ca, SELECT * FROM gateways, gateways);

... what's a good way to check if any rows were returned? Since one does
not have a $dbr(gateways=rows) value in this scenario, what should one do?

- is_avp_set($xavp(gateways=id))) does not appear to operate on
XAVPs, or at least, the fixup functions reject them:

ERROR: avpops [avpops.c:935]: fixup_is_avp_set(): bad attribute name
$xavp(gateways=id)

- the 'defined' operator does not appear to return a negative condition
here:

if(!defined $xavp(gateways=id))

This condition evaluates to true.

Much appreciated!

-- Alex



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Re: [SR-Users] sql_xquery() and xavp checks

[Fred Posner]

 I don't think so. As I understood the documentation, at least, $dbr
 doesn't get populated in this case; the rows just go straight to an
 xavp list. I suppose I should verify that.


Looks like you're right.

Tested various methods.

Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

On 04/05/2014 11:45 AM, Alex Balashov wrote:

On 04/05/2014 11:42 AM, Fred Posner wrote:


When I use xquery, I generally use it with a stored procedure that
returns a value if not matched... so I always return at least one row
with a variable of 'fail', -1, etc. to evaluate.


I actually do that too in many cases, but only because I often need to
pass back additional data about what went wrong (for logging) if no row
was found, so returning a row regardless, with a status column (with a
value like -1) and hijacking another column for some kind of
human-readable explanation.


I like this for a variety of reasons (ie changing sql without changing
the config)... but that being said...

Wouldn't this still work for you:

if($dbr(gateways=rows)0) { }


I don't think so. As I understood the documentation, at least, $dbr
doesn't get populated in this case; the rows just go straight to an xavp
list. I suppose I should verify that.



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Re: [SR-Users] sql_xquery() and xavp checks

[Fred Posner]

On 04/05/2014 09:01 PM, Alex Balashov wrote:

Does that work for SELECT queries? The documentation says it's only for
INSERT, UPDATE and DELETE.




It did not during my test this afternoon.

--fred


On 6 April 2014 02:14:24 CEST, Kelvin Chua kel...@gmail.com wrote:

dunno if this helps but i use $sqlrows(ca) to check whether there
are rows returned

Kelvin Chua


On Sat, Apr 5, 2014 at 9:15 AM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:

On 04/05/2014 12:14 PM, Fred Posner wrote:

   I don't think so. As I understood the documentation, at
least, $dbr
   doesn't get populated in this case; the rows just go
straight to an
   xavp list. I suppose I should verify that.
  

Looks like you're right.

Tested various methods.


This does make me wonder if there is a leak of result handles
here, since sql_result_free() is not possible, but presumably
there's an internal result handle still getting allocated. Or
maybe that was for the memory allocated to the $dbr(...) data
only and independent of the underlying DB API.


--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670 tel:%2B1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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--
Sent from my mobile, and thus lacking in the refinement one might expect
from a fully fledged keyboard.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com


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Re: [SR-Users] NAT Traversal issue

[Fred Posner]

It looks like you may be running Kamailio behind NAT as well, no?

Can you provide any traffic on the connections that fail?

Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

On 04/03/2014 08:44 AM, Ravi wrote:

Dear Kamailio'ns,

I am awaiting somebody's suggestions/hints/comments on this issue, with that
i can proceed further.

Please anybody help me in resolving this issue.

Any help will mean a lot and greatly appreciate.

Regards,
Ravi




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Re: [SR-Users] Caller ID name

[Fred Posner]

Just to add, besides the uac having some of the best example names... 
the callerid you mentioned is most likely set on your phone config; 
which kamailio is just passing along.


Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

On 03/19/2014 07:18 PM, Alex Balashov wrote:

Hello Abdul,

This depends on how you want to signal the caller ID.

If you want to indicate it with the P-Asserted-Identity header, which
overrides both Remote-Party-ID and the 'From' value, you can just append
your own header with a display name value:

append_hf(P-Asserted-Identity: \SHERIF MALIK\
sip:$fU@$Ri:$Rp\r\n);

If you want to actually override the From display value, that's a bit
more complex, since proxies aren't technically supposed to do that.
However, the 'uac' module gives you this capability:


http://kamailio.org/docs/modules/4.1.x/modules/uac.html#uac.f.uac_replace_from


e.g.

uac_replace_from(\SHERIF MALIK\, ); # Don't modify From URI.

-- Alex

On 03/19/2014 07:15 PM, malik sherif wrote:


Hello,
I set SIP users using kamctl add command  with username ( I put the
phone numbers) doamin (doamin name) and password. Is their a way to add
caller id name? when I make a call , i see the phone number but for
caller id name I displayline1.
Thank you for your help.
Abdul





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Re: [SR-Users] fresh kamailio installation unable to connect through jitsi

[Fred Posner]

On 2/27/14, 1:21 PM, Michelle Jun wrote:


m=audio 21064 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 101.
m=video 23134 RTP/AVP 105 99.


It does look like it's within the range. I would generally ensure that 
your firewall is forwarding the ports.


--
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http://qxork.com

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Re: [SR-Users] fresh kamailio installation unable to connect through jitsi

[Fred Posner]

On 02/26/2014 08:52 PM, Michelle Jun wrote:

hi

i just finished installation of kamailio on centos 6 64 bit following
the docs on

http://www.fredposner.com/voip/1457/kamailio-behind-nat/

the user able to chat through jitsi just fine, but when calling, one
user able to connect, while the other showing only connecting
(audio/video)

any idea what did i do wrong?

here is the /var/log./messages

thank you


If you're natted, make sure you have your firewall forwarded for the 
ports you've selected for rtp and sip.


Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

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Re: [SR-Users] fresh kamailio installation unable to connect through jitsi

[Fred Posner]

On 02/26/2014 09:11 PM, Michelle Jun wrote:

hi Fred

yes, i forwarded both TCP/UDP 5060 dan 2-3 like in your blog

but still having the issue

thanks



The rtp forwarding should be just udp. For the sip, that's up to how 
you're making the connections.


Did you specify a range when you started the rtpproxy?

Do you have any of the sip traffic (from ngrep, etc.)?

--fred

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Re: [SR-Users] Configuring Kamailio as an upstream proxy for FreeSwitch and which RTP proxy to choose

[Fred Posner]

On Mon Feb 24 10:50:07 CET 2014 Sean Kemball wrote:

 New to Kamailio and FreeSwitch, loosely familiar with SIP mechanics,
 and not a complete network idiot... but please be gentle. :)

Welcome!s,

 Questions:

 1.Should the proposed topology, with Kamailio + an RTP proxy
 behind a firewall, relaying to FS on an inside interface, work?
 (Can't see why not)

Yes, you said that your upstream is on the same private network. So it 
should be pretty straight forward.


 2.Does it need a local RTP proxy on the Kamailio box, particularly
 if we turn off the ASA SIP inspect stuff?

If you are all on the same private network, I would let FreeSWITCH 
handle the RTP, but you can do this a variety of ways.


 3.Can you recommend which RTP proxy to use? There seem to be at
 least 3 that work with Kamailio. The box is CentOS 6.5, and it would 
 be nice to use known-to-work packages rather than compile from source.

 (But eh, if I haveta).

On your scenario, I'd just use FreeSWITCH for the media proxy. Again, 
many different ways to go here.


 4.Can anyone point me to some docs to explain what ports need to
 be open between the Kamailio box and my upstream proxy/media server?
 I can be more liberal between inside and DMZ I guess.

Your upstream provider would generally tell you which rtp ports they 
would want opened.


 5.Is static NAT in this environment going to bite me, or should
 it be OK?

I've never had an upstream provider communicate with me on private nat.

 6.Is there any better documentation that we should be using to
 make this easier, or should I just man up and try harder?

Man up. =)
Practive makes perfect.

--
Fred Posner
The Palner Group, Inc.


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Re: [SR-Users] Replacing an ACME Packet Net-Net SBC

[Fred Posner]

Alex's article is one of my favorites. That being said, we switched out 
an Acme SBC for openser (at the time) and was immediately thrilled.


Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

On 02/20/2014 01:14 PM, Alex Balashov wrote:

Francesco,

Have a look at this blog post:

http://www.likewise.am/2013/03/kamailio-as-an-sbc-session-border-controller/


That said, I agree with Carsten's suggestion of SEMS.

On 02/20/2014 11:04 AM, Francesco Maria Magnini wrote:


Hi,

I would like to have some suggestions about a full replacement of an
ACME Packet Net-Net Session Border Controller.
By now, ACME SBC performs all the SBC functionalities, mainly:

- it is used as a SIP endpoint for SIP client registrations
- it is used as a SIP endpoint for interconnection to multiple SIP
carriers via SIP trunks
- it is used for NAT traversal

In this deployment, the SIP Server communicates only with the SBC and
this one takes care of the communication between the SIP Server and
the external SIP entities (UA clients, SIP Trunks).
In this scenario, can I consider to replace the SBC with Kamailio?


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Re: [SR-Users] Replacing an ACME Packet Net-Net SBC

[Fred Posner]

On 2/20/14, 6:25 PM, Francesco Maria Magnini wrote:

Fred,

in you ACME replacement, kamailio doesn’t rewrite headers for handling
RTP/SIGNALING and stay in the middle?


For nat it did. For others the media server did.

You can easily force all connections to use rtpproxy to do what you ask. 
We chose to do this for NAT and for media servers outside of kamailio to 
be the other choice. All calls were on one, the other, or both.


--
Fred Posner | The Palner Group, Inc.
http://qxork.com



Il giorno 21/feb/2014, alle ore 00:03, Fred Posner f...@palner.com
mailto:f...@palner.com ha scritto:


On 2/20/14, 5:55 PM, Francesco Maria Magnini wrote:

@Carsten
I looked athttp://www.iptel.org/semsand seems to be only broken links
to downloads.
Do you know if the project is still maintained?

@Fred
Are you using openser as a B2BUA?


No, because of course Kamailio is not a b2bua. =)

In the case of the ACME replacement, we used it to:

- handle NAT (rtpproxy)
- user regs
- load balance
- lcr
- security
- routing

and some other little hacks.

--
Fred Posner | The Palner Group, Inc.
http://qxork.com http://qxork.com/



Il giorno 20/feb/2014, alle ore 19:42, Fred Posner f...@palner.com
mailto:f...@palner.com ha scritto:


Alex's article is one of my favorites. That being said, we switched
out an Acme SBC for openser (at the time) and was immediately thrilled.

Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

On 02/20/2014 01:14 PM, Alex Balashov wrote:

Francesco,

Have a look at this blog post:

http://www.likewise.am/2013/03/kamailio-as-an-sbc-session-border-controller/


That said, I agree with Carsten's suggestion of SEMS.

On 02/20/2014 11:04 AM, Francesco Maria Magnini wrote:


Hi,

I would like to have some suggestions about a full replacement of an
ACME Packet Net-Net Session Border Controller.
By now, ACME SBC performs all the SBC functionalities, mainly:

- it is used as a SIP endpoint for SIP client registrations
- it is used as a SIP endpoint for interconnection to multiple SIP
carriers via SIP trunks
- it is used for NAT traversal

In this deployment, the SIP Server communicates only with the SBC and
this one takes care of the communication between the SIP Server and
the external SIP entities (UA clients, SIP Trunks).
In this scenario, can I consider to replace the SBC with Kamailio?


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Re: [SR-Users] Call forwarding from external did

[Fred Posner]

On 02/10/2014 04:00 PM, arun Jayaprakash wrote:

Hello, I have set up call forwarding in Kamailio using user_preference
table. When I make a call from a local extension the callfwd funcition
works. The call gets forwarded to an external did number. The problem
happens when and external call ( from a DID) comes to this extension the
call does not get forwarded. I had to comment out the following lines in
the config file to make it work:

# only local users allowed to call
#  if((from_uri!=myself)) {
#  sl_send_reply(403, Not Allowed);
#  exit;
# }

My question is if it is a risky think to comment out these lines? If so,
what are my options.

Thank you,
Arun



In short, yes, it is risky-- but in honesty, noone can tell from just 
that portion of the config.


You can test this perhaps by trying any number to your system. Is it 
also being forwarded?


A good way is to add a flag and check for the presence of that flag if 
allowing a non-registered user to make an outside call.


Fred Posner, @qxork
http://palner.com


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Re: [SR-Users] Kamailio behind NAT

[Fred Posner]

On 01/24/2014 09:51 AM, Andrew Pogrebennyk wrote:

On 01/23/2014 05:12 PM, Klaus Darilion wrote:

It is necessary to use the cwie / cwei flags in the rtpproxy_manage call?


If rtpproxy uses only a single listen-IP, then these flags are not
needed. Only if you operate rtpproxy in bridge mode, then you need these
flags. Bridge mode is necessary if you do not have IP routing between
the internal network and the virtual external network, or if you want
to bridge between IPv4 and IPv6.


John,
This function can be used to check the direction of every message:
http://kamailio.org/docs/modules/4.0.x/modules/rr.html#idp223296
You might also need to append the record-route parameters to remember
the flags you have passed to the manage_rtpproxy() initially.
Based on the direction of the request and initial flags you can
determine what flags to use when calling manage_rtpproxy() for a given
in-dialog requests and reply.

Hope this helps.
Andrew



Are the calls being bridged across two interfaces or is the Kamailio 
just natted? (or is it both?)


Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)

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Re: [SR-Users] Kamailio behind NAT

[Fred Posner]

With a patched version of rtpproxy you can advertise your private ip. 

http://www.fredposner.com/voip/1457/kamailio-behind-nat/

---Fred 


 On Jan 21, 2014, at 6:18 AM, John Smith jsmith...@mail.com wrote:
 
 Hello,
  
 I am currently deploying one Kamailio behind NAT with one Asterisk as 
 explained in the Asipto KB (Kamailio 4.0.x and Asterisk 11.3.0 using Asterisk 
 Database). The structure is deployed as described in that document, with the 
 only addition of one NAT between Kamailio and Internet:
  
 Phone ——— Nat ——— Kamailio —— Asterisk
  
 I have declared the private IP with the advertise option in order to support 
 the NAT, enabled WITH_NAT and I have installed rtpproxy using standard Debian 
 package configured as rtpproxy -l public_ip_ -s udp:localhost:7722
  
 After setting up two phones which register correctly at Asterisk, I have no 
 audio at all. 
  
 By placing tcpdumps between nodes I see at Kamailio node both audio from 
 public IP to internal Kamailio IP and from the latter to the Asterisk IP. In 
 Asterisk I see audio coming from the Kamailio private IP and then back to the 
 public IP of the phone.
  
 My guess is that audio should flow back into Kamailio and then to the phone, 
 not directly from Asterisk as it is right now.
  
 Can anyone hint at where I am wrong?
  
 Thank you
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Re: [SR-Users] kamdbctl create ?

[Fred Posner]

From the error, it looks like the tables, etc were created and the 
failure was in granting the privileges.


You can always grant them manually, or drop the kamailio database and 
try the script again.


Fred Posner | The Palner Group
direct: 503-914-0999 | fax: 954-472-2896

On 11/16/2013 04:10 PM, Thomas Secula wrote:

I seem to be in a catch 22, sure it’s something I’ve done wrong.

I set dbhost in kamctlrc to be the ip of my remote mysql server

I run kamdbctl create and I get

root@pcscf:/etc/kamailio# kamdbctl create

MySQL password for root:

database engine 'mysql' loaded

INFO: test server charset

INFO: creating database kamailio ...

INFO: granting privileges to database kamailio ...

ERROR 1044 (42000) at line 1: Access denied for user
'root'@'172.16.101.27' to database 'kamailio'

ERROR: granting privileges to database kamailio failed!

root@pcscf:/etc/kamailio#

Before I run the create from mysql:

mysql  select user,host from mysql.user;

+--+---+

| user | host  |

+--+---+

| root | % |

| root | 172.16.101.27 |

| root | 172.16.101.28 |

| root | 172.16.101.29 |

| root | localhost |

+--+---+

5 rows in set (0.00 sec)

mysql show grants for 'root'@'%';

+--+

| Grants for
root@%
|

+--+

| GRANT ALL PRIVILEGES ON *.* TO 'root'@'%' IDENTIFIED BY PASSWORD
'*FE04940F3F9B339AB9361B2D01AD3D940B215B52' |

+--+

1 row in set (0.00 sec)

If I rerun it the create scripts says:

root@pcscf:/etc/kamailio# kamdbctl create

MySQL password for root:

database engine 'mysql' loaded

INFO: test server charset

INFO: creating database kamailio ...

ERROR 1007 (HY000) at line 1: Can't create database 'kamailio'; database
exists

ERROR: Creating database kamailio failed!

root@pcscf:/etc/kamailio#

any ideas?



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Re: [SR-Users] [sr-dev] Kamailio VUC Session on Nov 15

[Fred Posner]

I will be joining the call and IRC but cannot do the hangout. 

---Fred 


 On Nov 14, 2013, at 9:33 AM, Daniel-Constantin Mierla mico...@gmail.com 
 wrote:
 
 Hello,
 
 you, and anyone else that want to join the VUC session, will have to be 
 tomorrow on irc (channel #vuc on freenode.net) and we will see if we can plug 
 extra participants in the hangout. It looks very busy, having a large number 
 (about 10) of developers joining the event and hangout cannot sustain too 
 many participants.
 
 However, you can watch the live streaming of the hangout and ask questions 
 via sip -- this is valid for everyone willing to watch -- VUC video session 
 will be streamed live, can be watched with a web browser. Questions can be 
 asked via IRC or audio calls.
 
 Here are again the links where to look for more information:
 
 - http://vuc.me
 - http://www.kamailio.org/w/2013/10/kamailio-update-on-vuc-nov-15-2013/
 
 Cheers,
 Daniel
 
 On 10/29/13 7:24 PM, Muhammad Shahzad wrote:
 I would like to join this session with google hangout. If that's not 
 available then i can call via SIP as well.
 
 Thank you.
 
 
 On Tue, Oct 29, 2013 at 1:15 PM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:
 Hello,
 
 we are preparing for a new VUC session to give an update about Kamailio 
 project - a perfect timing as we are just about to release a new major 
 version.
 
 We will like to get many developers involved to be able to 
 highlight properly what is new, especially those that did new development 
 for v4.1 - new modules or enhancement to existing modules. However, any 
 developer and community member is welcome to join, we will appreciate it 
 very much, in support of the project.
 
 Those that have a google hangout account can participate with video, 
 otherwise there are options to join via sip or pstn audio calls - you can 
 see more details at:
 
 - http://vuc.me
 
 The number of participants with video is limited, therefore if you plan to 
 do it, let me know to be able to coordinate and send you the invite link 
 when the session starts.
 
 No matter you participate with audio/video, you can join the IRC channel 
 #vuc on freenode.net for text chatting during the event.
 
 I made quickly a news about the event:
 
 - http://www.kamailio.org/w/2013/10/kamailio-update-on-vuc-nov-15-2013/
 
 I will add names of other participants as I get confirmations. Do suggest 
 topics to highlight/discuss there as well.
 
 Cheers,
 Daniel
 
 -- 
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
 Kamailio Advanced Trainings - Berlin, Nov 25-28
   - more details about Kamailio trainings at http://www.asipto.com -
 
 
 ___
 sr-dev mailing list
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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
 
 
 
 -- 
 Mit freundlichen Grüßen
 Muhammad Shahzad
 ---
 CISCO Rich Media Communication Specialist (CRMCS)
 CISCO Certified Network Associate (CCNA)
 Cell: +49 176 99 83 10 85
 MSN: shari_78...@hotmail.com
 Email: shaherya...@googlemail.com
 
 -- 
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
 Kamailio Advanced Trainings - Berlin, Nov 25-28
   - more details about Kamailio trainings at http://www.asipto.com -
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Re: [SR-Users] kamailio dialplan

[Fred Posner]

When you dial 43 you get a prompt or 41?

Also, do you see anything in the freeswitch logs or have a sip capture/

Fred Posner | The Palner Group
direct: 503-914-0999 | fax: 954-472-2896

On 11/08/2013 06:04 PM, Joli Martinez wrote:

I am new to Kamailio and am having an issue with the dialplan setup.  I
have Kamailio setup as an SBC to handle all user authentication and call
routing.  I need freeswitch to handle all conferences and voicemails.
  When I dial 433001 I would like to be transferred to freeswitch for
conferences.  Right now I have followed the following article and it
when I dial 433001 call hangs up and never reaches FS.  If I call 43
call does reach FS and I am able to hear FS play the VM prompt.

My system is CentoOS 6.4 and FS is installed via yum, but Kamailio is
complied. Both FS and Kamailio are on the same box.

What commands would you suggest I use to troubleshoot these issues in
the future.

http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms#dokuwiki__top

Also, since I am new could you give some pointers as far as security and
documentation.

thanks,




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Re: [SR-Users] TLS - No ServerHello

[Fred Posner]

There's a great debugging article posted to:
http://www.kamailio.org/wiki/tutorials/tls/testing-and-debugging

 What kind of response do you get from:

openssl s_client -connect IPADDRESS:5061 -no_ssl2 -bugs

Fred Posner | The Palner Group
direct: 503-914-0999 | fax: 954-472-2896

On 11/05/2013 02:09 PM, Coy Cardwell wrote:

Hi all.

We are using Kamailio 4.0.4 in an Edge Proxy to Central Registrar
configuration.

The system is in a functional state.

Oddly, when we enable TLS, the Kamailio server never responds with a
Server Hello to the Client Hello for TLS, so a connection is never
established.

Has anyone seen anything like this? There are no errors anywhere, the
server just never responds to the initial TLS Client Hello. We know the
packet arrives and is acknowledged at the TCP level.

Just a bit stumped.


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Re: [SR-Users] Kamailio network edge for registration and rtp pass-through

[Fred Posner]

Do you need the registration be local to the asterisk? 

I would have all the asterisks send calls to the Kamailio.

You can have a lookup on endpoint outbound to decide which asterisk should 
handle the outbound call for that did. 

Also a lookup for incoming DIDs, etc. 

---Fred 


 On Oct 25, 2013, at 5:43 PM, Jr Richardson jmr.richard...@gmail.com wrote:
 
 Hi All,
 
 Starting a new project, roll your own SBC, not a full SBC, just need some 
 minor functionality.  I'm interested in deploying Kamailio as a edge device 
 on a VSP for single entry point for hosted PBX's, Asterisk based.  I had some 
 wonderful and informative conversations at Astricon 2013, several folks 
 assuring me Kamailio w/rtpproxy was the tool for the job, so this is a follow 
 up to delve more into details.
 
 I've been researching configs, topology, modules needed, ect...  Most of the 
 examples I'm reading about for this scenario are spreading the registrations 
 across many PBX's without distinction.  One concept I'm struggling with is 
 having a specific phone register to a specific PBX.
 
 phone-customer-A-x101internetkamailioPBX-customer-A-x101
 phone-customer-B-x101internetkamailioPBX-customer-B-x101
 phone-customer-C-x101internetkamailioPBX-customer-B-x101
 
 I could add a unique identifier to some part of the registration of each 
 phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect.  What 
 I'm not clear on is when the request comes to kamailio, where would I 
 identify what PBX the phone should register to and how to re-write the 
 'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct PBX 
 and ensure rtp flows through kamailio.
 
 Could this function be derived using dbaliases or possibly using dispatcher 
 with group number for each customer PBX?
 
 So assuming I can get the registrations to work properly, would standard 
 invite for calling just work or would I also have to have specific config in 
 place to ensure an invite from customer A phone also reaches the correct 
 customer A PBX?
 
 A point in the right direction?
 
 Thanks.
 
 JR
 -- 
 JR Richardson
 Engineering for the Masses
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Re: [SR-Users] Kamailio failed to start

[Fred Posner]

It appears as though you might have multiple syntax errors in the script -- 
such as not opening or not closing certain tags. 

The best way to troubleshoot this (or at least the way I like best) is to go 
back to your last known good config and make the changes one at a time. 

--fred

Wingsravi R wingsravi...@gmail.com wrote:

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Re: [SR-Users] (no subject)

[Fred Posner]

Hello Sebastien,

On 10/11/13 2:10 PM, Sébastien Cramatte wrote:

Hello,
...

My question is hwo can I replace rtpproxy by ngcp-mediaproxy-ng  in
bridge mode ?

Does it possible ?

I use  rwie and rwei flags   but in ngcp-mediaproxy-ng  e and i
seems to be used for IPv4 / IPv6 ..


...


I don't believe that mediaproxy-ng can be used to bridge two ipv4 
networks; only bridging for ipv6 - ipv4.


--
Fred Posner
http://qxork.com


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Re: [SR-Users] ZRTP

[Fred Posner]

Hi Jonathan,

On 10/11/13 3:42 PM, Jonathan Brown wrote:

Hi,

Does Kamailio support ZRTP end point to end point encryption? If so how
is this configured?

Sincerely,

*Jonathan Brown*



Kamailio handles the sip processing of the call allowing the two 
endpoints to negotiate the zrtp.


With Kamailio, you wouldn't need to configure anything special to allow 
the clients to use zrtp. If both clients support the option (such as 
jitsi), they can use it to secure the media.


--
Fred Posner | The Palner Group, Inc.
http://qxork.com

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Re: [SR-Users] New Developer: Charles Chance

[Fred Posner]

Welcome Charles!

Fred Posner | Team Forrest / LOD
direct: 503-914-0999 | fax: 954-472-2896

On 09/13/2013 10:06 AM, Daniel-Constantin Mierla wrote:

Hello,

I want to announce that a new person got developer GIT write access to
repository: Charles Chance.

He is for long time in the community, sending patches in the past to
modules such as memcache. His immediate goal is to care of dmq module,
having a set of patches to be committed as well as plans to integrate it
within more modules.

His git commit id is: cchance

My warm welcome and looking forward to future work within the project!

Cheers,
Daniel



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