Re: [SR-Users] [SIP Transaction] Handle Cancel message

[Hai Bui Duc Ha]

Hi Daniel,

I send you 2 files pcap, capture on server side.
+ Server: 192.168.1.77
+ Client: 192.168.1.134
+ Scenario:
   User 102 call to ring group 777 - has user 100 (192.168.1.134) and user
101 (192.168.1.164).
   User 100 is offline but it's configured push notification from Apple to
wake up, register and receive INVITE.
   I pick up phone on 101 but 100 can not receive the CANCEL message to
stop the ringing tone.

1) Pusher_ok.pcap: User 100 can revice the CANCEL message.
2) Pusher_failed.pcap: As I told above, when user 101 pick up the phone,
Freeswitch send CANCEL message to user 100. But when message go to
Kamailio, Kamailio reply 481 - Call leg/transaction does not exist. (I
can't capture this packets).
I also include the log on Freeswitch send CANCEL message and revice the 481.

 Kamailio doesn't send 481 from the C code in the case of call, it either
receives it or it is something in config.
=> This is configure on kamailio: default.cfg, it check *t_check_trans()*
===
*route[PREPARE_INITIAL_REQUESTS]*
*{*
*if (is_method("CANCEL")) {*
*if (t_check_trans()) {*
*route(RELAY);*
*} else {*
*sl_send_reply("481", "Call leg/transaction does not exist");*
*}*
*exit();*
===

Thank for your support !

Regards,
Hai Bui

On Thu, Mar 9, 2017 at 8:27 PM, Daniel-Constantin Mierla 
wrote:

> Hello,
>
> can you provide pcap (or full ngrep output) for all messages of such a
> call, from initial invite, including all responses and the other requests?
> The sip trace has to be taken on kamailio server.
>
> Kamailio doesn't send 481 from the C code in the case of call, it either
> receives it or it is something in config.
>
> Cheers,
> Daniel
>
> On 09/03/2017 06:59, Hai Bui Duc Ha wrote:
>
> Hi all,
>
> I have problem when use module pusher on Kazoo.
> The user after receive call from pusher can not receive the Cancel message
> if have another people pick up this phone.
> I trace log and debug on Kamailio and see the Freeswitch sent Cancel
> message but Kamailio reply 481 - Call leg/transaction does not exist.
>
> The configure on Kamailio: default.cfg
> ===
> *route[PREPARE_INITIAL_REQUESTS]*
> *{*
> *if (is_method("CANCEL")) {*
> *if(t_lookup_cancel()) xlog("L_INFO","$ci|log|
> t_lookup_cancel()");*
> *xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CAPTURED IN
> MAIN---\n");*
> *if (t_check_trans()) {*
> *xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CHECK TRANS
> TRUE---\n");*
> *//xlog("L_INFO",
> "$ci|log|");*
> *route(RELAY);*
> *} else {*
> *xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CHECK TRANS
> FALSE---\n");*
> *sl_send_reply("481", "Call leg/transaction does not exist");*
> *}*
> *exit();*
> *} else if (is_method("ACK")) {*
> *if (t_check_trans()) {*
> *route(RELAY);*
> *}*
> *exit();*
> *}*
> ===
>
> As I read on http://kamailio.org/docs/modules/4.3.x/modules/tm.html#tm
> .f.t_check_trans
> Kamailio see the cancel message not same the transaction INVITE message.
>
> INVITE message:
> 
> INVITE sip:1...@quydang.htk.cvoice SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.77:11000;rport;branch=z9hG4bKK9jyp8mXUZHgS
> Route: 
> Max-Forwards: 49
> From: "02 quy" ;tag=04K2SappeyDtj
> To: 
> Call-ID: 19fa9228-ef95-11e6-b473-1f645951fd7b
> CSeq: 103030653 INVITE
> Contact: 
> User-Agent: Anttel
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 247
> X-AUTH-IP: 192.168.1.141
> X-AUTH-PORT: 64799
> X-KAZOO-AOR: sip:1...@quydang.htk.cvoice
> X-KAZOO-PUSHER-Token-Proxy: sip:192.168.1.77:5060
> X-KAZOO-PUSHER-Token-ID: 803F0D6610AC5979C6F0513A3A4BE6
> 2E0BAF1530DC7A98C2C0DC13D784585FBE
> X-KAZOO-PUSHER-Token-Type: apple
> X-KAZOO-PUSHER-Token-App: com.htkinc
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "02 quy" ;p
> arty=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1486708670 1486708671 IN IP4 192.168.1.77
> s=FreeSWITCH
> c=IN IP4 192.168.1.77
> t=0 0
> m=audio 24764 RTP/AVP 0 8 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> 
>
> CANCEL message:
> -

[SR-Users] Multiple parallel branches, one invalid

[Nathan Ward]

Hi,

I have a scenario I’m not sure of the best way to solve.

I use alias_db and have multiple destinations for a single number, and have 
append_branches set to 1.
I then use lookup_branches to get details from the location table for each 
branch.
I then do t_load_contacts and t_next_contacts, then t_relay to parallel fork 
the calls to the destination, and all the phones ring at once.

This all works great if all the destinations from the dbaliases table are 
online.

However, if one of them is offline, lookup_branches says "Not found in usrloc”. 
This branch is still there, and the call is attempted. In my environment, this 
then does a DNS lookup, can’t resolve the domain, and the branch fails. This is 
the first branch that is tried, and when this branch has an error like this, it 
causes t_relay() to fail - I check for the output of t_relay() and reply with 
an error - per the default config that Kamailio ships with.

I guess what I need to figure out, is if there’s a way to have t_relay fail 
only if all branches fail? Or, is there a more elegant way to handle this?

I should add, I only allow a single registration per subscriber - so, 
lookup_branches only returns one branch per alias from alias_db_lookup.

--
Nathan Ward


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Re: [SR-Users] Bounty for topos fix

[Trevor Peirce]

On 08/03/2017 10:32 AM, Trevor Peirce wrote:
I would like to offer a $100 USD bounty if a developer is able to 
provide a fix for the two topos problems I have experienced.  I will 
add another $50 if this can be solved within 48 hours of the time 
stamp of this message.


The bug report is at: https://github.com/kamailio/kamailio/issues/1005


Seems to be fixed, thanks Daniel.

Please contact me off-list at this address to sort out the details of 
the bounty.  Happy to support you and the Kamailio project.


Thanks,

--
Trevor Peirce
AcroVoice Solutions Inc


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Re: [SR-Users] Dispatcher Module - Hava limit?

[Daniel-Constantin Mierla]

Hello,

do you write log messages from config file when processing sip messages?

Cheers,
Daniel


On 09/03/2017 19:22, Rodrigo Moreira wrote:
> Thank you Alex Balashov!
>
> I read the link that suggested, very good the issue of child processes
> fit as described. I have a machine with the carrierroute module, it
> can handle 150 CPS without fail, when I increase this number for eg
> 175, there are countless losses. I have increased the shared memory,
> but I still can not understand the grinding of so many losses. Do you
> have any suggestions for anything you can check?
>
> I know the problem is not the network channel because it currently has
> 40Mbps.
>
>
> Regards.,
> Rodrigo M.
>
> On Wed, Mar 8, 2017 at 2:34 PM, Alex Balashov
> mailto:abalas...@evaristesys.com>> wrote:
>
> On Wed, Mar 08, 2017 at 02:24:20PM -0300, Rodrigo Moreira wrote:
>
> > I am experiencing the following problem: my dispatcher module is
> losing
> > many calls when the scenario for balancing is about 175 CPS or
> more. Is
> > there a limit that the module can do? Where do I find this
> information?
>
> The dispatcher has no such limits. Also, how did you determine
> that the
> dispatcher module is the essence of the "loss" that you are seeing? It
> is most likely happening in a more general context:
>
> 
> http://blog.csrpswitch.com/tuning-kamailio-for-high-throughput-and-performance/
> 
> 
>
> -- Alex
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800  / +1-800-250-5920
>  (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> ___
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> list
> sr-users@lists.sip-router.org 
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> 
>
>
>
>
> -- 
> Rodrigo M.
> (37) 9132-4539
> (34) 9889-3069
> rodrigo.moreira2007
>
>
>
>
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-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - 
www.asipto.com
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Re: [SR-Users] Dispatcher Module - Hava limit?

[Rodrigo Moreira]

Thank you Alex Balashov!

I read the link that suggested, very good the issue of child processes fit
as described. I have a machine with the carrierroute module, it can handle
150 CPS without fail, when I increase this number for eg 175, there are
countless losses. I have increased the shared memory, but I still can not
understand the grinding of so many losses. Do you have any suggestions for
anything you can check?

I know the problem is not the network channel because it currently has
40Mbps.


Regards.,
Rodrigo M.

On Wed, Mar 8, 2017 at 2:34 PM, Alex Balashov 
wrote:

> On Wed, Mar 08, 2017 at 02:24:20PM -0300, Rodrigo Moreira wrote:
>
> > I am experiencing the following problem: my dispatcher module is losing
> > many calls when the scenario for balancing is about 175 CPS or more. Is
> > there a limit that the module can do? Where do I find this information?
>
> The dispatcher has no such limits. Also, how did you determine that the
> dispatcher module is the essence of the "loss" that you are seeing? It
> is most likely happening in a more general context:
>
> http://blog.csrpswitch.com/tuning-kamailio-for-high-
> throughput-and-performance/
>
> -- Alex
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> ___
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> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Rodrigo M.
(37) 9132-4539
(34) 9889-3069
rodrigo.moreira2007
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Re: [SR-Users] Changes in LCR Target List doesn't takes place immediately

[Victor Seva]

2017-03-09 11:39 GMT+01:00 Nitesh Lohchab :
> I believe LCR Target list is stored in the database and any changes made
> here should affect the behaviour of the calls imedialtely.

If you're changing the db directly, you need to do lcr.reload

https://www.kamailio.org/docs/modules/4.4.x/modules/lcr.html#idp27407500

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[SR-Users] Changes in LCR Target List doesn't takes place immediately

[Nitesh Lohchab]

Hello,

I believe LCR Target list is stored in the database and any changes made
here should affect the behaviour of the calls imedialtely.

I have changed one of my Target List Rule , where I changed my gateway Id
to another gateway. But the calls are still being routed via previous
gateway.

Is there something I can do to make the changes in production hours.
Sometimes one of carriers delivers bad call quality I would like to switch
the carriers.


-- 
Regards,
Nitesh Lohchab
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Re: [SR-Users] rtpengine sending rtp to wrong endpoint after reinvite

[Daniel-Constantin Mierla]

Hello,

good to know -- useful information to have in mind.

Cheers,
Daniel


On 08/03/2017 09:49, Grant Bagdasarian wrote:
>
> Hi Daniel,
>
>  
>
> Thank you for the answer.
>
> I’ve also asked the same question on the rtpengine github page and
> they suggested to try the asymmetric flag and that fixed the issue.
>
> Another fix has been suggested, but I haven’t tried it yet.
>
>  
>
> For anyone else interested in the same issue:
>
> https://github.com/sipwise/rtpengine/issues/330
>
>  
>
> Regards,
>
>  
>
> Grant Bagdasarian
>
> CM
>
>  
>
> *From:*sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
> Behalf Of *Daniel-Constantin Mierla
> *Sent:* dinsdag 7 maart 2017 23:06
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* Re: [SR-Users] rtpengine sending rtp to wrong endpoint
> after reinvite
>
>  
>
> Hello,
>
>  
>
> On 07/03/2017 13:10, Grant Bagdasarian wrote:
>
> Hi,
>
>  
>
> One of our customers is using a SEMS box to place two outbound
> calls using our sip trunk.
>
> Once the first call is connected a second call is placed and when
> the second call answers their server sends a re-invite to switch
> audio ports so the rtp traffic doesn’t flow through their server
> anymore but is routed inside our platform.
>
> Basically, they just switch SDP’s of both calls.
>
> It seems like a random issue, and is not really reproducible,
> except for placing multiple calls and sometimes both parties can
> hear each other, other times they can’t, because rtpengine fails
> (I think) to update the endpoint and keeps sending rtp back to
> their server for one of the call legs.
>
>  
>
> We tried to reproduce the case using a freeswitch box and it
> worked every time. After the reinvite, the rtp remained within our
> platform.
>
> The signaling in both cases still goes through the freeswitch or
> sems for call control.
>
>  
>
> Does anyone have experience with this case? Or seen the issue
> before where rtpengine keeps sending rtp to the original endpoint?
>
>
> Have your checked to see if the sip messages are received/processed in
> the expected order?
>
> In some very rare situations, it happened that the re-invite was sent
> very fast by callee after just sending the 200ok, so that the
> re-invite arrived to the proxy/rtprelay before the 200ok, so at the
> end the sdp from 200ok was taken as the last relevant one for the
> peer. I put there rtprelay, because I faced this issue where I had
> rtpproxy, but maybe the issue is exposed by the rtpengine as well.
>
> Cheers,
> Daniel
>
> -- 
> Daniel-Constantin Mierla
> www.twitter.com/miconda  -- 
> www.linkedin.com/in/miconda 
> Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - 
> www.asipto.com 
> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com 
> 

-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - 
www.asipto.com
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Re: [SR-Users] [SIP Transaction] Handle Cancel message

[Daniel-Constantin Mierla]

Hello,

can you provide pcap (or full ngrep output) for all messages of such a
call, from initial invite, including all responses and the other
requests? The sip trace has to be taken on kamailio server.

Kamailio doesn't send 481 from the C code in the case of call, it either
receives it or it is something in config.

Cheers,
Daniel


On 09/03/2017 06:59, Hai Bui Duc Ha wrote:
> Hi all,
>
> I have problem when use module pusher on Kazoo.
> The user after receive call from pusher can not receive the Cancel
> message if have another people pick up this phone.
> I trace log and debug on Kamailio and see the Freeswitch sent Cancel
> message but Kamailio reply 481 - Call leg/transaction does not exist.
>
> The configure on Kamailio: default.cfg
> ===
> /route[PREPARE_INITIAL_REQUESTS]/
> /{/
> /if (is_method("CANCEL")) {/
> /if(t_lookup_cancel()) xlog("L_INFO","$ci|log|
> t_lookup_cancel()");/
> /xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CAPTURED IN
> MAIN---\n");/
> /if (t_check_trans()) {/
> /xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CHECK TRANS
> TRUE---\n");/
> ///xlog("L_INFO",
> "$ci|log|");/
> /route(RELAY);/
> /} else {/
> /xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CHECK TRANS
> FALSE---\n");/
> /sl_send_reply("481", "Call leg/transaction does not exist");/
> /}/
> /exit();/
> /} else if (is_method("ACK")) {/
> /if (t_check_trans()) {/
> /route(RELAY);/
> /}/
> /exit();/
> /}/
> ===
>
> As I read
> on http://kamailio.org/docs/modules/4.3.x/modules/tm.html#tm.f.t_check_trans
> Kamailio see the cancel message not same the transaction INVITE message.
>
> INVITE message:
> 
> INVITE sip:1...@quydang.htk.cvoice SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.77:11000;rport;branch=z9hG4bKK9jyp8mXUZHgS
> Route: http://192.168.1.77:5060>>
> Max-Forwards: 49
> From: "02 quy" ;tag=04K2SappeyDtj
> To: 
> Call-ID: 19fa9228-ef95-11e6-b473-1f645951fd7b
> CSeq: 103030653 INVITE
> Contact:  >
> User-Agent: Anttel
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event,
> dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 247
> X-AUTH-IP: 192.168.1.141
> X-AUTH-PORT: 64799
> X-KAZOO-AOR: sip:1...@quydang.htk.cvoice
> X-KAZOO-PUSHER-Token-Proxy: sip:192.168.1.77:5060
> 
> X-KAZOO-PUSHER-Token-ID:
> 803F0D6610AC5979C6F0513A3A4BE62E0BAF1530DC7A98C2C0DC13D784585FBE
> X-KAZOO-PUSHER-Token-Type: apple
> X-KAZOO-PUSHER-Token-App: com.htkinc
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "02 quy"
> ;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1486708670 1486708671 IN IP4 192.168.1.77
> s=FreeSWITCH
> c=IN IP4 192.168.1.77
> t=0 0
> m=audio 24764 RTP/AVP 0 8 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> 
>
> CANCEL message:
> 
> CANCEL sip:1...@quydang.htk.cvoice SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.77:11000;rport;branch=z9hG4bKK9jyp8mXUZHgS
> Route: http://192.168.1.77:5060>>
> Max-Forwards: 49
> From: "02 quy" ;tag=04K2SappeyDtj
> To: 
> Call-ID: 19fa9228-ef95-11e6-b473-1f645951fd7b
> CSeq: 103030653 CANCEL
> Reason: SIP;cause=200;text="Call completed elsewhere"
> Content-Length: 0
> 
>
> How can I know it same transaction ? We have any idea to solve this
> problem ?
> Thank for advice !
>
> Regards,
> Hai Bui
>
> -- 
> Hai Bui
> VoIP engineer, Cvoice team, HTK-HCM Office
> Mobile: +84-165-618-9876
>
>
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-- 
Daniel-Constantin Mierla
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Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - 
www.asipto.com
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Re: [SR-Users] Access avp between sip transactions

[Daniel-Constantin Mierla]

Hello,


On 09/03/2017 11:57, Grant Bagdasarian wrote:
>
> Hi,
>
>  
>
> Is it possible to access an avp variable from a different sip transaction?
>
> On an INVITE is set an avp, but on ReINVITE I would like to access
> this avp again and get the value. Currently the value is null, which
> makes sense.
>
> Not sure if this is possible and if it violates any rules inside the
> Kamailio engine?
>
>  
>
> I  guess I could use a htable, but I’m just wondering if there is a
> built-in way for doing this.
>
>
you cannot access an avp set for invite during a (later) re-invite,
because the avps are destroyed with the transaction -- so shortly after
200ok for first invite is received, that transaction is destroyed, along
with its avps.

You can use htable as you already said or dialog variables (dialog module).

Cheers,
Daniel

-- 
Daniel-Constantin Mierla
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Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - 
www.asipto.com
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[SR-Users] Access avp between sip transactions

[Grant Bagdasarian]

Hi,

Is it possible to access an avp variable from a different sip transaction?
On an INVITE is set an avp, but on ReINVITE I would like to access this avp 
again and get the value. Currently the value is null, which makes sense.
Not sure if this is possible and if it violates any rules inside the Kamailio 
engine?

I  guess I could use a htable, but I'm just wondering if there is a built-in 
way for doing this.

Regards,

Grant Bagdasarian
CM
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