sr-users

sr-users@lists.kamailio.org

42 participants
37519 discussions

Presence between different domains
by Manuel Perez 29 Feb '12

29 Feb '12

Hello, I've used kamailio 3.2.2, i have rls, simple presence and xcap support working. I've replicated this server into another one in a different domain, but presence between domains is not working. I'm using for presence presence.so, presence_xml.so and presence_mwi.so. In route[WITHINDLG] i have the following lines: ... if (is_method("SUBSCRIBE|NOTIFY") && uri==myself) route(PRESENCE); ... When NOTIFY from user B at domain B arrives to the server at domain A, the message is not forwarded to the user A, How can i achieve this? Thank you.

2 3

dialog doesn't clear calls that were re-INVITEd shortly after ACK
by Andrew Pogrebennyk 29 Feb '12

29 Feb '12

Hi, I noticed dialog module showing calls that have already been terminated with state=4. The kamailio version is 3.1.5, I'm aware of certain limitations of dialog module in this version (must use stateless replies, create dialog after 407), but this is something rather new. In kamailio config I'm using dlg_manage() for initial INVITEs and BYE/CANCEL and in-dialog requests are loose-routed, so nothing finicky. The call scenario is: A calls B, B answers the call, A immediately after sending ACK for the 200 OK issues re-INVITE. After some seconds A hangs up, and the dialog stays in memory in state=4, e.g.: dialog:: hash=3304:623200556 state:: 4 ref_count:: 2 timestart:: 1329938667 timeout:: 78590167 callid:: MGExMjRiYzMzN2Q5YWU4YjRiMjFjYTYwYWZlMGFlMzY. from_uri:: sip:sipwise-user1@67.202.62.202 from_tag:: 4207e429 caller_contact:: sip:sipwise-user1@80.108.64.151:46420 caller_cseq:: 3 caller_route_set:: <sip:127.0.0.1:5060;lr;r2=on>,<sip:67.202.62.202:5060;lr;r2=on> caller_bind_addr:: udp:127.0.0.1:5062 callee_bind_addr:: udp:127.0.0.1:5062 to_uri:: sip:43991002@67.202.62.202 to_tag:: 6C8F89E9-4F4540EA0001F575-AC557700 callee_contact:: sip:sipwise-user2@127.0.0.1:5080 callee_cseq:: 11 callee_route_set:: When I set dlg_match_mode=1 the problem goes away. I would like however to understand why the dialog fails to clear in default mode (DID_ONLY), because the dialog cookie seems to be preserved by caller in all in-dialog requests properly. There is nothing in the debug log and trace attached that catches my eye. Do you see anything wrong there? In trace file the proxy address is 127.0.0.1:5062. You may notice that the actual signaling flow is a little bit more complicated, I will gladly answer any questions about it. For this test I've used two lines in the eyeBeam softphone, when one line answers the other is automatically put on hold which is what I want. Then I un-hold the callee and clear the call. The customer is reporting multiple stale calls when re-INVITE is sent by Cirpack PSTN gw immediately after call connect, which is basically the same call flow I am talking about, it is just not practical for me to run kamailio with debug=3 there all the time. I am going to try theredlg_match_mode=1 though and see if the problem goes away. Any ideas?

1 1

Re: [SR-Users] sr-users Digest, Vol 81, Issue 95
by Yufei Tao 29 Feb '12

29 Feb '12

Hi Thanks for your reply! I already set the accounting flag like you say. And all normal BYEs are inserted into the acc table fine. Just internally generated BYEs don't get inserted in acc table. I've done a bit googling and added event_route: event_route[tm:local-request] { if (is_method("BYE")) { xlog("L_INFO", "event_route: BYE: callid=$ci, cseq=$cs - internally generated\n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } } When I have two clients in a call, I disconnect the network for both clients, so no RTP either direction, which triggered mediaproxy to send dlg_bye_dlg to Kamailio, which sent BYE to both clients. I can see from the log that event_route did get executed, and BYE sent to both clients, but still BYE isn't inserted to acc table. The code inside event_route() above is copied from my route[WITHINDLG], which for client-generated BYEs operates correctly (inserts BYEs into acc table). But in event_route it didn't work. I also tried to use sql_query to insert BYE into acc table in event_route and it causes Kamailio to crash. Using sql_query to select * from acc also causes Kamailio to crash. Any idea where I've done wrong? I'm very much lost. Thank you! Yufei > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 27 Feb 2012 13:21:50 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] How to do accounting for Kamailio-generated > BYEs? > To: sr-users(a)lists.sip-router.org > Message-ID: <4F4BC9BE.8000704(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 02/27/2012 12:58 PM, Yufei Tao wrote: >> Hi >> >> I've got a mediaproxy that detects no media, and sends dlg_end_dlg to >> Kamailio via mi_datagram socket. Then Kamailio sends BYE to both calling >> parties. It all works fine. But my question is, how to do accounting for >> these Kamailio-generated BYEs? They don't appear in the acc table. > In the loose_route() section of the config script, set the accounting > flag for BYEs, e.g. > > if(is_method("BYE")) > setflag(xx); > > -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

1 2

Re: [SR-Users] CallControl and MediaProxy
by Reda Aouad 29 Feb '12

29 Feb '12

Ok thanks Daniel. I'll do what you suggested and we'll see how to proceed. Thanks again Reda On 27 févr. 2012, at 17:46, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, On 2/26/12 11:01 AM, Reda Aouad wrote: Daniel, I sent you the log file by private email, for privacy concerns. ok, no time yet to look over them, but as you say it does not call mediaproxy explicitely from config and callcontrol, maybe you should look inside mediaproxy module, might be automatically executed on dialog or tm module callbacks -- then the patch has to be done for mediaproxy module. I took a quick look at the log files, and didn't see any call to mediaproxy module. Though, the mediaproxy log file indicates a media session engaged, and CDRTool shows media information for the established session. The mediaproxy module is loaded, but no function from the module is called at all. It can be via some internal callbacks registered to some modules such as tm and dialog, or even script execution callbacks... so look over the code of mediaproxy module. You can run kamailio with debug=3 and from the logs you should get some hints about what is the way media proxy is executed. Cheers, Daniel Thanks Reda On Sat, Feb 25, 2012 at 18:58, Reda Aouad <reda.aouad(a)gmail.com> wrote: > I looked in detail in the source of the call control module and didn't > find anything related to the mediaproxy module neither.. > I'll send you the output of cfgtrace soon. > > Reda > > > > On Fri, Feb 24, 2012 at 23:46, Daniel-Constantin Mierla <miconda(a)gmail.com > > wrote: > >> A quick grep over the callcontrol sources didn't reveal a point where >> mediaproxy is engaged. >> >> Can you load debugger module and enable cfgtrace, then run such a >> scenario and send out the output from cfgtrace to see all the config >> actions executed? >> >> Cheers, >> Daniel >> >> >> On 2/24/12 7:56 PM, Reda Aouad wrote: >> >> It would be great Daniel if you could do it. >> I will be more than happy to test it. >> A function parameter would be more flexible than a module parameter. >> I expect it shouldn't affect the behavior of the external application >> CallControl. >> >> Thanks in advance :) >> >> Reda >> >> >> >> On Fri, Feb 24, 2012 at 08:09, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> I am not using mediaproxy at all (but nathelper/rtpproxy), neither the >>> call control module, but making an option (module parameter or function >>> parameter) for call control to bind to another module like media proxy, >>> should not be big deal if it is all you are looking for -- I can look over >>> it and send a patch if you are going to help testing it. I cannot do it >>> these days, though, being out of the office. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 2/23/12 8:59 PM, Reda Aouad wrote: >>> >>> First, I am posting about the wrong behavior of CallControl (or most >>> probably Kamailio modules) which leaves no option. I should be the only >>> one deciding about how to handle timeouts. If I decide to take some risk, >>> no module should oblige me to do otherwise. >>> >>> Mediaproxy detects ONLY RTP timeouts from BOTH parties, because linux >>> conntrack rules it uses are bi-directional. If a single party stops sending >>> RTP for whatever reason (connection lost, codec with silence detection >>> used, ....), mediaproxy doesn't care and doesn't act upon it. This is a >>> feature, and a wanted one, to mainly support voice-detecting codecs. Think >>> also about conferences for example, in which only a single person talks for >>> a long time while others are silent and don't send RTP. >>> >>> Single-side RTP timeout because of a real problem (loosing network >>> connection for example) should be handled with other methods, such as SIP >>> session timers. >>> >>> MY POINT IS : I don't see it practical to handle RTP flows for EVERY >>> call to handle the least probable scenario: an RTP timeout from both (or >>> all) parties. >>> >>> If I understood well, mediaproxy updates the CDR when it detects an >>> RTP timeout from both parties. CallControl can look in the CDR to debit the >>> correct balance, instead of attaching itself to the dialog module to detect >>> dialog termination. >>> >>> This is an extract from the call_control module : >>> >>> Even when mediaproxy is unable to end the dialog because it was not >>> started with engage_media_proxy(), the callcantrol application is still >>> able to detect calls that did timeout sending media, by looking in the >>> radius accounting records for entries recorded by mediaproxy for calls that >>> did timeout. These calls will also be ended gracefully by the callcontrol >>> application itself. >>> >>> >>> Unless there is something I miss.. >>> >>> I also opened a bug about the issue because call_control doesn't have >>> the same behavior with OpenSips. It doesn't force mediaproxy. >>> >>> Reda >>> >>> >>> >>> On Thu, Feb 23, 2012 at 20:00, Jeff Brower <jbrower(a)signalogic.com>wrote: >>> >>>> Reda- >>>> >>>> > It's clear but not necessary. It can look at radius records fixed by >>>> > mediaproxy on RTP timeout to debit the correct balance as well. And >>>> why >>>> > also force it on postpaid calls which it doesn't control at all ? >>>> >>>> I don't understand how you plan to tear down Kamailio calls that >>>> suffer RTP time-out? >>>> >>>> -Jeff >>>> >>>> > What happens is cost and performance issues for additional calls >>>> passing >>>> > through my mediaproxy server, which I didn't plan for at first. No >>>> audio >>>> > issue at all. >>>> > >>>> > Reda >>>> > >>>> > >>>> > >>>> > On Thu, Feb 23, 2012 at 11:58, Sammy Govind <govoiper(a)gmail.com> >>>> wrote: >>>> > >>>> >> Reading from the module docs its clear why it needs to engage >>>> media/rtp >>>> >> proxy to start,stop billing or timer of a call. so what happens when >>>> it >>>> >> engages mediaproxy on unwanted calls !? audio-issues? >>>> >> >>>> >> >>>> >> On Thu, Feb 23, 2012 at 1:21 PM, Reda Aouad <reda.aouad(a)gmail.com> >>>> wrote: >>>> >> >>>> >>> Thanks Sammy. I didn't get any reply yet. >>>> >>> >>>> >>> CallControl is an application used with CDRTool for prepaid calls. >>>> It >>>> >>> calculates the maximum call duration based on the user's balance. >>>> Once the >>>> >>> call's duration limit is reached, it sends BYE to both calling >>>> parties, >>>> >>> terminating the call. At the end of a prepaid call, terminated >>>> either by >>>> >>> the user or by CallControl, it debits the user's balance according >>>> to the >>>> >>> call's duration. >>>> >>> >>>> >>> The Call_Control module interfaces with this external application. >>>> >>> >>>> >>> call_control function is called in Kamailio's cfg to check if the >>>> user >>>> >>> has prepaid or postpaid account, and get the max call duration for >>>> prepaid >>>> >>> users. CallControl controls only prepaid calls, not postpaid ones. >>>> >>> >>>> >>> So call control and NAT traversal using mediaproxy are two >>>> differents >>>> >>> things which i can't link, since I don't want mediaproxy for every >>>> call. >>>> >>> And since the function call_control is called on every invite before >>>> >>> knowing if the user has a prepaid account or not, it engages >>>> mediaproxy for >>>> >>> every call. >>>> >>> >>>> >>> CallControl relies on mediaproxy to detect RTP timeouts and debit >>>> the >>>> >>> correct balance from a prepaid account based on the last instant the >>>> >>> mediaproxy saw an RTP packet. >>>> >>> >>>> >>> But why to force using mediaproxy with no choice? And why to force >>>> it for >>>> >>> every call, whether it falls under CallControl's control or not? >>>> >>> >>>> >>> I am using Kamailio 3.2. >>>> >>> >>>> >>> >>>> >>> Reda >>>> >>> >>>> >>> On 23 févr. 2012, at 07:21, Sammy Govind <govoiper(a)gmail.com> >>>> wrote: >>>> >>> >>>> >>> Hi, >>>> >>> I can see you posting multiple times on both proxies listings so >>>> I'm sure >>>> >>> you havent heard back from anyone.I am not at all familiar with your >>>> >>> functions in email but could it be possible for you to determine on >>>> which >>>> >>> calls you need to engage mediaproxy and on which not to, then on >>>> the base >>>> >>> of that flag use the call_control function ! >>>> >>> your problem is complicated for me atleast. I hope somebody could >>>> answer >>>> >>> you accurately and precisely. >>>> >>> >>>> >>> btw, what are you using in real? opensips or kamailio, which >>>> version? and >>>> >>> in what context you need to use the call_control function? >>>> >>> >>>> >>> Thanks, >>>> >>> Sammy >>>> >>> >>>> >>> On Thu, Feb 23, 2012 at 12:45 AM, Reda Aouad <reda.aouad(a)gmail.com >>>> >wrote: >>>> >>> >>>> >>>> Hi, >>>> >>>> >>>> >>>> When I use the function call_control( ) of the call_control >>>> module, it >>>> >>>> automatically engages mediaproxy if it finds the mediaproxy module >>>> loaded. >>>> >>>> If the mediaproxy module is not loaded, call_control doesn't even >>>> try to >>>> >>>> engage it. >>>> >>>> >>>> >>>> I need mediaproxy for NAT traversal in some cases, but don't want >>>> it to >>>> >>>> be engaged on every call. >>>> >>>> >>>> >>>> How can I disable this behavior? >>>> >>>> >>>> >>>> Thanks >>>> >>>> Reda >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> >>>> sr-users(a)lists.sip-router.org >>>> >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>>> >>>> >>>> >>> _______________________________________________ >>>> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> >>> sr-users(a)lists.sip-router.org >>>> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >>>> >>> >>>> >>> _______________________________________________ >>>> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> >>> sr-users(a)lists.sip-router.org >>>> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >>>> >>> >>>> >> >>>> >> _______________________________________________ >>>> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> >> sr-users(a)lists.sip-router.org >>>> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >> >>>> >> >>>> > _______________________________________________ >>>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> > sr-users(a)lists.sip-router.org >>>> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> > >>>> >>>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>> >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda

2 4

obtaining htable array index
by Asgaroth 28 Feb '12

28 Feb '12

Hi All, Is it possible to asertain the array index of an htable entry based on its value? For example, say I have an htable called foo with the following entries: Entry:: 11 testing[0]:: hello-2 testing[1]:: 1 testing::size:: 2 is it possible to: [a] extract the array index based on a regex of the value, for example, something like value count: shtcv(foo=>hello-*) [b] if [a] is possible, how would I extract that value (from the associated array index returned in [a] above) from the htable using sht()? Is the above possible, or am I using the wrong tool for the job? I did have a look at the htable module docs, but i didnt see anything that stood out to me showing that i could achieve the above. I saw an shtval exported function, but i dont see it documented in the pseudo-var wiki page. I then tried it in the routing script but got an error message to the affect that it couldnt find the PV. I assume its not exported. All comments welcome and appreciated? Thanks

2 4

Dropping 183 and sending 180
by Fabian Borot 27 Feb '12

27 Feb '12

Hello We need to implement this scenario: client -> proxy --> provider The provider sends 183 with SDP and we need to drop it and send 180 instead. I managed to do it like this: 1- on_reply block, I catch the 183 if it comes, call a route block named "DROP_180" then drop it: if ($rs == "183") { route(DROP_180); drop(); } 2- on the new route block "DROP_180" I use the command send_reply(180, "Ringing") route[DROP_180] { xlog("L_INFO","mylog: On Route DROP_180."); send_reply(180,"Ringing"); } I had to do it on a separate block because the send_reply command could not be used on the same on_reply block, also I had to send the 180 first before the 183 is dropped because the drop() apparently stops the script execution and the 180 was not sent The idea works fine, BUT the To tag on the locally generated 180 is not the same as the one on the original 183 message. After the progress (183 message), it comes the 200 OK, which we forward downstream unaltered. Now, the client seems to ignore the 200 OK, (there is no ACK), we think that since the To tag on the original 183 and the 200 OK does not match the one on the 180 that is why the customer is processing the 200 OK. Any ideas here? txs a lot

3 4

XMLOPS xpath issues
by Olle E. Johansson 27 Feb '12

27 Feb '12

Friends, I'm having issues with the xpath support. If I run Daniel's example in the XMLOPS, xpath works as documented. But if I take the body of a PUBLISH from the and run xpath, I don't get expected results. Now, I'm no XPATH guru so I may be totally off the markup here... Here's the test script: $xml(pub=>doc) = $rb; $var(contact)=$xml(pub=>xpath:registration/contact@event); xlog("--- Publish contact $var(contact) \n"); xlog("--- Publish aor $xml(pub=>xpath:registration@aor) \n"); xlog("--- Publish aor: $xml(pub=>xpath://@aor) \n"); xlog("--- Publish reg state: $xml(pub=>xpath://reginfo@state) \n"); xlog("--- Publish uri: $xml(pub=>xpath:contact/uri/text()) \n"); I've tested with various paths, orginating from /reginfo or just picking an attribute. Here's the XML from Carsten's PUA_REGINFO module: <?xml version="1.0"?> <reginfo xmlns="urn:ietf:params:xml:ns:reginfo" version="0" state="full"> <registration aor="sip:oej@testnamn.se" id="0x80590a590" state="terminated"/> </reginfo> I wonder if it's something with using the request body that doesn't parse properly in the XMLOPS module? Error messages are a series of "XPath error : Invalid expression" /O

4 6

Using Voicemail strange behavior - git head 3.3
by David 27 Feb '12

27 Feb '12

I was testing the head version in GIT; I ran across a strange issue. The new config includes a "WITH_VOICEMAIL" definition I am not sure exactly how it is designed to work, but by default if the "usrloc" fails it creates a branch and goes to voicemail That part work correctly. It seems however if a phone is registered and the you get a busy, the failure route creates a branch but also retries the original RURI but I dont see a fork being created for the new branch to the voicemail server. Seems to just retry the original RURI. I am using the default config from the kamailio. repo. Any ideas. Thanks.

1 0

change DBURL without restarting kamailio?
by Uri Shacked 27 Feb '12

27 Feb '12

Hi, Is there a way to set a DBURL as a parametes that i can change in run time without restarting kamailio? It is usfull if i have a backup data base and i dont want to use mysql clustering... Cheers, Uri

2 1

CSEQ problem with SPA504G
by David 27 Feb '12

27 Feb '12

Hello, I am trying to connect to a kamailio server using a Cisco SPA504G phone using TCP instead of TCP. When in UDP, it works fine, but in TCP I get errors relating to CSEQ. I included the trace taking with ngrep after the email. I don't understand the CSeq error. I had a look at RFC3261, it says that the CSEQ number has to increase and that's what it has done. Can anyone tell me where the errorlies ? I tried changeing the check sanity to not check that CSeq is valid, but it screws up authentication as it mis detects the method type. Please advise, David interface: eth0 (SERVERIP/255.255.255.128) filter: (ip or ip6) and ( port 5060 and host 206.248.130.233 #### T 206.248.130.233:5075 -> SERVERIP:5060 [AP] REGISTER sip:SERVERNAME.TLD SIP/2.0. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-b03863b5;rport. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50446 REGISTER. Max-Forwards: 70. Contact: "104 Zone T" <sip:testzone.104@192.168.1.225:5075;transport=tcp>;expires=3600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE. Supported: replaces. . ## T SERVERIP:5060 -> 206.248.130.233:5075 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 192.168.1.225:5075;received=206.248.130.233;branch=z9hG4bK-b03863b5;rport=5075. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=a889ade7c63805944f01d11bb71a916b.6a29. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50446 REGISTER. WWW-Authenticate: Digest realm="SERVERNAME.TLD", nonce="4f4b87c31ea308c5fdf54c90cf2b30a400fa97a7". Server: OmniVigil 5.0a. Content-Length: 0. . ## T 206.248.130.233:5075 -> SERVERIP:5060 [AP] REGISTER sip:SERVERNAME.TLD SIP/2.0. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-dcfc09e;rport. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50447 REGISTER. Max-Forwards: 70. Authorization: Digest username="testzone.104",realm="SERVERNAME.TLD",nonce="4f4b87c31ea308c5fdf54c90cf2b30a400fa97a7",uri="sip:SERVERNAME.TLD",algorithm=MD5,response="77ddf2a1baafd33ed0c675fdc074bc03". Contact: "104 Zone T" <sip:testzone.104@206.248.130.233:5075;transport=tcp>;expires=3600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE. Supported: replaces. . ## T SERVERIP:5060 -> 206.248.130.233:5075 [AP] SIP/2.0 400 CSeq method does not match request method. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-dcfc09e;rport=5075;received=206.248.130.233. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=1feca66ab82b1f32a901c4cd8a4b7a05.e8e6. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50447 REGISTER. Server: kamailio (3.1.5 (x86_64/linux)). Content-Length: 0. .

2 1

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users