February 2023 - sr-users

Getting the name of an already set onreply_route
by Denys Pozniak 14 Oct '24

14 Oct '24

Hello! Is it possible to find out the name of the onreply_route that was set before? Something like this: t_on_reply("MANAGE_REPLY"); ... if ( t_is_set("onreply_route") ) { get_onreply_route_name(); ... } -- BR, Denys Pozniak

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Via header force change protocol to TLS?
by Anthony Alba 30 May '24

30 May '24

I have kamailio behind a TLS termination proxy so the sockets are correctly deduced to be TCP. However the clients only talk TLS to the proxy and are confused when the top Via header added by Kamailio is TCP. Is there a way for Kamailio to forcibly pretend its protocol is TLS? Like advertised_address but "advertised_protocol" instead. (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio and continues to use the persistent TLS transport to proxy. Linphone fails because it tries to honor TCP in Via and is unable to establish TCP transport). BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy as TCP even though the contact addresses specify transport=TLS.

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Dialog and DMQ on restart
by Björn Klasen 25 Mar '24

25 Mar '24

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

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Increase MAX_BRANCHES_LIMIT
by Lars Jansson 18 Sep '23

18 Sep '23

Hi, I am building a configuration script where for some traffic flows parallel forking will be needed, and in this case I need to go beyond the default max limit of the max amount of branches. Apart from this specific case the traffic load that kamailio will need to handle is very low, let's say just one second here and there with up to 5 concurrent calls at most. This limit is as far as I understand set here: usr/local/src/kamailio-5.6/kamailio/src/core/config.h And the default limit is: #define MAX_BRANCHES_LIMIT 32 /*!< limit of maximum number of branches per transaction */ Here are the questions I have related to this: 1) If I increase the value of this constant in config.h, how high is it reasonable to set this value and still have a stable system? 2) If I increase MAX_BRANCHES_LIMIT beyond 32, are there also other parameters that needs to be changed for the system to be able to cope, and if so whichparameters? Regards, Lars

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Is Min-SE check broken?
by harneet singh 31 May '23

31 May '23

Hi All, I am facing an issue in understanding how the min_se should be working in kamailio. As per the SST documentation, it seems like if the min_se is configured as 500, then any value of Session-Expires OR MIN-SE if lower than 500, can be responded to by a 422. However, I strangely see the reverse happening. To investigate further, I looked in to the ki_sst_check_min() code in the master, and these seems like a potential issue. Ref Code: Inside ki_sst_check_min(), there is an if condition like below: if (sst_min_se < MIN(minse, se.interval)) { However, shouldn't it be the other way around? ie if (sst_min_se > MIN(minse, se.interval)) { because we need to send 422 if the received value(in INVITE etc) is smaller than the sst configure min_se value? I also found a different documentation, at https://git.sgu.ru/oldssu/ex-opensips/blob/cb9df8d59dbb254a9d862569fd5d11f6… where the check is as below? if (sst_min_se > MIN(minse, se.interval)) { Can someone confirm if this is broken, or my understanding is incorrect? Regards, Harneet -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

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Performing async_http calls on stateless Loadbalancers
by Julius Flohr 09 May '23

09 May '23

Hi List, we are running several stateless SIP Loadbalancers using Kamaillio 5.1 and 5.6. Now we would like to perform an async_http request if an INVITE matches certain criteria. The request itself is performed very similar to how it's described here [1] This uses the transaction module in order to assure that the response can be mapped to the original request. However, this also means that the processing of the entire request becomes stateful which we would like to avoid. We tried to use t_release() in the callback route (in [1] this route is called "RELAY_API_RESPONSE"). However, documentation says that this is only allowed in the REQUEST_ROUTE. Is there an easy way to circumnavigate this issue? Cheers [1]: https://wazo-platform.org/blog/kamailio-routing-with-rtjson-and-http-async-… -- Julius Flohr - Voice Engineer - flohr(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Mobil: +49 (0)1579-000-0205 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

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Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 02 May '23

02 May '23

On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: > Sure. Attached. Problem appears to be that the topos query can't find > callid-totag (from the response). > > I'll try the same scenario with the mysql backend to see if it behaves > different. Config works fine with mysql as topos backend. So the bug is restricted to topos-redis.

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Rtpengine: no audio after kernelization.
by Michel Pelletier 17 Apr '23

17 Apr '23

Hello, I am proxying all RTP through RTPEngine. Everything works fine until about 5 seconds into the call, when rtpengine enters kernelization, after which all RTP forwarding ceases. I've checked the required iptables entries, and all looks good. Here is a description of my environment: # cat os-release PRETTY_NAME="Debian GNU/Linux 11 (bullseye)" NAME="Debian GNU/Linux" VERSION_ID="11" VERSION="11 (bullseye)" VERSION_CODENAME=bullseye ID=debian HOME_URL="https://www.debian.org/" SUPPORT_URL="https://www.debian.org/support" BUG_REPORT_URL="https://bugs.debian.org/" # kamailio -v version: kamailio 5.6.2 (x86_64/linux) 54a9c1 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 54a9c1 compiled on 16:02:49 Dec 1 2022 with gcc 10.2.1 # rtpengine -v Version: 11.1.1.3-1~bpo11+1 Michel Pelletier

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Dispatcher Incoming
by nutxase 14 Mar '23

14 Mar '23

Hi All I have setup kamailio using dispatcher to proxy registrations from the UAC to asterisk but when asterisk sends an incoming call it does not seem to keep the path header and therefore kamailio sends 404 is there anyway around this? Thanks Sent with [Proton Mail](https://proton.me/) secure email.

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Kamailio v5.6.3 Released
by Daniel-Constantin Mierla 05 Mar '23

05 Mar '23

Hello, Kamailio SIP Server v5.6.3 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.3. For more details about version 5.6.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/01/kamailio-v5-6-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

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Releasing Kamailio v5.5.6
by Daniel-Constantin Mierla 02 Mar '23

02 Mar '23

Hello, I am considering to release Kamailio v5.5.6 soon, branch on branch 5.5, likely on Thursday or Wednesday next week (Mar 2/3, 2023). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 1

Websocket closing
by David Villasmil 01 Mar '23

01 Mar '23

Hello guys, We're seeing corner cases where the following happens: On proxy-ws - IP1:PORT1 connects via websocket from Client1 - Registration happens on an upstream kamailio - for any reason, the TCP socket closes or times out. - IP1:PORT1 (same IP:PORT combination) connects via websocket from Client2 - Registration happens on an upstream kamailio Now a call comes in to Client1. Because the first connection was never cleaned up, it is sent to the proxy-ws and the proxy will send it to the IP1:PORT1 where Client2 is connected. Short story, proxy-ws doesn't check the IP1:PORT1 where it is sending the INVITE is the actual client it is supposed to be sending... It seems that when a socket is closed, the mapping IP:PORT to Address (i.e.: sip:d4f27e34@994f31243be9.invalid;transport=ws) doesn't seem to be cleared... is this by design? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Kamailio v5.6.4 Released
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, Kamailio SIP Server v5.6.4 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.4. For more details about version 5.6.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/02/kamailio-v5-6-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. An additional note to say that call for speakers and registration for Kamailio World Conference 2023 are now open, more details at: * https://www.kamailioworld.com Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Kamailio MS Teams TLS Issue
by iliusha.md＠gmail.com 27 Feb '23

27 Feb '23

Hello, We have one Kamailio Instance connected with MS Teams (based on this instruction: https://skalatan.de/en/blog/kamailio-sbc-teams), which worked fine for a while until recently we noticed that calls from teams are not working anymore. When I looked through the logs I found that Microsoft cannot establish a TLS connection to our server because of the cipher: TLS accept:error:1408A0C1:SSL routines:ssl3_get_client_hello:no shared cipher (sni: sbc.example.com - domain is obfuscated). Certificate is valid, the configuration is below: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt server_name = sbc1-teams.example.net ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt [client:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt We use a certificate from Sectigo, but I've tried with Let's Encrypt - and it's the same. Any idea what could be the reason?

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Releasing v5.6.4
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, I am considering to release Kamailio v5.6.4 (out of branch 5.6) next week (likely on Monday or Tuesday, Feb 27/28, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 1

Kamailio RPC configuration
by sadik.oualla.mohamed＠gmail.com 24 Feb '23

24 Feb '23

Dear Kamailio community, I am a newbie in Kamailio and I am currently working on a project that involves communicating with Kamailio using Python. I have been exploring the Kamailio documentation and I have come across RPC, JSONRPCS..., which I believe can help me achieve my goal. However, I am not sure about the steps to follow in the Kamailio configuration file to create my own script to communicate with Kamailio using Python (the script is on an other machine on the same LAN). I would be grateful if someone could guide me through the process. I have already set up Kamailio (builed it from the source code 5.5.5) and Python on my machines and I have basic knowledge of Kamailio configuration. I just need some guidance on how to set up the RPC module in Kamailio and how to create a Python script to communicate with it. I found that I can use RPyC library of python. I would appreciate any help or suggestions from the Kamailio community. Thank you in advance for your time and assistance. Best regards, Wild Coder.

2 1

OT: MS Teams source IP address ranges
by Jon Bonilla (Manwe) 24 Feb '23

24 Feb '23

Hi Sorry for the OT but I think here's the place where I an find a lot of Ms teams integrations I've been working on MS teams direct routing integration for PekePBX. It works. I guess I've done it as everybody else, using Henning's guide as base and extending it for multitenant setup (thanks Henning!) What I've realized is that the source IP address of calls coming from MS are not always matching dispatcher hosts. Sometimes they come from another source IP and failover to the dispatcher hosts when they receive no response. That makes some of the calls to have an additional latency Searching in the MS doc I see that they document these nets as source of their signaling: 52.112.0.0/14 52.120.0.0/14 But I've seen IP addresses outside of this range as source. In this blog https://erwinbierens.com/microsoft-teams-direct-routing-ip-addresses/ The ranges are listed as 52.112.0.0/16 52.113.0.0/16 52.114.0.0/16 52.115.0.0/16 52.120.0.0/16 52.121.0.0/16 52.122.0.0/16 52.123.0.0/16 which looks better but scares me out. Having no auth is it secure to bind so many ranges to MS? Do you use anything else than certificate verification for these calls? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

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Dispatcher failover when SRV failover is on?
by Jon Bonilla (Manwe) 23 Feb '23

23 Feb '23

Hi Again, fighting with MS Teams but this time it's a kamailio question. I have MSteams hosts in the dispatcher for OPTIONS and for destination. Acording to MS you need to try sip.pstnhub.foo then sip2.pstnhub.foo and sip3.pstnhub.foo 2 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 1 2 | sip:sip2.pstnhub.microsoft.com;transport=tls | 0 | 2 2 | sip:sip3.pstnhub.microsoft.com;transport=tls | 0 | 3 I was going to implement dispatcher failover as usual ds_select_domain (alg 8) with failure routes but checking the SRV of the first domain I see it already includes 4 hosts: host -t srv _sips._tcp.sip.pstnhub.microsoft.com _sips._tcp.sip.pstnhub.microsoft.com has SRV record 10 0 5061 sip.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 20 0 5061 sip2.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 30 0 5061 sip3.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 40 0 5061 sip4.pstnhub.microsoft.com. So... If I have dns srv failover enabled in kamailio... what would be the point of the dispatcher failover? In case I only want failover in case 408 or 5XX wouldn't the tm module do it for me? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

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What does l_username in UAC do?
by Дилян Палаузов 23 Feb '23

23 Feb '23

Hello, the uac module-registrations have the fields l_uuid - user for Contact: in REGISTER (before @). Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local unique id used to build and match contact addresses.”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local unique user id, e.g.,: 12345678” l_username - Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local username”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local user name, e.g.,: test ”. For me there is no difference between “local unique id” (as for l_uuid) and “local username“ (here). l_domain - used for Contact: in REGISTER (after @), described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “local domain” When is l_username used, what is it purpose? Does it solely serve as possible filter criterion for the RPC commands: uac.reg_info l_username abc uac.reg_enable l_username abc uac.reg_disable l_username abc, and uac.reg_unregister l_username abc Do I see correctly, that via UAC-RPC an unREGISTER can be triggered, but there is no way to trigger a REGISTER over UAC-RPC? Thanks for the clarifications Дилян

2 2

change default 477 response code per case
by Patrick Karton 22 Feb '23

22 Feb '23

Hello , i would like to know if there is a way to change the default 477 response code generated by kamailio in some cases. for example if kamalio generates 477 because its not able to reach a peer i would like to change it to another code before to forward it. Thanks.

2 1

Call History
by Anthony Wittig 21 Feb '23

21 Feb '23

Hello, I'd like to start tracking a call history/log as soon as I receive a call. I'd like the customer to be able to see that there's an incoming call to one of their numbers in real time. I'm thinking of using the RABBITMQ Module to manually emit an event from my main route when I receive an INVITE. Is that a bad idea? What if the broker isn't located on the same host (latency/availability)? I don't want to negatively affect the call processing. I was looking at the Dialog Module and it has three event routes: * event_route[dialog:start] * event_route[dialog:end] * event_route[dialog:failed] It's really close to what I want, but I'd like to capture the dialog before it starts (Unconfirmed or Early). Maybe another option is to scan the Dialog table with an external process and emit events from there. Has anyone had success in creating a realtime call history? Is there a better way to go about doing so? Thanks! -- Anthony Wittig Mango Voice Developer

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Info: Registration to Kamailio World 2023 is open!
by Daniel-Constantin Mierla 21 Feb '23

21 Feb '23

Hello, a quick note to the community forums to announce that the registration for the next Kamailio World Conference (June 5-7, 2022, in Berlin, Germany) is now open! The event returns to an in-person conference format, at the same wonderful location in the city center of Berlin! More details are available at: * https://www.kamailioworld.com/k2023/registration/ Call for presentations is also open, there is already a group of very interesting submissions, we aim to publish details about them in the near future. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

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topos breaking route set on hairpin calls?
by Benoît Panizzon 20 Feb '23

20 Feb '23

Hi Having this situation while elaborating topos in lab. * Kamailio Registration Instance with topos enabled where customers are registered. * Kamailio Core Instance with Dialogue enabled and handling interconnections. Calls in transit from customer to IC work flawlessly, topos does what it should, hides and restores headers needed on the core. But I have run into an issue with customer calling another customer on same registrar instance. To generate a CDR with diaglogue and for decision where to route the calls to (registrar is kept simple and only handles registrations and locations), those calls are also routed via core. This works as long as PRACK is disabled. When PRACK is enabled and CPE B is requesting 100rel, the PRACK send from CPE A has no route set (it was removed by topos). Therefore the PRACK only makes it to the core who then does not know where to route that PRACK. It looks like topos on the REG only restores ONE Route pointing to the core (maybe because the 2nd one pointed to itself?) header when restoring them. I have attempted to spam more record_route() here and there in the config of the core and the registrar to try to force creation of route headers, but failed. Is this a known limitation of using topos? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Re: OT: MS Teams source IP address ranges
by Henning Westerholt 20 Feb '23

20 Feb '23

Hi Jon, you could of course try to add some heuristic, in the end the used IPs are a smaller sub-set. But to just quote from the official docs, it seems to be pretty clear: "You need to open ports for all these IP address ranges in your firewall to allow incoming and outgoing traffic to and from the addresses for signaling." Cheers, Henning ________________________________ From: Jon Bonilla (Manwe) Sent: Monday, February 20, 2023 15:49 To: Henning Westerholt; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] OT: MS Teams source IP address ranges El Mon, 20 Feb 2023 13:06:30 +0000 Henning Westerholt <hw(a)gilawa.com> escribió: > Hello, > > correct me if I am wrong, but the second group of addresses ("/16") is > included in the first group ("/14"), right? > > Regarding the large IP scope, this is the way Microsoft designed it, > unfortunately. There is not that much what you can do about. > > Right. Didn't check the masks. they all were 16 in my mind :) So, if I add those ranges in the permissions table my only way of avoiding spoofed calls is certificate validation? I'm already doing that and also checking the company domain and ms account email address. Just wanted to be sure that nothing "essential" was missing. -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

sendig 4 byte udp nat pings
by Bernd Krueger-Knauber 18 Feb '23

18 Feb '23

Hi, maybe I'm stupid, but I can not find out how to send 4 byte UDP NAT packets to the sip server. loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 15) modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") I'm a absolute beginner with kamailio and the page about the nathelper Module is not really helpful. At the moment no pings are sendet. But I receive 4 byte packets from the sip server. After 15 minutes my calls are cancelled because of the missing pings. Without proxy, the phone send the pings directly to the sip server and the call stays alive. When I experiment with some other nathelper settings, kamailo start results in errors. Any help would be appreciated. Best regards, Bernd

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enable topology hiding on one side or dynamic ip in topoh module Contact and Via Header
by Patrick Karton 17 Feb '23

17 Feb '23

Hello , im using kamailio with two interfaces external and internal. i need a way either to : * enable topoh hiding only when outgoing interface is external ( mask contact and Via ip only when ougoing interface is external * or if it is not possible to enable it only in one direction.i want to know how to configure dynamic(for example in xavp or avp) ip to put in Contact and Via when topoh is enabled. i see that the 'mask_ip' parameter of topoh module is a string. so we can not set a dynamic value here unfortunately. Thanks

4 8

Calling uac_replace multiple times?
by Benoît Panizzon 17 Feb '23

17 Feb '23

Hey all I have started using the uac module to be able to mirror username changes on both legs of a call. But now I found an issue: Assume this situation: Call IC => Customer IC is e164, Customer wants 'local' numbers. IC From: +41991234567 => Customer From: 0991234567 uac_replace_from("sip:" + $avp(from_local) + "@" + $(fu{uri.host})); Customer is absent, after timeout in failure_route call is dispatched to Voicemail Server, which is configured for e164 numbers: uac_replace_from("sip:" + $avp(from_e164) + "@" + $(fu{uri.host})); But then uac complains: replace_uri(): Already called uac_replace for this dialog Why isn't just the most from version kept and used? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

TLS version and cipher from WSS
by Michal Hajek 17 Feb '23

17 Feb '23

Hi, I can log incoming SIPS TLS version and cipher with PV $tls_version and $tls_cipher_info. How can I log these variables on incoming HTTP/WebSocket. Thanks Michal H.

2 2

Identify error in log
by Евгений Мастратов 16 Feb '23

16 Feb '23

Hello colleagues. I'm starting to use Kamailio for educational purposes, as an IMS test zone. Registration of subscribers, calls go successfully, but the following errors are constantly pouring into the log: Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Apparently they don’t carry anything critical, but they clog the log, natively I can’t understand the meaning of the errors. Please tell me what they mean and how to eliminate their cause? If necessary, I will provide the pcscf config.

2 1

uac reg reload frequency
by Dmytro M 16 Feb '23

16 Feb '23

When you do "kamcmd uac.reg_reload" frequently enough, you'll get an error: # kamcmd uac.reg_reload error: 500 - Failed to shift records - check log messages Logs: ERROR: uac [uac_reg.c:416]: uac_reg_ht_shift(): shifting in-memory table is not possible in less than 150 secs What is the reason behind 150 sec limitation for how frequent reloads can be done? Is it configurable in any way? Use case: Remote app needs to disable uac registration. It deletes record in DB and calls kamcmd uac.reg_reload to activate the change. With 150 sec limit it can't apply DB changes immediately. Please advise. Thanks, Dmytro

2 2

ACK on RE-Invite is send to UAC NATed Internal port instead of public port
by Angelo Sipper 15 Feb '23

15 Feb '23

Hi, My setup has: (TLS)UAC 77.1.1.1 -> (TLS)Kamailio 17.17.17.211 (UDP) -> (UDP)Kamailio 17.17.17.181 (UDP) -> Carrier(UDP) Both Kamailio are 4.3 with minor modifications on default cfg, and Topoh and TLS modules are used. I am having an issue on some outbound calls from nated UACs. On these calls our SBC sometimes receives reverse invites from the carrier side within the call. When this happens, my SBC(17.17.17.211) sends the invite to UAC and the UAC sends back 200 OK (SDP). Unfortunately my SBC(17.17.17.211) sends back the ACK not to the correct UAC public port but instead to the UAC internal port making the call to crash with several retries from my SBC to send 200 OK as ACK was never received from UAC! If you look the below traces of reverse invite past within the call, the reinvite has port private 20125 and alias the public port 2314 of UAC but the ACK bellow is send to 20125 and not to 2314 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060 INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314 INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 68 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Content-Length: 0 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 276 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Any help will be much appreciated. Thank you! Regards, Angelo

2 5

SCA Presence : Record-Route Filed Not Update in sca_subscriptions Table
by Vinayak Makwana 14 Feb '23

14 Feb '23

Hello all, I am using the sca module in kamailio script.so when i subscribe with my ip phone for SCA Presence i get all the information in a sca_subscriptions table for the first time. When I reboot the device I get a new record-route header in trace but It does not update *record-route *Filed in sca_subscriptions table. Can anyone help me out with this issue ? *Here's modparam configuration * modparam("sca", "db_update_interval", 10 ) modparam("sca", "db_url", DBURL_LOCAL) modparam( "sca", "line_seize_max_expires", 30) modparam( "sca", "purge_expired_interval", 1800) -- -- * <https://www.ecosmob.com/itexpo/> * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

3 8

SCA Feature:: Record-route query.
by satyaprakash ch 13 Feb '23

13 Feb '23

Hello I have a query regarding the record-route field update in DB for SCA SUBSCRIBE. Here are my test cases *Initial SCA subscription,* We got the below SUBSCRIBE packet in kamailio SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081381rdb66 *Record-Route* : <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=CFFBAF19-19A68CBC Call-ID: bd4b50d9384523c14ea05c532d895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 also in the sca_subscriptions table record-route header is updated as per Initial SUBSCRIBE record-route. *record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> * But After rebooting the device, we got a new record-route in SUBSCRIBE Packet but in DB Record-route Field not updated after getting an updated record-route in Trace. *Kamailio Trace *SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081415rdb48 *Record-Route*: <sip:sansay1081415rdb48spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=35852031-CF749E20 Call-ID: 41fa031d79e35fae4a61c61984895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 as per logic it should update the record-route field in sca_subscriptions table . but it didn't update it : record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> So can you suggest to us how we can update the record-route field as per SUBSCRIBE's packet new record-route ?

2 1

Issues matching regex
by Nathan Stratton 11 Feb '23

11 Feb '23

I am having an issue matching toll-free. I tested my regex on regex101.com, and it looks good, but it does not match my if statement. Am I doing something wrong? if (method=="INVITE") { xlog("L_INFO", "Checking Routes $rU"); if ($rU=~"^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$") { xlog("L_INFO", "Tollfree call"); }; }; Logs show "<script>: Checking Routes +18002211212" so I don't get why it does not match. ><> Nathan Stratton Founder, CTO Vocinity, Inc. <https://www.vocinity.com/gabbi>

2 1

kemi equivalent config
by mohsen khashei 11 Feb '23

11 Feb '23

2 1

KEMI Equivalent of script
by mohsen khashei 10 Feb '23

10 Feb '23

Hi I have an existing kamailio config and I want to migrate to kemi. I want to know what is the equivalent of this code in kemi javascript ? $avp(s:contact) = $hdr(Contact); $avp(s:contact_fix) = $hdr(Contact); avp_subst("$avp(s:contact_fix)", "/@([0-9]+).([0-9]+).([0-9]+).([0-9]+)/@$sel(cfg_get.kamailio.ss_ip)/gi"); remove_hf("Contact"); append_hf("Contact: $avp(s:contact_fix)\r\n"); t_on_reply("REG_FIX"); thanks for your attention.

1 0

Return value from lua/python/js/bash/etc... function
by Dmytro M 10 Feb '23

10 Feb '23

What is the simplest way to return values from external application into Kamailio config and assign it to a variable? I'm looking for the most light weight mechanism to do it and to offload some logic to external app. Something like this would be ideal: $var(myVar), $var(err) = lua_run("lua_func1", "$rU"); OR $var(encString) = exec_cmd("echo 'abc $rU' | base64"); There is this method: http_client_query("http://...", "", "", "$var(result)"); but there is an overhead of networking and HTTP protocol. Please advise. Thanks, Dmytro

3 4

ACK on Reverse Invite is send to NATed Internal port instead of public port
by Angelo Sipper 09 Feb '23

09 Feb '23

Hi, I am having an issue on some outbound calls from nated UAC. On these calls our SBC receive reverse invites from the carrier side within the call. When this happens, my SBC(Kamailio1) sends the invite to UAC and the UAC sends 200 OK (SDP). Unfortunately my SBC(Kamailio1) sends back the ACK not to the correct UAC public port but instead to UAC internal port making the call to crash! My setup has: (TLS)UAC -> (UDP)Kamailio1(UDP) -> (UDP)Kamailio2(TLS) -> Carrier(UDP) Bellow you can find the trace and you can see the ACK send to wrong port on 2023/01/26 09:48:31.254591 Your help will be much appreciated! 2023/01/26 09:48:23.225326 77.1.1.1:2314 -> 17.17.17.211:5060 INVITE sip:300987654321@17.17.17.211:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport Max-Forwards: 70 Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060> From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Remote-Party-ID: <sip:301234567890>;party=called Content-Length: 283 v=0 o=3cxPS 792019937001472 2393157908037633 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:23.240310 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314;received=77.1.1.1 To: <sip:300987654321@17.17.17.211:5060> From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:23.301454 17.17.17.211:5060 -> 17.17.17.181:5060 INVITE sip:300987654321@sip.mydomain.com:5060 SIP/2.0 Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com> From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 300 P-Asserted-Identity: <sip:301234567890@sip.mydomain.com> v=0 o=3cxPS 792019937001472 2393157908037633 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:23.302899 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0;rport=5060 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 To: <sip:300987654321@sip.mydomain.com> From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:24.077941 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 183 Session Progress From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:24.138348 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 183 Session Progress From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:24.139050 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 183 Session Progress From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:24.200575 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 183 Session Progress From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.115324 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.336507 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.432457 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 194.154.142.193 s=SIP Media Capabilities c=IN IP4 194.154.142.196 t=0 0 m=audio 10764 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2023/01/26 09:48:30.433017 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 194.154.142.193 s=SIP Media Capabilities c=IN IP4 194.154.142.196 t=0 0 m=audio 10764 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2023/01/26 09:48:30.457268 77.1.1.1:2314 -> 17.17.17.211:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.481470 17.17.17.211:5060 -> 17.17.17.181:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.44ee18452b274c8de283aa3c27549f4e.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.555470 77.1.1.1:2314 -> 17.17.17.211:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.557824 17.17.17.211:5060 -> 17.17.17.181:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.44ee18452b274c8de283aa3c27549f4e.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060 INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314 INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 68 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Content-Length: 0 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 276 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.530836 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.531333 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.609667 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6cff4a8159c123f434b422c73e17a08c.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.612312 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.63edf375f6b899dfa9d2d9a2dcee4dfd.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6cff4a8159c123f434b422c73e17a08c.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:32.536512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:32.536839 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:32.623207 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48a5467fc248682a755e70a672268f79.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:32.641928 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.745a55762e847bc3cf56d27926f7e62d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48a5467fc248682a755e70a672268f79.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:34.540735 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:34.541310 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:34.590013 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6c3274e7b34c3b780cfbe9180a4d5eb1.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:34.595763 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.6c6654fa1c2f4b1991d5b6d792f2f081.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6c3274e7b34c3b780cfbe9180a4d5eb1.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:38.561756 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:38.562039 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:38.599703 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.ef7afaa26d39ee76ce9000b1203574e7.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:38.603466 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e022b4e31f8266525e62f213dc4dd8ae.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.ef7afaa26d39ee76ce9000b1203574e7.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:42.584025 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:42.584257 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:42.622809 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.03eec221779917d79f0daa500f7a0cc6.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:42.626545 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.0e211693731e67705c85dd4844e42ce8.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.03eec221779917d79f0daa500f7a0cc6.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:46.629501 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:46.629961 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:46.676968 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.701d02748c09dee63b88ca1e96a4e371.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:46.682458 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.cc80042ad9e1838346f4cf6d66a4c13f.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.701d02748c09dee63b88ca1e96a4e371.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:50.636998 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:50.637287 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:50.675272 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.041bbf55bf3941bfbd16df0f3ca92cf1.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:50.681861 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.be45a1f76f0fcd5ce85292397ff68acc.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.041bbf55bf3941bfbd16df0f3ca92cf1.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:54.677049 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:54.677633 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:54.721399 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6ce35eaf9a5805bf47c868cd80c09478.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:54.729950 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.365607f0beb6e7e5dcb047c1ad6fcf5f.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6ce35eaf9a5805bf47c868cd80c09478.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:58.681010 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:58.681307 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:58.723730 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.abb70cc570d323c867c112765ec50432.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:58.731143 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.12bbae879e42215b50512c04c4fc0214.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.abb70cc570d323c867c112765ec50432.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:02.684483 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:49:02.684985 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:49:02.727999 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.c2b81601667ffb2a688deba47abd85ef.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:02.732652 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.fb4ff819be250670479341711417a554.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.c2b81601667ffb2a688deba47abd85ef.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:03.028200 77.1.1.1:2314 -> 17.17.17.211:5060 BYE sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---7c113301b578bb42;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:49:03.090194 17.17.17.211:5060 -> 17.17.17.181:5060 BYE sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK92d8.6e9f4e9b390a591c1c54d79389ba087d.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:49:03.261453 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE Content-Length: 0 Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bK92d8.6e9f4e9b390a591c1c54d79389ba087d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:03.261941 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE Content-Length: 0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Thank you! Regards, Angelo

2 4

DB_CLUSTER failure with carrierroute (and others?)
by Kaufman 08 Feb '23

08 Feb '23

All, I've been trying to chase down an issue with db_cluster and permissions in combination with something else, where with my full configuration using the same cluster in multiple modules has failures. I've been trying to create a simplified configuration to consistently reproduce the problem without much success, however I did stumble upon THIS particular problem. The configuration here works fine as is - avpops uses the cluster URL, and carrierroute uses the direct SQLite URL. If I define WITH_DBCLUSTER, then carrierroute fails fetching rows using the cluster connection. Note that AVPOPS is only used as a general test to see if the cluster config works. If I remove references to avpops , carrierroute still fails loading from the cluster URL. Thoughts? #!KAMAILIO loadmodule "db_sqlite" loadmodule "pv" loadmodule "db_cluster" loadmodule "carrierroute" loadmodule "avpops" modparam("db_cluster", "connection", "db1=>sqlite:////etc/db.sqlite") modparam("db_cluster", "cluster", "cl1=>db1=9s9s") modparam("db_cluster", "inactive_interval", 30) #!ifdef WITH_DBCLUSTER modparam("carrierroute", "db_url", "cluster://cl1") #!else modparam("carrierroute", "db_url", "sqlite:////etc/db.sqlite") #!endif modparam("carrierroute", "config_source", "db") modparam("avpops", "db_url", "cluster://cl1") modparam("avpops", "avp_table", "usr_preferences") request_route { exit; } Failure log: 0(1) ERROR: carrierroute [cr_db.c:345]: load_route_data_db(): Fetching rows failed 0(1) ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data 0(1) ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data 0(1) ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module carrierroute (/usr/lib/kamailio/modules/carrierroute.so) Kaufman

3 5

Drops on async_route with delay
by Alex Balashov 07 Feb '23

07 Feb '23

Hi, Some percentage of requests processed with async_route("REQ_PROCESS", "5") seem to end up with a resumed transaction, though most do. Not sure what the exact percentage is. The ones that don't Is this due to excessive requests? Is there a limit on internal IPC queue depth? Is it conceptually similar to a generic shared blocking queue internally, along the lines of 'mqueue'? Is there any reasonable way to troubleshoot this? Thanks! -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

3 3

Dispatcher + ENUM
by Jawaid Bazyar 07 Feb '23

07 Feb '23

Hi, Can DISPATCHER be used in conjunction with ENUM? E.g., to avoid sending requests to SIP destinations that are returned by ENUM, but, which DISPATCHER knows are down? Thanks, Jawaid

2 1

nating private uas to public ip addr
by Jack Murphy 06 Feb '23

06 Feb '23

I have a multi-homed kam with a uas behind one private interface and a public interface for the uac+sip provider. When the uac initiates a call, kam sends the invite to the private uas which sets up a b-leg call back to kam with a Contact header that has the uas' private ip address. I want to change that private ip address to be Kamailio's advertised/public ip address before it gets forwarded to the sip provider. I am rewriting Contact in a route from request_route like this: > if ($rP != $null) { > $var(ct) = ";transport=" + $rP; > }else{ > $var(ct) = ""; > } > $var(cr) = "<sip:" + $fU + "@" + $RAi + ":" + $RAp + $var(ct) + ">"; > > remove_hf("Contact"); > append_hf("Contact: $var(cr)\r\n"); This seems to work fine for the INVITE going to the provider. There's also the 200 OK that the uas sends for the uac (a-leg) with a private ip address in Contact -- I want to rewrite that before sending it to the UAC, too. Using the same method as above (but inside onreply_route), what happens is the UAC receives the modified OK and sends back an ACK with the modified Contact (kam's public ip addr) in the ACK r-uri, and kam gets in a loop of sending an ACK from its private ip address to its public ip address and replying to itself. call flows: https://imgur.com/a/Yj2DJpn Is there a right way to do this?

2 2

htable: autoexpire vs updatedelete
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, https://www.kamailio.org/docs/modules/devel/modules/htable#htable.p.htable says for modparam('htable', 'htable', '…autoexpire … updateexpire…') autoexpire -time in seconds to delete an item from a hash table if no update was done to it. If is missing or set to 0, the items won't expire. updateexpire - if set to 1 (default), the time until expiration of an item is reset when that item is updated. Certain uses of htable may dictate that updates should not reset the expiration timeout, however, in which case this attribute can be set to 0. My reading is that, when a value is updated and autoexpire is positive, autoexpire will reset the timer, but updateexpire may or may not reset the same timer. That is: modparam('htable', 'htable', 'a=>autoexpire=5;updatedelete=1') modparam('htable', 'htable', 'b=>autoexpire=5;updatedelete=0') If b was not updated for 5s, will it be removed, as the documentation for autoxpire suggests, irrespective of updatedelete? Greetings Dilyan

2 2

Depending on topoh.mask_key Kamailio either cannot parse RURIs or reads invalid data
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, when I use dns_try_naptr=on corelog=-1 debug=-1 enable_tls=yes use_dns_cache=off dns_cache_init=off modparam("topoh", "mask_key", "TEAI32l)- eauiDEUIA!?()") and run Kamailio under valgrind, Kamailio logs: 20(21) ERROR: kemix [kemix_mod.c:229]: ki_kx_get_ruri_attr(): failed to parse the R-URI 20(21) ERROR: rr [loose.c:1011]: loose_route_mode(): failed to parse Request URI The workflow is: I start Kamailio. A UDP-client registers. Then a websocket client calls the UDP-client. Finally the websocket client hangs up. Valgrind does not report anything suspicious. When I use instead modparam("topoh", "mask_key", "TEAI32l") for the same workflow, with the same configuration, valgrind logs: ==14== Invalid read of size 4 ==14== at 0x4867E5: atomic_cmpxchg_int (atomic_x86.h:224) ==14== by 0x486830: futex_get (futexlock.h:99) ==14== by 0x490CE4: dns_hash_get (dns_cache.c:673) ==14== by 0x4972A6: dns_get_entry (dns_cache.c:2001) ==14== by 0x499288: dns_srv_get_he (dns_cache.c:2455) ==14== by 0x597AAD: no_naptr_srv_sip_resolvehost (resolve.c:1599) ==14== by 0x598332: naptr_sip_resolvehost (resolve.c:1675) ==14== by 0x5983C5: _sip_resolvehost (resolve.c:1707) ==14== by 0x49943B: dns_srv_sip_resolvehost (dns_cache.c:2516) ==14== by 0x49B3DD: dns_sip_resolvehost (dns_cache.c:2738) ==14== by 0x59846A: sip_hostport2su (resolve.c:1727) ==14== by 0x4CB951: forward_request (forward.c:515) ==14== by 0x99993CD: t_relay_to (t_funcs.c:300) ==14== by 0x99EBA32: _w_t_relay_to (tm.c:1764) ==14== by 0x99F4DEB: ki_t_relay (tm.c:2917) ==14== by 0xA80A94B: sr_kemi_lua_exec_func_ex (app_lua_api.c:1022) ==14== by 0xA81237D: sr_kemi_lua_exec_func (app_lua_api.c:1706) ==14== by 0xA81B93F: sr_kemi_lua_exec_func_209 (app_lua_kemi_export.c:1717) ==14== by 0xA8383C0: luaD_precall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA84C4CA: luaV_execute (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA838C90: luaD_callnoyield (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA837029: luaD_rawrunprotected (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8391DF: luaD_pcall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8333DE: lua_pcallk (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA806F0C: app_lua_run_ex (app_lua_api.c:773) ==14== by 0xA825F06: sr_kemi_config_engine_lua (app_lua_mod.c:119) ==14== by 0x5048BE: sr_kemi_route (kemi.c:3784) ==14== by 0x588909: receive_msg (receive.c:502) ==14== by 0xA609ADF: ws_frame_receive (ws_frame.c:644) ==14== by 0x4BCAAB: sr_event_exec (events.c:299) ==14== Address 0x0 is not stack'd, malloc'd or (recently) free'd ==14== ==14== ==14== Process terminating with default action of signal 11 (SIGSEGV): dumping core ==14== Access not within mapped region at address 0x0 ==14== at 0x4867E5: atomic_cmpxchg_int (atomic_x86.h:224) ==14== by 0x486830: futex_get (futexlock.h:99) ==14== by 0x490CE4: dns_hash_get (dns_cache.c:673) ==14== by 0x4972A6: dns_get_entry (dns_cache.c:2001) ==14== by 0x499288: dns_srv_get_he (dns_cache.c:2455) ==14== by 0x597AAD: no_naptr_srv_sip_resolvehost (resolve.c:1599) ==14== by 0x598332: naptr_sip_resolvehost (resolve.c:1675) ==14== by 0x5983C5: _sip_resolvehost (resolve.c:1707) ==14== by 0x49943B: dns_srv_sip_resolvehost (dns_cache.c:2516) ==14== by 0x49B3DD: dns_sip_resolvehost (dns_cache.c:2738) ==14== by 0x59846A: sip_hostport2su (resolve.c:1727) ==14== by 0x4CB951: forward_request (forward.c:515) ==14== by 0x99993CD: t_relay_to (t_funcs.c:300) ==14== by 0x99EBA32: _w_t_relay_to (tm.c:1764) ==14== by 0x99F4DEB: ki_t_relay (tm.c:2917) ==14== by 0xA80A94B: sr_kemi_lua_exec_func_ex (app_lua_api.c:1022) ==14== by 0xA81237D: sr_kemi_lua_exec_func (app_lua_api.c:1706) ==14== by 0xA81B93F: sr_kemi_lua_exec_func_209 (app_lua_kemi_export.c:1717) ==14== by 0xA8383C0: luaD_precall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA84C4CA: luaV_execute (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA838C90: luaD_callnoyield (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA837029: luaD_rawrunprotected (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8391DF: luaD_pcall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8333DE: lua_pcallk (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA806F0C: app_lua_run_ex (app_lua_api.c:773) ==14== by 0xA825F06: sr_kemi_config_engine_lua (app_lua_mod.c:119) ==14== by 0x5048BE: sr_kemi_route (kemi.c:3784) ==14== by 0x588909: receive_msg (receive.c:502) ==14== by 0xA609ADF: ws_frame_receive (ws_frame.c:644) ==14== by 0x4BCAAB: sr_event_exec (events.c:299) The stacktrace with the values of the variables is available at https://github.com/kamailio/kamailio/issues/3350 . Or at least these things seem very similar to me. Any idea? I can share the OCI-image and the full configuration. Greetings Дилян

3 4

topoh: th_mask_decode (th_mask_encode (x)) ≠ x
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, after receiving INVITE sip:test-gnome-calls@bapha.be SIP/2.0 Via: SIP/2.0/WSS bapha.be;branch=z9hG4bK974100 Max-Forwards: 70 To: <sip:test-gnome-calls@bapha.be> From: "Online https://sip.bapha.be" <sip:online@bapha.be>;tag=12q2efc423 Call-ID: fkhe8faq0fh80glmq59i CSeq: 5622 INVITE Contact: <sip:4q5ne5no@bapha.be;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.7.8 Content-Type: application/sdp Content-Length: 1964 v=0 o=mozilla...THIS_IS_SDPARTA-99.0 8263261229339246488 0 IN IP4 0.0.0.0 s=- … kamailio calls th_mask_encode( char *in = "sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60722~2>;+sip.instance=\"<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs=\"conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime\n"\ "Content-Type: application/sdp and so on", int ilen = 107, str prefix = { .len = 23, .s ="sip:127.3.4.84;line=sr-" }, int *olen). It returns olen=167, and the string "sip:127.3.4.84;line=sr- if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p-1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWO9-eo9*". Then Kamailio forwards (sends): Contact: <sip:127.3.4.84;line=sr-if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p- 1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWO9-eo9*>;+sip.instance="<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs="conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime" which at some point leads to ACK sip:127.3.4.84;line=sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0 and BYE sip:127.3.4.84;line=sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0 Below you can find th_test.c . The functions th_mask_encode() and th_mask_decode() there are identical to the same functions in src/modules/topoh/th_mask.c . main() provides some tests: Passing «sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60472~2>;+sip.instance="<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs="conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime» to th_mask_encode() returns sip:127.3.4.84;line=sr-if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p- 1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWXi-eo9* (this is what Kamailio does send out). Passing the last string to th_mask_decode() returns sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50-740fb9bdc562;alias=87.118.1 the ;alias= is incomplete, compared to the origin. The above ACK leads to passing "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p-1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0\n"\ "Route: <sip:Q+L06HX94DMPtQWQTI5OE8SQTI5O4MY=@144.76.142.78;transport=ws;r2=on;lr;nat=yes>\n"\ "Route: <sip:127.3.4.84;line=sr-if7s17IB5XEfVzn0WIgWwqggCTzwV5C2g5R5wCg4WwrQ5CuZ6XIQWvH3WUHkWX9tWOnyMq4RbmWsb34Q2YgoiXP-coTAboPlioPtpY6Z8dCO>\n"\ to th_mask_decode() and the result is callub.p.b;rn:s.d2lQg#cfa=bA07beQ7R409bcQ62;alac=87718*1P.15#~60R22~2 with the dots being unprintable characters. (result is garbage). I have seen that callub.p.b in the unparsable R-URI reports from Kamailio. That is an indication that th_mask_decode() does not decode the Contact encoded by th_mask_encode() correctly. Valgrind does not report improper memory usage. Greetings Dilyan gcc -g -o th-test ./th-test.c && ./th-test th-test.c #include <stdio.h> #include <stdlib.h> #include <string.h> #define TH_EB64I \ "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789.-" char _th_EB64[65] = "EFvXIzGq94xKcW126gV5wCdYpNSbiM8enRUou7LmhajBlPtAsk-OQTf3H0DyrZ.J"; int _th_DB64[256]; char *_th_PD64 = "*"; struct str_ { char* s; /**< Pointer to the first character of the string */ int len; /**< Length of the string */ }; typedef struct str_ str; str prefix = { .len = 23, .s ="sip:127.3.4.84;line=sr-" }; char* th_mask_encode(char *in, int ilen, const str *prefix, int *olen); char* th_mask_decode(char *in, int ilen, const str *prefix, int extra, int *olen); char* th_mask_encode(char *in, int ilen, const str *prefix, int *olen) { char *out; int left; int idx; int i; int r; char *p; int block; *olen = (((ilen+2)/3)<<2) + ((prefix!=NULL&&prefix->len>0)?prefix->len:0); out = (char*)malloc((*olen+1)*sizeof(char)); if(out==NULL) { fprintf(stderr, "malloc error\n"); *olen = 0; return NULL; } memset(out, 0, (*olen+1)*sizeof(char)); if(prefix!=NULL&&prefix->len>0) memcpy(out, prefix->s, prefix->len); p = out + (int)((prefix!=NULL&&prefix->len>0)?prefix->len:0); for(idx=0; idx<ilen; idx+=3) { left = ilen - idx - 1 ; left = (left>1)?2:left; block = 0; for(i=0, r=16; i<=left; i++, r-=8) block += ((unsigned char)in[idx+i]) << r; *(p++) = _th_EB64[(block >> 18) & 0x3f]; *(p++) = _th_EB64[(block >> 12) & 0x3f]; *(p++) = (left>0)?_th_EB64[(block >> 6) & 0x3f]:_th_PD64[0]; *(p++) = (left>1)?_th_EB64[block & 0x3f]:_th_PD64[0]; } return out; } char* th_mask_decode(char *in, int ilen, const str *prefix, int extra, int *olen) { char *out; int n; int block; int idx; int i; int j; int end; char c; for(n=0,i=ilen-1; in[i]==_th_PD64[0]; i--) n++; *olen = (((ilen-((prefix!=NULL&&prefix->len>0)?prefix->len:0)) * 6) >> 3) - n; if (*olen<=0) { return NULL; } out = (char*)malloc((*olen+1+extra)*sizeof(char)); if(out==NULL) { fprintf(stderr, "malloc error\n"); *olen = 0; return NULL; } memset(out, 0, (*olen+1+extra)*sizeof(char)); end = ilen - n; i = (prefix!=NULL&&prefix->len>0)?prefix->len:0; for(idx=0; i<end; idx+=3) { block = 0; for(j=0; j<4 && i<end ; j++) { c = _th_DB64[(int)in[i++]]; if(c<0) { free(out); *olen = 0; return NULL; } block += c << (18 - 6*j); } for(j=0, n=16; j<3 && idx+j< *olen; j++, n-=8) out[idx+j] = (char)((block >> n) & 0xff); } return out; } int main() { //this is from th_mask_init for(int i=0; i<256; i++) _th_DB64[i] = -1; for(int i=0; i<64; i++) _th_DB64[(int)_th_EB64[i]] = i; int olen, o2; char *end; char *input = "sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60472~2>;+sip.instance=\"<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>\";+org.linphone.specs=\"conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime\""; printf("input=[%s]\n", input); char* middle = th_mask_encode (input, 107, &prefix, &olen); printf("olen = %i, middle=[%s]\n", olen, middle); end = th_mask_decode(middle, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); char * b1 = "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9*"; o2 = 0; end = th_mask_decode(b1, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); char* b2 = "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p-1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0\n"\ "Route: <sip:Q+L06HX94DMPtQWQTI5OE8SQTI5O4MY=@144.76.142.78;transport=ws;r2=on;lr;nat=yes>\n"\ "Route: <sip:127.3.4.84;line=sr-if7s17IB5XEfVzn0WIgWwqggCTzwV5C2g5R5wCg4WwrQ5CuZ6XIQWvH3WUHkWX9tWOnyMq4RbmWsb34Q2YgoiXP-coTAboPlioPtpY6Z8dCO>\n"\ "Via: SIP/2.0/WSS bapha.be;branch=z9hG4bK9069155\n"\ "Max-Forwards: 70\n"\ "To: <sip:test-gnome-calls@bapha.be>;tag=Pn4npAj\n"\ "From: \"Online https://sip.bapha.be\" <sip:online@bapha.be>;tag=12q2efc423\n"\ "Call-ID: fkhe8faq0fh80glmq59i\n"\ "CSeq: 5623 ACK\n"\ "Supported: outbound\n"\ "User-Agent: SIP.js/0.7.8\n"\ "Content-Length: 0\n"\ "\n"; o2 = 0; end = th_mask_decode(b2, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); return 0; }

2 1

no autocomplete for kamcmd for latest debian packages
by Ovidiu Sas 04 Feb '23

04 Feb '23

Hello, The latest debian packages are built without autocomplete for kamcmd. Do we have the libreadline-dev package installed on the build machine? I know that we have kamctl rpc and kamcli, but if we are still shipping kamcmd, it should have the autocomplete feature enabled. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

3 2

Sip messages responses counters per interface
by Patrick Karton 02 Feb '23

02 Feb '23

2 1

rtpengine - SRTP <> RTP
by kdrewicz+kamailio＠cludo.pl 01 Feb '23

01 Feb '23

I try to workout if - currently it would work, or - where and how to debug more: I face - 2 interfacec - public internet (so, TLS + sRTP) is desired and private - old infrastructure - i mus only use plain RTP 172.23.9.70 - private ip - from this endpoint of kamailio and rtpengine should send only basic RTP 172.23.210.75:5060 private - target for kamailio 1.2.3.24 obfuscated public IP (TLS + sRTP required) kamailio 5.4.4 (x86_64/linux) rtpengine -v Version: 11.1.1.4-1~bpo11+1 all i do is: if (proto==TLS) { rtpengine_manage("RTP/AVP ICE=remove replace-session-connection replace-origin pad-crypto ptime=20 codec-transcode-PCMA record-call=on allow-transcoding direction=external direction=internal record-call=on"); } else if ($ru =~ "transport=tls") { rtpengine_manage("DTLS=on SRTP AVPF ICE=remove replace-session-connection replace-origin pad-crypto ptime=20 codec-transcode-PCMA record-call=on allow-transcoding direction=internal direction=external record-call=on media-address=1.2.3.24"); } # 1.2.3.24 obfuscated public IP 172.23.210.75:5060 is in dispatch.cfg, as '11' route[SBC_CORE] { append_hf("X-My-SRTP: removed31337\r\n"); ### i see this text in invtes from kamailio 172.23.9.70 towards 172.23.210.75:5060 ### i see only RTP, so as expected if (!ds_select_dst("11", "0")) { xwarn("I:$var(i) DROP(DOWN!) FWD:$rm [$fU->$tU] [SBCVIP] to $du\n"); sl_send_reply("503", "Destination down"); exit; } what i did: certificate is a paid one (the public party needs it) TLS works i deleted - entries in (not kamailo) cryptosuite that caused this: 13:08:05 localhost rtpengine[15140]: ERR: [51ad8758-b64d-4d2f-9fd0-41d03a38f74d]: [core] Failed to parse a=crypto attribute, ignoring: unknown crypto suite Tried to search for any "ready" examples for this - only found old threads and - that this should be possible, but - no example for woking config. what i see: Jan 19 19:00:57 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000038 sec Jan 19 19:00:58 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000036 sec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Closing call due to timeout Jan 19 19:00:59 localhost rtpengine[17301]: DEBUG: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Redis delete_async=0 Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Final packet stats: Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- Tag 'JVR5LTs', created 60:00 ago for branch '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscribed to '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscription for '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] ------ Media #1 (audio over RTP/SAVP) using unknown codec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 1.2.3.24:30136 <> 52.129.106.28:17030, SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 1.2.3.24:30137 <> 52.129.106.28:17031 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- Tag '', created 60:00 ago for branch '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscribed to 'JVR5LTs' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscription for 'JVR5LTs' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 172.23.9.70:30014 <> :0 , SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 172.23.9.70:30015 <> :0 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Moved metadata file "/var/spool/rtpengine/tmp/rtpengine-meta-c17bab16-5eea-492e-b1c4-ac9387f3e265-7003946f152c6c8d.tmp" to "/var/spool/rtpengine/metadata" Jan 19 19:00:59 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000828 sec Jan 19 19:01:00 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000053 sec route(SBC_CORE); maybe any hint or - someone has working exmaple of kamailio config + rtpengine settings ? i use only userspace daemon rtp forwarding (this is a test, dont need any performance here) Thanks,

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sr-users February 2023