[SR-Users] Re: ACK and Bye not relaying when record_route is not used
As mentioned above - use TOPOS BUT - return record_route() back. On Sun, Jun 2, 2024, 08:40 Nick Digalakis via sr-users < sr-users@lists.kamailio.org> wrote: > Hi! > > Check out the TOPOS module. > > It does topology hiding and it should do what you want. > > On Jun 2, 2024 08:13, Arun K R via sr-users > wrote: > > Issue : Not getting relay of ACK and BYE to the next hop after the call is > answered > my Scenario : Asterisk --->kamailio sip proxy---> > carrier (outgoing call) > > My carrier is not allowed to send the SIP packet with Record-Route header. > So that I have removed record_route(). After that the call is getting > connected. > I am getting 200 OK (SDP) from carrier side and forwarded that to the > Asterisk on the other side. As a response I am getting ACK from asterisk. > But the kamailio is not forwarding the ACK to the carrier side. I > understood this is because the record-route is not there. The same thing is > happening for BYE also. The Bye is not forwarding to carrier side. > > Kindly suggest me a solution for this for relaying ACK and bye without > Record-Route in kamailio > > Bellow is the 200 OK SDP I am sending back to asterisk > 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 > SIP/2.0 200 OK > Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 > Via: SIP/2.0/UDP 103.182.153.113:5060 > ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af > To: >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 > From: >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 > CSeq: 22823 INVITE > Allow: > INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE > Contact: ;alias=10.5.110.117~5060~1;x-afi=11> > Content-Type: application/sdp > Session-Expires: 7200;refresher=uas > Supported: timer > Content-Length: 248 > > v=0 > o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 > s=- > c=IN IP4 103.155.114.101 > t=0 0 > m=audio 12806 RTP/AVP 8 101 > > --- > > The ACK I am getting back from asterisk is > 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 > ACK > sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 > SIP/2.0 > Via: SIP/2.0/UDP 103.182.153.113:5060 > ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce > From: >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 > To: >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 > Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 > CSeq: 22823 ACK > Max-Forwards: 70 > User-Agent: Asterisk PBX 18.13.0 > Content-Length: 0 > > Thanks > Arun > > > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: compile without downtime
If you do not store dialogs in the memory and not rely on transactional variables used for passing data between request/reply handling there is nothing bad regarding the downtime as state handled by the UA but not by proxy If you depend on such things the migration is really depends on your infrastructure ( using DNS as failover or VRRP and HA, or build cluster with DMQ, or distribute data between nodes over database) so in general advice - have multiple instances and transit traffic to one while upgrading another... On Wed, Apr 17, 2024, 18:59 satyaprakash ch via sr-users < sr-users@lists.kamailio.org> wrote: > HI > > I currently have two proxy servers running in my environment, and I want > to upgrade Kamailio from version 5.4.7 to 5.6. > > Can I recompile the Kamailio service before restarting it without downtime > ? Will there be any impact on my production environment if I recompile > Kamailio to the newer version? If there is no impact, I expect > approximately 2 minutes of downtime during the service restart. > > Here's the scenario: > > *No Downtime:* > Clone the Kamailio repository with git clone > https://github.com/kamailio/kamailio.git > Navigate to the Kamailio directory with cd kamailio > Update the modules.lst file with any necessary additions > Compile with make > Install with make install > > *Downtime:*Restart the Kamailio service. > > > Please advise on this. > > Thanks. > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Memory leaks - KEMI Lua
Regarding lua vars - no.
LuaVM runs all the time. KEMI uses not the same approach FreeSWITCH or
asterisk uses
On Tue, Apr 16, 2024, 12:34 Muhammad Danish Moosa
wrote:
> Hi,
> Docs says
> It is not necessary to free a reply to use it again in a new redis_cmd
> function. When ndb_redis module closes, all pending replies are freed
> automatically..
>
> Question is when is ndb_redis module closed, if there are no active calls
> it should be closed, shouldn't it?
>
> I am using variable in each Lua function, even if they are global they
> should be wiped off after call is finished?
>
> I will see how to work collectgarbage("collect") as explained by Michael
> Hajek in another email.
>
> Thanks guys.
>
>
>
>
>
> On Tue, Apr 16, 2024 at 8:27 PM Yuriy G via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
>> You don't using redis_free
>> And also your lua bars seems to be globals...
>>
>> On Tue, Apr 16, 2024, 10:44 Michal Hajek via sr-users <
>> sr-users@lists.kamailio.org> wrote:
>>
>>> Hi,
>>>
>>> you can try calling collectgarbage("collect") in Lua code.
>>>
>>> Michal
>>>
>>> út 16. 4. 2024 v 9:25 odesílatel Muhammad Danish Moosa via sr-users <
>>> sr-users@lists.kamailio.org> napsal:
>>>
>>>> Hi,
>>>>
>>>> I am using kamailio with KEMI Lua. The script has some medium level
>>>> complexity as it stores some information from the invite in redis
>>>> cache during the call and uses that information during the call transfers
>>>> (REFER and Invite triggered by REFER).
>>>>
>>>> Lua script itself creates/sets some variables, and
>>>> sends/retrieves information during the flow for example
>>>> KSR.ndb_redis.redis_cmd("srvN", "SET coreCI-" .. varCI .. " " ..
>>>> KSR.pv.getw("$ci") .. " EX 3600", "r");
>>>> KSR.ndb_redis.redis_cmd("srvN", "GET refvar-" .. cid, "r");
>>>> reft=KSR.pv.getw("$redis(r=>value)");
>>>>
>>>> I made a test call and disconnected, what I am observing is that even
>>>> if the call is finished the used memory is slightly increased.
>>>>
>>>> before:
>>>> < real_used: 844240
>>>> < real_used: 844088
>>>> < real_used: 844088
>>>> < real_used: 844088
>>>> < real_used: 844088
>>>> < real_used: 844088
>>>> < real_used: 844088
>>>> < real_used: 844088
>>>> after:
>>>> > real_used: 846080
>>>> > real_used: 845864
>>>> > real_used: 846584
>>>> > real_used: 846304
>>>> > real_used: 846200
>>>> > real_used: 846120
>>>> > real_used: 845248
>>>> > real_used: 846168
>>>>
>>>>
>>>> My assumption is after finishing the call, memory used by variables and
>>>> redis ops is released and should be back to original values but its not
>>>> happening. In production this issue is causing some problems as I have to
>>>> restart the kamailio process after every 3-4 weeks so that it starts
>>>> breathing with more available memory.
>>>>
>>>> Initially I thought its Kamailio version issue as in my last posts in
>>>> this forum people advised me to upgrade to the latest kamailio version
>>>> (original one was 5.4.0) but the same issue is happening on 5.7.0.
>>>>
>>>> Can someone explain how to debug/troubleshoot this situation . I have a
>>>> parallel lab environment, I do not see anything interesting in syslogs.
>>>>
>>>>
>>>> --
>>>> Muhammad Danish Moosa
>>>>
>>>> " The core of mans' spirit comes from new experiences. "___ Christopher
>>>> McCandless
>>>> __
>>>> Kamailio - Users Mailing List - Non Commercial Discussions
>>>> To unsubscribe send an email to sr-users-le...@lists.kamailio.org
>>>> Important: keep the mailing list in the recipients, do not reply only
>>>> to the sender!
>>>> Edit mailing list options or unsubscribe:
>>>>
>>> __
>>> Kamailio - Users Mailing List - Non Commercial Discussions
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>>> Important: keep the mailing list in the recipients, do not reply only to
>>> the sender!
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>>>
>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
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>> Important: keep the mailing list in the recipients, do not reply only to
>> the sender!
>> Edit mailing list options or unsubscribe:
>>
>
>
> --
> Muhammad Danish Moosa
>
> " The core of mans' spirit comes from new experiences. "___ Christopher
> McCandless
>
__
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[SR-Users] Re: Memory leaks - KEMI Lua
You don't using redis_free
And also your lua bars seems to be globals...
On Tue, Apr 16, 2024, 10:44 Michal Hajek via sr-users <
sr-users@lists.kamailio.org> wrote:
> Hi,
>
> you can try calling collectgarbage("collect") in Lua code.
>
> Michal
>
> út 16. 4. 2024 v 9:25 odesílatel Muhammad Danish Moosa via sr-users <
> sr-users@lists.kamailio.org> napsal:
>
>> Hi,
>>
>> I am using kamailio with KEMI Lua. The script has some medium level
>> complexity as it stores some information from the invite in redis
>> cache during the call and uses that information during the call transfers
>> (REFER and Invite triggered by REFER).
>>
>> Lua script itself creates/sets some variables, and
>> sends/retrieves information during the flow for example
>> KSR.ndb_redis.redis_cmd("srvN", "SET coreCI-" .. varCI .. " " ..
>> KSR.pv.getw("$ci") .. " EX 3600", "r");
>> KSR.ndb_redis.redis_cmd("srvN", "GET refvar-" .. cid, "r");
>> reft=KSR.pv.getw("$redis(r=>value)");
>>
>> I made a test call and disconnected, what I am observing is that even if
>> the call is finished the used memory is slightly increased.
>>
>> before:
>> < real_used: 844240
>> < real_used: 844088
>> < real_used: 844088
>> < real_used: 844088
>> < real_used: 844088
>> < real_used: 844088
>> < real_used: 844088
>> < real_used: 844088
>> after:
>> > real_used: 846080
>> > real_used: 845864
>> > real_used: 846584
>> > real_used: 846304
>> > real_used: 846200
>> > real_used: 846120
>> > real_used: 845248
>> > real_used: 846168
>>
>>
>> My assumption is after finishing the call, memory used by variables and
>> redis ops is released and should be back to original values but its not
>> happening. In production this issue is causing some problems as I have to
>> restart the kamailio process after every 3-4 weeks so that it starts
>> breathing with more available memory.
>>
>> Initially I thought its Kamailio version issue as in my last posts in
>> this forum people advised me to upgrade to the latest kamailio version
>> (original one was 5.4.0) but the same issue is happening on 5.7.0.
>>
>> Can someone explain how to debug/troubleshoot this situation . I have a
>> parallel lab environment, I do not see anything interesting in syslogs.
>>
>>
>> --
>> Muhammad Danish Moosa
>>
>> " The core of mans' spirit comes from new experiences. "___ Christopher
>> McCandless
>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
>> To unsubscribe send an email to sr-users-le...@lists.kamailio.org
>> Important: keep the mailing list in the recipients, do not reply only to
>> the sender!
>> Edit mailing list options or unsubscribe:
>>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
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> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> Edit mailing list options or unsubscribe:
>
__
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[SR-Users] Re: Unable to remove/modify/replace "Contact SIP header'
Check the logs and ensure you are not doing anything else with contact (
like nat handling or so )
On Sun, Apr 7, 2024, 17:02 ramaseshi.kolli--- via sr-users <
sr-users@lists.kamailio.org> wrote:
> yes i have tried it but no luck what am i doing wrong here.
>
> KSR.hdr.append("test_hdr: "test"\r\n"); --> this works i can see new
> header added
> KSR.hdr.remove("Contact"); -- this is not being removed
> __
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[SR-Users] Re: Unable to remove/modify/replace "Contact SIP header'
It is a core function in KEMI.
Not textops.
https://kamailio.org/docs/tutorials/devel/kamailio-kemi-framework/core/#ksrhdrremove
On Sun, Apr 7, 2024, 16:23 ramaseshi.kolli--- via sr-users <
sr-users@lists.kamailio.org> wrote:
> Hi All,
> My requirement is when incoming call comes from internal user in i want to
> put kamailio ip and port in Contact header and forward or dispatch request
> to sip trunk carrier but i am unable to do so using KEMI frame work LUA.
> Can someone please give some pointer please?
>
> KSR.textops.remove_hf("Contact");
> KSR.textopsx.assign_hf_value("Contact","");
> KSR.rtpengine.rtpengine_manage0();
> if KSR.tm.t_relay()<0 then
> KSR.sl.sl_reply_error();
> end
> __
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[SR-Users] Re: webRTC Test Client?
https://tryit.jssip.net/ here is a demo just set up connect with your server вт, 2 янв. 2024 г. в 17:12, Andy Newlands via sr-users < sr-users@lists.kamailio.org>: > Hi, > > I have a Freeswitch installation, fronted by a Kamailio proxy > > I would like to configure Kamailio to bridge SIP between webRTC clients > and Freeswitch. > > The first issue I hit is how to set up a test webRTC client to make calls > into Kamailio/FS. > > Can anyone point me to a simple webRTC client I can use to make > test webRTC/SIP calls into my setup? Thanks > > BTW: I'm not JS experienced (I work with Python, C#, C/C++). > > Kind regards, > > Andy > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Setting up Kamailio as SIP Proxy
Hi. Use UAC module Examples of the configuration are in the docs https://www.kamailio.org/docs/modules/5.7.x/modules/uac.html#idm700 On Wed, Nov 29, 2023, 01:26 Saul Ibarra via sr-users < sr-users@lists.kamailio.org> wrote: > Hi, > I am needing some help setting up kamailio as a sip proxy between a > provider(trunk) and a test pbx > The trunk requires username and i would like to add that > information to the request within the Kamailio proxy, so that the PBX can > authenticate directly by IP (without username) through the > Kamailio proxy. > > As a programmer with a limited background in SIP (currently learning), I > am keen on understanding the intricacies of this setup at the SIP level. > Could you please direct me to resources or documentation that covers > scenarios similar to mine? > > Any help or guidance you could provide would be greatly appreciated. > > Thanks > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: tps_storage_fill_contact(): no r-uri user
Hi Denis. I think this is topos specific. When topos does topology striping it also affects contact field, so it adds some data/id username. So if you wish topos to be involved - the same form of the URI should be used. There is also API from topos module to handle contact on your own if needed. On Tue, Nov 28, 2023, 11:25 Denys Pozniak via sr-users < sr-users@lists.kamailio.org> wrote: > Hello! > > My topology is like this: > PBX -> Kamailio Topos -> User > > When calling the user, PBX does not add the user part in the Contact URI, > something like: > Contact: > > When the customer sends a reinvite, the topos turns off for this > transaction and the sip partially breaks down with error: > > topos [tps_storage.c:319]: tps_storage_fill_contact(): no r-uri user > topos [tps_storage.c:615]: tps_storage_record(): failed to store > > As far as I understand, the RFC allows the absence of the user part in the > URI and this occurs quite often. > I can provide the pcap file and debug log in a private message if this > helps identify the essence of the problem. > > # kamailio -v > version: kamailio 5.7.1 (x86_64/linux) 1cf389-dirty > > -- > > BR, > Denys Pozniak > > > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: SIP Trunk registration In Kamailio
Use UAC module https://www.kamailio.org/docs/modules/5.7.x/modules/uac.html On Fri, Nov 17, 2023, 06:20 tanmoykarmakar10--- via sr-users < sr-users@lists.kamailio.org> wrote: > I want to register SIP Trunk directly in Kamailio. How can I register the > SIP trunk in Kamailio. Please find the below sample SIP trunk details. > > Network IP 10.2.2.2 > Subnet Mask 255.255.255.252 > Customer IP 10.3.3.3 > Gateway IP 10.1.1.1 > SBCIP 10.4.4.4 > Username12345678 > Password1234 > > Please help me with configuration to register SIP trunk in kamailio. > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Issue with Dialog Handling in Kamailio Post 2-Minute Duration
Check your firewall. According the screenshot and your description - it seems it is not delivered to kamailio service. So very possible blocked by firewall. On Thu, 9 Nov 2023, 12:53 Aaron Soto, wrote: > I enabled CFGTRACE, but the logs don't show this BYE; the only BYE that > appears is after 5 minutes, which is sent by Asterisk to Kamailio. It's > confusing because it seems like Kamailio isn't processing the BYE when i > make a hangup with my phone. > El 9/11/23 a las 11:46, Yuriy G escribió: > > Screenshot doesn't help more than raw BYE request would. > But mostly you need to look into kamailio service logs. Better with > CFGTRACE debugger parameter enabled. It will show you on which step BYE > handling was dropped. Also it may show you some meaningful output regarding > the reason it wss dropped. > > > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Issue with Dialog Handling in Kamailio Post 2-Minute Duration
Screenshot doesn't help more than raw BYE request would. But mostly you need to look into kamailio service logs. Better with CFGTRACE debugger parameter enabled. It will show you on which step BYE handling was dropped. Also it may show you some meaningful output regarding the reason it wss dropped. On Thu, 9 Nov 2023, 11:11 Aaron Soto, wrote: > Thank you for your response, > > I have confirmed that the dialog module is indeed loaded. However, as I > mentioned before, after 2 minutes, the BYE messages are no longer > recognized by Kamailio, and I'm uncertain why this is occurring. Perhaps > sharing a screenshot from sngrep could offer more specifics. I suspect the > issue might lie in the configuration of the dialog module, although I am > not certain of this. > > El 9/11/23 a las 10:35, Yuriy G escribió: > > If you not involve dialog module, then kamailio won't care about dialog ( > as it is Proxy, not UA) > The Indialog requests processing is done over proxy routing mechanisms ( > rfc3261 paragraph 16 ) > > So, after mentioning above: > What you mean under "unrecognised" ? > > On Thu, 9 Nov 2023, 10:18 aaron.soto--- via sr-users, < > sr-users@lists.kamailio.org> wrote: > >> I'm encountering an issue with dialog handling in Kamailio. Initially, >> everything functions correctly, and if a connection terminates within two >> minutes, the BYE method works as expected. However, if the call exceeds two >> minutes, the BYE method doesn't seem to be recognized, and there are no >> related entries in the logs. >> >> As someone new to working with Kamailio, I'm finding this quite >> perplexing. Could anyone offer guidance or troubleshooting steps to resolve >> this issue? >> >> Thank you for your assistance. >> __ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-le...@lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Issue with Dialog Handling in Kamailio Post 2-Minute Duration
If you not involve dialog module, then kamailio won't care about dialog ( as it is Proxy, not UA) The Indialog requests processing is done over proxy routing mechanisms ( rfc3261 paragraph 16 ) So, after mentioning above: What you mean under "unrecognised" ? On Thu, 9 Nov 2023, 10:18 aaron.soto--- via sr-users, < sr-users@lists.kamailio.org> wrote: > I'm encountering an issue with dialog handling in Kamailio. Initially, > everything functions correctly, and if a connection terminates within two > minutes, the BYE method works as expected. However, if the call exceeds two > minutes, the BYE method doesn't seem to be recognized, and there are no > related entries in the logs. > > As someone new to working with Kamailio, I'm finding this quite > perplexing. Could anyone offer guidance or troubleshooting steps to resolve > this issue? > > Thank you for your assistance. > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Redis issue when getting results in Kamailio
Did you try to use not JSON but usual GET/SET? Do you have the same issue?
AFAIK - ndb_redis is a wrapper oven hiredis wrapper and doesn't implement
serialisation protocol for redis, so, whatever you send delivered to redis
being only wrapped via hirer's client library.
Btw if uou use lua < 5.2 you may try redis-lua library. It is easier to use
with lua.
On Thu, 9 Nov 2023, 09:34 Ali Taher via sr-users, <
sr-users@lists.kamailio.org> wrote:
> I’m still stuck… I appreciate any assistance here.
>
>
>
> *From:* Ali Taher via sr-users
> *Sent:* Tuesday, November 7, 2023 12:47 PM
> *To:* Kamailio (SER) - Users Mailing List
> *Cc:* Ali Taher
> *Subject:* [SR-Users] Redis issue when getting results in Kamailio
>
>
>
> Hello all,
>
>
>
> I have redis database where the data is inserted in JSON format:
>
>
>
> #Redis Command
>
>
>
> JSON.GET Index1J:12345678
>
>
>
> #Redis Result
>
> [
>
> {
>
> "SupplierID": "7",
>
> "BCode": "987654",
>
> "Rate": 0.159
>
> },
>
> {
>
> "SupplierID": "7",
>
> "BCode": "765432",
>
> "Rate": 0.159
>
> },
>
> {
>
> "SupplierID": "7",
>
> "BCode": "5432159",
>
> "Rate": 0.159
>
> }
>
> ]
>
>
>
> When I try to get the result using Lua script in Kamailio like the
> following:
>
> #kamailio command
>
> if KSR.ndb_redis.redis_cmd("srvN", "JSON.GET pricelistJ:441246890",
> "ra") then
>
> KSR.xlog.xinfo("array size: $redis(ra=>size)\n");
>
> KSR.xlog.xinfo("Redis Response: " .. tostring(ra) .. "\n")
>
> KSR.xlog.xinfo("first values: $redis(ra=>value[0]) ,
> $redis(ra=>value[1])\n");
>
> end
>
>
>
> Nov 7 07:58:15 868d08ee318d /usr/sbin/kamailio[72]: INFO: LUA {INVITE}:
>
[SR-Users] Re: Sequential fork with different Call-Ids
In general - it is not how forking meant to be according rfc3261, and it is noylt a job for the sip Proxy. But, if you would like to have X parallel calls you probably can do some work around with UAC module. However: meeging management is on you then On Tue, 7 Nov 2023, 19:04 Vanderlei Torres Batistela via sr-users, < sr-users@lists.kamailio.org> wrote: > Hello all, > > I have a scenario on wich requires sequential fork (up to 30 possible > destinations) with different > Call-ID for each leg. I am using topoh module enabled, but it always puts > the same Call-ID. > Someone have a solution for this scenario? > > Regards > > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: SIP Proxy : BYE next hop
loose_route returns result in the $rc variable
Return code should be handed like this:
https://www.kamailio.org/wikidocs/cookbooks/5.6.x/pseudovariables/#rc-returned-code
чт, 2 нояб. 2023 г. в 13:27, Ali Taher via sr-users <
sr-users@lists.kamailio.org>:
> Hi James,
>
> Here are my listen directives:
> listen=udp:192.168.1.2:5060
> listen=tcp: 192.168.1.2:5060
>
> knowing that:
> Kamailio IP: 192.168.1.2
> SBC IP: 192.168.1.1
> Caller IP: 192.168.1.5
>
> -Original Message-
> From: Ali Taher via sr-users
> Sent: Thursday, November 2, 2023 1:42 PM
> To: James Browne ; Kamailio (SER) - Users Mailing List <
> sr-users@lists.kamailio.org>
> Cc: Ali Taher
> Subject: [SR-Users] Re: SIP Proxy : BYE next hop
>
> Hi James,
>
> Yes I'm using loose_route() and it's returning 1, but BYE message is not
> routed anywhere.
>
> Frame 6213: 673 bytes on wire (5384 bits), 673 bytes captured (5384 bits)
> Ethernet II, Src: Cisco_f8:19:ff (00:22:bd:f8:19:ff), Dst: VMware_bf:27:0b
> (00:50:56:bf:27:0b) Internet Protocol Version 4, Src: 192.168.1.5, Dst:
> 192.168.1.2 User Datagram Protocol, Src Port: 16368, Dst Port: 5060 Session
> Initiation Protocol (BYE)
> Request-Line: BYE sip:192.168.1.1:5060;transport=udp SIP/2.0
> Message Header
> Via: SIP/2.0/UDP 192.168.1.5:16368
> ;branch=z9hG4bK-d87543-c147891e677a1f19-1--d87543-;rport
> Max-Forwards: 70
> Route:
> Route:
> Contact:
> To: "121";tag=gtyrzghr-CC-1047-OFC-39
> From: "69079514";tag=7d4a7c3c
> Call-ID:
> 085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.
> [Generated Call-ID:
> 085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.]
> CSeq: 2 BYE
> User-Agent: eyeBeam release 1003s stamp 31159
> Reason: SIP;description="User Hung Up"
> Content-Length: 0
>
> Here is routing block related to BYE:
> if (is_method("BYE")) {
> xlog("L_INFO","INSIDE BYE");
> $var(loose) = loose_route();
> xlog("L_INFO","LOOSE : $var(loose)");
> }
>
>
> Regards,
>
> -Original Message-
> From: James Browne
> Sent: Thursday, November 2, 2023 12:31 PM
> To: Kamailio (SER) - Users Mailing List
> Cc: Ali Taher
> Subject: Re: [SR-Users] SIP Proxy : BYE next hop
>
> Hello
> Did you call the loose_route() function for the BYE? It should just work;
> that's one of the basics of how kamailio works, so I'll be very surprised
> if there's somehow a bug in kamailio.
> If that does not work, perhaps send a list of the listen directives in the
> kamailio.cfg, along with the headers of the two SIP BYE messages (incoming
> and outgoing) and I'll see what I can figure out for you.
>
> James
>
> On Thu, 2 Nov 2023 at 09:35, Ali Taher via sr-users <
> sr-users@lists.kamailio.org> wrote:
> >
> > Hello all,
> >
> >
> >
> > I’m using Kamailio as SIP proxy between session manager and SBC.
> >
> > I came across an issue in BYE request where the SBC sent in the header
> > list of Routes where first route is Kamailio IP, and the second is SM
> > IP, in addition to other routes (last route is the caller party)
> >
> > What Kamailio did is to route the BYE to the address set in the
> Request-URI, which is the caller party.
> >
> > What should be done instead is that Kamailio should remove the Route
> header including its own IP, and route the BYE to the next hop which is in
> this case the SM IP.
> >
> >
> >
> > How to accomplish that?
> >
> >
> >
> > Following is the Route header in the BYE request coming from SBC:
> >
> > [truncated]Route:
> > , > port=udp;lr;av-asset-uid=xx>, > p;ibmsid=local._x_x;lr;ibmdrr>,sips:127.0.
> > 0.2:15061;ibmsid=local.xx_xx_;lr;ibmdrr,
> > sip:SM1@SM_IP;transport=tls;lr;av-asset-uid=xx,
> > sip:CALLER_PARTY_IP:5061;transport=tls;lr
> >
> >
> >
> > Regards,
> >
> > __
> > Kamailio - Users Mailing List - Non Commercial Discussions To
> > unsubscribe send an email to sr-users-le...@lists.kamailio.org
> > Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> > Edit mailing list options or unsubscribe:
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
> To unsubscribe send an email to sr-users-le...@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> Edit mailing list options or unsubscribe:
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
> To unsubscribe send an email to sr-users-le...@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> Edit mailing list options or unsubscribe:
>
__
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to
[SR-Users] Re: WEBRTC-to-SIP & Vice Versa
Most common rules are: Use Path https://datatracker.ietf.org/doc/html/rfc3327 As well as use add_contact_alias/handle_ruri_alias from the nathelper module or same pattern ( like add_path_receoved ) to not destroy original Contact information, and, at the same time to provide destination address where to send INVITE to... On Wed, 1 Nov 2023, 20:28 Vishal Pai via sr-users, < sr-users@lists.kamailio.org> wrote: > Hello All > > Good Day > > Here I am trying to achieve something like this > > WSS to UDP Ext registartion which works well using > https://github.com/havfo/WEBRTC-to-SIP/tree/master/etc > Note here i am forwarding the registartion to asterisk pbx server and > define WSS of kamailio > > Now for calls > > - Webrtc - SIP calls works > > - SIP - Webrtc calls not working > Here I am getting the request of the call on kamailio but kamailio send > 404 not found to Asterisk PBX > > Webrtc - Webrtc calls not working > > Can anyone guide me on what I am doing wrong? > > Thanks > > Vishal P > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Information about Kamailio operating as an SBC
As there is no such consistent thing as SBC you may start at first with the definition of the components you whish to have according "sbc" terminology. The good start is from here: https://datatracker.ietf.org/doc/html/rfc5853 Then, once you understand exactly what you would like to have on board the best good entry point will be https://www.kamailio.org/wiki/ Going through the config examples and kamailio docs you will be able to find the best way to achieve your needs. On Tue, 24 Oct 2023, 18:55 Alvaro Escobar via sr-users, < sr-users@lists.kamailio.org> wrote: > Hello, > > I have Kamailio version 5.6.4. > I have installed rtpengine version 11.5.0.0. > I have the RTP through the kamailio. > The machine has just one NIC and all is working fine. > > I would like to have two NICs: One NIC for the private network and other > NIC to the public network. > I am looking information about Kamailio operating as an SBC. > Where I can find more detailed technical information ? > > Regards > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: STIR/SHAKEN with Kamailio
I will add some clarifications and the purpose of the approach: I've been using a 3d-party service ( written in-house ) via restful API as there were no STIR/SHAKEN modules for the Kamailio at that moment. So in short: it was a matter of time to integrate functionality. ср, 18 окт. 2023 г. в 10:03, Olle E. Johansson via sr-users < sr-users@lists.kamailio.org>: > Hello Kamailians! > > As STIR/SHAKEN seems to cross the ocean and arrive on the European shores, > I’m curious on how you’ve implemented it with Kamailio. I asked on our > Matrix chat and got two responses that seems to not use the Kamailio > STIR/SHAKEN support but rather a 3rd party service that they’ve integrated > using restful APIs. > > Please help me and share how you’re implementing STIR/SHAKEN with Kamailio. > > I’m not shaken, nor stirred. yet. > > Regards, > /O > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: TOPOS + Forcing the send socket
In the header of the topic you talking about topos, but inside the messages
you talking about topoh.
They are 2 different modules. If you usr them together - they can
conflictin case how they affect message. Try use or just topoh, or just
topos.
On Wed, 18 Oct 2023, 00:45 Marrold via sr-users, <
sr-users@lists.kamailio.org> wrote:
> Hi all,
>
> I've dug into this a bit more. Firstly I enabled debug logs and spotted
> the following record-route header being loaded from redis:
>
> 21(28) DEBUG: PY3 {ACK}: topos_redis [topos_redis_storage.c:1079]:
> tps_redis_load_dialog(): r[5]: s[ ;sn=internal;r2=on;lr;ftag=202310172234552;did=c1f.18b1;nat=yes>,]
>
> 127.0.0.8 is the wrong IP which explains why the ACK was not being
> forwarded correctly. A quick look in the source shows it's related to topoh.
>
> I had modparam("topoh", "mask_callid", 1) in my config so I disabled it
> and sure enough the ACK worked as expected which gets me a bit closer to
> finding the issue.
>
> Looking at the docs for topos and topoh it looks like things have changed
> since I used it last and I should be using the following instead:
>
> modparam("topoh", "use_mode", 1)
> modparam("topos", "mask_callid", 1)
>
> But with those configured things go really wrong:
>
> 18(25) CRITICAL: PY3 {INVITE}: [core/mem/q_malloc.c:501]:
> qm_free(): BUG: bad pointer 0x7ffc73e3ca90 (out of memory block!) called
> from core: core/data_lump.c: free_lump(470) - ignoring
> 5(11) ERROR: [core/msg_translator.c:2241]:
> build_req_buf_from_sip_req(): could not allocate private memory from pkg
> pool
> 5(11) ERROR: topos [topos_mod.c:518]: tps_msg_received(): not enough pkg
> memory for new message
> 5(11) CRITICAL: [core/mem/q_malloc.c:501]: qm_free(): BUG: bad
> pointer 0x2 (out of memory block!) called from core: core/data_lump.c:
> free_lump(470) - ignoring
> 5(11) INFO: [core/parser/parse_fline.c:80]: parse_first_line():
> message too short: 0 []
> 5(11) ERROR: [core/parser/parse_fline.c:271]: parse_first_line():
> parse_first_line: bad message (offset: 0)
> 5(11) ERROR: [core/parser/msg_parser.c:748]: parse_msg(): ERROR:
> parse_msg: message=<>
> 5(11) ERROR: [core/receive.c:376]: receive_msg(): core parsing of
> SIP message failed (172.24.0.21:5070/1)
> 3(9) ERROR: [core/msg_translator.c:2241]:
> build_req_buf_from_sip_req(): could not allocate private memory from pkg
> pool
> 3(9) ERROR: topos [topos_mod.c:518]: tps_msg_received(): not enough pkg
> memory for new message
> 3(9) CRITICAL: [core/mem/q_malloc.c:501]: qm_free(): BUG: bad
> pointer 0x2 (out of memory block!) called from core: core/data_lump.c:
> free_lump(470) - ignoring
> 3(9) INFO: [core/parser/parse_fline.c:80]: parse_first_line():
> message too short: 0 []
>
> Does anyone have any ideas what's going on here?
>
> Thanks
> Matthew
>
>
>
> On Tue, Oct 17, 2023 at 6:30 PM Marrold wrote:
>
>> Hi all,
>>
>> I am using Kamailio (5.7.2, Debian 11, Python KEMI) in a
>> mutli-homed environment with the topos module to hide the topology.
>>
>> I have noticed that when I use `set_send_socket` or `$fsn` to force the
>> socket it's sent from, it breaks topos and the ACK is not proxied to the
>> other leg of the call. I am calling record_route() *after* forcing the
>> socket.
>>
>> If I disable TOPOS the ACK works as expected, and the signalling looks
>> correct.
>> Or, with TOPOS enabled, if I comment out set_send_socket TOPOS works as
>> expected.
>> If I disable enable_double_rr, TOPOS works as expected and the ACK is
>> forwarded, but without the double RR subsequent in-dialog requests use the
>> wrong socket.
>>
>> One other observation, when I change the send socket I also see these
>> warnings in the logs:
>>
>> 24(32) WARNING: PY3 {ACK}: dialog [dlg_handlers.c:1348]: dlg_onroute():
>> tight matching failed for ACK with
>> callid='!!:MByLMlFAM.NfWxFAM.cAMxyfWjyLz.yAO.y6MxF1MxVZWG4ZMy**'/55,
>> ftag='2023101714101800015'/19,
>> ttag='2f55349a-2c59-4e37-bf58-fd84fb69ece9'/36 and direction=0
>> 24(32) WARNING: PY3 {ACK}: dialog [dlg_handlers.c:1355]: dlg_onroute():
>> dialog identification elements are
>> callid='2023101714101800015@2900-0601-0284-80'/37, caller
>> tag='2023101714101800015'/19, callee
>> tag='2f55349a-2c59-4e37-bf58-fd84fb69ece9'/36
>>
>> Is anyone using TOPOS + forcing the socket, or could someone advise where
>> I am going wrong?
>>
>> Thanks in advance
>> Matthew
>>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
> To unsubscribe send an email to sr-users-le...@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> Edit mailing list options or unsubscribe:
>
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Important: keep the mailing list in the recipients, do not reply only to the
sender!
[SR-Users] Re: UDP Fragmentation webRTC/UDP Call
For the sdp cleanup purpose ( if possibilities of rrpengine are not enough) I use the next technic: - Set up write_avp param for the rtpengine module - call rtpengine manage: handled sdp will be then posted to write_avp pvar. - use lua/python/etc to call via lua_run(...) (this is for lua ) - handle value of write_avp there - apply the value to body of the message. On Mon, 16 Oct 2023, 18:11 Yuriy G, wrote: > They actually aren't for the udp leg > > On Mon, 16 Oct 2023, 18:09 Mirko Brankovic via sr-users, < > sr-users@lists.kamailio.org> wrote: > >> I'm maybe out of context here but i don't think you can remove ssrc >> headers from sdp since they are crucial in identification of the rtp streams >> >> On Mon, Oct 16, 2023, 14:41 Social Boh via sr-users < >> sr-users@lists.kamailio.org> wrote: >> >>> Hello, >>> >>> in the rtpengine LOG i can see: >>> >>> Oct 16 06:32:53 sip1 rtpengine[146001]: DEBUG: [28vs0pbtdtu7udefti8m]: >>> ... "*: [ "sdp-attr-remove-audio-ssrc::" ], "replace":* [ "origin", >>> "session-connection" ], "transport-protocol": "RTP/AVP", "rtcp-mux": [ >>> "demux" ], "SDES": [ "off" ], "call-id": "28vs0pbtdtu7udefti8m", >>> "via-branch": "z9hG4bK4978560", "received-from": [ "IP4", "186.98.231.105" >>> ], "from-tag": "h8l2da68sp", "command": "offer" } >>> >>> but the ssrc lines still presents in the INVITE sent to sip device. >>> >>> Regards >>> >>> --- >>> I'm SoCIaL, MayBe >>> >>> El 14/10/2023 a las 12:39 p. m., Social Boh via sr-users escribió: >>> >>> Hello, >>> >>> I'm trying to using sdp-attr without luck: >>> >>> sdp-attr-remove-audio-ssrc: >>> >>> in this sentence: >>> >>> $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux >>> DTLS=off SDES-off ICE=remove RTP/AVP sdp-attr-remove-audio-ssrc:"; >>> >>> this to remove two ssrc lines >>> >>> Regards >>> >>> --- >>> I'm SoCIaL, MayBe >>> >>> El 14/10/2023 a las 11:12 a. m., Richard Fuchs via sr-users escribió: >>> >>> On 14/10/2023 02.46, [EXT] Karsten Horsmann via sr-users wrote: >>> >>> Hi, >>> >>> did you pass over the ice stuff from webrtc to the udp side? You could >>> strip that of with rtpengine options. >>> >>> >>> With recent versions we even have explicit SDP manipulations options, so >>> you can use rtpengine to strip attributes that it itself doesn't understand >>> or process. >>> >>> Cheers >>> >>> __ >>> Kamailio - Users Mailing List - Non Commercial Discussions >>> To unsubscribe send an email to sr-users-le...@lists.kamailio.org >>> Important: keep the mailing list in the recipients, do not reply only to >>> the sender! >>> Edit mailing list options or unsubscribe: >>> >>> __ >>> Kamailio - Users Mailing List - Non Commercial Discussions >>> To unsubscribe send an email to sr-users-le...@lists.kamailio.org >>> Important: keep the mailing list in the recipients, do not reply only to >>> the sender! >>> Edit mailing list options or unsubscribe: >>> >>> __ >>> Kamailio - Users Mailing List - Non Commercial Discussions >>> To unsubscribe send an email to sr-users-le...@lists.kamailio.org >>> Important: keep the mailing list in the recipients, do not reply only to >>> the sender! >>> Edit mailing list options or unsubscribe: >>> >> __ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-le...@lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: UDP Fragmentation webRTC/UDP Call
They actually aren't for the udp leg On Mon, 16 Oct 2023, 18:09 Mirko Brankovic via sr-users, < sr-users@lists.kamailio.org> wrote: > I'm maybe out of context here but i don't think you can remove ssrc > headers from sdp since they are crucial in identification of the rtp streams > > On Mon, Oct 16, 2023, 14:41 Social Boh via sr-users < > sr-users@lists.kamailio.org> wrote: > >> Hello, >> >> in the rtpengine LOG i can see: >> >> Oct 16 06:32:53 sip1 rtpengine[146001]: DEBUG: [28vs0pbtdtu7udefti8m]: >> ... "*: [ "sdp-attr-remove-audio-ssrc::" ], "replace":* [ "origin", >> "session-connection" ], "transport-protocol": "RTP/AVP", "rtcp-mux": [ >> "demux" ], "SDES": [ "off" ], "call-id": "28vs0pbtdtu7udefti8m", >> "via-branch": "z9hG4bK4978560", "received-from": [ "IP4", "186.98.231.105" >> ], "from-tag": "h8l2da68sp", "command": "offer" } >> >> but the ssrc lines still presents in the INVITE sent to sip device. >> >> Regards >> >> --- >> I'm SoCIaL, MayBe >> >> El 14/10/2023 a las 12:39 p. m., Social Boh via sr-users escribió: >> >> Hello, >> >> I'm trying to using sdp-attr without luck: >> >> sdp-attr-remove-audio-ssrc: >> >> in this sentence: >> >> $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux >> DTLS=off SDES-off ICE=remove RTP/AVP sdp-attr-remove-audio-ssrc:"; >> >> this to remove two ssrc lines >> >> Regards >> >> --- >> I'm SoCIaL, MayBe >> >> El 14/10/2023 a las 11:12 a. m., Richard Fuchs via sr-users escribió: >> >> On 14/10/2023 02.46, [EXT] Karsten Horsmann via sr-users wrote: >> >> Hi, >> >> did you pass over the ice stuff from webrtc to the udp side? You could >> strip that of with rtpengine options. >> >> >> With recent versions we even have explicit SDP manipulations options, so >> you can use rtpengine to strip attributes that it itself doesn't understand >> or process. >> >> Cheers >> >> __ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-le...@lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> >> __ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-le...@lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> >> __ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-le...@lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Kamailio as SIP proxy
for other messages, you can find good examples in standard kamailio.conf
file ( it is quite a full example file that basically covers your question )
Also, you may need to read more about how uac module works and which
instruments it gives you to handle restoring
https://www.kamailio.org/docs/modules/5.7.x/modules/uac.html#uac.p.restore_mode
чт, 12 окт. 2023 г. в 11:23, Ali Taher via sr-users <
sr-users@lists.kamailio.org>:
> Hello,
>
>
>
> I’m using Kamailio as SIP proxy between Session Manager and SBC
>
>
>
> So here is the scenario :
>
> Session Manager sends a call (Invite) to Kamailio, kamailio needs to relay
> it to SBC after doing some modification.
>
> (-Session Manager-) --- > (Kamailio) --- >
> (-SBC-)
>
>
>
>
>
> My routing logic is similar to the below:
>
> *route{*
>
> * if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) {*
>
> *options_reply();*
>
> * }*
>
> *if (is_method("INVITE")) {*
>
> *sql_query("cb", "select number from pool1 order by random() limit 1",
> "ra");*
>
> *$var(rand)=$dbr(ra=>[0,0]);*
>
> *uac_replace_from($var(rand),"sip:$var(rand)@192.168.1.1
> ");*
>
> *xlog("L_INFO","Random: $var(rand)");*
>
> *$td = "192.168.1.10";*
>
> *$rd = "192.168.1.10";*
>
> *t_relay();*
>
> *}*
>
> *}*
>
> Where:
>
> 192.168.1.1 is Kamailio IP
>
> 192.168.1.10 is SBC IP
>
>
>
> The issue is that I’m not sure to handle the other SIP packets like
> 183,CANCEL, BYE etc that can be received from both SBC and session Manager?
>
> What to add to my configuration to handle it properly?
>
>
>
> Regards,
> __
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[SR-Users] Re: kamailio Behind nat. How Record-Route should be added ？
There are many possibilities.
One of - use listen names and advertised address with allow_double_rr
option.
Once you do that - you'll be able to add 2 rr headers, and kamailio will be
able to handle gateway functionality between interfaces via RR/Route headers
On Thu, 28 Sept 2023, 11:53 codenot--- via sr-users, <
sr-users@lists.kamailio.org> wrote:
> UA1(192.168.0.1) --> kamailio (internal:192.168.0.2 public:1.1.1.1) -->
> UA2(2.2.2.2)
> if in UA1 INVITE, I write KSR.rr.record_route().
> so UA2(2.2.2.2) received sip msg (Record-Route: ), so
> Kamailio can received BYE/ACK from UA2.
> but，Kamailio can't received BYE/ACK from UA1.
>
> if in UA1 INVITE, i write
> KSR.rr.record_route_advertised_address("192.168.0.2"), then Kamailio can
> received BYE/ACK from UA1, but not from UA2.
>
> what can i do, let Kamailio receive BYE/ACK from both sides ？
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[SR-Users] Re: failure_route, change_reply_status, and append_hf: Not Working as Expected
Failure route doesn't handle replies
It handles requests to be which must be processed one more time if >299
reply received
On Tue, 26 Sept 2023, 19:13 sadik.oualla.mohamed--- via sr-users, <
sr-users@lists.kamailio.org> wrote:
> Hello Kamailio Community,
>
> I'm currently working on a project that involves complex SIP routing and
> handling. I've encountered some difficulties while using
> change_reply_status, and append_hf functions in failure_route route in
> Kamailio. Although the documentation does provide guidance, the behaviors I
> am experiencing aren't aligning with the expected outcomes.
>
> Issue 1: failure_route
> I've set up a failure_route to handle specific SIP response codes (>4xx).
> However, I've observed that it get trigered and lunched, but I can't use
> change_reply_status or append_hf, I was using this first in onreply_route.
>
> Issue 2: change_reply_status
> I tried to use the change_reply_status function to modify the status of
> SIP replies, but it raises an error that this command cannot be executed
> here.
>
> Issue 3: append_hf
> I've also attempted to use append_hf in my failure_route block to add
> headers to replies, but the headers are not being added as expected.
>
> As a workaround, I've been using send_reply and append_to_reply for adding
> custom headers, and they seem to be working as expected. However, I found
> some old messages in kamailio mailing list, I can summerize what I learned
> like this ```
> #Functions and Context
> - append_hf(): Adds a header to the currently processed SIP message. If
> you are in a route {...} block, then this adds the header to the request.
> If you are in an onreply_route, it adds the header to the reply. This
> function always acts on the message currently under scrutiny.
>
> - append_to_reply(): This function is used to add a header to a reply that
> will be generated by Kamailio at a later time. This function only affects
> replies generated by Kamailio itself and is not applicable for messages
> that are simply being forwarded.
>
> #Contexts
> - route {...}: This is typically where the incoming SIP request gets
> processed. Here, append_hf() will add headers to that request, and
> append_to_reply() will add headers to any replies that Kamailio might
> generate for this request.
>
> - onreply_route: This is where incoming SIP replies are processed. Here,
> append_hf() will act on the reply, not the request.
> ``` I'd like to know if there is a way to handle sip replies in
> failure_route, because send_reply make a new reply from kamailio, instead I
> want to keep some important headers from the original reply.
>
> I'd appreciate any insights or guidance the community can offer.
>
>
>
> Best regards,
> __
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