[OpenSIPS-Users] pua_dialog info with expired PUBLISH

Hi, I've an OpenSIPS server that triggers PUBLISHes for call states to a remote presence server, it all works great for trying, ringing, answered states, however, when a call ends it sends out a PUBLISH with header Expires: 0 That causes the remote presence-server to immediately remove the presen

Re: [OpenSIPS-Users] OpenSips Configuration with RTP Proxy

Hi, Could you share further details as whats not working? check the following: 1 - OpenSIPS is able to connect tot he rtpproxy socket 2 - OpenSIPS is calling the right rtpproxy function from the script 3 - the RTPproxy function have correct parameters 4 - Check the SDP for incoming leg and the out

Re: [OpenSIPS-Users] Help with rtpproxy on a multihomed host.

rtpproxy_engage("ies"); route(TO_LAN); } route(RELAY); There are multiple variations depending on how you use rtpproxy and in which route you use them i,.e branch_route or main route etc. Regards, Sammy On Wed, Jan 8, 2020 at 2:16 PM Matthew Schumacher wrote: > SamyGo, > > T

Re: [OpenSIPS-Users] Dialog profiles replication issues

rascu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 1/7/20 10:40 PM, SamyGo wrote: > > Hi Vlad, > Update: it wasn't the issue with dialog module or bin_proto. When I used > avp_db_query() inside async() that led to script processing to halt, I've > removed th

Re: [OpenSIPS-Users] Help with rtpproxy on a multihomed host.

Hi, if *a.a.a.a* is PublicIP and *b.b.b.b* is Private IP ; where c.c.c.c is another Private IP address then you just need to enable multihome param " *mhomed=1" *in your opensips.cfg script and OpenSIPS should take care of relaying the packet our with proper SIP headers, the selection of the interf

Re: [OpenSIPS-Users] Dialog profiles replication issues

ion. Best Regards, Sammy On Tue, Jan 7, 2020 at 11:38 AM SamyGo wrote: > Thanks Vald, > The problem I'm facing with profile sharing is strange, when I make a call > the whole flow freeze in that route with dialog profile operations i.e > get/set. I can wait for as many as 15+

Re: [OpenSIPS-Users] Dialog profiles replication issues

sterdam, May 2020 > https://www.opensips.org/events/Summit-2020Amsterdam/ > OpenSIPS Bootcamp, Miami, March 2020 > https://opensips.org/training/OpenSIPS_Bootcamp_2020/ > > On 1/6/20 4:21 AM, SamyGo wrote: > > Hi Bogdan, > Thanks for taking time out, here are the

[OpenSIPS-Users] Dialog profiles replication issues

Hi All, Im trying to wrap my head around a problem with OpenSIPS 3.0 dialog profile replication. I've a cluster of OpenSIPS instances sharing dialog profiles with each other over clusterer/binary_proto layer. As soon as I initiate a call on first instance they all start exchanging the dialog prof

Re: [OpenSIPS-Users] Rest Client Async operation

Hi Liviu, Is it possible to suspend the transaction and resume it once we somehow get the event from libcurl as soon as the connect is done? Im imagining the way usually APNS is done these days. The only thing missing here is the event from the rest_client ! Can this mechanism help with the probl

Re: [OpenSIPS-Users] Is it possible to generate Notify from OpenSIPS on Freeswitch NOTIFY ESL event

Hi Sagar, So if I can glue info provided by Bogdan then yes you can pull data from incoming ESL event and create a new NOTIFY and send it to (where ) ! I would send this new NOTIFY to myself(127.0.0.1:5060) so opensips can find the location of the desired registered client and relay it swiftly. Ho

Re: [OpenSIPS-Users] RFC2833 to INFO

That sounds like some custome development for rtpengine/rtpproxy. During random duration of the call any DTMF can pop uo into rtpengine requiring it to send atleast an interrupt signal of some sort to OpenSIPS to generate an INFO and insert into dialog. On Mon, Mar 18, 2019, 11:45 AM Alain Bieuze

Re: [OpenSIPS-Users] Integrate OpenSIPS with VG202 Audio Gateway

Hi Vikash, I assume your CIsco gateway support SIP ? can you register your cisco g/w with opensips ? or is it possible for the gateway to receive/send calls from a statis IP (of opensips) ? You might need to provide some more insights to your cisco gateway on it's capabilities and supported protoc

Re: [OpenSIPS-Users] I deleted an entry from clusterer table , but still opensips try to ping that node .

Hi, Did you restart OpenSIPS process on both node1, and 2 simultaneously ? The way I look at this is one of the two nodes kept the 3rd one in the memory and restarting both nodes one at a time resulted in both sharing their node structure and hence node3 stayed visible. I think possible way to remo

Re: [OpenSIPS-Users] ACC module with JSON events

(1 << 26) While other event routes have unique value. Is this intentional? can I change this flag value to something unused i.e 1 << 31 ? On Wed, Nov 14, 2018 at 12:20 PM SamyGo wrote: > Howdy again, > > Thanks team for creating the event_jsonrpc module in 2.4 - I

Re: [OpenSIPS-Users] ACC module with JSON events

odule loaded as well, (which I can't remove). loadmodule "event_routing.so" loadmodule "event_route.so" Any insight as to why these error show up? Big thanks, Sammy. On Mon, Feb 26, 2018 at 12:07 PM SamyGo wrote: > Hi DanB, > I can certainly try CGRates, but

Re: [OpenSIPS-Users] redis cache sharing for dialog storage?

I have a strong feeling that you're using an old version of opensips to expect it to share dialog states/profiles. I think you'll need to use newer opensips 2.4+ having dialog sharing capability using proto_bin and clusterer module: http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#param_

Re: [OpenSIPS-Users] async(wait_for_event()) statement in a branch_route[]

Hi Bogdan, Intrigued by this thread, I recall in older threads I was told that I can do something like this, I'm going to use my example. event_route[ABCD]{ route(DO_SOMETHING); } and then in that DO_SOMETHING route I can use the redis/mysql functions or anything that is not acceptable t

Re: [OpenSIPS-Users] Raise event from MI/FIFO interface(externally)

identify the RURI in script and do a raise_event() in script. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > >

[OpenSIPS-Users] Raise event from MI/FIFO interface(externally)

Hi, I'm trying to find out document pages regarding raising events from outside the OpenSIPS via fifo/mi_* modules. All I have read so far is opensips can send events and their data OUT to external "subscribing" applications. There is even a fifo command to subscribe for an event from FIFO layer.

[OpenSIPS-Users] Callcenter module with dynamic/realtime agents+flows

Hi, Is it possible to have agents and flows created in realtime w/o them being pre-defined into the DB ? I'm thinking that I login agents to a flow when they Register and handle calls in a flow w/o DB engaged. ! something like: loadmodule "call_center.so" modparam("call_center", "db_mode", 0)

Re: [OpenSIPS-Users] 1XX, 2XX not relayed to caller

nsips.org/events/Summit-2018Amsterdam > > On 04/12/2018 08:21 PM, SamyGo wrote: > > Hi, > Seems like I'm stuck with a very basic situation. I've 2 OpenSIPS boxes > and 2 Users registered on each Proxy. Caller is on UDP, Destination is on > TCP. Call made from A to B

[OpenSIPS-Users] 1XX, 2XX not relayed to caller

Hi, Seems like I'm stuck with a very basic situation. I've 2 OpenSIPS boxes and 2 Users registered on each Proxy. Caller is on UDP, Destination is on TCP. Call made from A to B will not have its 1XX and 2XX relayed back to the originating Proxy: see this sngrep flow. So, naturally OpenSIPS-B tri

Re: [OpenSIPS-Users] MWI light on cisco phones

Hi Brian, What do you mean by "sip file" you mean the sipsak is sending that sample packet, are you sure that the phone is receiving and NOTIFY with the Message-Waiting=yes and Voicemail=1/0/43 data in them. It seems form your sample packet that phone wont blink a light unless there is an actual m

Re: [OpenSIPS-Users] Choosing the latest version

Hi Liviu, So, this is great, you're saying that individual opensips servers(different Public IPs) can still synchronize the user-locations (or dialog ) states using internal replication mode right !? Regards, Sammy On Wed, Mar 7, 2018 at 10:08 AM, Liviu Chircu wrote: > Hi John, > > Yes, we have

Re: [OpenSIPS-Users] ACC module with JSON events

Hi DanB, I can certainly try CGRates, but...surely enough its an overkill for simple CDRs. Anyways, my request is for an event_JSONRPC which maybe very helpful in events outside of ACC. Best Regards, Sammy On Mon, Feb 26, 2018 at 6:13 AM, Khalil Khamlichi < khamlichi.kha...@gmail.com> wrote:

Re: [OpenSIPS-Users] ACC module with JSON events

o_user)= $avp(dnis); > ... > $avp(json_acc)= $json(acc); #deserialize for raising the > event > } > > Obviously there are other parameters involved for the acc module but this > should provide the gist for using aqmp or any other method you want once > you de

Re: [OpenSIPS-Users] ACC module with JSON events

> On Thu, Feb 22, 2018 at 10:14 PM SamyGo wrote: > >> Hi, >> >> I'm using ACC module to trigger accounting events which land up in >> event_xmlrpc and ultimately shipped to remote CDR collection server. The >> sent data is obviously an XML - now, can I have

[OpenSIPS-Users] ACC module with JSON events

Hi, I'm using ACC module to trigger accounting events which land up in event_xmlrpc and ultimately shipped to remote CDR collection server. The sent data is obviously an XML - now, can I have an event_json like module so data is sent as JSON or is there any other way I can achieve the same ? Rega

Re: [OpenSIPS-Users] I have some doubts on contact fix in opensips-1.11 .

:D @ Bogdan. Good one. On Wed, Jan 31, 2018 at 10:51 AM, Bogdan-Andrei Iancu wrote: > Hi Sasmita, > > Of course it fixes only the first contact - afterall the function is > called fix_contact() and not fix_contactS() > > But please ignore me, this was a silly joke :D.. > > Yes, it seems that fix

Re: [OpenSIPS-Users] How to route calls between SIP domains

Hi, I'm pretty sure you'll need to write domain-translation rules (custom logic) in order to allow inter-domain calls. There can be many ways to do that but lets stick to old-school-approach...prefixes. Use prefixes to differentiate if user want to do inter-domain calling. Prefix: Domain 1 : aaa.c

Re: [OpenSIPS-Users] OpenSIPS crash with core dump

ncu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Summit May 2017 Amsterdam > http://www.opensips.org/events/Summit-2017Amsterdam.html > > On 05/02/2017 11:57 PM, SamyGo wrote: > > Hi Razvan, > Can you guide me on how to do that

Re: [OpenSIPS-Users] InterProxy messaging

] https://github.com/OpenSIPS/opensips/issues/ > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 05/03/2017 06:51 PM, SamyGo wrote: > > Hi All, > > I'm toying with the idea of having a cluster of opensips sending events &

[OpenSIPS-Users] InterProxy messaging

Hi All, I'm toying with the idea of having a cluster of opensips sending events between each other and also capable of "querying" other servers in the cluster to retrieve the required data. This idea is not any different than having a shared Redis sentinel between the cluster to set/get data but

Re: [OpenSIPS-Users] OpenSIPS crash with core dump

gt; > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 05/02/2017 02:22 AM, SamyGo wrote: > >> Hi All, >> I've been seeing some crashes in my opensips servers: >> >> >> *_version:_*

[OpenSIPS-Users] OpenSIPS crash with core dump

Hi All, I've been seeing some crashes in my opensips servers: *version:* opensips 2.2.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_S

Re: [OpenSIPS-Users] AVP + rest_post with utf-8 values

erything is okay on the OpenSIPS side. > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 04.04.2017 20:42, SamyGo wrote: > > Hi Liviu, > > I confirm that the same happening on my side as well. but take a look at > the last two lines of your

Re: [OpenSIPS-Users] AVP + rest_post with utf-8 values

stated. ! A bit confused, do you mean that conversion of Chinese characters to "..." is the right behavior ? Looking for some details. Thanks, Sammy On Tue, Apr 4, 2017 at 9:49 AM, Liviu Chircu wrote: > Hi SamyGo, > > The rest_post() function seems to properly put _any_ given

[OpenSIPS-Users] AVP + rest_post with utf-8 values

Hi, I've a specific problem with avps containing values in language other than English. For example an avp(test) holding a Chinese character gets converted to ? while passing to some rest_post URL. *code:* $avp(test) = $fn; xlog("L_INFO","Got Display Name: $avp(test) \n"); rest_post("$avp(url)",

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

_UL_CONTACT_INSERT]{ # everything route[DO_SOMETHING] could do plus events-overhead, still some limitations } With event route in async mode event_route[E_UL_CONTACT_INSERT,async]{ # route[DO_SOMETHING] with xlog capability only. :D } Thanks, Sammy. On Wed, Jun 22, 2016 at

Re: [OpenSIPS-Users] Replace the old registration record with new registration

Hi Aqs, I think there is a mod param for usrloc or registrar which states the max number of Contacts per AoR. You can set it to 1 and hence you will only be keeping the latest Registering device. I hope thats what you needed. Thanks. Sammy On Jan 28, 2017 14:09, "Aqs Younas" wrote: Greetings

Re: [OpenSIPS-Users] [NEW] Mid-registrar: scalable registration and call forking

Hi Liviu, This looks very neat. This was somewhat possible previously as well using customized opensips.cfg, but now its more of a built-in function and I like it. However, couple of questions came to my mind after going through this. - Is this mid-registrar compatible with proto_bin ? Like what

Re: [OpenSIPS-Users] Topology_Hiding adding extra VIA header

", "myvoip_box1") modparam("topology_hiding", "th_passed_contact_uri_params", "account_id") modparam("topology_hiding", "th_passed_contact_params", "+mediabx1.wholevoip.se;device;caller") Looking for some answers thanks, Reg

Re: [OpenSIPS-Users] Topology_Hiding adding extra VIA header

this resolved. Thanks, Sammy. On Mon, Nov 14, 2016 at 1:30 PM, SamyGo wrote: > Hi Razvan, > > > Here is the requested data. > > > > *INITIAL INVITE: *Via: SIP/2.0/TLS 123.123.212.123:5061;branch= > z9hG4bK442.8373b213.0;i=35f5 > > > *200 OK from the B party as

Re: [OpenSIPS-Users] Topology_Hiding adding extra VIA header

lso, can > you also post the Via headers of the initial INVITE and for the 200 OK > received by OpenSIPS? > > Best regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 11/12/2016 12:33 AM, SamyGo wrote: > > Hi, > > I'm usin

[OpenSIPS-Users] Topology_Hiding adding extra VIA header

Hi, I'm using OpenSIPS 2.2.1 version and I'm facing a weird situation where OpenSIPS is adding a duplicated VIA header to the 200 OK, This only happens when I've topology_hiding() engaged into the call. The scenario is very simple; two users making call to each other on the same OpenSIPS but with

Re: [OpenSIPS-Users] BYE with different transport

routed. Do you have record route header? Get Outlook for iOS <https://aka.ms/o0ukef> On Fri, Oct 7, 2016 at 12:47 AM +0200, "SamyGo" wrote: Hi All, > > I have a opensips 2.2 with residential script loaded. A TCP client makes a > call and that call gets forwarded to Fre

[OpenSIPS-Users] BYE with different transport

Hi All, I have a opensips 2.2 with residential script loaded. A TCP client makes a call and that call gets forwarded to FreeSWITCH over UDP. The call establishes just fine and everything works smooth untill the B party sends the BYE. That BYE comes over UDP and hence opensips tries to send the BYE

Re: [OpenSIPS-Users] Monitoring end to end solution.

Hi Ziv, You should create scripts based on SIPP to execute these scenarios, and based on results evaluate status of the servers. Regards, Sammy On Aug 8, 2016 05:59, "Ziv Gabel" wrote: > Hi, > > I’m running several opensips servers and I want to monitor the health of > the servers. > > I can ch

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

Hi Nabeel, Point 1 I cant imagine how those lines possibly relate to no media error in asterisk, I guess it depends on your config setup. The logical answer to your point 2 would be Asterisk realtime. However this is not going to be as staraight forward as making asterisk use subscriber table for

Re: [OpenSIPS-Users] OpenSIPS 2.2 SIP ping not triggering

s -V) and where did you get it ? > There is a very similar bug which was already fixed. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 28.06.2016 22:59, SamyGo wrote: > > Hi All, > > I've OpenSIPS 2.2

[OpenSIPS-Users] OpenSIPS 2.2 SIP ping not triggering

Hi All, I've OpenSIPS 2.2 and doing some tests with it I can't seem to find any OPTIONS triggering for the registered users. I've following settings for usrloc module and nathelper loadmodule "usrloc.so" modparam("usrloc", "nat_bflag", "NAT") modparam("usrloc", "db_mode", 2) modparam("usrloc",

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

st regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/22/2016 07:39 PM, SamyGo wrote: > > Yes correct. Async event route even stops to be executed. > On Jun 22, 2016 12:37, "Răzvan Crainea" wrote: > >> So the patch

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

Yes correct. Async event route even stops to be executed. On Jun 22, 2016 12:37, "Răzvan Crainea" wrote: > So the patch doesn't do anything but stops triggering the event? > > Regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > &g

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

2, 2016 at 3:10 AM, Răzvan Crainea wrote: > Hi, Sammy! > > Does this happen only at startime, or happens during runtime too? > > Regards, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06/21/2016 10:57 PM, SamyGo wrote: > > Hi , > >

Re: [OpenSIPS-Users] Async event_route and cachdb_redis

On Tue, Jun 21, 2016 at 3:40 AM, Răzvan Crainea wrote: > Hi, Sammy! > > Could you try this patch: > > https://gist.github.com/razvancrainea/9d239c82474bb0f1c403b6459dbdb647 > > Thanks, > > Răzvan Crainea > OpenSIPS Solutionswww.opensips-solutions.com > > On 06

[OpenSIPS-Users] Async event_route and cachdb_redis

Hi, I'm seeing errors from cachedb_redis module when called in an event route in async mode. event_route[E_UL_CONTACT_INSERT,async] { ... cache_raw_query("redis:group1","SET ABC"); .. } OpenSIPS throws error stating that redis group1 unavailable DBG:core:cachedb_raw_query: from script [redis] -

Re: [OpenSIPS-Users] Break in while loop

Hi Husnain, Simple *return* should work fine in your scenario but if everything fails you can do some additional condition check in while() loop to break it ! for example: $avp(flag) = 0; while( ($var(i) < 10) && $avp(flag) != 1) { if(condition){ $avp(flag) = 1; } } I chec

Re: [OpenSIPS-Users] Userloc, Bin Interface, and Clusterer combo

; > Once replicated, the registration records are independent and they do not > depend on the what is happening with the original node (where the > registration was pushed from). > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com &

[OpenSIPS-Users] Userloc, Bin Interface, and Clusterer combo

Hi, I've a cluster setup with userloc replication enabled for 3+ servers. As soon as a user register anywhere the other servers which receive this replicated contact display these errors: ERROR:usrloc:receive_ucontact_insert: non-local socket ERROR:usrloc:receive_ucontact_insert: failed to proces

Re: [OpenSIPS-Users] OpenSIPS 2.2 Documentation needs update !

> > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 13.06.2016 23:11, SamyGo wrote: > > Hi, > > It seems like mentioning of clusterer module as a dependency is missing > form module docs: > > http://www.opensips.org/html/

[OpenSIPS-Users] OpenSIPS 2.2 Documentation needs update !

Hi, It seems like mentioning of clusterer module as a dependency is missing form module docs: http://www.opensips.org/html/docs/modules/2.2.x/usrloc.html#id293640 Although it is obvious that the write refers to "valid cluster id" in function accept_replicated_contact()

Re: [OpenSIPS-Users] Rely forbidden

Did you modify something. Now the error from your opensips seem like 500 Service Full. That usually is configured with load balancer. Means that it was unable to find any active server to route the call to. On Apr 15, 2016 06:04, "Francjos" <35...@heb.be> wrote: > Here is the output of :ngrep -d e

Re: [OpenSIPS-Users] Rely forbidden

Hi, So all you've gotta do with sngrep is select an INVITE and press Enter...it will open up the whole signalling flow, you can use arrow keys up and down to inspect individual packet as well. I was expecting that it'll be easier for you to just start using it right away. *Idx:*[ ] 1 *SIP From

Re: [OpenSIPS-Users] Rely forbidden

Hi, So it seems that the RURI contains a $rd (request-domain) which is not recognizable at your opensips and hence it says I won't relay the call from one external domain to another external domain. In simple words, how come that there is req-domain "localhost" is the RURI for a call coming from P

Re: [OpenSIPS-Users] Openisp and freeswitch

Hi, Once you are able to differentiate calls based on IP you will be able to send them wherever you want them to go. So Like Johan hinted about using load balancer you can use another function of that module to determine that if the source IP is your FreeSWITCH or not. Could you please share your

[OpenSIPS-Users] Memcache Operations other than fetch/store

Hi All, I'm interesting in knowing if I can tell memcache to do INCR/DECR on some key. I understand that this is doable via cache_raw_query() but thats only for Redis. For now I can only do fetch/store/remove operations on memcache backend. Regards, Sammy

Re: [OpenSIPS-Users] Opensips IVR

Hi, OpenSIPS is a SIP proxy and it ma ynot be the best tool for making it do IVR, just wondering how would opensips capture a DTMF? and play a new file !!. It is recommended that you use any Media-Server to handle IVR with easy, like SEMS, FreeSwitch, Asterisk. Regards, Sammy On Fri, Mar 25, 20

Re: [OpenSIPS-Users] Two opensips servers and one sip trunk

Hi Francos, If both of the servers don't have to be active all the time then maybe a LinuxHA or VIP based approach, but I feel that its not the case. I'd suggest to get help from the two mi functions of uac_registrant module, one to monitor and other to reload. I assume both servers have their o

Re: [OpenSIPS-Users] Register to Cisco call manager on behalf of clients

Hi, I am wondering that even if you accomplish it one way or another wouldn't you be limited by the number of ports a system can handle per OpenSIPS box ? Regards, Sammy On Mar 13, 2016 7:25 PM, "Adrian Georgescu" wrote: > If you get rid of CUCM, your life would be much much easier. > > Adrian >

Re: [OpenSIPS-Users] mhome asymmetric port issue

Hi Satish, I can't understand this: "*...use t_relay() in dispatcher then its consuming REGISTER packet and sending AUTH challenge to client instead **of sending that REGISTER to backend dispatcher..*" t_relay has nothing to do with "consuming" REGISTER packets. I've done it so many times, the A

Re: [OpenSIPS-Users] dispatcher with t_relay performance

the Sender will get a 302, with a new Contact header and then it is supposed to talk to that new server. Again its the business logic and depends from usage to usage. This is not the only way to increase capacity or efficiency. Sometimes UACs/UASs don't like Redirecting and this method flop

Re: [OpenSIPS-Users] dispatcher with t_relay performance

our setup separately to know what are your capabilities. Regards, Sammy On Mon, Mar 7, 2016 at 8:54 AM, Satish Patel wrote: > We have 200,000 CPS and more in future. Just worried about t_relay() and > its performance. Any idea? > > -- > Sent from my iPhone > > On Mar 6,

Re: [OpenSIPS-Users] dispatcher with t_relay performance

I'd ask you to read difference between Load_balancer and Dispatcher module. Dispatcher module is not an accurate measure but it is the only option when it comes to load balancing REGISTER requests. Dispatcher is hence very light weight as compared to Load Balancer. For a 200 CPS calls Load Balance

Re: [OpenSIPS-Users] Listening ips and sending call from them

Hey Miha, See if this thread helps you: http://lists.opensips.org/pipermail/users/2010-October/015150.html Regards, Sammy On Tue, Mar 1, 2016 at 9:35 AM, Miha wrote: > HI. > > If you have two ips on your server, let say X in Y. > When you route calls to providers is it possible to use ip > X f

Re: [OpenSIPS-Users] ACK did not Receive on Proxy and get in loop

I beleive you have to engage NAT corrections in reply route for 200OK. I havent seen your SIP trace but if im Comtaact header the IP is of UAS and it goes to the caller side ACK will get directly to that IP. Just my guess. Need more info from you. Regards, Sammy On Sep 2, 2015 10:24 AM, "Hamid Ha

Re: [OpenSIPS-Users] extension status monitoring

nd Developerhttp://www.opensips-solutions.com > > On 17.08.2015 18:10, SamyGo wrote: > > Hi Bogdan, > > Can you recommend this approach for a huge number of ul records ? Is there > any better approach in your opinion ? > > I Push Registering Users into Redis from cfg file, and on

Re: [OpenSIPS-Users] extension status monitoring

org/html/docs/modules/2.1.x/pua_usrloc.html > > -Eric > > > On 08/17/2015 09:10 AM, SamyGo wrote: > > Hi Bogdan, > > Can you recommend this approach for a huge number of ul records ? Is there > any better approach in your opinion ? > > I Push Registering Users into Redis fro

Re: [OpenSIPS-Users] extension status monitoring

Hi Bogdan, Can you recommend this approach for a huge number of ul records ? Is there any better approach in your opinion ? I Push Registering Users into Redis from cfg file, and on un-registering delete them from redis List of online users. This wont give me accurate results but I believe much l

Re: [OpenSIPS-Users] 408 Request Timeout with UDP

force the NAT pinhole to stay open. Nevertheless, using the SIP pinging > with UDP should also have fixed the problem. > > I will update the default cfg to use SIP pinging rather than simple UDP > pinging. > > Thanks and Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder

Re: [OpenSIPS-Users] 408 Request Timeout with UDP

12 and both phones are >> registered to OpenSIPS. In this case, the callee is connected to Wi-Fi >> (hence 192.xx IP address) and the caller is connected to a mobile network. >> >> The opensips.cfg I am using was generated from 'make menuconfig', except >>

Re: [OpenSIPS-Users] 408 Request Timeout with UDP

attached my config file > at this link: > > http://pastebin.com/0QRyC938 > > > > On 6 August 2015 at 05:00, SamyGo wrote: > >> Hi Nabeel, >> Quick question; what is this destination ip? 192.168.0.19:60912 ? - >> Destination >> User Agent Registered o

Re: [OpenSIPS-Users] 408 Request Timeout with UDP

Hi Nabeel, Quick question; what is this destination ip? 192.168.0.19:60912 ? - Destination User Agent Registered on OpenSIPS? Can you share the opensips.cfg code snippet for this call ? On Wed, Aug 5, 2015 at 11:55 PM, Nabeel wrote: > Hi, > > I am using the residential script generated by 'make

Re: [OpenSIPS-Users] What sould I do/run to avoid rtpproxy ERROR while starting OpenSIPS?

Hi, If you need media passing through your OpenSIPS box then you need RTPproxy, if not then comment this rtpproxy line as well as any other rtpproxy module functions. OpenSIPS don't run this rtpproxy automatically and 12221 is not a standard port for this. Regards, Sammy On Mon, Jul 20, 2015 at

Re: [OpenSIPS-Users] Dialplan module and priority

Hi, Swapping the priority works the way you want ? I have a feeling this makes sense (just like an ACL or firewall rules) ^888444* should get called before ^888* . On Thu, Jul 16, 2015 at 5:47 PM, Ali Pey wrote: > Hello, > > Let's say I have the two following rules in my dialplan table: > > Ru

Re: [OpenSIPS-Users] Simple questions before adopting OpenSIPS.

See reply inline. On Wed, Jul 15, 2015 at 10:43 AM, Rodrigo Pimenta Carvalho < pime...@inatel.br> wrote: > > > > > Dear OpenSIPS-users, > > This is my first contact in this mailing list. > > > Yesterday I was watching the video OpenSIPS Kick Start, from Youtube, and > I have collected all needed

Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

Thats sound like you need to ask for script doing the flood blocking and security rather than IVR call control etc. On Sat, Mar 14, 2015 at 1:11 PM, mahan77 wrote: > Hello again Danilo, > > Thank you for the quick replay. > > I have asterisk server running at public IP. > > I have to use IVR,

Re: [OpenSIPS-Users] Installing OpenSIPS 2.1.0 gives error

Thanks again worked this time. (Y) On Fri, Mar 6, 2015 at 7:21 PM, SamyGo wrote: > Thanks for the fix, will test it now. > > On Fri, Mar 6, 2015 at 5:39 PM, Liviu Chircu wrote: > >> Hello Samy, >> >> I moved some allocator-specific global variables in their own f

Re: [OpenSIPS-Users] Installing OpenSIPS 2.1.0 gives error

and thanks for reporting! > > Cheers, > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 06.03.2015 23:04, SamyGo wrote: > > Hi All, > > Im trying to install new OpenSIPS 2.1.0 from git. Im getting compilation > errors; something

[OpenSIPS-Users] Installing OpenSIPS 2.1.0 gives error

Hi All, Im trying to install new OpenSIPS 2.1.0 from git. Im getting compilation errors; something like this. Compiling net/proto_tcp/proto_tcp.c Compiling net/proto_udp/proto_udp.c Compiling lex.yy.c lex.yy.c:4142:12: warning: redundant redeclaration of ‘isatty’ [-Wredundant-decls] /usr/include/

Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

Hi Danilo, Can you just use application *Dial(SIP/OpenSips/${EXTEN})* at the IVR to send call back to the opensips server.Thats how I'd do to send call back to OpenSIPS. BR, Sammy On Thu, Mar 5, 2015 at 11:10 AM, danilo...@tin.it wrote: > Hi there, > I'm working on this scenario to manage som

Re: [OpenSIPS-Users] mediaproxy, force trusting SDP

Hi, Here is what it looks like to me; NAT'd Client<=>OpenSIPS+Media-Proxy<==>Second Proxy??<=>Upstream >From what I understand, your main issue is that after failover/failure_route your call goes to second Carrier and since the use_media_proxy has been called for first one, hand shake

Re: [OpenSIPS-Users] MediaProxy behaviour on Heavy call volume.

ius requests, media reservations can cause this. > > The farther away each component is the poorest the performance as that > child cannot process any new packet until is done with the previous. > > Adrian > > On Apr 7, 2013, at 3:27 PM, SamyGo wrote: > > Hi Sir, > Yes, we'

Re: [OpenSIPS-Users] MediaProxy behaviour on Heavy call volume.

; > > > On Fri, Apr 5, 2013 at 8:42 PM, SamyGo wrote: > >> Hello, >> >> I'm working with opensips with heavy CPS, recently I added >> mediaproxy-dispatcher on the server and couple of relays on different >> servers in different Data Center. Everyth

[OpenSIPS-Users] MediaProxy behaviour on Heavy call volume.

Hello, I'm working with opensips with heavy CPS, recently I added mediaproxy-dispatcher on the server and couple of relays on different servers in different Data Center. Everything worked fine until we observed that opensips SIP UDP port 5060 got huge queued packets. That obviously impacts the c

Re: [OpenSIPS-Users] Max storable size in $dlg_val(name)

ds, > > Vlad Paiu > OpenSIPS Developerhttp://www.opensips-solutions.com > > > On 03/26/2013 02:26 PM, SamyGo wrote: > > Hello, > > I'm using dlg_val or store_dlg_value() functions in my script and its > working fine, but I want to know what is the maximum

[OpenSIPS-Users] Max storable size in $dlg_val(name)

Hello, I'm using dlg_val or store_dlg_value() functions in my script and its working fine, but I want to know what is the maximum size of data I can store in it before it chops off part of data before storing. !! Thanks, Sammy ___ Users mailing list Us

Re: [OpenSIPS-Users] RTPProxy Support - Not prefilling callees address

Hi Nick, When I first hit the same issue where media was to be sent/received from a different IP other than signalling I used 'r' flag in the engage_rtpproxy() function. I don't think you are using this function rather using offer and answer functions which I'd say a manual approach. Tell me if us

Re: [OpenSIPS-Users] Sipcapture with different DB backend

db_xx > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 02/07/2013 04:29 PM, SamyGo wrote: > > Hello, > > I've been working with sipcapture module to rpelicate traffic to another > server. A

[OpenSIPS-Users] Sipcapture with different DB backend

Hello, I've been working with sipcapture module to rpelicate traffic to another server. Also made it to work with mysql db backend, now I need to know if I can push the sip traces to some other backend ! Is there support for using any other back end. Thanks, Sammy

Re: [OpenSIPS-Users] TM module Minor branching

Minor correction: Its "183 - Session Progress" instead of 182 in first line. On Thu, Dec 20, 2012 at 12:56 PM, SamyGo wrote: > Hi, > > I'm using the minor_branch flag for my calls to inject 182 session > progress for the calls to carriers. > > http://www.ope

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