[OpenSIPS-Devel] [opensips] UAC_REGISTRANT module: fixed expiration time calculation (#236)
You can merge this Pull Request by running: git pull https://github.com/nikbyte/nikbyte-opensips uac_registrant Or you can view, comment on it, or merge it online at: https://github.com/OpenSIPS/opensips/pull/236 -- Commit Summary -- * UAC_REGISTRANT module: fixed expiration time calculation -- File Changes -- M modules/uac_registrant/registrant.c (13) -- Patch Links -- https://github.com/OpenSIPS/opensips/pull/236.patch https://github.com/OpenSIPS/opensips/pull/236.diff --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/pull/236 ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [opensips] uac_registrant and external expires parameter (in 200OK) (#217)
Recreated pull request. The proper one: https://github.com/OpenSIPS/opensips/pull/236 --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/issues/217#issuecomment-43589176___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [opensips] UAC_REGISTRANT module: fixed expiration time calculation (#234)
Closed #234. --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/pull/234#event-122760818___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [opensips] RTPPROXY module: compatibility with topology hiding (#237)
RTPPROXY module: workaround to run rtpproxy when we use dialog topology hiding and thus we have not 2nd via in reply It fixes autobridging for rtpproxy when using topology hiding. The problem is reply doesn#39;t have second via. We used this patch on production system about 7 months without any problems. You can merge this Pull Request by running: git pull https://github.com/nikbyte/nikbyte-opensips rtpproxy Or you can view, comment on it, or merge it online at: https://github.com/OpenSIPS/opensips/pull/237 -- Commit Summary -- * RTPPROXY module: workaround to run rtpproxy when we use dialog topology hiding and thus we have not 2nd via in reply -- File Changes -- M modules/rtpproxy/rtpproxy.c (22) -- Patch Links -- https://github.com/OpenSIPS/opensips/pull/237.patch https://github.com/OpenSIPS/opensips/pull/237.diff --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/pull/237 ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [opensips] RTPPROXY module: compatibility with topology hiding (#237)
} else {
- LM_ERR(can't extract 2nd via
found reply\n);
+ if (parse_headers(msg,
HDR_VIA2_F, 0) != -1
You could make the `} else {` above an `} else if {`, that saves you some
indentation.
Feel free to squash and push -f.
---
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https://github.com/OpenSIPS/opensips/pull/237/files#r12830105___
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Re: [OpenSIPS-Devel] [opensips] RTPPROXY module: compatibility with topology hiding (#237)
@@ -3229,6 +3229,8 @@ force_rtp_proxy(struct sip_msg* msg, char* str1, char* str2, char *setid, char * struct force_rtpp_args args; struct force_rtpp_args *ap; union sockaddr_union to; + struct socket_info *si; + struct dlg_cell * dlg; Please stay consistent with local code style: join the `*` to the `dlg`. --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/pull/237/files#r12830089___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] Send rtp packet between two rtpproxy server
Hello, Is it possible to send media rtp packets from one rtpproxy server to another rtpproxy server? In my scenario , i am registering voip account via opensips proxy server. We have rtpproxy and opensips server hosted on same place. opensips changes c= and m= lines of SDP accordingly but when packet goes to voip switch , its rtpproxy server also changes SDP. So Peer1 is sending packets to hout hosted rtpproxy server and peer2 sending rtp packets to voip switch's rtpproxy server. SIP packets : Peer -- opensips -- asterisk RTP packets peer1 -- rtpproxy1 =X= rtpproxy2 --peer2 here no connection between rtpproxy1 and rtpproxy 2 so no media transfer between peer1 and peer2. Please help to solve this problem. -- Kind regards, Kaushik Parmar ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Send rtp packet between two rtpproxy server
Hi, Kaushik! I don't see any problem with your scenario, it should work without any issues - peer2 should send RTP to rtpproxy2 which relays them to rtpproy1 and then peer1, and the other way around. Are you sure both proxies advertise the correct IP in the SDP, for both INVITE and 200 OK? Best regards, PS: this is not a devel question. Please register on us...@lists.opensips.org and move the thread there. Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 05/20/2014 10:17 AM, kaushik parmar wrote: Hello, Is it possible to send media rtp packets from one rtpproxy server to another rtpproxy server? In my scenario , i am registering voip account via opensips proxy server. We have rtpproxy and opensips server hosted on same place. opensips changes c= and m= lines of SDP accordingly but when packet goes to voip switch , its rtpproxy server also changes SDP. So Peer1 is sending packets to hout hosted rtpproxy server and peer2 sending rtp packets to voip switch's rtpproxy server. SIP packets : Peer -- opensips -- asterisk RTP packets peer1 -- rtpproxy1 =X= rtpproxy2 --peer2 here no connection between rtpproxy1 and rtpproxy 2 so no media transfer between peer1 and peer2. Please help to solve this problem. -- Kind regards, Kaushik Parmar ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Send rtp packet between two rtpproxy
Thank you Razvan, As you said rtpproxy2 will send rtp to rtpproxy1. How can i implement this? What are configuration in opensips or rtpproxy to make this possible? -- Kind regards, Kaushik Parmar ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [OpenSIPS/opensips] 6dc726: dispatcher: add priority column
Branch: refs/heads/master Home: https://github.com/OpenSIPS/opensips Commit: 6dc726158f4042e07bd019c1fbbf8288a1b655b4 https://github.com/OpenSIPS/opensips/commit/6dc726158f4042e07bd019c1fbbf8288a1b655b4 Author: Razvan Crainea raz...@opensips.org Date: 2014-05-20 (Tue, 20 May 2014) Changed paths: M db/schema/dispatcher.xml M modules/dispatcher/dispatch.c M modules/dispatcher/dispatch.h M modules/dispatcher/dispatcher.c M scripts/db_berkeley/opensips/dispatcher M scripts/dbtext/opensips/dispatcher M scripts/mysql/dispatcher-create.sql M scripts/oracle/dispatcher-create.sql M scripts/pi_http/dispatcher-mod M scripts/pi_http/dispatcher-table M scripts/pi_http/pi_framework.xml M scripts/postgres/dispatcher-create.sql Log Message: --- dispatcher: add priority column This is useful for first entry algorithm, where you can now order the entries regardless the database order Commit: 310dafa013b1f127c9cf60b4f7a5efac639b1aed https://github.com/OpenSIPS/opensips/commit/310dafa013b1f127c9cf60b4f7a5efac639b1aed Author: Razvan Crainea raz...@opensips.org Date: 2014-05-20 (Tue, 20 May 2014) Changed paths: M modules/dispatcher/dispatch.c Log Message: --- dispatcher: fix possible bogus gateways In case the domain name can not be resolved, the destination should be droped and the number of destinations within the set should not be increased. Compare: https://github.com/OpenSIPS/opensips/compare/79249f8ce0ac...310dafa013b1___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [OpenSIPS/opensips] 46327c: dispatcher: fix possible bogus gateways
Branch: refs/heads/1.11 Home: https://github.com/OpenSIPS/opensips Commit: 46327c188685ad775f5a15c3312f77a14961cfcf https://github.com/OpenSIPS/opensips/commit/46327c188685ad775f5a15c3312f77a14961cfcf Author: Razvan Crainea raz...@opensips.org Date: 2014-05-20 (Tue, 20 May 2014) Changed paths: M modules/dispatcher/dispatch.c Log Message: --- dispatcher: fix possible bogus gateways In case the domain name can not be resolved, the destination should be droped and the number of destinations within the set should not be increased. (cherry picked from commit 310dafa013b1f127c9cf60b4f7a5efac639b1aed) Conflicts: modules/dispatcher/dispatch.c ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Devel Digest, Vol 71, Issue 42
rtpproxy
Or you can view, comment on it, or merge it online at:
https://github.com/OpenSIPS/opensips/pull/237
-- Commit Summary --
* RTPPROXY module: workaround to run rtpproxy when we use dialog
topology hiding and thus we have not 2nd via in reply
-- File Changes --
M modules/rtpproxy/rtpproxy.c (22)
-- Patch Links --
https://github.com/OpenSIPS/opensips/pull/237.patch
https://github.com/OpenSIPS/opensips/pull/237.diff
---
Reply to this email directly or view it on GitHub:
https://github.com/OpenSIPS/opensips/pull/237
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Message: 5
Date: Tue, 20 May 2014 00:00:54 -0700
From: Walter Doekes notificati...@github.com
Subject: Re: [OpenSIPS-Devel] [opensips] RTPPROXY module:
compatibility with topology hiding (#237)
To: devel@lists.opensips.org
Message-ID: OpenSIPS/opensips/pull/237/r12830...@github.com
Content-Type: text/plain; charset=utf-8
@@ -3229,6 +3229,8 @@ force_rtp_proxy(struct sip_msg* msg, char* str1,
char* str2, char *setid, char *
struct force_rtpp_args args;
struct force_rtpp_args *ap;
union sockaddr_union to;
+ struct socket_info *si;
+ struct dlg_cell * dlg;
Please stay consistent with local code style: join the `*` to the `dlg`.
---
Reply to this email directly or view it on GitHub:
https://github.com/OpenSIPS/opensips/pull/237/files#r12830089
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Message: 6
Date: Tue, 20 May 2014 00:01:40 -0700
From: Walter Doekes notificati...@github.com
Subject: Re: [OpenSIPS-Devel] [opensips] RTPPROXY module:
compatibility with topology hiding (#237)
To: devel@lists.opensips.org
Message-ID: OpenSIPS/opensips/pull/237/r12830...@github.com
Content-Type: text/plain; charset=utf-8
} else {
- LM_ERR(can't extract 2nd
via found reply\n);
+ if (parse_headers(msg,
HDR_VIA2_F, 0) != -1
You could make the `} else {` above an `} else if {`, that saves you some
indentation.
Feel free to squash and push -f.
---
Reply to this email directly or view it on GitHub:
https://github.com/OpenSIPS/opensips/pull/237/files#r12830105
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Message: 7
Date: Tue, 20 May 2014 12:47:27 +0530
From: kaushik parmar androidj...@gmail.com
Subject: [OpenSIPS-Devel] Send rtp packet between two rtpproxy server
To: devel@lists.opensips.org
Message-ID:
CAGzPE1-QJdJ4-LKhwn4rVWeEFGUJQ0RZG92kquWyw+c=
vhp...@mail.gmail.com
Content-Type: text/plain; charset=utf-8
Hello,
Is it possible to send media rtp packets from one rtpproxy server to
another rtpproxy server?
In my scenario , i am registering voip account via opensips proxy server.
We have rtpproxy and opensips server hosted on same place. opensips changes
c= and m= lines of SDP accordingly but when packet goes to voip switch ,
its rtpproxy server also changes SDP. So Peer1 is sending packets to hout
hosted rtpproxy server and peer2 sending rtp packets to voip switch's
rtpproxy server.
SIP packets : Peer -- opensips -- asterisk
RTP packets peer1 -- rtpproxy1 =X= rtpproxy2 --peer2
here no connection between rtpproxy1 and rtpproxy 2 so no media transfer
between peer1 and peer2. Please help to solve this problem.
--
Kind regards,
Kaushik Parmar
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Message: 8
Date: Tue, 20 May 2014 10:34:42 +0300
From: R?zvan Crainea raz...@opensips.org
Subject: Re: [OpenSIPS-Devel] Send rtp packet between two rtpproxy
server
To: devel@lists.opensips.org
Message-ID: 537b0592.8060...@opensips.org
Content-Type: text/plain; charset=UTF-8; format=flowed
Hi, Kaushik!
I don't see any problem with your scenario, it should work without any
issues - peer2 should send RTP to rtpproxy2 which relays them to
rtpproy1 and then peer1, and the other way around.
Are you sure both proxies advertise the correct IP in the SDP, for both
INVITE and 200 OK?
Best regards,
PS: this is not a devel question. Please register on
us...@lists.opensips.org and move the thread there.
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/20/2014 10:17 AM, kaushik parmar wrote
Re: [OpenSIPS-Devel] [opensips] uac_registrant and external expires parameter (in 200OK) (#217)
Both hash size and timer interval are configurable so the admin can tune them to fit their needs. One reason for having a hash is to avoid running over all records when the timer fires. When the timer fires, the modules checks only one entry into the hash table, that's why the timer_interval is divided by reg_hsize during initialization. If you want to use expire=120 with a hash_size =4, then set: modparam(uac_registrant, timer_interval, 32) modparam(uac_registrant, hash_size, 4) This will result in 16 branches in the hash table. Each record will be checked every 32s and every 2s the timer will fire checking one branch in the hash table. --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/issues/217#issuecomment-43646676___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] OpenSIPS ignores CANCEL request
Hi, I’m using PJSIP Library for Android client, when call is initiated by A Party, and before B Party answers the call, if A Party sends CANCEL request, OpenSIPS does neither send ACK nor forward it to B Party. This works fine if we do testing with X-Lite or Zoiper. Please let me know what could be the issue? Maqbul A Khan Reform InfoTech email: maq...@reforminfotech.com mobile: +91 99988 97686 skype: maqbul.a ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] Is OpenSIPS source code architecture independent?
I want to port OpenSIPs on a RISC platform. is it architecture independent? I don't see a configure script in the source code directory. What is the best way to port it? ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [opensips] /tmp/opensips_fifo intermittently disappears (#235)
Closed #235. --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/issues/235#event-122986649___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [opensips] /tmp/opensips_fifo intermittently disappears (#235)
Indeed, OpenSIPS itself does not delete the fifo file at all. Also the code cannot detect if the FIFO file was deleted by mistake from outside (in order to re-open the file). As @jalung said, it will be wiser to move your FIFO file away from /tmp, which is subject the cleanup. I will close this ticket as it is not an bug or so - I put on the NICE-2-HAVE list the ability to detect and reopen a deleted fifo file. --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/issues/235#issuecomment-43650774___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [opensips] uac_registrant and external expires parameter (in 200OK) (#217)
I think this moment not well documented. And also something wrong there. If I have timer_interval 60 with hash_size 5 then for registration with expires=120 we have rec-registration_timeout = now + 120 - 60 = now + 60 So, we will update every 60 seconds. It's not good. Maybe this logic will be better? rec-registration_timeout = now + 120 - 120*0.8 (80% of expires) rec-registration_timeout = now + rec-expires - rec-expires*0.8; instead of rec-registration_timeout = now + rec-expires - timer_interval; When you have many registrants and database with many different intervals, you will have many inaccuracies with re-register intervals. Or we should better document this moment. For example, we should recommend using as small timer_interval as user can. --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/issues/217#issuecomment-43670320___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [opensips] /tmp/opensips_fifo intermittently disappears (#235)
Many thanks for the help! --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/issues/235#issuecomment-43673278___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [OpenSIPS/opensips] f51752: dialplan: return 404 No translation for failed M...
Branch: refs/heads/master Home: https://github.com/OpenSIPS/opensips Commit: f517529a86a2d77a9645e341153b76384e56f4fb https://github.com/OpenSIPS/opensips/commit/f517529a86a2d77a9645e341153b76384e56f4fb Author: Ovidiu Sas o...@voipembedded.com Date: 2014-05-20 (Tue, 20 May 2014) Changed paths: M modules/dialplan/dialplan.c Log Message: --- dialplan: return 404 No translation for failed MI dp_translate command ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [OpenSIPS/opensips] 2923d4: dialplan: dp_translate(): accept id param as STR...
Branch: refs/heads/master Home: https://github.com/OpenSIPS/opensips Commit: 2923d4178484af48d1f3f30c9fa1b8b7464c72f9 https://github.com/OpenSIPS/opensips/commit/2923d4178484af48d1f3f30c9fa1b8b7464c72f9 Author: Ovidiu Sas o...@voipembedded.com Date: 2014-05-20 (Tue, 20 May 2014) Changed paths: M modules/dialplan/dialplan.c Log Message: --- dialplan: dp_translate(): accept id param as STR PVAR - convert it to INT id STR PVAR (if possible) ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
