Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hi, Yavari! Yes, I think you should call this method when you receive the INVITE. Also, you should probably convert the 180 to 183. Also, try to do some traces on the RTPProxy machine, to see if it tries to generate media. Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 02/26/2014 03:08 PM, H Yavari wrote: Hi Razvan, I want Ringing replaced with Playing media for user that called (UAC). Yet caller hear nothing. I should drop 180 ? I should call this method when I received INVITE? Best Regards, H.Yavari Hi, Yavari! In your scenario, you want to play media to UAC, not UAS, right? Does it work properly now, can the caller hear the ringing? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 02/26/2014 01:34 PM, H Yavari wrote: Hi, Your hint solved the problem of the playing and now I see this in logs: INFO:handle_play: -1 times playing prompt /var/rtpproxy/prompts/092 codec 8. Shouldn't you use the rtpproxy_stream2uac() function for playing ringback tone? Why? and How I should this? Hi, Yavari! That error indicates that RTPProxy couldn't find any available codecs for your client. Please check the files in /var/rtpproxy/prompts/092.* to make sure you have the proper files installed. I find something weird in your examples: you're saying that you are calling the function like this, which is correct: rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1); However, RTPProxy prints the comand as: /var/rtpproxy/prompts/092.8, which is incorrect, since the codec should not be appended to the file. Are you using different configure versions? Also, shouldn't you use the rtpproxy_stream2uac() function for playing ringback tone? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/http://www.opensips-solutions.com/ On 02/26/2014 09:22 AM, H Yavari wrote: Dear Razvan, Thanks for your reply. RTPproxy and OpenSIPS have communication with each other and all of call's RTP streams passed from RTPproxy. So there isn't any problem between them.In logs only exist this info: DBG:rtpproxy:force_rtp_proxy_body: proxy reply: 57548 192.168.1.20#012 and nothing for : rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1) or any errors. What do you mean the function does not work properly? I mean this function not work for me and not send any command to RTPproxy. If you can, give me a choice to use your patch. Thanks so. Best Regards, H.Yavari PS: I enabled the RTPproxy logging and this is result of calling the method :rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1) ::: DBUG:handle_command: received command 6387_7 P-1 562e15da4cdd600161336f3235ce8213@192.168.1.116 mailto:562e15da4cdd600161336f3235ce8213@192.168.1.116 mailto:562e15da4cdd600161336f3235ce8213@192.168.1.116 mailto:562e15da4cdd600161336f3235ce8213@192.168.1.116 /var/rtpproxy/prompts/092.8 session as5f218e73;1 ERR:handle_play: can't create player DBUG:doreply: sending reply 6387_7 E6 ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hi, Yavari!
That error indicates that RTPProxy couldn't find any available codecs
for your client. Please check the files in /var/rtpproxy/prompts/092.*
to make sure you have the proper files installed.
I find something weird in your examples: you're saying that you are
calling the function like this, which is correct:
rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1);
However, RTPProxy prints the comand as: /var/rtpproxy/prompts/092.8,
which is incorrect, since the codec should not be appended to the file.
Are you using different configure versions?
Also, shouldn't you use the rtpproxy_stream2uac() function for playing
ringback tone?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/26/2014 09:22 AM, H Yavari wrote:
Dear Razvan,
Thanks for your reply. RTPproxy and OpenSIPS have communication with
each other and all of call's RTP streams passed from RTPproxy. So there
isn't any problem between them.In logs only exist this info:
DBG:rtpproxy:force_rtp_proxy_body: proxy reply: 57548 192.168.1.20#012
and nothing for : rtpproxy_stream2uas(/var/rtpproxy/prompts/test,
-1) or any errors.
What do you mean the function does not work properly? I mean this
function not work for me and not send any command to RTPproxy.
If you can, give me a choice to use your patch.
Thanks so.
Best Regards,
H.Yavari
PS:
I enabled the RTPproxy logging and this is result of calling the method
:rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1) :::
DBUG:handle_command: received command 6387_7 P-1
562e15da4cdd600161336f3235ce8213@192.168.1.116
/var/rtpproxy/prompts/092.8 session as5f218e73;1
ERR:handle_play: can't create player
DBUG:doreply: sending reply 6387_7 E6
Hi!
What do you mean the function does not work properly? It doesn't send
any command to RTPProxy? Can you trace the communication between
OpenSIPS and RTPProxy?
Also, have you checked the RTPProxy logs for errors? I am not sure how
you can detect this, but if I remember correctly, I had to patch
RTPProxy to properly inject media in early stage. If you want me, I can
provide that patch for you so you can give it a try.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com http://www.opensips-solutions.com/
On 02/24/2014 08:03 AM, H Yavari wrote:
Hi,
I used Opensips with RTPproxy and now RTP goes from RTPproxy. But
rtpproxy_stream2uac() function not work properly. my cfg:
if (is_method(INVITE)) {
rtpproxy_offer(ro);
rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1);
}
There isn't any info in logs about this function. what is the problem?
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Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hi, Your hint solved the problem of the playing and now I see this in logs: INFO:handle_play: -1 times playing prompt /var/rtpproxy/prompts/092 codec 8. Shouldn't you use the rtpproxy_stream2uac() function for playing ringback tone? Why? and How I should this? Hi, Yavari! That error indicates that RTPProxy couldn't find any available codecs for your client. Please check the files in /var/rtpproxy/prompts/092.* to make sure you have the proper files installed. I find something weird in your examples: you're saying that you are calling the function like this, which is correct: rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1); However, RTPProxy prints the comand as: /var/rtpproxy/prompts/092.8, which is incorrect, since the codec should not be appended to the file. Are you using different configure versions? Also, shouldn't you use the rtpproxy_stream2uac() function for playing ringback tone? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 02/26/2014 09:22 AM, H Yavari wrote: Dear Razvan, Thanks for your reply. RTPproxy and OpenSIPS have communication with each other and all of call's RTP streams passed from RTPproxy. So there isn't any problem between them.In logs only exist this info: DBG:rtpproxy:force_rtp_proxy_body: proxy reply: 57548 192.168.1.20#012 and nothing for : rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1) or any errors. What do you mean the function does not work properly? I mean this function not work for me and not send any command to RTPproxy. If you can, give me a choice to use your patch. Thanks so. Best Regards, H.Yavari PS: I enabled the RTPproxy logging and this is result of calling the method :rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1) ::: DBUG:handle_command: received command 6387_7 P-1 562e15da4cdd600161336f3235ce8213@192.168.1.116 /var/rtpproxy/prompts/092.8 session as5f218e73;1 ERR:handle_play: can't create player DBUG:doreply: sending reply 6387_7 E6___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hi, Yavari! In your scenario, you want to play media to UAC, not UAS, right? Does it work properly now, can the caller hear the ringing? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 02/26/2014 01:34 PM, H Yavari wrote: Hi, Your hint solved the problem of the playing and now I see this in logs: INFO:handle_play: -1 times playing prompt /var/rtpproxy/prompts/092 codec 8. Shouldn't you use the rtpproxy_stream2uac() function for playing ringback tone? Why? and How I should this? Hi, Yavari! That error indicates that RTPProxy couldn't find any available codecs for your client. Please check the files in /var/rtpproxy/prompts/092.* to make sure you have the proper files installed. I find something weird in your examples: you're saying that you are calling the function like this, which is correct: rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1); However, RTPProxy prints the comand as: /var/rtpproxy/prompts/092.8, which is incorrect, since the codec should not be appended to the file. Are you using different configure versions? Also, shouldn't you use the rtpproxy_stream2uac() function for playing ringback tone? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 02/26/2014 09:22 AM, H Yavari wrote: Dear Razvan, Thanks for your reply. RTPproxy and OpenSIPS have communication with each other and all of call's RTP streams passed from RTPproxy. So there isn't any problem between them.In logs only exist this info: DBG:rtpproxy:force_rtp_proxy_body: proxy reply: 57548 192.168.1.20#012 and nothing for : rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1) or any errors. What do you mean the function does not work properly? I mean this function not work for me and not send any command to RTPproxy. If you can, give me a choice to use your patch. Thanks so. Best Regards, H.Yavari PS: I enabled the RTPproxy logging and this is result of calling the method :rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1) ::: DBUG:handle_command: received command 6387_7 P-1 562e15da4cdd600161336f3235ce8213@192.168.1.116 mailto:562e15da4cdd600161336f3235ce8213@192.168.1.116 /var/rtpproxy/prompts/092.8 session as5f218e73;1 ERR:handle_play: can't create player DBUG:doreply: sending reply 6387_7 E6 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hi Razvan, I want Ringing replaced with Playing media for user that called (UAC). Yet caller hear nothing. I should drop 180 ? I should call this method when I received INVITE? Best Regards, H.Yavari Hi, Yavari! In your scenario, you want to play media to UAC, not UAS, right? Does it work properly now, can the caller hear the ringing? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 02/26/2014 01:34 PM, H Yavari wrote: Hi, Your hint solved the problem of the playing and now I see this in logs: INFO:handle_play: -1 times playing prompt /var/rtpproxy/prompts/092 codec 8. Shouldn't you use the rtpproxy_stream2uac() function for playing ringback tone? Why? and How I should this? Hi, Yavari! That error indicates that RTPProxy couldn't find any available codecs for your client. Please check the files in /var/rtpproxy/prompts/092.* to make sure you have the proper files installed. I find something weird in your examples: you're saying that you are calling the function like this, which is correct: rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1); However, RTPProxy prints the comand as: /var/rtpproxy/prompts/092.8, which is incorrect, since the codec should not be appended to the file. Are you using different configure versions? Also, shouldn't you use the rtpproxy_stream2uac() function for playing ringback tone? Best regards, Razvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 02/26/2014 09:22 AM, H Yavari wrote: Dear Razvan, Thanks for your reply. RTPproxy and OpenSIPS have communication with each other and all of call's RTP streams passed from RTPproxy. So there isn't any problem between them.In logs only exist this info: DBG:rtpproxy:force_rtp_proxy_body: proxy reply: 57548 192.168.1.20#012 and nothing for : rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1) or any errors. What do you mean the function does not work properly? I mean this function not work for me and not send any command to RTPproxy. If you can, give me a choice to use your patch. Thanks so. Best Regards, H.Yavari PS: I enabled the RTPproxy logging and this is result of calling the method :rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1) ::: DBUG:handle_command: received command 6387_7 P-1 562e15da4cdd600161336f3235ce8213@192.168.1.116 mailto:562e15da4cdd600161336f3235ce8213@192.168.1.116 /var/rtpproxy/prompts/092.8 session as5f218e73;1 ERR:handle_play: can't create player DBUG:doreply: sending reply 6387_7 E6 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
Dear Razvan,
Thanks for your reply. RTPproxy and OpenSIPS have communication with each other
and all of call's RTP streams passed from RTPproxy. So there isn't any problem
between them.In logs only exist this info: DBG:rtpproxy:force_rtp_proxy_body:
proxy reply: 57548 192.168.1.20#012
and nothing for : rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1) or
any errors.
What do you mean the function does not work properly? I mean this function
not work for me and not send any command to RTPproxy.
If you can, give me a choice to use your patch.
Thanks so.
Best Regards,
H.Yavari
Hi!
What do you mean the function does not work properly? It doesn't send
any command to RTPProxy? Can you trace the communication between
OpenSIPS and RTPProxy?
Also, have you checked the RTPProxy logs for errors? I am not sure how
you can detect this, but if I remember correctly, I had to patch
RTPProxy to properly inject media in early stage. If you want me, I can
provide that patch for you so you can give it a try.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/24/2014 08:03 AM, H Yavari wrote:
Hi,
I used Opensips with RTPproxy and now RTP goes from RTPproxy. But
rtpproxy_stream2uac() function not work properly. my cfg:
if (is_method(INVITE)) {
rtpproxy_offer(ro);
rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1);
}
There isn't any info in logs about this function. what is the problem?
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Re: [OpenSIPS-Users] Initializing SIP messages from routing
Dear Razvan,
Thanks for your reply. RTPproxy and OpenSIPS have communication with each other
and all of call's RTP streams passed from RTPproxy. So there isn't any problem
between them.In logs only exist this info: DBG:rtpproxy:force_rtp_proxy_body:
proxy reply: 57548 192.168.1.20#012
and nothing for : rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1) or
any errors.
What do you mean the function does not work properly? I mean this function
not work for me and not send any command to RTPproxy.
If you can, give me a choice to use your patch.
Thanks so.
Best Regards,
H.Yavari
PS:
I enabled the RTPproxy logging and this is result of calling the method
:rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1) :::
DBUG:handle_command: received command 6387_7 P-1
562e15da4cdd600161336f3235ce8213@192.168.1.116
/var/rtpproxy/prompts/092.8 session as5f218e73;1
ERR:handle_play: can't create player
DBUG:doreply: sending reply 6387_7 E6
Hi!
What do you mean the function does not work
properly? It doesn't send
any command to RTPProxy? Can you trace the communication between
OpenSIPS and RTPProxy?
Also, have you checked the RTPProxy logs for errors? I am not sure how
you can detect this, but if I remember correctly, I had to patch
RTPProxy to properly inject media in early stage. If you want me, I can
provide that patch for you so you can give it a try.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/24/2014 08:03 AM, H Yavari wrote:
Hi,
I used Opensips with RTPproxy and now RTP goes from
RTPproxy. But
rtpproxy_stream2uac() function not work properly. my cfg:
if (is_method(INVITE)) {
rtpproxy_offer(ro);
rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1);
}
There isn't any info in logs about this function. what is the problem?
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Re: [OpenSIPS-Users] Initializing SIP messages from routing
Sorry, but if not more difficult, please help.
I need configured Rtpproxy or mediaproxy B2B and to work with such a scheme:
UAC---si,rtp(IP1 Opensips IP2)sip,rtp-(MGW accepts all
INVITE with prefix)TDM---PSTN
Please helpwith CFG-file and rtp-proxy configurate.
26.02.2014 11:22, H Yavari пишет:
Dear Razvan,
Thanks for your reply. RTPproxy and OpenSIPS have communication with
each other and all of call's RTP streams passed from RTPproxy. So
there isn't any problem between them.In logs only exist this info:
DBG:rtpproxy:force_rtp_proxy_body: proxy reply: 57548 192.168.1.20#012
and nothing for : rtpproxy_stream2uas(/var/rtpproxy/prompts/test,
-1) or any errors.
What do you mean the function does not work properly? I mean this
function not work for me and not send any command to RTPproxy.
If you can, give me a choice to use your patch.
Thanks so.
Best Regards,
H.Yavari
PS:
I enabled the RTPproxy logging and this is result of calling the
method :rtpproxy_stream2uas(/var/rtpproxy/prompts/092, -1) :::
DBUG:handle_command: received command 6387_7 P-1
562e15da4cdd600161336f3235ce8213@192.168.1.116
/var/rtpproxy/prompts/092.8 session as5f218e73;1
ERR:handle_play: can't create player
DBUG:doreply: sending reply 6387_7 E6
Hi!
What do you mean the function does not work properly? It doesn't send
any command to RTPProxy? Can you trace the communication between
OpenSIPS and RTPProxy?
Also, have you checked the RTPProxy logs for errors? I am not sure how
you can detect this, but if I remember correctly, I had to patch
RTPProxy to properly inject media in early stage. If you want me, I can
provide that patch for you so you can give it a try.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com http://www.opensips-solutions.com/
On 02/24/2014 08:03 AM, H Yavari wrote:
Hi,
I used Opensips with RTPproxy and now RTP goes from RTPproxy. But
rtpproxy_stream2uac() function not work properly. my cfg:
if (is_method(INVITE)) {
rtpproxy_offer(ro);
rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1);
}
There isn't any info in logs about this function. what is the problem?
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Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hi!
What do you mean the function does not work properly? It doesn't send
any command to RTPProxy? Can you trace the communication between
OpenSIPS and RTPProxy?
Also, have you checked the RTPProxy logs for errors? I am not sure how
you can detect this, but if I remember correctly, I had to patch
RTPProxy to properly inject media in early stage. If you want me, I can
provide that patch for you so you can give it a try.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/24/2014 08:03 AM, H Yavari wrote:
Hi,
I used Opensips with RTPproxy and now RTP goes from RTPproxy. But
rtpproxy_stream2uac() function not work properly. my cfg:
if (is_method(INVITE)) {
rtpproxy_offer(ro);
rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1);
}
There isn't any info in logs about this function. what is the problem?
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Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hi,
I used Opensips with RTPproxy and now RTP goes from RTPproxy. But
rtpproxy_stream2uac() function not work properly. my cfg:
if (is_method(INVITE)) { rtpproxy_offer(ro);
rtpproxy_stream2uas(/var/rtpproxy/prompts/test, -1); }
There isn't any info in logs about this function. what is the problem?
Hello,
You can simply do ring back tones with OpenSIPS and RTPProxy -
it can inject media in a call in early stage. See the
rtpproxy_stream2uac() functions:
http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html#rtpproxy_stream2xxx
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer http://www.opensips-solutions.com
On 13.02.2014 20:45, Jayesh Nambiar wrote:
Hi,
CRBT is caller ring back tone. What you are primarily looking at is sending
the INVITE to some b2bua like FreeSWITCH or Asterisk where you control both
legs of the call. So when you get a ringing signal from the B-leg, you play
some media file on the A-leg.
--- Jayesh
On Friday, February 7, 2014, Bogdan-Andrei Iancu bog...@opensips.org wrote:
Hello,
No sure what CRBT stands for, but it looks to me that
you need to use B2B module - what you are trying to do
is something more than simply proxying a call.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer http://www.opensips-solutions.com
On 07.02.2014 19:00, H Yavari wrote:
Hi Bogdan,
thanks for your answer. I want to implement CRBT. For this I want when the
invite received,I send an invite to media server and play something.
I can do this with B2BUA? I write a module for this or do with script?
script running has side effect on performance when load is high?
Regards,
H.Yavari
Hello,
Typically you process in OpenSIPS script
an incoming request (and you fwd or reply
it). It is unusual to generate a new
request while processing another one.
May I ask about your scenario ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer http://www.opensips-solutions.com
On 28.01.2014 10:02, H Yavari wrote:
Hi to all openSIPSer,
I want to initialize a sip message from routing, is it possible?
If only way that do this is writing script, is this efficient when load on
openSIPS is high?
If answer is yes, can you give me some example to how do this?
Regards,
H.Yavari
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Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hello, RTPproxy can be installed on a different machine (and you can have multiple rtpproxies under a single OpenSIPS) - the communication between is done via IP. Yes, the idea is to start media injection and to stop it at 200 OK (to allow the end-to-end media). Using a B2BUA for this purpose is like using a truck to carry a feather :). It can be done, but not the optimal solution. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.02.2014 23:30, H Yavari wrote: Dear Bogdan, thanks for your guide. this solution is dependent to RTPProxy and I think this should be on the same server with opensips. this thing have any impact on performance? in this solution, we can play something when we received 180 or 183? and after the call answered we should care about RTP stream of the call ? Have you any idea how do this with B2BUA like freeswitch or asterisk? Regards, H.Yavari *From:* Bogdan-Andrei Iancu bog...@opensips.org *To:* OpenSIPS users mailling list users@lists.opensips.org; Jayesh Nambiar jayesh1...@gmail.com *Sent:* Thursday, 13 February 2014, 23:26:27 *Subject:* Re: [OpenSIPS-Users] Initializing SIP messages from routing Hello, You can simply do ring back tones with OpenSIPS and RTPProxy - it can inject media in a call in early stage. See the rtpproxy_stream2uac() functions: http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html#rtpproxy_stream2xxx Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 13.02.2014 20:45, Jayesh Nambiar wrote: Hi, CRBT is caller ring back tone. What you are primarily looking at is sending the INVITE to some b2bua like FreeSWITCH or Asterisk where you control both legs of the call. So when you get a ringing signal from the B-leg, you play some media file on the A-leg. --- Jayesh On Friday, February 7, 2014, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello, No sure what CRBT stands for, but it looks to me that you need to use B2B module - what you are trying to do is something more than simply proxying a call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 07.02.2014 19:00, H Yavari wrote: Hi Bogdan, thanks for your answer. I want to implement CRBT. For this I want when the invite received,I send an invite to media server and play something. I can do this with B2BUA? I write a module for this or do with script? script running has side effect on performance when load is high? Regards, H.Yavari Hello, Typically you process in OpenSIPS script an incoming request (and you fwd or reply it). It is unusual to generate a new request while processing another one. May I ask about your scenario ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 28.01.2014 10:02, H Yavari wrote: Hi to all openSIPSer, I want to initialize a sip message from routing, is it possible? If only way that do this is writing script, is this efficient when load on openSIPS is high? If answer is yes, can you give me some example to how do this? Regards, H.Yavari ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hi, CRBT is caller ring back tone. What you are primarily looking at is sending the INVITE to some b2bua like FreeSWITCH or Asterisk where you control both legs of the call. So when you get a ringing signal from the B-leg, you play some media file on the A-leg. --- Jayesh On Friday, February 7, 2014, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hello, No sure what CRBT stands for, but it looks to me that you need to use B2B module - what you are trying to do is something more than simply proxying a call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 07.02.2014 19:00, H Yavari wrote: Hi Bogdan, thanks for your answer. I want to implement CRBT. For this I want when the invite received,I send an invite to media server and play something. I can do this with B2BUA? I write a module for this or do with script? script running has side effect on performance when load is high? Regards, H.Yavari -- Hello, Typically you process in OpenSIPS script an incoming request (and you fwd or reply it). It is unusual to generate a new request while processing another one. May I ask about your scenario ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 28.01.2014 10:02, H Yavari wrote: Hi to all openSIPSer, I want to initialize a sip message from routing, is it possible? If only way that do this is writing script, is this efficient when load on openSIPS is high? If answer is yes, can you give me some example to how do this? Regards, H.Yavari ___ Users mailing listus...@lists.opensips.org javascript:_e(%7B%7D,'cvml','Users@lists.opensips.org');http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
You can develop something using perl and Net::SIP module. Il 13/02/2014 19.45, Jayesh Nambiar ha scritto: Hi, CRBT is caller ring back tone. What you are primarily looking at is sending the INVITE to some b2bua like FreeSWITCH or Asterisk where you control both legs of the call. So when you get a ringing signal from the B-leg, you play some media file on the A-leg. --- Jayesh On Friday, February 7, 2014, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello, No sure what CRBT stands for, but it looks to me that you need to use B2B module - what you are trying to do is something more than simply proxying a call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.02.2014 19:00, H Yavari wrote: Hi Bogdan, thanks for your answer. I want to implement CRBT. For this I want when the invite received,I send an invite to media server and play something. I can do this with B2BUA? I write a module for this or do with script? script running has side effect on performance when load is high? Regards, H.Yavari Hello, Typically you process in OpenSIPS script an incoming request (and you fwd or reply it). It is unusual to generate a new request while processing another one. May I ask about your scenario ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 28.01.2014 10:02, H Yavari wrote: Hi to all openSIPSer, I want to initialize a sip message from routing, is it possible? If only way that do this is writing script, is this efficient when load on openSIPS is high? If answer is yes, can you give me some example to how do this? Regards, H.Yavari ___ Users mailing list Users@lists.opensips.org javascript:_e(%7B%7D,'cvml','Users@lists.opensips.org'); http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hello, You can simply do ring back tones with OpenSIPS and RTPProxy - it can inject media in a call in early stage. See the rtpproxy_stream2uac() functions: http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html#rtpproxy_stream2xxx Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.02.2014 20:45, Jayesh Nambiar wrote: Hi, CRBT is caller ring back tone. What you are primarily looking at is sending the INVITE to some b2bua like FreeSWITCH or Asterisk where you control both legs of the call. So when you get a ringing signal from the B-leg, you play some media file on the A-leg. --- Jayesh On Friday, February 7, 2014, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello, No sure what CRBT stands for, but it looks to me that you need to use B2B module - what you are trying to do is something more than simply proxying a call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.02.2014 19:00, H Yavari wrote: Hi Bogdan, thanks for your answer. I want to implement CRBT. For this I want when the invite received,I send an invite to media server and play something. I can do this with B2BUA? I write a module for this or do with script? script running has side effect on performance when load is high? Regards, H.Yavari Hello, Typically you process in OpenSIPS script an incoming request (and you fwd or reply it). It is unusual to generate a new request while processing another one. May I ask about your scenario ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 28.01.2014 10:02, H Yavari wrote: Hi to all openSIPSer, I want to initialize a sip message from routing, is it possible? If only way that do this is writing script, is this efficient when load on openSIPS is high? If answer is yes, can you give me some example to how do this? Regards, H.Yavari ___ Users mailing list Users@lists.opensips.org javascript:_e(%7B%7D,'cvml','Users@lists.opensips.org'); http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hi, Thank you for your reply. I want do this exactly same as Jayesh said. Jayesh did you implement this? how? Stefano, scripting with perl is a good solution with high performance?Have you any example that help me? Regards, H.Yavari From: Stefano Pisani stefano.pis...@omnianet.it To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, 13 February 2014, 22:53:23 Subject: Re: [OpenSIPS-Users] Initializing SIP messages from routing You can develop something using perl and Net::SIP module. Il 13/02/2014 19.45, Jayesh Nambiar ha scritto: Hi, CRBT is caller ring back tone. What you are primarily looking at is sending the INVITE to some b2bua like FreeSWITCH or Asterisk where you control both legs of the call. So when you get a ringing signal from the B-leg, you play some media file on the A-leg. --- Jayesh On Friday, February 7, 2014, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hello, No sure what CRBT stands for, but it looks to me that you need to use B2B module - what you are trying to do is something more than simply proxying a call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.02.2014 19:00, H Yavari wrote: Hi Bogdan, thanks for your answer. I want to implement CRBT. For this I want when the invite received,I send an invite to media server and play something. I can do this with B2BUA? I write a module for this or do with script? script running has side effect on performance when load is high? Regards, H.Yavari Hello, Typically you process in OpenSIPS script an incoming request (and you fwd or reply it). It is unusual to generate a new request while processing another one. May I ask about your scenario ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.01.2014 10:02, H Yavari wrote: Hi to all openSIPSer, I want to initialize a sip message from routing, is it possible? If only way that do this is writing script, is this efficient when load on openSIPS is high? If answer is yes, can you give me some example to how do this? Regards, H.Yavari ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
Dear Bogdan, thanks for your guide. this solution is dependent to RTPProxy and I think this should be on the same server with opensips. this thing have any impact on performance? in this solution, we can play something when we received 180 or 183? and after the call answered we should care about RTP stream of the call ? Have you any idea how do this with B2BUA like freeswitch or asterisk? Regards, H.Yavari From: Bogdan-Andrei Iancu bog...@opensips.org To: OpenSIPS users mailling list users@lists.opensips.org; Jayesh Nambiar jayesh1...@gmail.com Sent: Thursday, 13 February 2014, 23:26:27 Subject: Re: [OpenSIPS-Users] Initializing SIP messages from routing Hello, You can simply do ring back tones with OpenSIPS and RTPProxy - it can inject media in a call in early stage. See the rtpproxy_stream2uac() functions: http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html#rtpproxy_stream2xxx Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 13.02.2014 20:45, Jayesh Nambiar wrote: Hi, CRBT is caller ring back tone. What you are primarily looking at is sending the INVITE to some b2bua like FreeSWITCH or Asterisk where you control both legs of the call. So when you get a ringing signal from the B-leg, you play some media file on the A-leg. --- Jayesh On Friday, February 7, 2014, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hello, No sure what CRBT stands for, but it looks to me that you need to use B2B module - what you are trying to do is something more than simply proxying a call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.02.2014 19:00, H Yavari wrote: Hi Bogdan, thanks for your answer. I want to implement CRBT. For this I want when the invite received,I send an invite to media server and play something. I can do this with B2BUA? I write a module for this or do with script? script running has side effect on performance when load is high? Regards, H.Yavari Hello, Typically you process in OpenSIPS script an incoming request (and you fwd or reply it). It is unusual to generate a new request while processing another one. May I ask about your scenario ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.01.2014 10:02, H Yavari wrote: Hi to all openSIPSer, I want to initialize a sip message from routing, is it possible? If only way that do this is writing script, is this efficient when load on openSIPS is high? If answer is yes, can you give me some example to how do this? Regards, H.Yavari ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Initializing SIP messages from routing
Hello, No sure what CRBT stands for, but it looks to me that you need to use B2B module - what you are trying to do is something more than simply proxying a call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.02.2014 19:00, H Yavari wrote: Hi Bogdan, thanks for your answer. I want to implement CRBT. For this I want when the invite received,I send an invite to media server and play something. I can do this with B2BUA? I write a module for this or do with script? script running has side effect on performance when load is high? Regards, H.Yavari Hello, Typically you process in OpenSIPS script an incoming request (and you fwd or reply it). It is unusual to generate a new request while processing another one. May I ask about your scenario ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 28.01.2014 10:02, H Yavari wrote: Hi to all openSIPSer, I want to initialize a sip message from routing, is it possible? If only way that do this is writing script, is this efficient when load on openSIPS is high? If answer is yes, can you give me some example to how do this? Regards, H.Yavari ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Initializing SIP messages from routing
Hi to all openSIPSer, I want to initialize a sip message from routing, is it possible? If only way that do this is writing script, is this efficient when load on openSIPS is high? If answer is yes, can you give me some example to how do this? Regards, H.Yavari ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
