Re: [OpenSIPS-Users] No Redirection to VM on Busy/Decline
This seems to happen due to a missing line - revert_uri(); the failure
route snippet needs to be:
if (t_check_status("487|408|486|480|603")){
revert_uri();
rewritehostport("VM.IP:5090");
t_relay();
exit;
On Thu, Jul 11, 2024 at 6:54 PM HS wrote:
> Hi all,
>
> Need some collective wisdom. I am using Opensips 3.0 and have the
> following snippet for failure_route:
>
> failure_route[missed_call] {
> if (t_was_cancelled()) {
> rtpengine_delete();
> exit;
> }
>
>
> # redirect the failed to a different VM system
> if (t_check_status("487|408|486|480|603")){
> rewritehostport("VM.IP:5090");
> t_relay();
> exit;
>
> # do not set the missed call flag again
> }
> }
>
> The call is redirected successfully to the VM if the user is not online.
> However, if the user is busy/declines/times out - the call is not
> redirected. I get the following error. I have searched quite a few answers,
> but don't seem to cover what I seek. Thanks for the help in advance.
>
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
> ERROR:proto_tls:tls_conn_init: no TLS client domain found
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:core:tcp_conn_new:
> failed to do proto 3 specific init for conn 0x7f6f38860760
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
> ERROR:proto_tls:tls_sync_connect: tcp_conn_create failed, closing the socket
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
> ERROR:proto_tls:proto_tls_send: connect failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:msg_send: send() to
> VM.IP:5090 for proto tls/3 failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:t_forward_nonack:
> sending request failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:w_t_relay:
> t_forward_nonack failed
>
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[OpenSIPS-Users] No Redirection to VM on Busy/Decline
Hi all,
Need some collective wisdom. I am using Opensips 3.0 and have the following
snippet for failure_route:
failure_route[missed_call] {
if (t_was_cancelled()) {
rtpengine_delete();
exit;
}
# redirect the failed to a different VM system
if (t_check_status("487|408|486|480|603")){
rewritehostport("VM.IP:5090");
t_relay();
exit;
# do not set the missed call flag again
}
}
The call is redirected successfully to the VM if the user is not online.
However, if the user is busy/declines/times out - the call is not
redirected. I get the following error. I have searched quite a few answers,
but don't seem to cover what I seek. Thanks for the help in advance.
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
ERROR:proto_tls:tls_conn_init: no TLS client domain found
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:core:tcp_conn_new:
failed to do proto 3 specific init for conn 0x7f6f38860760
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
ERROR:proto_tls:tls_sync_connect: tcp_conn_create failed, closing the socket
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
ERROR:proto_tls:proto_tls_send: connect failed
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:msg_send: send() to
VM.IP:5090 for proto tls/3 failed
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:t_forward_nonack:
sending request failed
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:w_t_relay:
t_forward_nonack failed
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[OpenSIPS-Users] Adding PSTN Gateway in Opensips 3.0
Hi all. Thanks a lot @Bogdan-Andrei Iancu , once you confirmed that it appends the correct domain, I had to play around with the dialplan and config and made it work. @Ronald - thanks. Hope it works for you too. Best. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Adding PSTN Gateway in Opensips 3.0
Dear all. Greetings and a Happy New Year. I want to add a PSTN gateway and the carrier expects something like: [PREFIX]phonenum...@sbc.anveo.com. I think I have figured out the [PREFIX] part, but can't seem to add "@sbc.anveo.com" via the dialplan module. There seems to be a validation error on the Opensips CP. According to some old documentation it used to be possible to add something like: sip:\\1...@sbc.anveo.com - but not any more. I have searched about Opensips/Kamailio, read Flavio's book. Is it possible to append the correct domain/URI/@sbc.whatever.com to a phone number dependant on the gateway used? Would this happen in the config? Dialplan? Dynamic Routing? Dispatcher module? I am sure this is really basic - references to any tutorials would be helpful. Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Push Notification - Location table has flag 0 v3.3
Hi again. Appreciate the offer to share the code - look forward to it. Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Push Notification - Location table has flag 0 v3.3
Hello sugar! Thanks a lot again. Most certainly would appreciate if you can share the code you used. Best of luck. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Push Notification - Location table has flag 0 v3.3
Hi again. Thanks a lot. I am trying to keep it as self-contained as possible too. Using the rest_client, with this project: https://github.com/sunaoka/push-notifications-php Unsure if I need to though :) Thanks again. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Push Notification - Location table has flag 0 v3.3
Hi Sugar. Major bow of gratitude! Thanks a lot. That's really helpful. I managed to test with Mizudroid and I get flag 4. I will find and make the relevant changes with regards to +sip.pnsreg. Another quick one, would you recall what you had implemented for the exec/rest_client (to actually send out the push?) pls? I have been using sylk-pushserver for Opensips v3.0. Best wishes once again. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Push Notification - Location table has flag 0 v3.3
Hi all.
I am starting to implement push notifications on Opensips v3.3, using an
old Linphone version that sends out parameters in the following format:
pn-type instead of pn-provider
firebase instead of fcm
app-id instead of pn-param
pn-tok instead of pn-prid
Location table has flag 0 (I think it should be 4). I tried adding pn-type,
app-id in pn_ct_match_params, but didn't seem to work (flag is still 0).
modparam("registrar", "pn_providers", "apns, fcm, firebase")
modparam("registrar", "pn_ct_match_params", "pn-provider, pn-prid,
pn-param, pn-type, pn-tok, app-id")
Is v3.3 incompatible with the old nomenclature or am I missing something?
Thanks.
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[OpenSIPS-Users] Push Notification Implementation on Opensips v3.3
Dear all, I have implemented push notifications on Opensips v3.0 with push-sylkserver and seems to work well. I am now working with v3.3 and used the following blog post: https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ Are there and best practices or tuts for implementation of the actual exec/rest_client? Anyone done this? Any tips please? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Mobile Phones and Multiple Invites
Hi all.
I see that mobile phone UACs don't unregister sometimes and multiple
invites are sent out. There's some discussion here:
https://github.com/OpenSIPS/opensips/issues/1316
And a snippet that mentions some code:
https://github.com/OpenSIPS/opensips/issues/1316#issuecomment-524024253
Has anyone tried it? Or is there some other preferred method pls? Also,
can't seem to make the following work:
modparam("usrloc", "save_iflag", "enable");
Save iflag?
Thanks
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Re: [OpenSIPS-Users] invalid contact wss
Nutxcase, With ref to invalid wss contact - can you confirm Opensips version please? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] LATE SDP Script on Opensips 3.3
Hi,
It seems I have fixed issue number 2. But issue 1 is still there (I call a
browser (Firefox or chrome), the call is rejected and I get the error:
"Failed to get local SDP".). Can someone please confirm if NAT/IP detection
works on Opensips 3.3.3? I have tested the exact same config and it works
fine on 3.0.5.
I have df7jal23ls0d.invalid in the SDP on Opensips 3.3.3.
Best wishes.
On Thu, Jan 26, 2023 at 10:44 PM HS wrote:
> Hi all,
>
> Here's is a link to the log of 2 calls. First part is a call that has
> one-way audio and the second is a call that has great bi-directional audio.
> https://pastebin.com/ggUFUmsp
>
> Appreciate if someone can identify if opensips config is ok pls, or
> suggestions on how to fix pls.
>
> On Sat, Jan 21, 2023 at 3:34 PM HS wrote:
>
>> Hi all.
>>
>> I am trying to setup an Opensips 3.3 instance on AWS and have rtpengine
>> installed. I am using a modified version of the script here:
>>
>> http://www.opensips.org/pub/docs/tutorials/websockets/opensips-late.cfg
>>
>> Two issues.
>> 1. When I call a browser (Firefox or chrome), the call is rejected and I
>> get the error: "Failed to get local SDP".
>> 2. One-way audio in specific circumstances - log below. Investigating
>> further I found that if I use "t_on_reply("handle_nat");" the calls go
>> through fine. However, if I use "t_on_branch("handle_nat");" I see
>> behaviour below.
>>
>> Thanks in advance.
>>
>> I am using 2 UA - Microsip and Linphone. When I call from Microsip
>> (Laptop) to Linphone (Android) I have great audio and the logs show this:
>>
>> Jan 20 16:11:06 ip-172-31-13-220 rtpengine[42387]: INFO:
>> [f75f1e374747457ca016e3a3a62ff4b3]: [control] Received command 'answer'
>> from 127.0.0.1:33919
>> Jan 20 16:11:06 ip-172-31-13-220 rtpengine[42387]: INFO:
>> [f75f1e374747457ca016e3a3a62ff4b3]: [control] Replying to 'answer' from
>> 127.0.0.1:33919 (elapsed time 0.000268 sec)
>> Jan 20 16:11:06 ip-172-31-13-220 /usr/sbin/opensips[79601]: incoming reply
>> Jan 20 16:11:07 ip-172-31-13-220 dhclient[451]: DHCPREQUEST for
>> 172.31.13.220 on ens5 to 172.31.0.1 port 67
>> Jan 20 16:11:07 ip-172-31-13-220 dhclient[451]: DHCPACK of 172.31.13.220
>> from 172.31.0.1
>> Jan 20 16:11:07 ip-172-31-13-220 dhclient[451]: bound to 172.31.13.220 --
>> renewal in 1607 seconds.
>> Jan 20 16:11:10 ip-172-31-13-220 rtpengine[42387]: INFO:
>> [f75f1e374747457ca016e3a3a62ff4b3 port 13215]: [core] Confirmed peer
>> address as My.Pub.IP.Add:4011
>> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
>> [f75f1e374747457ca016e3a3a62ff4b3]: [control] Received command 'delete'
>> from 127.0.0.1:33919
>> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
>> [f75f1e374747457ca016e3a3a62ff4b3]: [core] Scheduling deletion of call
>> branch 'bGUEC~8' (via-branch '') in 30 seconds
>> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
>> [f75f1e374747457ca016e3a3a62ff4b3]: [core] Scheduling deletion of call
>> branch '33a25bec39c742678ce74836e0ff40d1' (via-branch '') in 30 seconds
>> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
>> [f75f1e374747457ca016e3a3a62ff4b3]: [core] Scheduling deletion of entire
>> call in 30 seconds
>> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
>> [f75f1e374747457ca016e3a3a62ff4b3]: [control] Replying to 'delete' from
>> 127.0.0.1:33919 (elapsed time 0.000179 sec)
>>
>>
>> However, when I call from Linphone to Microsip, there's one-way audio and
>> the following in the logs:
>>
>> Jan 20 16:02:29 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
>> [control] Received command 'offer' from 127.0.0.1:40022
>> Jan 20 16:02:29 ip-172-31-13-220 rtpengine[42387]: NOTICE: [CjBB-HZqpe]:
>> [core] Creating new call
>> Jan 20 16:02:29 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
>> [control] Replying to 'offer' from 127.0.0.1:40022 (elapsed time
>> 0.000500 sec)
>> Jan 20 16:02:37 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
>> [control] Received command 'answer' from 127.0.0.1:40022
>> Jan 20 16:02:37 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
>> [control] Replying to 'answer' from 127.0.0.1:40022 (elapsed time
>> 0.000209 sec)
>> Jan 20 16:02:37 ip-172-31-13-220 /usr/sbin/opensips[79600]: incoming reply
>> Jan 20 16:02:38 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
>> [control] Received command 'answer' from 127.0.0.1:40022
>> Jan 20 16:02:38 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
>> [control] Repl
Re: [OpenSIPS-Users] LATE SDP Script on Opensips 3.3
Hi all,
Here's is a link to the log of 2 calls. First part is a call that has
one-way audio and the second is a call that has great bi-directional audio.
https://pastebin.com/ggUFUmsp
Appreciate if someone can identify if opensips config is ok pls, or
suggestions on how to fix pls.
On Sat, Jan 21, 2023 at 3:34 PM HS wrote:
> Hi all.
>
> I am trying to setup an Opensips 3.3 instance on AWS and have rtpengine
> installed. I am using a modified version of the script here:
>
> http://www.opensips.org/pub/docs/tutorials/websockets/opensips-late.cfg
>
> Two issues.
> 1. When I call a browser (Firefox or chrome), the call is rejected and I
> get the error: "Failed to get local SDP".
> 2. One-way audio in specific circumstances - log below. Investigating
> further I found that if I use "t_on_reply("handle_nat");" the calls go
> through fine. However, if I use "t_on_branch("handle_nat");" I see
> behaviour below.
>
> Thanks in advance.
>
> I am using 2 UA - Microsip and Linphone. When I call from Microsip
> (Laptop) to Linphone (Android) I have great audio and the logs show this:
>
> Jan 20 16:11:06 ip-172-31-13-220 rtpengine[42387]: INFO:
> [f75f1e374747457ca016e3a3a62ff4b3]: [control] Received command 'answer'
> from 127.0.0.1:33919
> Jan 20 16:11:06 ip-172-31-13-220 rtpengine[42387]: INFO:
> [f75f1e374747457ca016e3a3a62ff4b3]: [control] Replying to 'answer' from
> 127.0.0.1:33919 (elapsed time 0.000268 sec)
> Jan 20 16:11:06 ip-172-31-13-220 /usr/sbin/opensips[79601]: incoming reply
> Jan 20 16:11:07 ip-172-31-13-220 dhclient[451]: DHCPREQUEST for
> 172.31.13.220 on ens5 to 172.31.0.1 port 67
> Jan 20 16:11:07 ip-172-31-13-220 dhclient[451]: DHCPACK of 172.31.13.220
> from 172.31.0.1
> Jan 20 16:11:07 ip-172-31-13-220 dhclient[451]: bound to 172.31.13.220 --
> renewal in 1607 seconds.
> Jan 20 16:11:10 ip-172-31-13-220 rtpengine[42387]: INFO:
> [f75f1e374747457ca016e3a3a62ff4b3 port 13215]: [core] Confirmed peer
> address as My.Pub.IP.Add:4011
> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
> [f75f1e374747457ca016e3a3a62ff4b3]: [control] Received command 'delete'
> from 127.0.0.1:33919
> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
> [f75f1e374747457ca016e3a3a62ff4b3]: [core] Scheduling deletion of call
> branch 'bGUEC~8' (via-branch '') in 30 seconds
> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
> [f75f1e374747457ca016e3a3a62ff4b3]: [core] Scheduling deletion of call
> branch '33a25bec39c742678ce74836e0ff40d1' (via-branch '') in 30 seconds
> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
> [f75f1e374747457ca016e3a3a62ff4b3]: [core] Scheduling deletion of entire
> call in 30 seconds
> Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
> [f75f1e374747457ca016e3a3a62ff4b3]: [control] Replying to 'delete' from
> 127.0.0.1:33919 (elapsed time 0.000179 sec)
>
>
> However, when I call from Linphone to Microsip, there's one-way audio and
> the following in the logs:
>
> Jan 20 16:02:29 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
> [control] Received command 'offer' from 127.0.0.1:40022
> Jan 20 16:02:29 ip-172-31-13-220 rtpengine[42387]: NOTICE: [CjBB-HZqpe]:
> [core] Creating new call
> Jan 20 16:02:29 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
> [control] Replying to 'offer' from 127.0.0.1:40022 (elapsed time 0.000500
> sec)
> Jan 20 16:02:37 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
> [control] Received command 'answer' from 127.0.0.1:40022
> Jan 20 16:02:37 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
> [control] Replying to 'answer' from 127.0.0.1:40022 (elapsed time
> 0.000209 sec)
> Jan 20 16:02:37 ip-172-31-13-220 /usr/sbin/opensips[79600]: incoming reply
> Jan 20 16:02:38 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
> [control] Received command 'answer' from 127.0.0.1:40022
> Jan 20 16:02:38 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
> [control] Replying to 'answer' from 127.0.0.1:40022 (elapsed time
> 0.000178 sec)
> Jan 20 16:02:38 ip-172-31-13-220 /usr/sbin/opensips[79600]: incoming reply
> Jan 20 16:02:39 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
> [control] Received command 'answer' from 127.0.0.1:40022
> Jan 20 16:02:39 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
> [control] Replying to 'answer' from 127.0.0.1:40022 (elapsed time
> 0.000139 sec)
> Jan 20 16:02:39 ip-172-31-13-220 /usr/sbin/opensips[79600]: incoming reply
> Jan 20 16:02:44 ip-172-31-13-220 dhclient[524]: XMT: Solicit on ens5,
> interval 111260ms.
> Jan 20 16:02:45 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe port
> 13139]: [core] Confirmed peer address as My.Pub.IP.Add:
[OpenSIPS-Users] LATE SDP Script on Opensips 3.3
Hi all.
I am trying to setup an Opensips 3.3 instance on AWS and have rtpengine
installed. I am using a modified version of the script here:
http://www.opensips.org/pub/docs/tutorials/websockets/opensips-late.cfg
Two issues.
1. When I call a browser (Firefox or chrome), the call is rejected and I
get the error: "Failed to get local SDP".
2. One-way audio in specific circumstances - log below. Investigating
further I found that if I use "t_on_reply("handle_nat");" the calls go
through fine. However, if I use "t_on_branch("handle_nat");" I see
behaviour below.
Thanks in advance.
I am using 2 UA - Microsip and Linphone. When I call from Microsip (Laptop)
to Linphone (Android) I have great audio and the logs show this:
Jan 20 16:11:06 ip-172-31-13-220 rtpengine[42387]: INFO:
[f75f1e374747457ca016e3a3a62ff4b3]: [control] Received command 'answer'
from 127.0.0.1:33919
Jan 20 16:11:06 ip-172-31-13-220 rtpengine[42387]: INFO:
[f75f1e374747457ca016e3a3a62ff4b3]: [control] Replying to 'answer' from
127.0.0.1:33919 (elapsed time 0.000268 sec)
Jan 20 16:11:06 ip-172-31-13-220 /usr/sbin/opensips[79601]: incoming reply
Jan 20 16:11:07 ip-172-31-13-220 dhclient[451]: DHCPREQUEST for
172.31.13.220 on ens5 to 172.31.0.1 port 67
Jan 20 16:11:07 ip-172-31-13-220 dhclient[451]: DHCPACK of 172.31.13.220
from 172.31.0.1
Jan 20 16:11:07 ip-172-31-13-220 dhclient[451]: bound to 172.31.13.220 --
renewal in 1607 seconds.
Jan 20 16:11:10 ip-172-31-13-220 rtpengine[42387]: INFO:
[f75f1e374747457ca016e3a3a62ff4b3 port 13215]: [core] Confirmed peer
address as My.Pub.IP.Add:4011
Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
[f75f1e374747457ca016e3a3a62ff4b3]: [control] Received command 'delete'
from 127.0.0.1:33919
Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
[f75f1e374747457ca016e3a3a62ff4b3]: [core] Scheduling deletion of call
branch 'bGUEC~8' (via-branch '') in 30 seconds
Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
[f75f1e374747457ca016e3a3a62ff4b3]: [core] Scheduling deletion of call
branch '33a25bec39c742678ce74836e0ff40d1' (via-branch '') in 30 seconds
Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
[f75f1e374747457ca016e3a3a62ff4b3]: [core] Scheduling deletion of entire
call in 30 seconds
Jan 20 16:11:21 ip-172-31-13-220 rtpengine[42387]: INFO:
[f75f1e374747457ca016e3a3a62ff4b3]: [control] Replying to 'delete' from
127.0.0.1:33919 (elapsed time 0.000179 sec)
However, when I call from Linphone to Microsip, there's one-way audio and
the following in the logs:
Jan 20 16:02:29 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
[control] Received command 'offer' from 127.0.0.1:40022
Jan 20 16:02:29 ip-172-31-13-220 rtpengine[42387]: NOTICE: [CjBB-HZqpe]:
[core] Creating new call
Jan 20 16:02:29 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
[control] Replying to 'offer' from 127.0.0.1:40022 (elapsed time 0.000500
sec)
Jan 20 16:02:37 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
[control] Received command 'answer' from 127.0.0.1:40022
Jan 20 16:02:37 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
[control] Replying to 'answer' from 127.0.0.1:40022 (elapsed time 0.000209
sec)
Jan 20 16:02:37 ip-172-31-13-220 /usr/sbin/opensips[79600]: incoming reply
Jan 20 16:02:38 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
[control] Received command 'answer' from 127.0.0.1:40022
Jan 20 16:02:38 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
[control] Replying to 'answer' from 127.0.0.1:40022 (elapsed time 0.000178
sec)
Jan 20 16:02:38 ip-172-31-13-220 /usr/sbin/opensips[79600]: incoming reply
Jan 20 16:02:39 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
[control] Received command 'answer' from 127.0.0.1:40022
Jan 20 16:02:39 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe]:
[control] Replying to 'answer' from 127.0.0.1:40022 (elapsed time 0.000139
sec)
Jan 20 16:02:39 ip-172-31-13-220 /usr/sbin/opensips[79600]: incoming reply
Jan 20 16:02:44 ip-172-31-13-220 dhclient[524]: XMT: Solicit on ens5,
interval 111260ms.
Jan 20 16:02:45 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe port
13139]: [core] Confirmed peer address as My.Pub.IP.Add:4007
Jan 20 16:02:46 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe port
13146]: [core] Confirmed peer address as My.Pub.IP.Add:7078
Jan 20 16:02:46 ip-172-31-13-220 rtpengine[42387]: INFO: [CjBB-HZqpe port
13146]: [core] Kernelizing media stream: My.Pub.IP.Add:7078 ->
172.31.13.220:13146 | 172.31.13.220:13138 -> My.Pub.IP.Add:4006
Jan 20 16:02:51 ip-172-31-13-220 rtpengine[42387]: INFO:
[8fbde711f8864432ba7a37290d224a9a]: [core] Final packet stats:
Jan 20 16:02:51 ip-172-31-13-220 rtpengine[42387]: INFO:
[8fbde711f8864432ba7a37290d224a9a]: [core] --- Tag
'de22d68c1c7e4e759933a940e2f9eaa8', created 1:30 ago for branch ''
Jan 20 16:02:51 ip-172-31-13-220 rtpengine[42387]: INFO:
[8fbde711f8864432ba7a37290d224a9a]: [core] --- subscribed to 'PCjqSrP'
Jan 20 16:02:51 ip-172-31-13-220
Re: [OpenSIPS-Users] DBAliases with Late SDP Negotiation
Bogdan-Andrei.
Thanks again for the help. Yes, it doesn't transfer to VM on "timeout upon
ringing" or "temporary unavailable".
Best wishes.
On Thu, Aug 11, 2022 at 9:02 PM Bogdan-Andrei Iancu
wrote:
> Hi,
>
> By "in all other cases" you mean "timeout upon ringing" ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
> https://www.opensips.org/events/Summit-2022Athens/
>
> On 8/10/22 4:20 PM, HS wrote:
>
> Hi again.
>
> Thanks. Exact opposite :) When using an alias, after the call to users
> device is rejected, it is redirecting successfully to VM. However, in all
> other cases it doesn't redirect to VM (when using an alias) and times out.
> On the other hand, everything works as per snippet if using subscriber
> table details.
>
> Does that make sense please?
>
> Thanks.
>
> On Wed, Aug 10, 2022 at 5:25 PM Bogdan-Andrei Iancu
> wrote:
>
>> Hi,
>>
>> So, your issue is that, after the call to user's device is rejected, and
>> redirecting to VM (via failure route) you have some wrong RURI, right ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>> https://www.opensips-solutions.com
>> OpenSIPS Summit 27-30 Sept 2022, Athens
>> https://www.opensips.org/events/Summit-2022Athens/
>>
>> On 8/10/22 1:51 PM, HS wrote:
>>
>> Hi again.
>>
>> Thanks a lot for the quick response. My cfg is a bit of a trial-and-error
>> based, hence will email it (if that's ok). However, dbaliases support is
>> inserted as per default residential script:
>>
>> if ($rU==NULL) {
>> # request with no Username in RURI
>> send_reply(484,"Address Incomplete");
>> exit;
>> }
>>
>> # apply DB based aliases
>> alias_db_lookup("dbaliases");
>>
>> Thanks again.
>>
>> HS.
>>
>> On Wed, Aug 10, 2022 at 3:31 PM Bogdan-Andrei Iancu
>> wrote:
>>
>>> Hi,
>>>
>>> Could you share the cfg WITH the dbaliases support, to check where the
>>> alias lookup was inserted in the cfg logic ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>> https://www.opensips-solutions.com
>>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>> https://www.opensips.org/events/Summit-2022Athens/
>>>
>>> On 8/10/22 1:28 PM, HS wrote:
>>>
>>> Hi again Bogdan,
>>>
>>> Thanks for responding. I *tried* to add support for aliases. What is
>>> interesting is that the following snippet works fine on decline (but not
>>> otherwise):
>>>
>>> if (t_check_status("487|408|486|480|603")){
>>> rewritehostport("INT.IP.XX.XXX:5091");
>>> t_relay();
>>> exit;
>>>
>>> Any thoughts on how to fix please?
>>>
>>> Thanks.
>>>
>>> On Tue, Aug 9, 2022 at 2:45 PM Bogdan-Andrei Iancu
>>> wrote:
>>>
>>>> Hi HS,
>>>>
>>>> The aliases and late SDP negotiation has noting in common.
>>>>
>>>> And the mentioned script has no support for aliase - I guess you added
>>>> such support ?
>>>>
>>>> Regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>>
>>>> OpenSIPS Founder and Developer
>>>> https://www.opensips-solutions.com
>>>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>>> https://www.opensips.org/events/Summit-2022Athens/
>>>>
>>>> On 6/20/22 5:21 PM, HS wrote:
>>>>
>>>> Hi all.
>>>>
>>>> Just wondering if anyone has "dbaliases" working with the Late SDP
>>>> negotiation script below:
>>>>
>>>> https://opensips.org/pub/docs/tutorials/websockets/opensips-late.cfg
>>>>
>>>> If we call an alias, it works fine if the user is online. However, in
>>>> case of offline users I want the call to be forwarded to another
>>>> instance/server. If we use the ID (from subscriber table) calls are being
>>>> forwarded. But not if we dial an alias.
>>>>
>>>> Appreciate the help.
>>>>
>>>> ___
>>>> Users mailing
>>>> listUsers@lists.opensips.orghttp://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] DBAliases with Late SDP Negotiation
Hi again.
Thanks. Exact opposite :) When using an alias, after the call to users
device is rejected, it is redirecting successfully to VM. However, in all
other cases it doesn't redirect to VM (when using an alias) and times out.
On the other hand, everything works as per snippet if using subscriber
table details.
Does that make sense please?
Thanks.
On Wed, Aug 10, 2022 at 5:25 PM Bogdan-Andrei Iancu
wrote:
> Hi,
>
> So, your issue is that, after the call to user's device is rejected, and
> redirecting to VM (via failure route) you have some wrong RURI, right ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
> https://www.opensips.org/events/Summit-2022Athens/
>
> On 8/10/22 1:51 PM, HS wrote:
>
> Hi again.
>
> Thanks a lot for the quick response. My cfg is a bit of a trial-and-error
> based, hence will email it (if that's ok). However, dbaliases support is
> inserted as per default residential script:
>
> if ($rU==NULL) {
> # request with no Username in RURI
> send_reply(484,"Address Incomplete");
> exit;
> }
>
> # apply DB based aliases
> alias_db_lookup("dbaliases");
>
> Thanks again.
>
> HS.
>
> On Wed, Aug 10, 2022 at 3:31 PM Bogdan-Andrei Iancu
> wrote:
>
>> Hi,
>>
>> Could you share the cfg WITH the dbaliases support, to check where the
>> alias lookup was inserted in the cfg logic ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>> https://www.opensips-solutions.com
>> OpenSIPS Summit 27-30 Sept 2022, Athens
>> https://www.opensips.org/events/Summit-2022Athens/
>>
>> On 8/10/22 1:28 PM, HS wrote:
>>
>> Hi again Bogdan,
>>
>> Thanks for responding. I *tried* to add support for aliases. What is
>> interesting is that the following snippet works fine on decline (but not
>> otherwise):
>>
>> if (t_check_status("487|408|486|480|603")){
>> rewritehostport("INT.IP.XX.XXX:5091");
>> t_relay();
>> exit;
>>
>> Any thoughts on how to fix please?
>>
>> Thanks.
>>
>> On Tue, Aug 9, 2022 at 2:45 PM Bogdan-Andrei Iancu
>> wrote:
>>
>>> Hi HS,
>>>
>>> The aliases and late SDP negotiation has noting in common.
>>>
>>> And the mentioned script has no support for aliase - I guess you added
>>> such support ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>> https://www.opensips-solutions.com
>>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>> https://www.opensips.org/events/Summit-2022Athens/
>>>
>>> On 6/20/22 5:21 PM, HS wrote:
>>>
>>> Hi all.
>>>
>>> Just wondering if anyone has "dbaliases" working with the Late SDP
>>> negotiation script below:
>>>
>>> https://opensips.org/pub/docs/tutorials/websockets/opensips-late.cfg
>>>
>>> If we call an alias, it works fine if the user is online. However, in
>>> case of offline users I want the call to be forwarded to another
>>> instance/server. If we use the ID (from subscriber table) calls are being
>>> forwarded. But not if we dial an alias.
>>>
>>> Appreciate the help.
>>>
>>> ___
>>> Users mailing
>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>
___
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Users@lists.opensips.org
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Re: [OpenSIPS-Users] DBAliases with Late SDP Negotiation
Hi again.
Thanks a lot for the quick response. My cfg is a bit of a trial-and-error
based, hence will email it (if that's ok). However, dbaliases support is
inserted as per default residential script:
if ($rU==NULL) {
# request with no Username in RURI
send_reply(484,"Address Incomplete");
exit;
}
# apply DB based aliases
alias_db_lookup("dbaliases");
Thanks again.
HS.
On Wed, Aug 10, 2022 at 3:31 PM Bogdan-Andrei Iancu
wrote:
> Hi,
>
> Could you share the cfg WITH the dbaliases support, to check where the
> alias lookup was inserted in the cfg logic ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
> https://www.opensips.org/events/Summit-2022Athens/
>
> On 8/10/22 1:28 PM, HS wrote:
>
> Hi again Bogdan,
>
> Thanks for responding. I *tried* to add support for aliases. What is
> interesting is that the following snippet works fine on decline (but not
> otherwise):
>
> if (t_check_status("487|408|486|480|603")){
> rewritehostport("INT.IP.XX.XXX:5091");
> t_relay();
> exit;
>
> Any thoughts on how to fix please?
>
> Thanks.
>
> On Tue, Aug 9, 2022 at 2:45 PM Bogdan-Andrei Iancu
> wrote:
>
>> Hi HS,
>>
>> The aliases and late SDP negotiation has noting in common.
>>
>> And the mentioned script has no support for aliase - I guess you added
>> such support ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>> https://www.opensips-solutions.com
>> OpenSIPS Summit 27-30 Sept 2022, Athens
>> https://www.opensips.org/events/Summit-2022Athens/
>>
>> On 6/20/22 5:21 PM, HS wrote:
>>
>> Hi all.
>>
>> Just wondering if anyone has "dbaliases" working with the Late SDP
>> negotiation script below:
>>
>> https://opensips.org/pub/docs/tutorials/websockets/opensips-late.cfg
>>
>> If we call an alias, it works fine if the user is online. However, in
>> case of offline users I want the call to be forwarded to another
>> instance/server. If we use the ID (from subscriber table) calls are being
>> forwarded. But not if we dial an alias.
>>
>> Appreciate the help.
>>
>> ___
>> Users mailing
>> listUsers@lists.opensips.orghttp://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] DBAliases with Late SDP Negotiation
Hi again Bogdan,
Thanks for responding. I *tried* to add support for aliases. What is
interesting is that the following snippet works fine on decline (but not
otherwise):
if (t_check_status("487|408|486|480|603")){
rewritehostport("INT.IP.XX.XXX:5091");
t_relay();
exit;
Any thoughts on how to fix please?
Thanks.
On Tue, Aug 9, 2022 at 2:45 PM Bogdan-Andrei Iancu
wrote:
> Hi HS,
>
> The aliases and late SDP negotiation has noting in common.
>
> And the mentioned script has no support for aliase - I guess you added
> such support ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
> https://www.opensips.org/events/Summit-2022Athens/
>
> On 6/20/22 5:21 PM, HS wrote:
>
> Hi all.
>
> Just wondering if anyone has "dbaliases" working with the Late SDP
> negotiation script below:
>
> https://opensips.org/pub/docs/tutorials/websockets/opensips-late.cfg
>
> If we call an alias, it works fine if the user is online. However, in case
> of offline users I want the call to be forwarded to another
> instance/server. If we use the ID (from subscriber table) calls are being
> forwarded. But not if we dial an alias.
>
> Appreciate the help.
>
> ___
> Users mailing
> listUsers@lists.opensips.orghttp://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] DBAliases with Late SDP Negotiation
Hi all. Just wondering if anyone has "dbaliases" working with the Late SDP negotiation script below: https://opensips.org/pub/docs/tutorials/websockets/opensips-late.cfg If we call an alias, it works fine if the user is online. However, in case of offline users I want the call to be forwarded to another instance/server. If we use the ID (from subscriber table) calls are being forwarded. But not if we dial an alias. Appreciate the help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No audio on TLS connection - Opensips 3.1
Hi again. Appreciate if someone could take a look at the log and confirm if its an Opensips bug or UA issue please. Subsequent calls have been possible, but intermittent. Thanks. On Sun, Jun 5, 2022 at 7:04 PM HS wrote: > Hi again. > > I discovered something interesting with regards to the TLS connection with > a specific UA when it connects via TLS. In our scenario we forward a call > via Freeswitch (on another server) to another call centre provider. The UA > works fine on UDP + TCP mode. However, on TLS we do not get to hear audio > from the other end. (It seems that Opensips sends out a local address). > > Interestingly, when I restart Opensips (3.1) and connect within a minute > or so, I DO get to hear audio. But subsequent calls do NOT have any audio. > This is really beyond my understanding. Is this an Opensips issue or user > agent? Following is a link to level 7 log - appreciate if someone can help. > There are two calls - the first one has audio, the second one doesn't. > > https://pastebin.pl/view/abc908c1 > > Much appreciate the help. > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No audio on TLS connection - Opensips 3.1
Hi again. I discovered something interesting with regards to the TLS connection with a specific UA when it connects via TLS. In our scenario we forward a call via Freeswitch (on another server) to another call centre provider. The UA works fine on UDP + TCP mode. However, on TLS we do not get to hear audio from the other end. (It seems that Opensips sends out a local address). Interestingly, when I restart Opensips (3.1) and connect within a minute or so, I DO get to hear audio. But subsequent calls do NOT have any audio. This is really beyond my understanding. Is this an Opensips issue or user agent? Following is a link to level 7 log - appreciate if someone can help. There are two calls - the first one has audio, the second one doesn't. https://pastebin.pl/view/abc908c1 Much appreciate the help. On Thu, May 26, 2022 at 1:04 PM HS wrote: > Hi all. > > Just testing out Opensips 3.1 and run into a confusing issue. Users can > connect and make calls successfully on UDP and TCP. However, some users are > facing a problem with TLS - they connect fine, but don't hear any audio. > > Looking at the logs it seems that a private IP is registered. We have > tried nat_uac_test 23 and 119 (REGISTER) and 1 and 2 (ONReply). We are > using the standard residential script. Any thoughts on what else to test? > > Thanks for the help. > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] No audio on TLS connection - Opensips 3.1
Hi all. Just testing out Opensips 3.1 and run into a confusing issue. Users can connect and make calls successfully on UDP and TCP. However, some users are facing a problem with TLS - they connect fine, but don't hear any audio. Looking at the logs it seems that a private IP is registered. We have tried nat_uac_test 23 and 119 (REGISTER) and 1 and 2 (ONReply). We are using the standard residential script. Any thoughts on what else to test? Thanks for the help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Billing And Invoicing for Opensips 3.0
Thanks Alberto. Just looked at CDRTool - don't think it has invoicing. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Billing And Invoicing for Opensips 3.0
Hi all. I am looking to add billing + invoicing to the setup. Came across CGrates for rate cards etc, but is there a billing + invoicing setup that works great with Opensips CDRs? I have looked at astpp so far and searched quite a bit, but nothing much available. Any suggestions? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] WebRTC Implementation on Opensips 3.0
Callum. Thanks a lot for the help. I seem to have been able to compile the module and add it. Now working on port 443 on apache2 + Opensips. The fun never ends :D ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] WebRTC Implementation on Opensips 3.0
Thanks a lot Callum. Haven't compiled a module ever (yet) and upgrade scares me :) Anyone have a copy that I can use please or any other suggestions? Or would the module have changed significantly - what if I use the 3.1 build version? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] WebRTC Implementation on Opensips 3.0
Apologies, I have already tried: apt-get install opensips-wss-module ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] WebRTC Implementation on Opensips 3.0
Callum. Thanks for the quick response. I have the other two modules installed, I think this is missing: opensips-wss-module How does one get this module? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] WebRTC Implementation on Opensips 3.0
Dear all. Trying to following the tutorials, however, when I restart opensips after adding proto_wss. to the config file - I get an error message saying the module can't be loaded and it isn't in the modules folder. I recall having downloaded a module previously, but have no recollection of it. Any pointers please? Thanks again. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 3.0 + Letsencrypt + TLS issues
Dear all.
Been trying to make TLS work for our setup. With the following
configuration:
loadmodule "tls_mgm.so"
modparam("tls_mgm", "server_domain", "ourdomain.com")
modparam("tls_mgm","verify_cert", "[ourdomain.com]0")
modparam("tls_mgm","require_cert", "[ourdomain.com]0")
modparam("tls_mgm","tls_method", "[ourdomain.com]TLSv1")
modparam("tls_mgm", "certificate", "[ourdomain.com
]/etc/opensips/tls/tls_cnf/fullchain.pem")
modparam("tls_mgm", "private_key", "[ourdomain.com
]/etc/opensips/tls/tls_cnf/privkey.pem")
Things work fine if I use the default/testing file path (rootCA), however,
when I change to using the private key from letsencrypt I get the following
error:
Sep 21 10:39:50 ip-172-31-27-223 /usr/sbin/opensips[3240]:
ERROR:tls_mgm:load_private_key: unable to load private key file
'/etc/opensips/tls/tls_cnf/privkey.pem'. #012Retry (0 left) (check password
case)
Sep 21 10:39:50 ip-172-31-27-223 /usr/sbin/opensips[3240]:
ERROR:tls_mgm:load_private_key: unable to load private key file
'/etc/opensips/tls/tls_cnf/privkey.pem'
Sep 21 10:39:50 ip-172-31-27-223 /usr/sbin/opensips[3240]:
ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'ourdomain.com'
Sep 21 10:39:50 ip-172-31-27-223 /usr/sbin/opensips[3240]:
ERROR:core:init_mod: failed to initialize module tls_mgm
I have tried removing the password (
https://github.com/OpenSIPS/opensips/issues/987), but letsencrypt keys
don't have a password - it didn't work.
Any thoughts please?
Many thanks.
___
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[OpenSIPS-Users] From header from dbaliases
Dear all, Hope everyone is doing great. Had a quick query (I hope). Our setup on Opensips 3.0 involves an internal username (subscriber table) and an external (dbaliases). I would like to ensure that the from header is always based on the information (alias_username + alias_domain). In case of mismatch, either the calls don't go through or the server always uses "From" information from the dbaliases table. Which one might be better and what would be the best way to do it please? Thanks in advance. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SQL Query Variables
Hi Kingsley. Thanks a lot for your comment. I realised that the following was missing before the query: $var(user) = $rU; $var(domain) = $rd; It seems to be working for usernames now. Probably been the toughest bit of self-learning. Next step, need to figure out how to make this work for aliases. Thanks again. > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] SQL Query Variables
Hi all.
In my quest to make the push server work on my Opensips 3.0 installation. I
have figured out that the following query isn't working correctly:
$avp(query) = "SELECT token, app, platform, sip_instance from push_tokens
WHERE username='" + $(var(user){s.escape.common}) + "' AND domain='" +
$(var(domain){s.escape.common}) + "'";
When I replace $(var(user...)/$var(domain...) with $(tU..) and $(td..), the
query seems to work. I have tried multiple combinations, but the
original query isn't working.
What do I need to change to make the original query work? I did find
details of what tU is, but nothing about var(user).
Thanks for the help.
___
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[OpenSIPS-Users] Script variable for dbaliases
Gents. My setup works on dbaliases as well as usernames. I understand that the variable for usernames is du and/or ru. Is there something comparable for dbaliases. I looked here and couldn't find anything: https://www.opensips.org/Documentation/Script-CoreVar-3-0 Any suggestions please? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Calling aliases + usernames
Hello all. I have been moderately successful in getting AG's push notifications ( https://github.com/AGProjects/sylk-pushserver) to work. Quick-ish question. Our setup works on aliases (external) as well as usernames (internal). The normal opensips.cfg file works great to route calls in both scenarios. However, the push server code sends calls specifically to the usernames and completely ignores the aliases. I was thinking we do this via the database (have 2 sets of information in the push_tokens table) or the config file. >From the little that I know about opensips, db calls are not the preferred way - correct? Any tips on how to modify the push server code to route calls to both aliases + username? Hope not too confusing - thanks for the help. Best wishes. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Header difference between "if(is_method("INVITE"))" vs "if(!is_method("INVITE"))"
Hi all,
I had setup an Opensips (3.0) instance (as Registrar) on an Amazon EC2
instance and freeswitch (media server) on another instance. I had tested it
thoroughly and it seems to work great on the following code (Opensips.cfg):
if (!is_method("INVITE")) {
return;
}
# if the called number begins with "star" (*) then strip it and redirect to
freeswitch
# (if it begins with two stars, eg: **, then one will be passed to FS)
if ($rU=~"^\*") {
strip(1);
$du = "sip:172.31.23.7:5090";
route(relay);
}
}
I dial the IVR - everything works ok for MicroSIP (laptop) and Linphone
(Android phone).
However, in the following snippet - MicroSIP (call disconnects immediately)
and Linphone has no audio (disconnects after 32 seconds). Pcap shows no
connection on MicroSIP but on Linphone I can play the audio stream.
if(is_method("INVITE")) {
if (t_check_status("408|486")) {
$du = "sip:172.31.23.7:5090";
# do not set the missed call flag again
route(relay);
}
Is there a difference in the headers/routing sent out in
"if(is_method("INVITE"))" vs "if(!is_method("INVITE"))". And how to fix
please?
Please let me know if any other information is required.
Thanks again.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DTMF/Freeswitch Integration with Opensips 3.0
Liviu, Thanks a lot. Much appreciate the help, I will try both options. I have setup the IVR on freeswitch, but curious about something. In the Opensips Call Center module, one adds the agent/support contact to a "skill" for example. And an entry in the opensips config to route to that agent. How does it work in the Opensips + Freeswitch scenario? I didn't see any entry about the support number being called - or have I misunderstood? Thanks again. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Push notification implementation Opensips 3.0
Liviu, Thanks again. You are looking at the fixed version. This is the ticket I had raised: https://github.com/AGProjects/sylk-pushserver/issues/3 Don't think anyone who had used the repo could have missed it :) Best wishes. On Wed, Jun 23, 2021 at 10:09 AM Liviu Chircu wrote: > May I ask what sample config you used? Because I'm looking at > AGProjects/sylk-pushserver [1] and the query seems correct, you just > have to scroll to the right a bit to correctly grab all of it. Or just > download the entire file. > > [1]: > > https://github.com/AGProjects/sylk-pushserver/blob/ff41d72fa9c353e6e07296867dd881f92904b403/config/opensips.cfg#L83 > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > OpenSIPS Summit 2021 Distributed | www.opensips.org/events > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Push notification implementation Opensips 3.0
Liviu, Thanks a lot for taking the time to respond. Correct, the queries were unterminated in the sample config on github. They are correct now. However, considering no one had pointed out in almost a year - makes me feel there's some other methodology used for push notifications by the community. Would be great to know any best practices. Best wishes. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DTMF/Freeswitch Integration with Opensips 3.0
Dear Liviu, Thanks a lot, I am honored that you took the time to respond. However, I am actually a business guy (no tech background) trying to make the script work. I didn't want to reply on the list - for fear of completely embarrassing myself. However, in the spirit of entertaining you all, I will confess, I still have no clue what to do next :D > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DTMF/Freeswitch Integration with Opensips 3.0
Hello all, I have figured out the changes required for Opensips 3.0 from the How-To here: https://blog.opensips.org/2018/01/17/how-to-script-advanced-freeswitch-integrations-with-opensips-2-4/ However, the following is a problem area: if (do_routing(0, "$var(flags)")) return(1); This gives the error: script function "do_routing" (types=43) does not support route type (256) Essentially do_routing is possible only in the REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE and LOCAL_ROUTE. However, the route is named [goes_to_support]. What do I need to change here please? Thanks. > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Push notification implementation Opensips 3.0
Hello all
Just came across this post:
https://opensips.org/pipermail/users/2020-June/043294.html
I am trying to implement this on Opensips 3.0/Amazon EC2, just wondering if
anyone has done this already? Clearly I am not a dev and getting the
following errors:
Jun 11 11:40:47 [15463] Traceback (last included file at the bottom):
Jun 11 11:40:47 [15463] 0. /etc/opensips/opensips.cfg
Jun 11 11:40:47 [15463] CRITICAL:core:yyerror: parse error in
/etc/opensips/opensips.cfg:331:2-7: syntax error
Jun 11 11:40:47 [15463] # save data required later for push
notifications
Jun 11 11:40:47 [15463] if ($avp(pn_type) and $avp(pn_token) and
$avp(pn_app)) {
Jun 11 11:40:47 [15463] $avp(query) = "SELECT token from
push_tokens where username = '" + $(tU{s.escape.common}) + "' and domain =
'" + $(td{s.escape.common}) + "' and app
Jun 11 11:40:47 [15463] ^~
Jun 11 11:40:47 [15463] xlog("L_DBG", "[CONFIG] REGISTER push
SQL query: $avp(query)\n");
Jun 11 11:40:47 [15463] avp_db_query($avp(query),
"$avp(old_pn_token)");
Jun 11 11:40:47 [15463] CRITICAL:core:yyerror: parse error in
/etc/opensips/opensips.cfg:331:2-7: bad command!)
Jun 11 11:40:47 [15463] CRITICAL:core:yyerror: parse error in
/etc/opensips/opensips.cfg:331:8-9: bare word found, command calls
need '()'
Jun 11 11:40:47 [15463] CRITICAL:core:yyerror: parse error in
/etc/opensips/opensips.cfg:331:8-9: bad command: missing ';'?
Thanks.
___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] DTMF/Freeswitch Integration with Opensips 3.0
Hi again. Was working on DTMF + Freeswitch Integration with Opensips 3.0 and found the following How-to: https://blog.opensips.org/2018/01/17/how-to-script-advanced-freeswitch-integrations-with-opensips-2-4/ Just wondering if anyone had implemented this on Opensips 3.0/Amazon EC2 - as there are changes in the .cfg file syntax. Thanks a lot. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Config script - Opensips + RTPProxy + Freeswitch on EC2
This made me laugh (Try-and-cry)! However, agree completely with Tomi. A lot of the times, people don't respond here - but typing stuff out helps loads. So calls are going through to the IVR and conference. But having the usual issues: 1. IVR/calls to Freeswitch disconnect after 30s. 2. Unsure if users are being pulled from the database or the profiles. 3. DTMF input isn't working (system doesn't respond to input.) So far so good :) > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Config script - Opensips + RTPProxy + Freeswitch on EC2
Hi Alexey. Thanks a lot for taking the time to respond. Incidentally, I had read the book a few months ago and took a look at it again yesterday. It has good details on scripting and integrating with Asterisk. Will have to do some trial-and-error, as usual. Thanks again. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Config script - Opensips + RTPProxy + Freeswitch on EC2
Hi all. I keep writing from time to time asking for help as a newbie to Opensips. Somehow trial and error makes things work. I wanted to know if there's a book/tutorial available to understand how the .cfg file and script works and all the variables involved. Specifically trying to do the following on an Amazon EC2 setup: 1. Call centre implementation (I have the modules etc. setup), 2. Enable IVR, 3. Enable Conferencing, 4. Enable Voicemail, 5. Enable Push Notifications. I understand a lot of it has to do with Freeswitch (I have read the old tutorial) - and would like to understand how the routing aspect would work. Thanks again. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications
Bogdan-Andrei, Thanks a lot for the help. Much appreciated. Best wishes. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications
Bogdan-Andrei, Thanks a lot for sharing. Appreciate the help. Do we need to make any changes for the script to work with iOS? Apologies, but really difficult to find help with Opensips where I am located. Much appreciate your time and help. Best wishes. On Thu, Apr 15, 2021 at 11:01 AM Bogdan-Andrei Iancu wrote: > The last link is for OpenSIPS 2.3 and higher. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/14/21 7:33 PM, HS wrote: > > Bogdan-Andrei, > > Thanks a lot for sharing these links. I had actually looked at them > previously, all are for Opensips 3.1 - any possibility of getting PN to > work with Openips 3.0 please? > > Thanks. > > On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu > wrote: > >> See >> >> >> https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ >> >> >> https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ >> >> >> https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp 2021 online >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 4/9/21 12:39 PM, HS wrote: >> >> Hello all. >> >> We are using Opensips 3.0 (with rtpproxy) and been looking to implement >> Push Notifications for iOS devices. Did come across quite a few resources, >> but they seem either too old ~2013 time or about V3.1. Has anyone >> implemented something to make calls to devices with the UAC in background >> mode on V3.0 pls? >> >> Or suggestions on how to make this work on v3.0? >> >> http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ >> >> Appreciate the help. >> >> Thanks. >> >> ___ >> Users mailing >> listUsers@lists.opensips.orghttp://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] Opensips 3.0 + iOS Push Notifications
Bogdan-Andrei, Thanks a lot for sharing these links. I had actually looked at them previously, all are for Opensips 3.1 - any possibility of getting PN to work with Openips 3.0 please? Thanks. On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu wrote: > See > > > https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ > > > https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/ > > > https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/ > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2021 online > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 4/9/21 12:39 PM, HS wrote: > > Hello all. > > We are using Opensips 3.0 (with rtpproxy) and been looking to implement > Push Notifications for iOS devices. Did come across quite a few resources, > but they seem either too old ~2013 time or about V3.1. Has anyone > implemented something to make calls to devices with the UAC in background > mode on V3.0 pls? > > Or suggestions on how to make this work on v3.0? > > http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ > > Appreciate the help. > > Thanks. > > ___ > Users mailing > listUsers@lists.opensips.orghttp://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] Opensips 3.0 + iOS Push Notifications
Hello all. We are using Opensips 3.0 (with rtpproxy) and been looking to implement Push Notifications for iOS devices. Did come across quite a few resources, but they seem either too old ~2013 time or about V3.1. Has anyone implemented something to make calls to devices with the UAC in background mode on V3.0 pls? Or suggestions on how to make this work on v3.0? http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/ Appreciate the help. Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips+SIPREC on Amazon EC2/S3
Hi Răzvan, Thanks for the note. We tried quite a bit and don't think this worked out. I think there's quite a few IP/NAT related issues that we couldn't fix. Would be great if anyone who has tried doing this could help. Do we need to change anything on rtpproxy config also? Thanks again. On Tue, Mar 2, 2021 at 11:08 AM HS wrote: > Hello all! > > I came across this Opensips blog ( > https://blog.opensips.org/2017/11/08/call-recording-in-opensips-2-4-using-siprec/). > To clarify, we are using Opensips (3.0) + Rtpproxy. Just want to confirm: > > 1. Would it work on a cloud based setup (AWS)? I was thinking one EC2 > instance running opensips+rtpproxy+orkaudio. Another instance running > orktrack+orkweb? > 2. We could use S3 to store backups only? > > Anyone tried it? Watchouts? > > Thx a lot. > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Invalid SDP Descriptor
Hi all. I am using Opensips (3.0) on Amazon EC2 and have implemented RTPProxy and it seems to do NAT Traversal fine. I am now trying to setup Active SIPREC recording and get the following error when I try and make calls: Bad SDP 399.. "Invalid SDP Descriptor (PJMEDIA_SDP_EINSDP). I think there's something wrong with my Opensips config - any suggestions please? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to install uac_auth module
Nevermind, fixed! On Thu, Mar 4, 2021 at 4:36 PM HS wrote: > Hi all. > > Trying to implement SIPREC on the cloud. From the blog here: > > After restarting Opensips (3.0), getting the following error: > > module b2b_entities depends on module uac_auth, but it was not loaded! > > I have tried multiple combos, with apt install, but no luck, anyone know > how to install uac_auth please? > > Thanks, > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to install uac_auth module
Hi all. Trying to implement SIPREC on the cloud. From the blog here: After restarting Opensips (3.0), getting the following error: module b2b_entities depends on module uac_auth, but it was not loaded! I have tried multiple combos, with apt install, but no luck, anyone know how to install uac_auth please? Thanks, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips+SIPREC on Amazon EC2/S3
Hello all! I came across this Opensips blog ( https://blog.opensips.org/2017/11/08/call-recording-in-opensips-2-4-using-siprec/). To clarify, we are using Opensips (3.0) + Rtpproxy. Just want to confirm: 1. Would it work on a cloud based setup (AWS)? I was thinking one EC2 instance running opensips+rtpproxy+orkaudio. Another instance running orktrack+orkweb? 2. We could use S3 to store backups only? Anyone tried it? Watchouts? Thx a lot. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips.cfg on Amazon EC2 - no audio issues
Dear all.
We have a fairly basic setup on Opensips 3.0 on Amazon EC2. We allow users
to make and receive voice calls only. We are not using any proxy or STUN as
all our calls route completely fine within our country. However, following
are some issues:
1. When we make calls outside our country, there is no audio (both ways).
2. When our users try and make calls between themselves (outside our
country, the calls don't go through at all).
We have tried multiple configurations that we found on github and personal
blogs, however, can't seem to solve this issue - and unable to find someone
who can.
Following is our opensips.cfg file, appreciate if someone can clarify what
is wrong:
#
# OpenSIPS residential configuration script
# by OpenSIPS Solutions
#
# This script was generated via "make menuconfig", from
# the "Residential" scenario.
# You can enable / disable more features / functionalities by
# re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
# https://opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#
### Global Parameters #
log_level=3
log_stderror=no
#log_facility=LOG_LOCAL0
log_facility=LOG_LOCAL7
udp_workers=8
/* uncomment the following lines to enable debugging */
#debug_mode=yes
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes
/* comment the next line to enable the auto discovery of local aliases
based on reverse DNS on IPs */
auto_aliases=no
advertised_address=XX.XXX.XXX.XX
listen=udp:XXX.XX.XX.XXX:5060 # CUSTOMIZE ME
listen=tcp:XXX.XX.XX.XXX:5060 # CUSTOMIZE ME
### Modules Section
#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"
SIGNALING module
loadmodule "signaling.so"
StateLess module
loadmodule "sl.so"
Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)
MAX ForWarD module
loadmodule "maxfwd.so"
SIP MSG OPerationS module
loadmodule "sipmsgops.so"
FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)
MYSQL module
loadmodule "db_mysql.so"
HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", )
USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
modparam("registrar", "received_avp", "$avp(received_nh)")/* uncomment the
next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", 0)
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")
ALIAS module
loadmodule "alias_db.so"
modparam("alias_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
DOMAIN module
loadmodule "domain.so"
modparam("domain", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("domain", "db_mode", 0) # Use caching
modparam("auth_db|usrloc", "use_domain", 1)
PRESENCE modules
loadmodule "xcap.so"
loadmodule "presence.so"
loadmodule "presence_xml.so"
modparam("xcap|presence", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("presence_xml", "force_active", 1)
modparam("presence", "fallback2db", 0)
DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600) # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
NAT modules
loadmodule
[OpenSIPS-Users] Opensips with RTPEngine
Hi all.
I just tried to replace RTPProxy with RTPEngine. I did the change as I was
getting one-way audio previously. However, one connection seems
inconsistent (One-way audio) sometimes and the other connection does not
actually even ring a bell. I have tried with both devices on the same Wi-Fi
and one on Wi-Fi and the other on a cellular network. There's no error in
opensips or rtpengine logs, so it seems that the routing is incorrect. I am
using Opensips 3.0. Please let me know what is incorrect or if further
information is needed. Following is my opensips.cfg file:
Thanks for the help in advance.
#
# OpenSIPS residential configuration script
# by OpenSIPS Solutions
#
# This script was generated via "make menuconfig", from
# the "Residential" scenario.
# You can enable / disable more features / functionalities by
# re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
# https://opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#
### Global Parameters #
log_level=3
log_stderror=no
#log_facility=LOG_LOCAL0
log_facility=LOG_LOCAL7
udp_workers=8
/* uncomment the following lines to enable debugging */
#debug_mode=yes
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes
/* comment the next line to enable the auto discovery of local aliases
based on reverse DNS on IPs */
auto_aliases=no
advertised_address=
listen= # CUSTOMIZE ME
listen= # CUSTOMIZE ME
# Set up listeners
#listen=ws:
#listen=wss:
#listen=tls:
### Modules Section
#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"
SIGNALING module
loadmodule "signaling.so"
StateLess module
loadmodule "sl.so"
Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)
MAX ForWarD module
loadmodule "maxfwd.so"
SIP MSG OPerationS module
loadmodule "sipmsgops.so"
FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)
MYSQL module
loadmodule "db_mysql.so"
HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", )
USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
modparam("registrar", "received_avp", "$avp(received_nh)")/* uncomment the
next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", 0)
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")
ALIAS module
loadmodule "alias_db.so"
modparam("alias_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
DOMAIN module
loadmodule "domain.so"
modparam("domain", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("domain", "db_mode", 0) # Use caching
modparam("auth_db|usrloc", "use_domain", 1)
PRESENCE modules
loadmodule "xcap.so"
loadmodule "presence.so"
loadmodule "presence_xml.so"
modparam("xcap|presence", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("presence_xml", "force_active", 1)
modparam("presence", "fallback2db", 0)
DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600) # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
NAT modules
loadmodule "nathelper.so"
modparam("registrar|nathelper", "received_avp", "$avp(rcv)")
[OpenSIPS-Users] Authenticate with email_address
Hi all. Still struggling with this. We want to manage users via dbaliases so that one subscriber can have multiple aliases. We have an internal username in the subscriber table. We authenticate with the internal username, the call goes through with the internal username. Is there anyway to specify: 1. Caller-ID to be the actual email_address (added to subscriber table), 2. Authenticate with dbaliases? Any other thoughts please? Appreciate the help. Best wishes. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Authenticate with dbaliases or change Caller ID
Hello all. I am trying to either: 1. Authenticate with alias_name+alias_domain against password in subscribers. 2. If above is not possible, be able to change the Caller ID (I have tried changing RPID - and that did not work). Any suggestions? Thx. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips on Amazon EC2
Hi all. First time poster here, so apologies if something is incorrect. I am trying to setup a small production SIP server. I have read quite a bit but just want to be sure that I have done everything correctly. I followed the following guide: http://www.powerpbx.org/content/opensips-v30-debian-v10-mariadb-apache-v1 I can make and receive calls, everything is working correctly. I have deployed on Amazon EC2, thinking I need to do the following for voice calls only: 1. Setup TURN/STUN server (or not needed any more) 2. TLS Certificates. 3. fail2ban. 4. Db replication. 5. Setup IPtables. 6. Do I need an RTP Proxy yet? Thanks for the help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Users Digest, Vol 146, Issue 5
Anything else? I changed the database_url in the config file - but that
didn't make a difference.
> Message: 2
> Date: Fri, 4 Sep 2020 10:42:58 +0300
> From: Răzvan Crainea
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] How to change default password -
> opensipsrw
> Message-ID: <03974fcb-8306-973a-167c-d2af9fc88...@opensips.org>
> Content-Type: text/plain; charset=utf-8; format=flowed
>
> Make sure you specify a different `database_url` in your configuration
> file[1].
>
> [1]
>
> https://github.com/OpenSIPS/opensips-cli/blob/master/opensipscli/defaults.py#L70
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 9/2/20 11:52 AM, HS wrote:
> > Hi there.
> >
> > I am using version 3.1 of opensips. I tried changing the database
> > password from the config files, however, it does not work.
> >
> > Can someone please share a list of locations of where the passwords
> > (incl. UI) needs to be changed? I am on a debian machine.
> >
> > Thanks.
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
>
> --
>
> Message: 3
> Date: Fri, 4 Sep 2020 10:44:39 +0300
> From: Răzvan Crainea
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] Dispatcher algorithm 9
> Message-ID: <86147906-cd9f-6a0e-4fa0-11c85696a...@opensips.org>
> Content-Type: text/plain; charset=utf-8; format=flowed
>
> Depending on the BYE's direction, the `$dd` variable may point to the
> gateway or to the caller. If the BYE comes from the gateway, you should
> decrease the statistic based on $si, not $dd.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 9/3/20 7:10 PM, Social Boh wrote:
> > Hello,
> >
> > I'm trying to using DISPATCHER module with algorithm 9 together with
> > STATISTICS module:
> >
> > My configuration:
> >
> > STATISTIC module
> > loadmodule "statistics.so"
> > modparam("statistics", "stat_groups", "load_sip")
> >
> > DISPATCHER module
> > loadmodule "dispatcher.so"
> > modparam("dispatcher",
> > "db_url","mysql://opensips:password@localhost/opensips")
> > modparam("dispatcher", "algo_route", "disproute")
> > modparam("dispatcher", "ds_ping_method", "OPTIONS")
> > modparam("dispatcher", "ds_ping_from", "sip:proxy@1.2.3.4")
> > modparam("dispatcher", "ds_ping_interval", 30)
> > modparam("dispatcher", "ds_probing_threshhold", 2)
> > modparam("dispatcher", "ds_probing_mode", 1)
> > modparam("dispatcher", "dst_avp", "$avp(271)")
> > modparam("dispatcher", "pvar_algo_pattern", "$stat(load_%u)")
> >
> > script:
> >
> > on the BYE block:
> >
> > update_stat("load_sip:$dd", -1)
> >
> > On the DISPATCHER route:
> >
> > route[DISP] {
> > if(!ds_select_dst(1,9)) {
> > send_reply(404, "No destination");
> > exit;
> > }
> > xlog("L_NOTICE", "script: Call to $ru via $du\n");
> > update_stat("load_sip:$dd", +1);
> > t_on_failure("DISPATCHER_FAILURE");
> > t_relay();
> > exit;
> > }
> >
> > The problem is the when the call terminate, the load stay on 1. If I
> > make 6 calls and hangup:
> >
> > {
> > "URI": "sip:5.6.7.8:5060",
> > "state": "Active",
> > "first_hit_counter": 6
> > },
> >
> > The calls never go to second Gateway and never change the load.
> >
> > Any help is really appreciate.
> >
> > Regards
> >
>
>
>
> --
>
> Message: 4
> Date: Fri, 4 Sep 2020 03:30:17 -0500
> From: Social Boh
> To: OpenSIPS users mailling list , Răzvan
> Crainea
> Subject: Re: [OpenSIPS-Users] How to change default password -
> opensipsrw
> Mess
[OpenSIPS-Users] How to change default password - opensipsrw
Hi there. I am using version 3.1 of opensips. I tried changing the database password from the config files, however, it does not work. Can someone please share a list of locations of where the passwords (incl. UI) needs to be changed? I am on a debian machine. Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
