Re: [OpenSIPS-Users] Run custom PHP script to send SMS upon receiving invite
Sounds like a bad idea :) You can try to play with the exec module, see http://www.opensips.org/html/docs/modules/1.6.x/exec.html -Laszlo 2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk Hi, using OpenSIPs 1.6, how does one run a custom php script (or whatever code) upon receiving incoming INVITE. We need to perform some custom action ie send SMS message when receiving an INVITE. With asterisk one can use the System function in DialPlan what is the equivalent way with OpensSIPS. Thanks ___ 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] Run custom PHP script to send SMS upon receiving invite
The exec module has a huge performance penalty. I'd avoid it if high performance is required. I don't know anything about your application, but if the time delay between the invite and the SMS isn't too sensitive, you could consider using avp_db_query to insert a queued notification to a SMS message. Alternatively, you could use avp_db_query with db_http to post a realtime SMS notification, but you'd need to create a db adapter per db_http; it shouldn't be too hard at all. The perl module is also very good, but I'm not sure how up-to-date it is. -Brett On Sat, Jul 24, 2010 at 7:48 AM, Laszlo las...@voipfreak.net wrote: Sounds like a bad idea :) You can try to play with the exec module, see http://www.opensips.org/html/docs/modules/1.6.x/exec.html -Laszlo 2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk Hi, using OpenSIPs 1.6, how does one run a custom php script (or whatever code) upon receiving incoming INVITE. We need to perform some custom action ie send SMS message when receiving an INVITE. With asterisk one can use the System function in DialPlan what is the equivalent way with OpensSIPS. Thanks ___ 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] Run custom PHP script to send SMS upon receiving invite
Thank you for the respnses --- On Sat, 24/7/10, Brett Nemeroff br...@nemeroff.com wrote: From: Brett Nemeroff br...@nemeroff.com Subject: Re: [OpenSIPS-Users] Run custom PHP script to send SMS upon receiving invite To: OpenSIPS users mailling list users@lists.opensips.org Date: Saturday, 24 July, 2010, 15:59 The exec module has a huge performance penalty. I'd avoid it if high performance is required. I don't know anything about your application, but if the time delay between the invite and the SMS isn't too sensitive, you could consider using avp_db_query to insert a queued notification to a SMS message. Alternatively, you could use avp_db_query with db_http to post a realtime SMS notification, but you'd need to create a db adapter per db_http; it shouldn't be too hard at all. The perl module is also very good, but I'm not sure how up-to-date it is. -Brett On Sat, Jul 24, 2010 at 7:48 AM, Laszlo las...@voipfreak.net wrote: Sounds like a bad idea :) You can try to play with the exec module, see http://www.opensips.org/html/docs/modules/1.6.x/exec.html -Laszlo 2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk Hi, using OpenSIPs 1.6, how does one run a custom php script (or whatever code) uponreceiving incoming INVITE. We need to perform some custom action ie send SMS message when receiving an INVITE. With asterisk one can use the System function in DialPlanwhat is the equivalent way with OpensSIPS. Thanks ___ 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 -Inline Attachment Follows- ___ 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
[OpenSIPS-Users] Opensips as outbound proxy for asterisk client
Hi, We have Asterisk 1.6.2.9 and Opensips 1.6.2 in the same Linux box and we
want
to have OpenSIPS as the out/inbound proxy for Asterisk when running as UAC.
We have Asterisk registered with Sipgate ok when sending REGISTER directly to
the Sipgate
domain but would like to send the Register via OpenSIPS so it's in the path for
incoming invites
Opensips.cfg as follows
listen=udp:192.168.0.20:5060
register code: we have removed the authentication for now to get this working
first
if (is_method(REGISTER))
{
if (!db_check_to())
{
sl_send_reply(403,Forbidden auth ID);
exit;
}
if (!save(location))
sl_reply_error();
exit;
}
we have astersik sip.conf file as follows
[general]
port=5062
bindaddr=0.0.0.0
context=default
outboundproxy=192.168.0.20
outboundproxyport=5060
register = username:sec...@sipgate.co.uk/username
[username]
type=friend
context=incoming
username=username
secret=secret
host=sipgate.co.uk
host=dynamic
fromdomain=sipgate.co.uk
insecure=port,invite
nat=yes
disallow=all
allow=alaw
canreinvite=no
However the REGISTER messages are sent by Astersik but ther is no response fron
OpenSIPS and it is not routing them either . Is it because they are both on the
same IP?
We do have then on different ports so though it would be ok
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Re: [OpenSIPS-Users] Run custom PHP script to send SMS upon receiving invite
Brett is right, but avp_db_query can be a performance killer too (just imagine it with 200 new calls per second = 200 insert queries/sec) -Laszlo 2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk Thank you for the respnses --- On *Sat, 24/7/10, Brett Nemeroff br...@nemeroff.com* wrote: From: Brett Nemeroff br...@nemeroff.com Subject: Re: [OpenSIPS-Users] Run custom PHP script to send SMS upon receiving invite To: OpenSIPS users mailling list users@lists.opensips.org Date: Saturday, 24 July, 2010, 15:59 The exec module has a huge performance penalty. I'd avoid it if high performance is required. I don't know anything about your application, but if the time delay between the invite and the SMS isn't too sensitive, you could consider using avp_db_query to insert a queued notification to a SMS message. Alternatively, you could use avp_db_query with db_http to post a realtime SMS notification, but you'd need to create a db adapter per db_http; it shouldn't be too hard at all. The perl module is also very good, but I'm not sure how up-to-date it is. -Brett On Sat, Jul 24, 2010 at 7:48 AM, Laszlo las...@voipfreak.nethttp://mc/compose?to=las...@voipfreak.net wrote: Sounds like a bad idea :) You can try to play with the exec module, see http://www.opensips.org/html/docs/modules/1.6.x/exec.html -Laszlo 2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.ukhttp://mc/compose?to=nauman762-h...@yahoo.co.uk Hi, using OpenSIPs 1.6, how does one run a custom php script (or whatever code) upon receiving incoming INVITE. We need to perform some custom action ie send SMS message when receiving an INVITE. With asterisk one can use the System function in DialPlan what is the equivalent way with OpensSIPS. Thanks ___ Users mailing list Users@lists.opensips.org http://mc/compose?to=us...@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://mc/compose?to=us...@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -Inline Attachment Follows- ___ Users mailing list Users@lists.opensips.org http://mc/compose?to=us...@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] Opensips as outbound proxy for asterisk client
You can relay the register to sipgate, so when auth is done, then you can
forward it.
consume_credentials();
$du = sip:sipgate_ip_goes_here:port;
t_relay();
exit;
so instead of doing if (!save(location)) etc
-Laszlo
2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk
Hi, We have Asterisk 1.6.2.9 and Opensips 1.6.2 in the same Linux box and
we want
to have OpenSIPS as the out/inbound proxy for Asterisk when running as UAC.
We have Asterisk registered with Sipgate ok when sending REGISTER directly
to the Sipgate
domain but would like to send the Register via OpenSIPS so it's in the path
for incoming invites
Opensips.cfg as follows
listen=udp:192.168.0.20:5060
register code: we have removed the authentication for now to get this
working first
if (is_method(REGISTER))
{
if (!db_check_to())
{
sl_send_reply(403,Forbidden auth ID);
exit;
}
if (!save(location))
sl_reply_error();
exit;
}
we have astersik sip.conf file as follows
[general]
port=5062
bindaddr=0.0.0.0
context=default
outboundproxy=192.168.0.20
outboundproxyport=5060
register = username:sec...@sipgate.co.uk/username
[username]
type=friend
context=incoming
username=username
secret=secret
host=sipgate.co.uk
host=dynamic
fromdomain=sipgate.co.uk
insecure=port,invite
nat=yes
disallow=all
allow=alaw
canreinvite=no
However the REGISTER messages are sent by Astersik but ther is no response
fron OpenSIPS and it is not routing them either . Is it because they are
both on the same IP?
We do have then on different ports so though it would be ok
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Re: [OpenSIPS-Users] Opensips as outbound proxy for asterisk client
Hi, many thanks
we don't follow this line
$du = sip:sipgate_ip_goes_here:port
where do we get the sipgate ip, should this just be
$du = sip:sipgate.co.uk:5060
at least it explains why we saw the REGISTER being gobbled
--- On Sat, 24/7/10, Laszlo las...@voipfreak.net wrote:
From: Laszlo las...@voipfreak.net
Subject: Re: [OpenSIPS-Users] Opensips as outbound proxy for asterisk client
To: nauman762-h...@yahoo.co.uk, OpenSIPS users mailling list
users@lists.opensips.org
Date: Saturday, 24 July, 2010, 19:28
You can relay the register to sipgate, so when auth is done, then you can
forward it.
consume_credentials();
$du = sip:sipgate_ip_goes_here:port;
t_relay();
exit;
so instead of doing if (!save(location)) etc
-Laszlo
2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk
Hi, We have Asterisk 1.6.2.9 and Opensips 1.6.2 in the same Linux box and we
want
to have OpenSIPS as the out/inbound proxy for Asterisk when running as UAC.
We have Asterisk registered with Sipgate ok when sending REGISTER directly to
the Sipgate
domain but would like to send the Register via OpenSIPS so it's in the path for
incoming invites
Opensips.cfg as follows
listen=udp:192.168.0.20:5060
register code: we have removed the authentication for now to get this working
first
if (is_method(REGISTER))
{
if (!db_check_to())
{
sl_send_reply(403,Forbidden auth ID);
exit;
}
if (!save(location))
sl_reply_error();
exit;
}
we have astersik sip.conf file as follows
[general]
port=5062
bindaddr=0.0.0.0
context=default
outboundproxy=192.168.0.20
outboundproxyport=5060
register =
username:sec...@sipgate.co.uk/username
[username]
type=friend
context=incoming
username=username
secret=secret
host=sipgate.co.uk
host=dynamic
fromdomain=sipgate.co.uk
insecure=port,invite
nat=yes
disallow=all
allow=alaw
canreinvite=no
However the REGISTER messages are sent by Astersik but ther is no response fron
OpenSIPS and it is not routing them either . Is it because they are both on the
same IP?
We do have then on different ports so though it would be ok
___
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Re: [OpenSIPS-Users] Mediaproxy not updating with new SDPs
Saul (and everyone) good afternoon. I think I've come across a call flow that Mediaproxy would be expected to handle but is not. Umm, it's like sacrilege I know; but I think Mediaproxy may have a bug. 1) The originator sends INVITE with a couple of codec choices. 2) engage_media proxy is called and the INVITE is forwarded 3) 200 comes back with a couple of codec choices and m line indicates rtp will source from port 35210 (and indeed rtp starts sourcing from there 4) The ACK comes back and for 10 milliseconds life is good Then the Originator reinvites with only the single codec choice that it believes was negotiated. Standard procedure for this device and allowed and all that but really, 10 milliseconds?? Anyway... 5) reinvite is forwarded without any mediaproxy calls or anything from the script 6) The destination sends back a 200 but this time claims it will source from port 1082. 7) The destination rtp continues to source from the original port 35210 So, the result here is that rtp comes from the origination and is forwarded, by Mediaproxy, to port 35210 at the destination where it is happily accepted. Media also flows from the destination from port 35210 but it is not forwarded which results in one way audio on the call. It will be Monday before I am able to request more calls to be made but I'm kind of thinking the firewall may be the cause of the port not changing and the UA is actually sending from a new port. Maybe not though. In either case, should the Mediaproxy have waited for a rtp packet on any port from the (call) destination IP after the 200 to the reinvite and set a new connection track rule after it saw one? I'm using Mediaproxy 2.4.2. Richard On Jul 21, 2010, at 5:14 PM, Saúl Ibarra Corretgé wrote: Hi Jeff, On 21/07/10 21:50, Jeff Pyle wrote: Hello, I am using Mediaproxy 2.3.8 with Opensips 1.6 r6702. I use the engage_media_proxy() function. Most things work pretty well except for the following scenario: Call comes in from CPE, hits the engage_media_proxy() function, and t_relay's to the far end. The far-end sends a 503, but before they do, they send some media. I'm working with them now to determine the true source of this unwanted media. Anyway, Opensips does a serial fork to the next carrier, who sends a 183 w/ SDP, and finally a 200 OK. But the media relay has already locked down to the media the first carrier sent. How come the media relay doesn't learn about the new media information from the SDPs in the 183 and the 200? MediaProxy uses the information from the SDP from both ends and then it waits to receive a RTP packet from each endpoint. Once this happens it inserts a conntrack rule in the kernel so that the relaying is performed. I could judge better with a SIP trace but my guess is that the dialog could get to an inconsistent state, so mediaproxy loses track. I'd like to be able to wait for an SDP to come back on a 18x or a 200 before starting the relay. It appears the engage_media_proxy() function is available only from the REQUEST_ROUTE. Perhaps it would be smarter in this case to use the use_media_proxy()/end_media_session() functions in the REPLY_ROUTE and FAILURE_ROUTEs. You are right. Engage_media_proxy covers most general purpose scenarios but it's not suitable for everyone, so in this case you may want yo call the individual functions manually. Regards, -- Saúl Ibarra Corretgé AG Projects ___ 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] Opensips as outbound proxy for asterisk client
Hi, making progress
we now have this so the REGISTER is passed to the Sipgate proxy transparently
it could be any proxy server so
if (is_method(REGISTER))
{
##$du = sip:sipgate.co.uk:5060;
t_relay();
exit();
}
We can see via Wireshark trace that REGISTER is reaching the sipgate proxy but
OpenSIPS is returning a 403 to our Asterisk (acting as the SIP client)
its because of this code, so then upon receiving the 403 from Opensips it
doesn't respond to the 401 from the sipgate proxy. What should we be doing below
if (!is_uri_host_local())
{
if(is_from_local()) {
route(1);
}
else {
sl_send_reply(403,Not here);
}
##append_hf(P-hint: outbound\r\n);
# if you have some interdomain connections via TLS
##if($rd==tls_domain1.net) {
## t_relay(tls:domain1.net);
## exit;
##} else if($rd==tls_domain2.net) {
## t_relay(tls:domain2.net);
## exit;
##}
##route(1);
}
Thanks
--- On Sat, 24/7/10, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote:
From: Nauman Sulaiman nauman762-h...@yahoo.co.uk
Subject: Re: [OpenSIPS-Users] Opensips as outbound proxy for asterisk client
To: OpenSIPS users mailling list users@lists.opensips.org, Laszlo
las...@voipfreak.net
Date: Saturday, 24 July, 2010, 19:48
Hi, many thanks
we don't follow this line
$du = sip:sipgate_ip_goes_here:port
where do we get the sipgate ip, should this just be
$du = sip:sipgate.co.uk:5060
at least it explains why we saw the REGISTER being gobbled
--- On Sat, 24/7/10, Laszlo las...@voipfreak.net wrote:
From: Laszlo las...@voipfreak.net
Subject: Re: [OpenSIPS-Users] Opensips as outbound proxy for asterisk client
To: nauman762-h...@yahoo.co.uk, OpenSIPS users mailling list
users@lists.opensips.org
Date: Saturday, 24 July, 2010, 19:28
You can relay the register to sipgate, so when auth is done, then you can
forward it.
consume_credentials();
$du =
sip:sipgate_ip_goes_here:port;
t_relay();
exit;
so instead of doing if (!save(location)) etc
-Laszlo
2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk
Hi, We have Asterisk 1.6.2.9 and Opensips 1.6.2 in the same Linux box and we
want
to have OpenSIPS as the out/inbound proxy for Asterisk when running as UAC.
We have Asterisk registered with Sipgate ok when sending REGISTER directly to
the Sipgate
domain but would like to send the Register via OpenSIPS so it's in the path for
incoming invites
Opensips.cfg as follows
listen=udp:192.168.0.20:5060
register code: we have removed the authentication for now to get this working
first
if (is_method(REGISTER))
{
if (!db_check_to())
{
sl_send_reply(403,Forbidden auth ID);
exit;
}
if (!save(location))
sl_reply_error();
exit;
}
we have astersik sip.conf file as follows
[general]
port=5062
bindaddr=0.0.0.0
context=default
outboundproxy=192.168.0.20
outboundproxyport=5060
register =
username:sec...@sipgate.co.uk/username
[username]
type=friend
context=incoming
username=username
secret=secret
host=sipgate.co.uk
host=dynamic
fromdomain=sipgate.co.uk
insecure=port,invite
nat=yes
disallow=all
allow=alaw
canreinvite=no
However the REGISTER messages are sent by Astersik but ther is no response fron
OpenSIPS and it is not routing them either . Is it because they are both on the
same IP?
We do have then on different ports so though it would be ok
___
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[OpenSIPS-Users] DNS issues
Hi all, I wanted to collect some ideas on how do you solve DNS connectivity problems. I've run into those issues a couple of times already and don't see a perfect solution so far. Maybe I can trigger some discussion: Some background: - opensips blocks the child process while resolving a domain / querying ENUM - standard resolver has minimum timeout = 1s - standard resolver does only one query at a time and can cycle nameservers, but does not save state I believe these are not real problems - just ugly legacy :) that we can work around. The implication is that if you don't use a caching nameserver on your side and you allow users to use routing based on a domain name (not very hard - do you handle 302s, record-routes, registration?), you're basically screwed: 1. If you don't cache, any domain which times out will block a child for at least 1s. If you use retries, you block for at least Ns where N = number of nameservers. You can be DoS-ed with ~8 packets per second, in standard configuration. 2. If you cycle N nameservers and one of them is down, you're processing N-1 packets correctly, then block until timeout on the last one, then processing N-1, etc. - not good for a high-traffic proxy. 3. If you cache results, you're safe from random failures, but only if you cache timeouts as negative results and keep the state of servers being down, so you don't try to query them again. (nothing apart from `dnsmasq` does that, AFAIK) 4. What solves half of the problem for me, is `dnsmasq` - as far as I know it's the only caching dns server which allows to query all nameservers in parallel. I get 4 times the needed DNS traffic, but I'm never timing out connections if one of the servers is down. Also some results come from cache, so it's only 2 times the traffic in reality. The problem with `dnsmasq` is that it doesn't cache SRV and NAPTR requests (doesn't cache the timeouts / NX responses for them either), only A//PTR/ 5. So even if you have a local caching and backup resolver in `resolv.conf`, minimal timeout, parallel querying from the local cache, saving the state of upstream resolvers being down and route all internal traffic via IPs... it takes only one person with custom NAPTR sending you to custom SRV address which times out to kill all the traffic. So... what's your experience with this? Do you have some better protection in place? I'm considering adding negative caching of dns timeouts and general caching of SRV and NAPTR records into `dnsmasq` to complete my protection. Do you know of any software which would solve those problems out-of-box? Thanks, Stan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
