[OpenSIPS-Users] Setting Extensions via Rest API or Database writes
Hi, Just wondering if Opensips has something similar to Asterisk where one can setup extensions, queues etc via realtime database? Alternatively could it be done via RestAPI. If so, is all functionality configurable from remote or just some? Regards Nauman ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] REGISTER destination issue
Hi We are using Opensips as proxy, when Opensips sends a REGISTER message it resolves the domain of the RURI however if the registrar has a bank of servers this could result in the 2nd REGISTER message(with auth details) being sent to a different server. Some registrars don't seem to keep auth state across servers. The simple solution is to send the 2nd REGISTER to the same address as the first. I thought of storing the si variable from the 401 response and just send the 2nd register to this ip address. But i can't use avps to store nor the Dialog state stuff. Can anyone say what is the best way to fix this. Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] REGISTER destination issue
Hi, many thanks, we'll give that a try --- On Tue, 6/12/11, Vlad Paiu vladp...@opensips.org wrote: From: Vlad Paiu vladp...@opensips.org Subject: Re: [OpenSIPS-Users] REGISTER destination issue To: users@lists.opensips.org Date: Tuesday, 6 December, 2011, 15:29 Hi, Just a suggestion, you could use caching. When you receive a 401 response, you could save in localcache, for the key register_$tu the value $si, with a short expiry. When you receive a new Register request, check whether you have register_$tu in cache. If you do, then route the Register to that value, the old $si. Regards, Vlad Paiu OpenSIPS Developer On 12/06/2011 04:33 PM, Nauman Sulaiman wrote: Hi We are using Opensips as proxy, when Opensips sends a REGISTER message it resolves the domain of the RURI however if the registrar has a bank of servers this could result in the 2nd REGISTER message(with auth details) being sent to a different server. Some registrars don't seem to keep auth state across servers. The simple solution is to send the 2nd REGISTER to the same address as the first. I thought of storing the si variable from the 401 response and just send the 2nd register to this ip address. But i can't use avps to store nor the Dialog state stuff. Can anyone say what is the best way to fix this. 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] REGISTER destination issue
Unfortunately when we get the 401 the $si value gives the source address of the REGISTER request not the of the 401 response. How dow we get the ip address of the 401 response? --- On Tue, 6/12/11, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote: From: Nauman Sulaiman nauman762-h...@yahoo.co.uk Subject: Re: [OpenSIPS-Users] REGISTER destination issue To: OpenSIPS users mailling list users@lists.opensips.org Date: Tuesday, 6 December, 2011, 15:55 Hi, many thanks, we'll give that a try --- On Tue, 6/12/11, Vlad Paiu vladp...@opensips.org wrote: From: Vlad Paiu vladp...@opensips.org Subject: Re: [OpenSIPS-Users] REGISTER destination issue To: users@lists.opensips.org Date: Tuesday, 6 December, 2011, 15:29 Hi, Just a suggestion, you could use caching. When you receive a 401 response, you could save in localcache, for the key register_$tu the value $si, with a short expiry. When you receive a new Register request, check whether you have register_$tu in cache. If you do, then route the Register to that value, the old $si. Regards, Vlad Paiu OpenSIPS Developer On 12/06/2011 04:33 PM, Nauman Sulaiman wrote: Hi We are using Opensips as proxy, when Opensips sends a REGISTER message it resolves the domain of the RURI however if the registrar has a bank of servers this could result in the 2nd REGISTER message(with auth details) being sent to a different server. Some registrars don't seem to keep auth state across servers. The simple solution is to send the 2nd REGISTER to the same address as the first. I thought of storing the si variable from the 401 response and just send the 2nd register to this ip address. But i can't use avps to store nor the Dialog state stuff. Can anyone say what is the best way to fix this. 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 1.7 uac_registrant issue
Hi The link below details an issue Chris Martineau reported back in July this year. Ovidiu Sas replied and asked a bug be reported. Checked the bug tracker but can't find it. Does anyone know if this issue was fixed or if not raised as a bug is there a workaround to solve the problem. http://www.mail-archive.com/users@lists.opensips.org/msg16929.html Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Where in script can one catch 200OK to NOTIFY
Hi, using Opensips 1.6.3 Where in the script can I catch the 200OK response to a NOTIFY sent in response to in dialog REFER (call transfer) . The UA receiving the NOTIFY is sending a 200OK with a private IP in the contact header. I wish to call fix nated contact on this header. Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Rewriting destination uri
Hi, I have found writing and reading these dlg vars work fine with the good
proxies but the buggy ones (which have the Opensips IP as the RURI) that I need
to use this method with I can't read the dlg values so it appears the dialog is
not being matched or something
--- On Mon, 17/10/11, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote:
From: Nauman Sulaiman nauman762-h...@yahoo.co.uk
Subject: Re: [OpenSIPS-Users] Rewriting destination uri
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Monday, 17 October, 2011, 12:12
Hi,
I have tried using dialog vals to store the contact ip in
on_reply_route
the create dialog call succeeds, so is there anything else
you need to set up to use dialog vals, do you need to use
the Database??
if (status=~(180)|(183)|2[0-9][0-9])
{
store_dlg_value(contact_ip,$si);
store_dlg_value(contact_port,$sp);
}
}
but then i can't access them in the sequential route for
the ACK
after loose_route return true
route[2]
{
...
fetch_dlg_value(contact_ip,$var(ip));
fetch_dlg_value(contact_port,$var(port));
xlog(New ACK destination= $var(ip)
\n);
xlog(New
port= $var(port) \n);
}
--- On Fri, 14/10/11, Nauman Sulaiman nauman762-h...@yahoo.co.uk
wrote:
From: Nauman Sulaiman nauman762-h...@yahoo.co.uk
Subject: Re: [OpenSIPS-Users] Rewriting destination
uri
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Friday, 14 October, 2011, 20:57
Hi,
Thanks, yes its a buggy proxy. We won't have the
scenario
UAC ProxyA OpenSIPS ProxyB
so how best to rewrite the $du in this case
Regards
--- On Fri, 14/10/11, Vlad Paiu vladp...@opensips.org
wrote:
From: Vlad Paiu vladp...@opensips.org
Subject: Re: [OpenSIPS-Users] Rewriting
destination
uri
To: users@lists.opensips.org
Date: Friday, 14 October, 2011, 10:19
Hello,
First of all, why does the 200 OK have in R-URI
the
OpenSIPS IP ?
It should have as R-URI the Contact URI in the
200 OK
? Are
you dealing
with a buggy client/proxy behind OpenSIPS or is
there
some
other kind of
miss-configuration ?
The solution that you are trying to achieve is
not
generic,
because it
would not work
in case of
UAC ProxyA OpenSIPS ProxyB
Regards,
Vlad Paiu
OpenSIPS Developer
On 10/14/2011 01:02 AM, Nauman Sulaiman wrote:
Hi,
We have the following set up , inbound call
to
UAC via
Opensips
UAC- Opensips--
Proxy
1
INVITE
INVITE
-
200OK
200OK
--
ACK RURI=
Opensips IP
We would like to set the $du variable after
Opensips
receives the
final ACK to be IP:port of the UAC.
When Opensips receives the 200OK from the
UAC is
it
possible to store the received ip and port
somewhere
so we
can rewrite the $du when we receive the ACK.How
to do
this?
Then how to rewrite the $du
Currently the ACK is not routed back to the
UAC
as the
RURI of the ACK has Opensips address.
Regards
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Re: [OpenSIPS-Users] Rewriting destination uri
Hi,
I have tried using dialog vals to store the contact ip in on_reply_route
the create dialog call succeeds, so is there anything else you need to set up
to use dialog vals, do you need to use the Database??
if (status=~(180)|(183)|2[0-9][0-9])
{
store_dlg_value(contact_ip,$si);
store_dlg_value(contact_port,$sp);
}
}
but then i can't access them in the sequential route for the ACK
after loose_route return true
route[2]
{
...
fetch_dlg_value(contact_ip,$var(ip));
fetch_dlg_value(contact_port,$var(port));
xlog(New ACK destination= $var(ip) \n);
xlog(New port= $var(port) \n);
}
--- On Fri, 14/10/11, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote:
From: Nauman Sulaiman nauman762-h...@yahoo.co.uk
Subject: Re: [OpenSIPS-Users] Rewriting destination uri
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Friday, 14 October, 2011, 20:57
Hi,
Thanks, yes its a buggy proxy. We won't have the
scenario
UAC ProxyA OpenSIPS ProxyB
so how best to rewrite the $du in this case
Regards
--- On Fri, 14/10/11, Vlad Paiu vladp...@opensips.org
wrote:
From: Vlad Paiu vladp...@opensips.org
Subject: Re: [OpenSIPS-Users] Rewriting destination
uri
To: users@lists.opensips.org
Date: Friday, 14 October, 2011, 10:19
Hello,
First of all, why does the 200 OK have in R-URI the
OpenSIPS IP ?
It should have as R-URI the Contact URI in the 200 OK
? Are
you dealing
with a buggy client/proxy behind OpenSIPS or is there
some
other kind of
miss-configuration ?
The solution that you are trying to achieve is not
generic,
because it
would not work
in case of
UAC ProxyA OpenSIPS ProxyB
Regards,
Vlad Paiu
OpenSIPS Developer
On 10/14/2011 01:02 AM, Nauman Sulaiman wrote:
Hi,
We have the following set up , inbound call to
UAC via
Opensips
UAC- Opensips-- Proxy
1
INVITE
INVITE
-
200OK
200OK
--
ACK RURI=
Opensips IP
We would like to set the $du variable after
Opensips
receives the
final ACK to be IP:port of the UAC.
When Opensips receives the 200OK from the UAC is
it
possible to store the received ip and port somewhere
so we
can rewrite the $du when we receive the ACK.How to do
this?
Then how to rewrite the $du
Currently the ACK is not routed back to the UAC
as the
RURI of the ACK has Opensips address.
Regards
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Re: [OpenSIPS-Users] Rewriting destination uri
Hi, Thanks, yes its a buggy proxy. We won't have the scenario UAC ProxyA OpenSIPS ProxyB so how best to rewrite the $du in this case Regards --- On Fri, 14/10/11, Vlad Paiu vladp...@opensips.org wrote: From: Vlad Paiu vladp...@opensips.org Subject: Re: [OpenSIPS-Users] Rewriting destination uri To: users@lists.opensips.org Date: Friday, 14 October, 2011, 10:19 Hello, First of all, why does the 200 OK have in R-URI the OpenSIPS IP ? It should have as R-URI the Contact URI in the 200 OK ? Are you dealing with a buggy client/proxy behind OpenSIPS or is there some other kind of miss-configuration ? The solution that you are trying to achieve is not generic, because it would not work in case of UAC ProxyA OpenSIPS ProxyB Regards, Vlad Paiu OpenSIPS Developer On 10/14/2011 01:02 AM, Nauman Sulaiman wrote: Hi, We have the following set up , inbound call to UAC via Opensips UAC- Opensips-- Proxy 1 INVITE INVITE - 200OK 200OK -- ACK RURI= Opensips IP We would like to set the $du variable after Opensips receives the final ACK to be IP:port of the UAC. When Opensips receives the 200OK from the UAC is it possible to store the received ip and port somewhere so we can rewrite the $du when we receive the ACK.How to do this? Then how to rewrite the $du Currently the ACK is not routed back to the UAC as the RURI of the ACK has Opensips address. Regards ___ 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
[OpenSIPS-Users] Not writing to Opensips log
Hi, I deleted the 1.6 opensips log i had redirected logging to /var/log/opensips.log as it had become huge. However opensips is not logging to it anymore. I've done a chmod 777 on the new log file but that still does not help. Can anyone tell me how to get it logging again without having to reboot the whole server. Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Rewriting destination uri
Hi, We have the following set up , inbound call to UAC via Opensips UAC- Opensips -- Proxy 1 INVITE INVITE - 200OK200OK -- ACK RURI= Opensips IP We would like to set the $du variable after Opensips receives the final ACK to be IP:port of the UAC. When Opensips receives the 200OK from the UAC is it possible to store the received ip and port somewhere so we can rewrite the $du when we receive the ACK.How to do this? Then how to rewrite the $du Currently the ACK is not routed back to the UAC as the RURI of the ACK has Opensips address. Regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Any limit on registrants in uac module
Hi, thanks just a couple of quick questions 1) Can module parameters to uac_registrant be loaded from avp (database) rather than hard code in cfg 2) Can uac_registrant module parameters be dynamically changed? Regards --- On Sat, 8/10/11, Ovidiu Sas o...@voipembedded.com wrote: From: Ovidiu Sas o...@voipembedded.com Subject: Re: [OpenSIPS-Users] Any limit on registrants in uac module To: nauman762-h...@yahoo.co.uk, OpenSIPS users mailling list users@lists.opensips.org Date: Saturday, 8 October, 2011, 22:53 Try to increase the size of the hash table to better distribute the load: http://www.opensips.org/html/docs/modules/devel/uac_registrant.html#id249100 If you encounter any issues, let me know. Regards, Ovidiu Sas On Sat, Oct 8, 2011 at 2:40 PM, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote: Hi, With Opensips 1.7 Is there any limit on the number of registrants you can have with uac_registrant module. Could it handle 1000? 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
[OpenSIPS-Users] Any limit on registrants in uac module
Hi, With Opensips 1.7 Is there any limit on the number of registrants you can have with uac_registrant module. Could it handle 1000? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips crashing due to out of memory error
Hi, I'm getting random crashes of Opensips 1.6.2, here are the entries in the log, seems to be out of memory, how should i try to solve this issue. ERROR:tm:_reply_light: failed to allocate shmem buffer ERROR:tm:_reply_light: failed to allocate shmem buffer ERROR:tm:relay_reply: no more share memory ERROR:core:new_avp: no more shm mem ERROR:core:add_avp: Failed to create new avp structure Regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Lookup contact from user part of RURI
Hi, using opensips 1.6.2. I am trying to use the user part of incoming RURI
to look up a contact, reason being is full RURI is incorrect, this is due to
bogus proxy upstream so need a workaround.
lookup(location) seems to be only if you use AOR.
For exmaple i need to reroute incoming ACK to real address of UA
So i would like to lookup 1234 user part of RURI below and rewrite the
RURI with the correct ip. 12.34.56.78 is Opensips IP rather than end UA ip.
1...@domain.com is stored in db. How do i lookup contact just with user part
and rewrite the RURI.
ie ACK sip:1234@12.34.56.78;rinstance=A89B5393
Need something for below
if(method==ACK)
{
xlog(ACK received \n);
if( $rd == 12.34.56.78) // check if opensips ip
{
lookup(user); // ??? // need to lookup with user or rinstance
// rewrite RURI with correct address
}
}
Hope its clear, thanks
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[OpenSIPS-Users] Fw: Re: Opensips outbound proxy problem
Hi, Does anyone have any suggestions for this? Many thanks --- On Tue, 18/1/11, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote: From: Nauman Sulaiman nauman762-h...@yahoo.co.uk Subject: Re: [OpenSIPS-Users] Opensips outbound proxy problem To: Bogdan-Andrei Iancu bog...@opensips.org Date: Tuesday, 18 January, 2011, 21:03 Hi, yes Registar receives correct contact but then changes it to the ip address it came from. For example Voxalot do this, its incorrect behaviour. We just want to know best way to workaround it. Regards --- On Tue, 18/1/11, Bogdan-Andrei Iancu bog...@opensips.org wrote: From: Bogdan-Andrei Iancu bog...@opensips.org Subject: Re: [OpenSIPS-Users] Opensips outbound proxy problem To: nauman762-h...@yahoo.co.uk, OpenSIPS users mailling list users@lists.opensips.org Date: Tuesday, 18 January, 2011, 11:53 Hi Nauman, First you need to see who's changing the Contact hdr (from UA IP to opensips IP) - does the REGISTER received by REGISTRAR server carry the right Contact? Regards, Bogdan Nauman Sulaiman wrote: Hi, we are using opensips as outbound proxy. We have a problem when a UA registers with certain servers via opensips but the contact address returned by the registrar is that of opensips rather than the UA. Both the UA and opensips are on public ip and we have opensips inserting path header. Just some registrars behave incorrectly. UA - Opensips REGISTRAR Are there any standard tricks/techniques we can use in opensips to route any INVITES/ACK/BYE back to the UA as the RURI are set to the opensips proxy address rather than the UA. Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips outbound proxy problem
Hi, we are using opensips as outbound proxy. We have a problem when a UA registers with certain servers via opensips but the contact address returned by the registrar is that of opensips rather than the UA. Both the UA and opensips are on public ip and we have opensips inserting path header. Just some registrars behave incorrectly. UA - Opensips REGISTRAR Are there any standard tricks/techniques we can use in opensips to route any INVITES/ACK/BYE back to the UA as the RURI are set to the opensips proxy address rather than the UA. Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips altering RTP port in SDP??
Hi, using Opensips 1.6.2. I have observed that soemtimes Opensips changes the RTP port in the SDP description. This is when using Bria Counterpath client with another softphone. It seems to be because the Bria client has an RTP port of 4000 which gets remapped to something higher eg 4100. This then results in one way audio. Are there any reasons that Opensips would alter the SDP m=audio line. We are using a Netgear router but don't believe this is modifying anything. Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Fw: Using same DNS resolved ip
Hi, Unfortunately adding Path header did not work :( Regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Fw: Using same DNS resolved ip
Just to clarify, i have the following setup
UA Opensips --- Callcentric Voip Provider(for example)
I wish to have the UA register with Callcentric via Opensips as outbound proxy
so that all future invites come via it. I have not being able to get any UA (
Bria etc) to register with CallCentric which leads me to think there is a
problem with my script. Registering with providers who do not do load balancing
is straight forward. It's just with providers such as Callcentric, what happens
is as i am just using Opensips as a relay for registration the 407 from
Callcentric is passed back to the UA which sends another REGISTER request, this
is then sent to a different IP (different callcentric proxy) by Opensips,
presumably because it does a fresh look up.
here is my script which deals with register requests:
if (!uri==myself)
{
route(1);
}
In route[1]
if (method==REGISTER)
{
if (!t_relay()) {
sl_reply_error();
}
exit;
}
This works with most providers but not those doing load balancing.
Thanks
--- On Fri, 1/10/10, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote:
From: Nauman Sulaiman nauman762-h...@yahoo.co.uk
Subject: [OpenSIPS-Users] Fw: Using same DNS resolved ip
To: users@lists.opensips.org
Date: Friday, 1 October, 2010, 17:17
Hi Anca
I've tried 2 different User Agent behind Opensips issuing
the REGISTER, Opensips is just proxying the request. The
problem is each time it sends to a different IP.So
Callcentric returns 407 with stale = true
Regards
--- On Thu, 30/9/10, Nauman Sulaiman nauman762-h...@yahoo.co.uk
wrote:
From: Nauman Sulaiman nauman762-h...@yahoo.co.uk
Subject: [OpenSIPS-Users] Using same DNS resolved ip
To: users@lists.opensips.org
Date: Thursday, 30 September, 2010, 22:34
Hi, using Opensips 1.6.2. We were
wondering if it was possible to force Opensips to use
the
same IP address when issuing REGISTER request to
certain
VoIP providers such as CallCentric which do load
balancing
on their servers. Currently we are using Opensips as
outboundproxy each time it issues a REGISTER request
it does
a round robin of all DNS address got from an SRV
lookup.
Because i think there is a bug in CallCentric and
others
that if it receives a REGISTER with auth info at a
different
ip that issued the challenge it sends another 407
challenge.
Is there anyway to force Opensips to use the same ip?
Thanks
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[OpenSIPS-Users] Downstream proxy rewriting contact header
Hi, using Opensips 1.6.2 we have the following setup. UA1Opensips Proxy --- P1 -- P2 -- UA2 Proxy P1 is rewriting contact header sent by opensips in 180 and 200K response to INVITE from UA2 to UA1. All proxies are record-routing. P1 seems to assume all requests are from behind NAT and rewrites the contact header with source IP ie that of Opensips proxy. The result is UA2 ACK is routed only to Opensips proxy and does not reach UA1. Is there any i can handle this in the Opensips proxy to account for behaviour of P1. Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Fw: Using same DNS resolved ip
Hi Anca I've tried 2 different User Agent behind Opensips issuing the REGISTER, Opensips is just proxying the request. The problem is each time it sends to a different IP.So Callcentric returns 407 with stale = true Regards --- On Thu, 30/9/10, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote: From: Nauman Sulaiman nauman762-h...@yahoo.co.uk Subject: [OpenSIPS-Users] Using same DNS resolved ip To: users@lists.opensips.org Date: Thursday, 30 September, 2010, 22:34 Hi, using Opensips 1.6.2. We were wondering if it was possible to force Opensips to use the same IP address when issuing REGISTER request to certain VoIP providers such as CallCentric which do load balancing on their servers. Currently we are using Opensips as outboundproxy each time it issues a REGISTER request it does a round robin of all DNS address got from an SRV lookup. Because i think there is a bug in CallCentric and others that if it receives a REGISTER with auth info at a different ip that issued the challenge it sends another 407 challenge. Is there anyway to force Opensips to use the same ip? 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
[OpenSIPS-Users] Using same DNS resolved ip
Hi, using Opensips 1.6.2. We were wondering if it was possible to force Opensips to use the same IP address when issuing REGISTER request to certain VoIP providers such as CallCentric which do load balancing on their servers. Currently we are using Opensips as outboundproxy each time it issues a REGISTER request it does a round robin of all DNS address got from an SRV lookup. Because i think there is a bug in CallCentric and others that if it receives a REGISTER with auth info at a different ip that issued the challenge it sends another 407 challenge. Is there anyway to force Opensips to use the same ip? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Problem with Opensips routing Reinvite ACK
Hi, we have a set up as follows using Opensips 1.6.2 UA - OpenSIPS - SIP Provider TLSTLS UDP UDP We can set up a call from Sip Provider to UA. However when we issue a Reinvite due to (Hold) from UA the Sip provider gets the invite and sends 200K to which the UA responds with an ACK , however Opensips does not seem to be routing this ACK back to the provider. Strangely the Reinvite which initally started this has the same route set, yet that reaches the provider. ACK sip:anonym...@217.10.79.23:5060 SIP/2.0 Via: SIP/2.0/TLS 192.168.0.4;branch=z9hG4bKyQatKQmaHK0Br Route: sip:178.230.138.190:5061;transport=tls;r2=on;lr=on;ftag=as7615d7df Route: sip:178.230.138.190;r2=on;lr=on;ftag=as7615d7df Route: sip:217.10.79.23;lr=on;ftag=as7615d7df Route: sip:172.20.40.3;lr=on Route: sip:217.10.79.23;lr=on;ftag=as7615d7df Max-Forwards: 70 From: sip:00441619082...@sipgate.co.uk;tag=t8B2BFQDUgNvc To: anonymous sip:anonym...@sipgate.co.uk;tag=as7615d7df Call-ID: 27a6f665224ffd7b7c5e0d2616be4...@sipgate.co.uk CSeq: 2556588 ACK Content-Length: 0 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips ACK routing problem
Maybe one way round this problem would be for us to replace the incoming RURI on the ACK with a sips address maybe to force Opensips to forward it on the tls connection Not sure how to do this however or is there better way? --- On Mon, 23/8/10, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote: From: Nauman Sulaiman nauman762-h...@yahoo.co.uk Subject: [OpenSIPS-Users] Opensips ACK routing problem To: users@lists.opensips.org Date: Monday, 23 August, 2010, 14:42 Hi, using Opensips 1.6.2. We are using Opensips as outbound proxy using TLS just for final hop between UAC and Opensips. Other legs of the call will be udp. We have a test set up with Sipgate (but same occurs with other providers) where an incoming INVITE to the UAC via opensips results in 200ok generated by UAC and then the ACK coming back from sipgate server is routed over udp connection as opposed to tls, opensips seems to be ignoring the Route header in the ACK If the contact header has a sips uri opensips will route it correctly but some voip providers do not like sips uri in contact header and we are only using tls for the final hop so are not using a sips uri for contact header. Do we have our understanding of the spec wrong or should opensips be routing according to the route header or should it use the contact header. Opensips ip 172.230.135.190 UAC ip 81.13.94.206 U 217.10.79.23:5060 - 172.230.135.190:5060 ACK sip:9082...@81.13.94.206:59053 SIP/2.0. Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.226;branch=z9hG4bK6a423428. Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK6a423428;rport=5060. Route: sip:172.230.135.190;r2=on;lr=on;ftag=as126bf37e,sip:172.230.135.190:5061;transport=tls;r2=on;lr=on;ftag=as126bf37e. From: anonymous sip:anonym...@sipgate.co.uk;tag=as126bf37e. To: sip:00441519082...@sipgate.co.uk;tag=94gc3N42r9X5D. Contact: sip:anonym...@217.10.66.71. Call-ID: 181c48541a5ad00d04322ccc0752c...@sipgate.co.uk. CSeq: 102 ACK. Max-Forwards: 67. Content-Length: 0. X-hint: rr-enforced. . U 172.230.135.190:5060 - 81.13.94.206:59053 ACK sip:9082...@81.13.94.206:59053 SIP/2.0. Via: SIP/2.0/UDP 172.230.135.190;branch=z9hG4bK59c7.6f77cf2.3. Via: SIP/2.0/UDP 217.10.79.23:5060;rport=5060;received=217.10.79.23;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.226;branch=z9hG4bK6a423428. Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK6a423428;rport=5060. From: anonymous sip:anonym...@sipgate.co.uk;tag=as126bf37e. To: sip:00441519082...@sipgate.co.uk;tag=94gc3N42r9X5D. Contact: sip:anonym...@217.10.66.71. Call-ID: 181c48541a5ad00d04322ccc0752c...@sipgate.co.uk. CSeq: 102 ACK. Max-Forwards: 66. Content-Length: 0. X-hint: rr-enforced. ___ 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 ACK routing problem
Hi, using Opensips 1.6.2. We are using Opensips as outbound proxy using TLS just for final hop between UAC and Opensips. Other legs of the call will be udp. We have a test set up with Sipgate (but same occurs with other providers) where an incoming INVITE to the UAC via opensips results in 200ok generated by UAC and then the ACK coming back from sipgate server is routed over udp connection as opposed to tls, opensips seems to be ignoring the Route header in the ACK If the contact header has a sips uri opensips will route it correctly but some voip providers do not like sips uri in contact header and we are only using tls for the final hop so are not using a sips uri for contact header. Do we have our understanding of the spec wrong or should opensips be routing according to the route header or should it use the contact header. Opensips ip 172.230.135.190 UAC ip 81.13.94.206 U 217.10.79.23:5060 - 172.230.135.190:5060 ACK sip:9082...@81.13.94.206:59053 SIP/2.0. Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.226;branch=z9hG4bK6a423428. Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK6a423428;rport=5060. Route: sip:172.230.135.190;r2=on;lr=on;ftag=as126bf37e,sip:172.230.135.190:5061;transport=tls;r2=on;lr=on;ftag=as126bf37e. From: anonymous sip:anonym...@sipgate.co.uk;tag=as126bf37e. To: sip:00441519082...@sipgate.co.uk;tag=94gc3N42r9X5D. Contact: sip:anonym...@217.10.66.71. Call-ID: 181c48541a5ad00d04322ccc0752c...@sipgate.co.uk. CSeq: 102 ACK. Max-Forwards: 67. Content-Length: 0. X-hint: rr-enforced. . U 172.230.135.190:5060 - 81.13.94.206:59053 ACK sip:9082...@81.13.94.206:59053 SIP/2.0. Via: SIP/2.0/UDP 172.230.135.190;branch=z9hG4bK59c7.6f77cf2.3. Via: SIP/2.0/UDP 217.10.79.23:5060;rport=5060;received=217.10.79.23;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.226;branch=z9hG4bK6a423428. Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK6a423428;rport=5060. From: anonymous sip:anonym...@sipgate.co.uk;tag=as126bf37e. To: sip:00441519082...@sipgate.co.uk;tag=94gc3N42r9X5D. Contact: sip:anonym...@217.10.66.71. Call-ID: 181c48541a5ad00d04322ccc0752c...@sipgate.co.uk. CSeq: 102 ACK. Max-Forwards: 66. Content-Length: 0. X-hint: rr-enforced. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] 302 not being handled by failure route
Hi, we have a setup where we are using Opensips to redirect an invite from 3rd
party client registered at address 172.228.136.190:5060 to our Asterisk server
at 172.228.136.190:5062
We use the little bit of code below. The problem we have is if our Asterisk
server sends a 302 redirect(after original INVITE redirection) it is no longer
being handled by failure route! So it actually ends up being relayed along
rather than handled locally by uac_redirects.
If we did not use a 3rd party client and Asterisk was the target of the
original invite (so we did not have the code below as no need to redirect)
then when Asterisk sends a 302 everything works properly ie in failure route we
call uac_redirects etc
So the question is why is the 302 stuff not being called if the original invite
was redirected.
if (is_method(INVITE)) {
xlog(method invite in route 1 \n);
#lookup(location);
if(uri=~sip:@172.228.136.190:5060 )
{
xlog( Forwarding to Asterisk \n);
rewritehostport(172.228.136.190:5062);
xlog(rewritten RURI [$ru] \n);
t_relay();
exit;
}
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
}
Thanks
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Re: [OpenSIPS-Users] 302 not being handled by failure route
Please ignore this post , the bug is obvious i had the t_relay before the
t_branch_failure in the code
--- On Sat, 21/8/10, Nauman Sulaiman nauman762-h...@yahoo.co.uk wrote:
From: Nauman Sulaiman nauman762-h...@yahoo.co.uk
Subject: [OpenSIPS-Users] 302 not being handled by failure route
To: users@lists.opensips.org
Date: Saturday, 21 August, 2010, 14:42
Hi, we have a setup where we are
using Opensips to redirect an invite from 3rd party client
registered at address 172.228.136.190:5060 to our Asterisk
server at 172.228.136.190:5062
We use the little bit of code below. The problem we have is
if our Asterisk server sends a 302 redirect(after original
INVITE redirection) it is no longer being handled by failure
route! So it actually ends up being relayed along rather
than handled locally by uac_redirects.
If we did not use a 3rd party client and Asterisk was the
target of the original invite (so we did not have the code
below as no need to redirect)
then when Asterisk sends a 302 everything works properly ie
in failure route we call uac_redirects etc
So the question is why is the 302 stuff not being called if
the original invite was redirected.
if (is_method(INVITE)) {
xlog(method invite in route 1 \n);
#lookup(location);
if(uri=~sip:@172.228.136.190:5060 )
{
xlog( Forwarding to Asterisk \n);
rewritehostport(172.228.136.190:5062);
xlog(rewritten RURI [$ru] \n);
t_relay();
exit;
}
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
}
Thanks
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[OpenSIPS-Users] Fixed nated contact problem when using TLS
Hi, we have a setup where we are using OpenSIPS as an out/inbound proxy. Te connection to the UAC is a TLS (port 5061) one and the connection fron OpenSIPS to third party voip provider is UDP (5060). We have the TLS connection working and the UAC can successfully register with the provider. OpenSIPS is record routing twice once for the TLS route and another for the UDP. It seems to be bridging too. The problem we have is when we have an incoming invite (from voip provider) the contact header returned in the 200 OK from the UAC to OpensIPS has a private address say 192.168.1.20:5061, when opensips bridges this and the fixed nated contact is applied the correct external ip is sent 172.175.130.156:51056 but the port has changed. So the ACK when sent from provider back to OpenSIPS has the above address as req URI and then opensips can't route it back to the UAC because of the incorrect port. So main issue is OpenSIPS can't get the ACK back to UAC to establish the dialog. When there was no bridging (ie no TLS) the port was 5060 and fixed nated contact mapped it like this 172.175.130.156:5060 ie didn't change the port How to get round this? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Fixed nated contact problem when using TLS
it turns out we just need to rewrite the ipadress and not the port. There was some discussion on here some time back of a variant of fix_nated_contact_f that would allow a rewrite of ipaddress only, did this get implemented? If not is there a patch. Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS TLS client certificates
Hi, using version 1.6.2. I created the rootCA and server certificate using the opensipsctl tls rootCA and opensipsctl tls userCERT commands but what file am i supposed to give to the client, which is custom softphone UA? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Location info when using OpenSIPS as outbound proxy
Hi Bogan
Thanks for the replies, we are basically registering with a third party
provider and the save location on the Opensips isn't really to keep track of
registration state, its just to keep track of the location of the UAC
irrespective of whether its registered or not.
we now have the following code in route(1) of sample script
if (method==REGISTER)
{
xlog(saving AoR non local domain \n);
if (!save(location,mr))
{
xlog(couldn't save location in route 1 \n);
sl_reply_error();
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
we are not challenging the UAC if the domain is not the opensips proxy, we just
let it go through and save location. We take it if you issue a challenge here
you need a copy of the credentials of the UAC on the opensips proxy, as we will
not have this, is the above valid?
--- On Sun, 1/8/10, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Location info when using OpenSIPS as outbound
proxy
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Sunday, 1 August, 2010, 19:14
Hi Andrew,
what you are doing is dangerous :)
Even if the contacts+expires values will be properly
extracted from the
reply, other data will be bogus, like:
- user agent
- socket info (only if you do change it
before relaying the register)
- path info
- received value + branch flags (if some
forking is done)
Regards,
Bogdan
Andrew Pogrebennyk wrote:
On 01.08.2010 20:37, Bogdan-Andrei Iancu wrote:
2) ideally, for an outbound proxy, you should do
the registration
processing at reply time, once the main registrar
accepted the
registration and eventually made all the changes
over it. But right
now
opensips does not accept registration processing
for replies.
Just in case - some time ago I did something like:
onreply_route[3] {
# Here we handle REGISTER replies
xlog(L_INFO, [$mi] [$rs
$rr]\n);
if (status=~200) {
route(3);
};
route[3] {
# workaround for location saving
xlog(L_INFO, saving
location\n);
save(location,0x02);
}
0x02 - do not generate a SIP reply to the current
REGISTER request.
--
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro
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[OpenSIPS-Users] What does save(location) actually do?
Hi, we are using opensips 1.6.2. We are using OpenSIPS as an outbound proxy in
open relay mode for time being. For all register requests to provider beyond
opensips we wish to save the AoR. So we are calling save(location).
This for local routing for 302 later.
However the very act of calling save(location) is causing a 401 to go out??
We are not using authentication on the opensips outbound proxy. There is
nothing in the docs to suggest this.
If we remove the save then the register completes fine, code below
added to route[1] of sample script
called from here
if (!uri==myself)
{
xlog( Route 1 due to non local domain \n);
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
if (is_method(INVITE)) {
xlog(method invite in route 1 \n);
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
}
if (method==REGISTER)
{
xlog(saving AoR non local domain \n);
save(location);
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
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[OpenSIPS-Users] Contact - private IP problem
Hi, we are using OpenSIPS 1.6.2 as an outbound proxy, we have registered with Voipfone to test but the same problem occurs with other Voip providers Once registered we can't receive incoming calls, if we use STUN on the UAC then we are ok. However we would prefer to handle the nat traversal on the proxy. We have implemented all the code in Flavio's book. However the contact header sent from the proxy to the Voipfone server has a private ip although it has the nat=yes appended. So something is working please see REGISTER trace below. We think we are missing something but unsure what. UAC-Proxy REGISTER sip:voipfone.co.uk SIP/2.0. Via: SIP/2.0/UDP 192.168.0.2;rport;branch=z9hG4bKeyaZHZ6DrXptj. Max-Forwards: 70. From: sip:30146...@voipfone.co.uk;tag=44U5mK3gB9ZvH. To: sip:30146...@voipfone.co.uk. Call-ID: 102ce92b-166c-122e-a99f-37fea39c3827. CSeq: 134125887 REGISTER. Contact: sip:30146...@192.168.0.2:5060. Expires: 600. User-Agent: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, NOTIFY. Supported: path. Authorization: Digest username=30146027, realm=asterisk, nonce=hhvuahjm, algorithm=MD5, uri=sip:voipfone.co.uk, response=4d8dbfef$ Content-Length: 0. . Proxy-Voipfone REGISTER sip:voipfone.co.uk SIP/2.0. Via: SIP/2.0/UDP 178.235.138.190;branch=z9hG4bK23c.505fb9b2.0. Via: SIP/2.0/UDP 192.168.0.2;received=88.148.7.122;rport=5060;branch=z9hG4bKeyaZHZ6DrXptj. Max-Forwards: 69. From: sip:30146...@voipfone.co.uk;tag=44U5mK3gB9ZvH. To: sip:30146...@voipfone.co.uk. Call-ID: 102ce92b-166c-122e-a99f-37fea39c3827. CSeq: 134125887 REGISTER. Contact: sip:30146...@192.168.0.2:5060;nat=yes. Expires: 600. User-Agent: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, NOTIFY. Supported: path. Authorization: Digest username=30146027, realm=asterisk, nonce=hhvuahjm, algorithm=MD5, uri=sip:voipfone.co.uk, response=4d8dbfef$ Content-Length: 0. Voipfone-Proxy SIP/2.0 200 OK. Via: SIP/2.0/UDP 178.235.138.190;branch=z9hG4bK23c.505fb9b2.0;received=178.235.138.190;rport=5060. Record-Route: sip:195.189.173.10:5060;lr=on. Via: SIP/2.0/UDP 192.168.0.2;received=88.148.7.122;rport=5060;branch=z9hG4bKeyaZHZ6DrXptj. From: sip:30146...@voipfone.co.uk;tag=44U5mK3gB9ZvH. To: sip:30146...@voipfone.co.uk. Call-ID: 102ce92b-166c-122e-a99f-37fea39c3827. CSeq: 134125887 REGISTER. Contact: sip:30146...@192.168.0.2:5060;nat=yes;expires=60. Expires: 60. Date: Fri, 30 Jul 2010 15:16:15 GMT. Min-Expires: 60. User-Agent: Voipfone. Content-Length: 0. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Location info when using OpenSIPS as outbound proxy
Hi, using opensips 1.6.2. We are using Opensips as an outbound proxy which also does local routing of 302 responses. IF we have a client UAC1 which registers with a provider using OpenSIPS only as an outbound proxy is it possible to save the location info of UAC1, reason being if there is a call to UAC2 for instance from the provider which is redirected with 302, then our Opensips can handle the redirection because it knows where UAC1 is without having it to go back to the provider's proxy server. Maybe we need to do 2 registrations one with OpenSIPS and one with the provider but if we can do this with just the one it would be a neat solution. Secondly if we do save the location info of UAC1 at the opensips proxy, what url would need to go in contact field of the 302, would it be u...@provider.com - opensips needs soemthing ot be able to send to UAC1 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] 302 redirect problem
Hi , we have a set up where we are using a SIP proxy( OpenSIPS) to handle a 302
redirect locally but it doesn't seem to be working with voip providers here is
the call flow. Is it legal, if so we'll bother you with a SIP trace. The 302 is
handled locally by the OpenSIPS proxy and generates a new call leg but can't
establish a dialog with Sipgate or Voipfone
INVITE INVITE
VoipFone Server - Proxy UAC1
Trying
VoipFone Server Proxy
180 Ring
180 Ring - UAC1
-- Proxy 302 Redir
-- UAC1
INVITE
--- UAC2
180 Ring
UAC2
180 Ring
VoipFone Server --
CANCEL
200 OK
200 OK --- UAC2
THe VoipFone server sends a cancel straight after the 180 ring, Sipgate
doesn't do this however it never sends an ACK for the 200K, anyway both
don't like what we are doing.
We need to do the 302 redirect locally on our proxy as not all Voip providers
support it so we can't let it go all the way back. Hope its clear what we are
trying to, and is there any way forward.
Here is the code in our opensips.cfg file
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
get_redirects(3:1);
t_relay();
}
Thanks
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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] 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();
$du = sip:sipgate_ip_goes_here:port;
t_relay();
exit;
so instead of doing if (!save(location)) etc
-Laszlo
2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk
Hi, We have Asterisk 1.6.2.9 and Opensips 1.6.2 in the same Linux box and we
want
to have OpenSIPS as the out/inbound proxy for Asterisk when running as UAC.
We have Asterisk registered with Sipgate ok when sending REGISTER directly to
the Sipgate
domain but would like to send the Register via OpenSIPS so it's in the path for
incoming invites
Opensips.cfg as follows
listen=udp:192.168.0.20:5060
register code: we have removed the authentication for now to get this working
first
if (is_method(REGISTER))
{
if (!db_check_to())
{
sl_send_reply(403,Forbidden auth ID);
exit;
}
if (!save(location))
sl_reply_error();
exit;
}
we have astersik sip.conf file as follows
[general]
port=5062
bindaddr=0.0.0.0
context=default
outboundproxy=192.168.0.20
outboundproxyport=5060
register =
username:sec...@sipgate.co.uk/username
[username]
type=friend
context=incoming
username=username
secret=secret
host=sipgate.co.uk
host=dynamic
fromdomain=sipgate.co.uk
insecure=port,invite
nat=yes
disallow=all
allow=alaw
canreinvite=no
However the REGISTER messages are sent by Astersik but ther is no response fron
OpenSIPS and it is not routing them either . Is it because they are both on the
same IP?
We do have then on different ports so though it would be ok
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Re: [OpenSIPS-Users] Opensips as outbound proxy for asterisk client
Hi, making progress
we now have this so the REGISTER is passed to the Sipgate proxy transparently
it could be any proxy server so
if (is_method(REGISTER))
{
##$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] OpenSIPS as UAC and Proxy
Hi, we have a design requirement where we have a 'home network' which acts a UA (temporary) and will proxy INVITE etc to mobile SIP endpoints. The mobile SIP endpointswont be registered and will need to be woken up by the home network using non SIP mechanism. They will inform the home network of their location once woken and the home network should then proxy the INVITE to them. We are considering usng OpenSIPS for this but would like to know if it possible for us to do this. In order to receive the initial INVITE the OpenSIPS server will need to register on behalf of the mobile UA's and then once they are woken and their location known it needs to proxy the SIP messages to them. Hope its clear, asterisk we don't think works for this as it creates another call leg rather than proxying the messages. Can OpensSIPS do this? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Run custom PHP script to send SMS upon receiving invite
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 messagewhen 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
