Re: [OpenSIPS-Users] pua_send_publish feature or bug?
Hi Damien, The behavior that you get is the correct one. If you publish with MI the presence server will consider as if there is another device publishing for the same account. So when sending Notify, it will aggregate what you have sent with what it has received from the phone. Regards, Anca ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] T38
Just created a Google+ page for T38. I am hoping to start a circle for people who love to chat and help with SIP T38. https://plus.google.com/117160247884486247346 Have fun! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Problem with hanging dialogs
Hi, I'm having problem with hanging dialogs. Nov 23 20:46:18 V0P034-VoIP-LB /usr/local/sbin/opensips[16945]: New request - M=INVITE RURI=sip:ora517xxx...@lb-gw.sip.int.ccig.pl F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.0.130.161 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:18 V0P034-VoIP-LB /usr/local/sbin/opensips[16946]: Reply - S=100 R= D=Trying F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.18.2.1 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:22 V0P034-VoIP-LB /usr/local/sbin/opensips[16946]: Reply - S=183 R= D=Session Progress F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.18.2.1 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:22 V0P034-VoIP-LB /usr/local/sbin/opensips[16943]: Reply - S=180 R= D=Ringing F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.18.2.1 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:42 V0P034-VoIP-LB /usr/local/sbin/opensips[16946]: Reply - S=200 R= D=OK F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.18.2.1 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:42 V0P034-VoIP-LB /usr/local/sbin/opensips[16944]: New request - M=ACK RURI=sip:517xx@10.18.2.1:5060 F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.0.130.161 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:42 V0P034-VoIP-LB /usr/local/sbin/opensips[16945]: New request - M=INVITE RURI=sip:517xx@10.18.2.1:5060 F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.0.130.161 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:42 V0P034-VoIP-LB /usr/local/sbin/opensips[16945]: Reply - S=200 R= D=OK F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.18.2.1 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:42 V0P034-VoIP-LB /usr/local/sbin/opensips[16944]: New request - M=ACK RURI=sip:517xx@10.18.2.1:5060 F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.0.130.161 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:42 V0P034-VoIP-LB /usr/local/sbin/opensips[16946]: New request - M=INVITE RURI=sip:517xx@10.18.2.1:5060 F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.0.130.161 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:46:42 V0P034-VoIP-LB /usr/local/sbin/opensips[16945]: Reply - S=100 R= D=Trying F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.18.2.1 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:48:43 V0P034-VoIP-LB /usr/local/sbin/opensips[16948]: failure_route(1) - S=100 R=sip:517xx@10.18.2.1:5060 D=Trying F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.0.130.161 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 Nov 23 20:48:43 V0P034-VoIP-LB /usr/local/sbin/opensips[16943]: New request - M=ACK RURI=sip:517xx@10.18.2.1:5060 F=sip:519xx@10.0.130.161 T=sip:ora517xxx...@lb-gw.sip.int.ccig.pl IP=10.0.130.161 ID=704b35af1ca1a25171204e963b815e1d@10.0.130.161:5060 My config file is following: http://pastebin.com/JYqp7mmZ Call flow is following Asterisk -> Opensips -> Patton SmartNode gateway. Once every minute or two, I have ~200 active calls to ora destination. I have totally no idea where problem lies. Can anybody guide me? Regards, Marcin ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2B Not routing 200 OK for BYE
Hi Ryan, My first guess is that it is something wrong with the 200 OK you receive for the BYE - could you post somewhere the trace (or send it offline to me) ? Regards, Bogdan On 11/23/2011 07:52 PM, Ryan Bullock wrote: I am use the B2B modules with the topology hiding scenario and periodically see the following error in my opensips log: ERROR:b2b_entities:b2b_tm_cback: No dialog found reply 200 for method BYE I did a capture to find out what was happening and it appears that after receiving a BYE opensips will correctly forward the BYE but not the subsequent 200 OK to the BYE (and I get the error message above). This causes the side that sent the BYE to retransmit and I can then see these retransmits in the main opensips route. Currently I have the main route sending a 200 OK for any BYE with a totag to catch these retransmits. I have also seen a few occurrence where if a 200 OK for an INVITE happens right before a BYE from the other side that the ACK for the 200 OK will also not be correctly routed using the B2B. It seems like the B2B is tearing down the session a bit to quickly, is there any way to adjust how long a B2B session will linger? Regards, Ryan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer OpenSIPS solutions and "know-how" ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] B2B Not routing 200 OK for BYE
I am use the B2B modules with the topology hiding scenario and periodically see the following error in my opensips log: ERROR:b2b_entities:b2b_tm_cback: No dialog found reply 200 for method BYE I did a capture to find out what was happening and it appears that after receiving a BYE opensips will correctly forward the BYE but not the subsequent 200 OK to the BYE (and I get the error message above). This causes the side that sent the BYE to retransmit and I can then see these retransmits in the main opensips route. Currently I have the main route sending a 200 OK for any BYE with a totag to catch these retransmits. I have also seen a few occurrence where if a 200 OK for an INVITE happens right before a BYE from the other side that the ACK for the 200 OK will also not be correctly routed using the B2B. It seems like the B2B is tearing down the session a bit to quickly, is there any way to adjust how long a B2B session will linger? Regards, Ryan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Transfer problem with Opensips as a load balancer
Hi Schneur, What you have to do is to change the way you distribute the call among the asterisk boxes in such a way that all calls in which a user is involved to be on the same box (so that the transfers will work). How to do that? with a mixed routing logic. When you receive a new call, do: - check if caller or callee are already involved into an existing call on a certain box. if so, route to that box - default is to do LB as you do now. For the check part, you need to use the dialog module (to be dialog stateful), set in some dialog variables the caller / callee / box (to be remembered later) and query via get_dialog_info() function - http://www.opensips.org/html/docs/modules/1.7.x/dialog.html#id294051 Regards, Bogdan On 11/23/2011 06:48 PM, Schneur Rosenberg wrote: I'm using Opensips as a Load balancer and as a registrar, so basically all phones are registered to the Opensips, all Incoming calls hit the opensips server which forwards the call to asterisk with load balancing, asterisk decides what to do with the call ie IVR voicemail etc and if the call needs to be sent to a phone asterisk will send it back to opensips and opensips will send it to the phone. Outgoing calls are sent to asterisk via load balancing and asterisk decides how to terminate the call. This setup helps me load balance all calls and also removes the registrar load from asterisk which does not handle registrations fine when there are approx 300 peers on my asterisk system. My problem is that sometimes when I do a transfer I get back from asterisk "SIP/2.0 481 Call leg/transaction does not exist.". The test call I've done was done by calling from phone 1 a phone number which hits our system, so what happened is phone invited opensips to the DID, opensips sent the call to Asterisk server 1, then the DID called in and opensips sent it to Asterisk server 2, Asterisk server 2 saw that this did should ring on a phone so it sent it back to opensips which properly terminated the call to phone 2, then phone 1 wanted to transfer call to a outside phone, so it sent a invite to opensips with the phone number to call, opensips sent call to Asterisk server 2, then when user on phone 1 hit transfer, phone sent a refer to Asterisk 1, and asterisk 1 retuned a NOTIFY with Subscription-state: terminated;reason=noresource. and SIP/2.0 481 Call leg/transaction does not exist. Can anyone please help me solve this problem. thank you S. Rosenberg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer OpenSIPS solutions and "know-how" ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 1.7.1 minor release
Hi all, *OpenSIPS 1.7.1* - a minor release on the 1.7 main branch - was release today. This release contains only bug fixing (additional to 1.7.0) - crashes, malfunctions and compliance issues - but no functionality, scripting or interfacing were changed. *OpenSIPS 1.7.1* contains important fixes in critical modules (like dialog and registration support) and in core part - it is highly recommended to upgrade - **OpenSIPS* 1.7.1* is now available for download on project web site and SF download system. The full Changelog is available http://opensips.org/pub/opensips/1.7.1/src/ChangeLog To get the *OpenSIPS 1.7.1* version, see the download page - http://www.opensips.org/Resources/Downloads Many thanks to all people who contributed with bug reports, troubleshooting and debugging, fixings and packaging. Best regards, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer OpenSIPS solutions and "know-how" ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Users Digest, Vol 40, Issue 81
Sarò assente fino al 25 Novembre compreso. Per urgenze rivolgersi direttamente ad assiste...@longwave.eu o chiamare lo 0522375500. Saluti I will be out of office till November 25th 2011. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Transfer problem with Opensips as a load balancer
I'm using Opensips as a Load balancer and as a registrar, so basically all phones are registered to the Opensips, all Incoming calls hit the opensips server which forwards the call to asterisk with load balancing, asterisk decides what to do with the call ie IVR voicemail etc and if the call needs to be sent to a phone asterisk will send it back to opensips and opensips will send it to the phone. Outgoing calls are sent to asterisk via load balancing and asterisk decides how to terminate the call. This setup helps me load balance all calls and also removes the registrar load from asterisk which does not handle registrations fine when there are approx 300 peers on my asterisk system. My problem is that sometimes when I do a transfer I get back from asterisk "SIP/2.0 481 Call leg/transaction does not exist.". The test call I've done was done by calling from phone 1 a phone number which hits our system, so what happened is phone invited opensips to the DID, opensips sent the call to Asterisk server 1, then the DID called in and opensips sent it to Asterisk server 2, Asterisk server 2 saw that this did should ring on a phone so it sent it back to opensips which properly terminated the call to phone 2, then phone 1 wanted to transfer call to a outside phone, so it sent a invite to opensips with the phone number to call, opensips sent call to Asterisk server 2, then when user on phone 1 hit transfer, phone sent a refer to Asterisk 1, and asterisk 1 retuned a NOTIFY with Subscription-state: terminated;reason=noresource. and SIP/2.0 481 Call leg/transaction does not exist. Can anyone please help me solve this problem. thank you S. Rosenberg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] LRN dips with Dynamic Routing
Hi Jeff,
Well, according to RFC3261, a Contact hdr must carry a valid SIP URI -
now, in dip lookups, the answer is added as params to the SIP URI or to
the CT SIP hdr...depending...
If you uac_redirect does not server your purpose (like answer in in CT
hdr params), you can access the hdr directly like getting params :
$(ct.fields(params))
Regards,
Bogdan
On 11/23/2011 03:25 PM, Jeff Pyle wrote:
Bogdan,
I don't think the uac_redirect module in this case is helpful. The
Contact data that comes back from an LRN DIP's 302 isn't a real SIP
URI, but rather just some routing data that happens to be using a
302's Contact field as a transport mechanism.
Kent,
Sorry for the late reply... I do not. I use string transformations
to yank out the portions I need. Something like this in the
onreply_route from a 302 dip:
$var(lrnct) = $ct;
if !($(var(lrnct){param.value,rn}) == '') {
$var(call_lrn1) = $(var(lrnct){param.value,rn});
# more processing to clean up any leading 1's or +1s
}
I do not use the drouting module because my config is too old. I use
a combination of the lcr module to load gateways for a particular
carrier, or if a carrier has only one IP/hostname, I pull it directly
from a DB into an AVP and route to it.
I take the LRN data from the DIP along with some other items
(jurisdiction/ani/etc) and feel it to a Perl script to do the actual
LCR carrier selection. I keep all my carriers' rates in separate DB
tables and use mysql stored functions to normalize the rate lookup
into a standard format used by a while loop in the Perl script. The
rate function name that gets called is another usr_preference per carrier.
I certainly don't pretend this is the most efficient way, but it does
all our needs better than anything else I have been able to come up with.
- Jeff
On Tue, Nov 22, 2011 at 9:27 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:
Hi Kpirlo,
When sending the call to the dip provider, use a failure route in
order to catch the 3xx reply you get back. In the failure route,
use the uac_redirect module with the get_redirects() function
(http://www.opensips.org/html/docs/modules/1.7.x/uac_redirect.html#id250367)
in order to extract the redirect contacts from the reply and push
them as new destinations.
Regards,
Bogdan
On 11/20/2011 08:04 PM, Kpirlo wrote:
We are currently using the Dynamic routing module for our least
cost routing.
Now we are looking at implementing an LRN dipping service, where
we will send the call to a dip provider first and receive a 302
redirect back which will have the LRN returned in the contact
header as "rn=" if the number has been ported or will include
";npdi" in the contact header if it has not been ported.
Im asking for any advice anyone has on how to implement this and
how it could work with dymanic routing to choose the route based
on rn if available, but actually send the call using the original
"to" number.
Thank you in advance for any help.
Kent
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OpenSIPS Founder and Developer
OpenSIPS solutions and "know-how"
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Re: [OpenSIPS-Users] BYE on 180 causing dialog loop
Hello, I committed a fix for this in trunk and 1.7 branch, so while using topology hiding, requests should be properly routed even when the dialog is in the early state. Regards, Vlad Paiu OpenSIPS Developer On 11/22/2011 04:18 PM, Saul Ibarra Corretge wrote: Hi Bogdan, On Nov 22, 2011, at 3:17 PM, Bogdan-Andrei Iancu wrote: Hi Saul, Just to clarify - while the call is still in early stage, the control is done at transaction level (the INVITE transaction) - if transaction is successful (200OK) -> call established; if transaction fails (negative reply) -> call fails. So, the dialog module is not interested in the CANCEL -> it will wait to see the feedback on the INVITE level, like the 487 reply (as a result of the CANCEL being accepted). The BYE (instead of CANCEL) works in a similar way - the dialog module will simply wait to see what will happen with the INVITE. So, from standard dialog state, the dialog module does not care about the CANCELs or BYEs in early state. Of course, things are a bit different when using topology hiding with dialog module - there you have the "topo hide" the BYE also ;).and this needs to be fixed Thanks for the detailed explanation! Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] LRN dips with Dynamic Routing
Bogdan,
I don't think the uac_redirect module in this case is helpful. The Contact
data that comes back from an LRN DIP's 302 isn't a real SIP URI, but rather
just some routing data that happens to be using a 302's Contact field as a
transport mechanism.
Kent,
Sorry for the late reply... I do not. I use string transformations to
yank out the portions I need. Something like this in the onreply_route
from a 302 dip:
$var(lrnct) = $ct;
if !($(var(lrnct){param.value,rn}) == '') {
$var(call_lrn1) = $(var(lrnct){param.value,rn});
# more processing to clean up any leading 1's or +1s
}
I do not use the drouting module because my config is too old. I use a
combination of the lcr module to load gateways for a particular carrier, or
if a carrier has only one IP/hostname, I pull it directly from a DB into an
AVP and route to it.
I take the LRN data from the DIP along with some other items
(jurisdiction/ani/etc) and feel it to a Perl script to do the actual LCR
carrier selection. I keep all my carriers' rates in separate DB tables and
use mysql stored functions to normalize the rate lookup into a standard
format used by a while loop in the Perl script. The rate function name
that gets called is another usr_preference per carrier.
I certainly don't pretend this is the most efficient way, but it does all
our needs better than anything else I have been able to come up with.
- Jeff
On Tue, Nov 22, 2011 at 9:27 AM, Bogdan-Andrei Iancu wrote:
> **
> Hi Kpirlo,
>
> When sending the call to the dip provider, use a failure route in order to
> catch the 3xx reply you get back. In the failure route, use the
> uac_redirect module with the get_redirects() function (
> http://www.opensips.org/html/docs/modules/1.7.x/uac_redirect.html#id250367)
> in order to extract the redirect contacts from the reply and push them as
> new destinations.
>
> Regards,
> Bogdan
>
>
> On 11/20/2011 08:04 PM, Kpirlo wrote:
>
> We are currently using the Dynamic routing module for our least cost
> routing.
>
> Now we are looking at implementing an LRN dipping service, where we will
> send the call to a dip provider first and receive a 302 redirect back which
> will have the LRN returned in the contact header as "rn=" if the number has
> been ported or will include ";npdi" in the contact header if it has not
> been ported.
>
> Im asking for any advice anyone has on how to implement this and how it
> could work with dymanic routing to choose the route based on rn if
> available, but actually send the call using the original "to" number.
>
> Thank you in advance for any help.
>
> Kent
>
>
> ___
> Users mailing
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS solutions and "know-how"
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] dp_reload
Hi Bogdan That's clear. Thanks for your support. I think I'll try to create a mi datagram interface and execute the dp_reload command remotely. Regards, Carmelo -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/dp-reload-tp7023627p7024030.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Users Digest, Vol 40, Issue 80
Sarò assente fino al 25 Novembre compreso. Per urgenze rivolgersi direttamente ad assiste...@longwave.eu o chiamare lo 0522375500. Saluti I will be out of office till November 25th 2011. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dp_reload
Hi Carmelo, This command is particular to dialplan module. On the other hand, most of the modules that load and cache data from DB do have a similar command, like Dynamic Routing module has dr_reload, permissions module has address_reload, nathelper has nh_reload, etc... You need to check the documentation of the modules you are using. Regards, Bogdan On 11/23/2011 11:56 AM, wüber wrote: Hi Bogdan Thank you very much for your answer. Could you please tell me if this is the only possibility to load the new translation rules (or in general the new database info) from the database? Regards, Carmelo -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/dp-reload-tp7023627p7023852.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer OpenSIPS solutions and "know-how" ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dp_reload
Hi Bogdan Thank you very much for your answer. Could you please tell me if this is the only possibility to load the new translation rules (or in general the new database info) from the database? Regards, Carmelo -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/dp-reload-tp7023627p7023852.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dp_reload
Hi,
dp_reload is not a script function, but a MI (Management Interface)
function - such functions are called from outside opensips.
You can trigger the functions like:
opensipsctl fifo do_reload
Regards,
Bogdan
On 11/23/2011 10:27 AM, wüber wrote:
Hi all
I'm using a database with some translation rules in the dialplan table. How
can I reload the updated information in the db without restarting Opensips?
I tried
dp_reload();
dp_translate("1","$rU/$rU");
but dp_reload() in the opensips.cfg seems to be not allowed ...
Any suggestion?
Thanks!
--
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Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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Re: [OpenSIPS-Users] using dialog based topology_hiding in failover scenario
Hi Jayesh, The dialog topology hiding works at dialog level and you cannot use it in a per-branch manner. So, you cannot set or not topo hiding for some branches. Solutions are: 1) simply do topo hiding when routing, so you will do it for all branches. 2) set a separate opensips instance (or spiral on the same opensips) to do topo hiding. Regards, Bogdan On 11/23/2011 09:57 AM, Jayesh Nambiar wrote: Hi All, I tested the topology_hiding function in dialog module and it works well. Now my scenario requires that I use topology_hiding function based on the carrier where the call is supposed to go. And I use failure_route to failover between multiple carriers. So the condition is, if 1st carrier requires topology_hiding I enable it and route the call and if that call fails, the next carrier might not need topology_hiding enabled so I need to somehow undo the topology_hiding that I called while routing to the first carrier. Moreover, if I call the topology_hiding again for the second carrier, the contact header gets malformed since the contact header is appended again and the call fails because of invalid contact header. One possible solution I thought of was calling the topology_hiding function in the branch_route, but opensips would not start and logs an error saying "Command cannot be used in the block". Anyone with any possible workaround for this? Any help is appreciated !! Thanks, --- Jayesh ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer OpenSIPS solutions and "know-how" ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] dp_reload
Hi all
I'm using a database with some translation rules in the dialplan table. How
can I reload the updated information in the db without restarting Opensips?
I tried
dp_reload();
dp_translate("1","$rU/$rU");
but dp_reload() in the opensips.cfg seems to be not allowed ...
Any suggestion?
Thanks!
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