[OpenSIPS-Users] 401 and 407 in REPLY ROUTE

. My first problem is that 407 and 401 go to the REPLY Route and not the FAILURE Route. Should they not go to FAILURE route? What am I missing? Thanks, Ali Pey ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman

Re: [OpenSIPS-Users] TLS - Certificate Validation Failure error on SIP Phones - OpenSIPS version 1.11.5

ertificate. Apparently, eyeBeam and Zoiper cannot or do not handle wild cards (*) in a certificate. Best regards, Ali Pey On Fri, Apr 8, 2016 at 10:48 AM, Rodrigo Pimenta Carvalho <pime...@inatel.br > wrote: > Hi. > > > I got the same problem in softphone ZOIPER. > > I

Re: [OpenSIPS-Users] TLS - Certificate Validation Failure error on SIP Phones - OpenSIPS version 1.11.5

Hello Hamid, The parameters below don't have any effects. In my scenario, the sip phones are rejecting the tls connection by saying "Certificate Validation Failure". Neither of parameters below had any effects. Anyone else has any idea what I need to look for? Regards, Ali Pey On

[OpenSIPS-Users] TLS - Certificate Validation Failure error on SIP Phones - OpenSIPS version 1.11.5

nection, they reject the connection saying "Certificate Validation Failure". My certificate is valid and works fine on the https website. What am I missing? What should I look for? Regards, Ali Pey ___ Users mailing list Users@lists.opensips.org

Re: [OpenSIPS-Users] Connection IP in SDP is messed up after RTPPRoxy Offer

Hello Aqs, Yes, that was my problem. I didn't need fix_nated_sdp. Thanks, Ali On Fri, Feb 26, 2016 at 4:23 PM, Aqs Younas wrote: > Calling fix_nated_sdp() and rtpproxy_offer() after one another does not > make sense since both do same things in some cases. Make sure you

Re: [OpenSIPS-Users] Connection IP in SDP is messed up after RTPPRoxy Offer

Hello Alex, Good to hear from you. Yes, I did have fix_nated sdp and that was causing conflict. Copy and paste issues:) Thanks, Ali On Fri, Feb 26, 2016 at 3:39 PM, Alex Balashov wrote: > Ali, > > Is there any danger that you are calling rtpproxy_offer() twice,

[OpenSIPS-Users] Connection IP in SDP is messed up after RTPPRoxy Offer

instead of replacing the external IP, it adds the internal IP to the end of connection line in the SDP. I'm using opensips version 1.11.5. It works with most clients but this happens time to time. Has anyone experienced this problem? How can I fix it? Regards, Ali Pey

Re: [OpenSIPS-Users] apt.opensips.org not reachable

tory for some period, so if somebody could contact me > about that, that would be great. > > > > *---* > > *Best regards* > > > > Frederik Bjerggard Nielsen > > Technical Specialist > > > > *Firstcom A/S* > > > > *Fra:* users-boun..

[OpenSIPS-Users] apt.opensips.org not reachable

ttp://apt.opensips.org/> http://apt.opensips.org <https://contactmonkey.com/api/v1/tracker?cm_session=77e8c2b6-04e1-4b8a-adab-352f7984cb88_type=link_link=1d66d86e-2591-4d04-9f0a-cf07e4310a99_destination=http://apt.opensips.org> Regards, Ali Pey ___ Users

Re: [OpenSIPS-Users] Dialplan module and priority

...@gmail.com wrote: Hi, Swapping the priority works the way you want ? I have a feeling this makes sense (just like an ACL or firewall rules) ^888444* should get called before ^888* . On Thu, Jul 16, 2015 at 5:47 PM, Ali Pey ali...@gmail.com wrote: Hello, Let's say I have the two following

[OpenSIPS-Users] Dialplan module and priority

. If I use dp_translate for 8884441234, it still matches rule 1 and that's not good. It should match rule 2. Is this a bug or expected behaviour? Is there a way I can work around this? Thanks, Ali Pey ___ Users mailing list Users@lists.opensips.org http

Re: [OpenSIPS-Users] DRouting and Range of numbers

Hi Razvan. Thank you for the response. Regards, Ali Pey On Fri, Mar 27, 2015 at 4:29 AM, Răzvan Crainea raz...@opensips.org wrote: Hi, Ali! Unfortunately drouting works only prefix based, so unless you write specific rules for the 5 numbers (i.e. add rules for each number from

[OpenSIPS-Users] DRouting and Range of numbers

Hello, Is it possible to have a rule with a range of numbers in Dynamic routing? For instance I want 8881231231 to 5 to be routed to a specific gw. Can I do this with one rule only? I don't want 8881231236 to 9 to be routed to that gateway. Thanks, Ali Pey

Re: [OpenSIPS-Users] Block user from registration

You can also consider using the permissions module. If the src IP is there, then you can accept the request, otherwise, drop the message. Regards, Ali Pey On Wed, Dec 31, 2014 at 1:30 PM, Duane Larson duane.lar...@gmail.com wrote: My logic saves the user that is registering into the location

Re: [OpenSIPS-Users] RTP Proxy and Re-Invites

Thank you Jeff. Don't I need to do unforce before doing a new offer? Why? Regards, Ali Pey On Thu, Nov 13, 2014 at 8:30 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Ali, This is what I use within loose_route() to handle rtpproxy. In my particular case I'm bridging between two interfaces

Re: [OpenSIPS-Users] RTP Proxy and Re-Invites

Hi Razvan, Thank you for your response and it makes sense. I will search for a work around for media re-negotiation rejection and will post my results here. Best regards, Ali Pey On Fri, Nov 14, 2014 at 4:45 AM, Răzvan Crainea raz...@opensips.org wrote: Hi, Ali! The reINVITES should

[OpenSIPS-Users] RTP Proxy and Re-Invites

Hello, What's the best way of handling rtpproxy with re-invites? Should I do unforce and then offer/answer? What if the re-invite gets rejected? Any help appreciated. Thanks, Ali Pey ___ Users mailing list Users@lists.opensips.org http

Re: [OpenSIPS-Users] OpenSIPS/RTpProxy BridgeMode after failure route

Hello Răzvan Salman, Thank you for your responses. I was able to fix it by moving rtpproxy_offer to branch route instead of having it in the main route. In failure route, I only needed to do unforce. Regards, Ali Pey On Thu, Oct 9, 2014 at 3:31 AM, Răzvan Crainea raz...@opensips.org wrote

Re: [OpenSIPS-Users] OpenSIPS/RTpProxy BridgeMode after failure route

Hello Salman, Can you please elaborate on how you got this working? I have the same problem and can't get it to work. In failure route, I do: unforce_rtp_proxy() Then when I have a new destination, I do: rtpproxy_offer(rocie); However, I end up with messed up SDP, in my second invite. It

Re: [OpenSIPS-Users] engage_rtp_proxy and unforce_rtp_proxy

Thank you Razvan. Great info. Regards, Ali Pey On Wed, Jul 16, 2014 at 3:28 AM, Răzvan Crainea raz...@opensips.org wrote: Hi, Ali! Rtpproxy offers an interface to communicate with it over network, through the communication socket (the -s parameter or the RTPProxy server). You can send

[OpenSIPS-Users] engage_rtp_proxy and unforce_rtp_proxy

Hello, If I use engage_rtp_proxy, do I still need to do unforce_rtp_proxy when the call ends? Also, what are the differences between using engage_rtp_proxy vs rtpproxy_offer and rtpproxy_answer? Thanks, Ali Pey ___ Users mailing list Users

[OpenSIPS-Users] Load balancer module: lb_count_call - what if I don't know the group id

Hello, In load balancer module, I can use lb_count_call(ip,port,grp,resources) to add the current call to the resource of the specified server. What if I don't know the group id, that this server belongs to, if I pass -1, would it add the call to the right group? Thanks, Ali

Re: [OpenSIPS-Users] OPTIONS over TCP

, Ali Pey On Tue, Apr 8, 2014 at 5:40 AM, Vincent DOCQUOIS v.docquois.netvi...@gmail.com wrote: Hello all, I am using Opensips 1.10 for SIP trunking purposes. I use DR module in order to route calls to external gateways. One of destination gateways is only handling SIP over TCP. By adding

Re: [OpenSIPS-Users] Issue in VIA Header

Have a look at your system network configuration and aliases in opensips? Where do these messages come from and where do they go? Have you checked if they are set properly before they reach opensips. Regards, Ali Pey On Tue, Apr 15, 2014 at 7:14 AM, B. Buitenhuis bern...@buitenhuis.nuwrote

Re: [OpenSIPS-Users] dlg_end_dlg ALL?

I second this as well. +1 Thanks, Ali Pey On Tue, Apr 15, 2014 at 10:48 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Very useful indeed. +1 - Jeff On Tue, Apr 15, 2014 at 10:28 AM, Brett Nemeroff br...@nemeroff.comwrote: Hi Liviu, I think this would be a very useful feature

Re: [OpenSIPS-Users] Issue in VIA Header

Are you running on a VM environment? If so, try your script on a bare metal hardware. VMs mess with OS quite a bit. Regards, Ali Pey On Tue, Apr 15, 2014 at 1:33 PM, B. Buitenhuis bern...@buitenhuis.nuwrote: Hi , I'm not using any aliases in my config and I've tested the config

Re: [OpenSIPS-Users] Query about RTP flow

to re-send the packets to Client B and vise versa. So if the SIP server is in the middle of RTP communications that's proxy RTP. If RTP packets don't reach your SIP server and is only between the Clients, that's Direct speech path. Regards, Ali Pey On Tue, Feb 25, 2014 at 3:46 AM, Nandini madhu

Re: [OpenSIPS-Users] Query about RTP flow

of information you are looking for? I suggest you do a bit of reading on IP routing and the 7 networking layers. Regards, Ali Pey On Tue, Feb 25, 2014 at 12:49 PM, Nandini madhu sermj2...@gmail.com wrote: Dear Ali Pey, Thank you for the reply, I am agreeing with your answer, but I am

Re: [OpenSIPS-Users] Query about RTP flow

they are routed to their destination IP is all in IP protocol and has nothing to do with SIP. The destination IP address in your RTP packets, that comes from SDP negotiation through SIP. A different story. Regards, Ali Pey On Tue, Feb 25, 2014 at 2:20 PM, Nandini sermj2...@gmail.com wrote: Dear Ali Pey

Re: [OpenSIPS-Users] 200 OK retransmission

Hello, Yes, 200 OK must be retransmitted if Ack is not received. opensips can't do that on behalf of the SIP trunk. Retransmission is the very basic requirement and I've never heard that a sip trunk to not be able to do so. Discuss with your provider. Regards, Ali Pey On Wed, Feb 19, 2014 at 4

Re: [OpenSIPS-Users] STUN

replies from more knowledgeable people. Please post updates when you find a solution. Regards, Ali Pey On Wed, Feb 19, 2014 at 5:47 AM, Nick Altmann nick.altm...@gmail.comwrote: Hi! How to correctly configure STUN module if we have two interfaces with internal addresses where NATed two

Re: [OpenSIPS-Users] [Freeswitch-users] Brainstorming load balancing feature - dispatcher FS

This would be a pretty cool feature and I know that OpenSIPS guys have discussed such a feature with FreeSWITCH dev. This would be a great addition for load balancing. Adding OpenSIPS user mailing list. Regards, Ali Pey On Mon, Feb 17, 2014 at 8:15 AM, Karsten Horsmann khorsm

Re: [OpenSIPS-Users] 407 Proxy authorization required

. This seems to be Bria on a mobile with a very slow network connectivity. Regards, Ali Pey On Mon, Feb 17, 2014 at 10:34 AM, Chandra Prakash chandraprak...@virtualemployee.com wrote: Hi, I cannot establish a call, Can someone pls help to identify the problem? This is the log.. XX.XX.XX

Re: [OpenSIPS-Users] Users Digest, Vol 64, Issue 70

Please always respond on a proper thread with a proper subject line. No one can tell what you are talking about and to what thread you are responding when you reply to a Users Digest email. Regards, Ali Pey On Tue, Dec 3, 2013 at 4:47 AM, Chandra Prakash chandraprak...@virtualemployee.com

Re: [OpenSIPS-Users] Firing gwlist in specific order with failover

Hello Nick, What is the question then? In rules, you can list gateways, carriers and assign wait. How is that different from what you want? Regards, Ali Pey On Wed, Jan 8, 2014 at 1:09 PM, Nick Cameo sym...@gmail.com wrote: Hello Everyone, We are currently using the dr module

Re: [OpenSIPS-Users] Firing gwlist in specific order with failover

). Or you can have multiple groups and have your perl script return a specific group id for a particular call. There are many ways you can do this. You can do this through your Perl script as well. Just route the call manually to one destination at a time!! Regards, Ali Pey On Wed, Jan 8, 2014 at 8:49

Re: [OpenSIPS-Users] BYE not reaching UAC

. This is how things are routed in SIP. Here is a good reference: http://www.in2eps.com/fo-sip/tk-fo-sip-ex3261.html Regards, Ali Pey On Fri, Dec 20, 2013 at 1:23 PM, Zoho Junk m...@mmilburn.com wrote: Greetings, I have OpenSips installed on an Amazon EC2 with an elastic IP (54.242.85.140). All non

Re: [OpenSIPS-Users] Checking for Initial INVITE in branch route

. Things should not be complicated. I'm not sure what you are trying to do, but when an invite reaches opensips, it's easy to tell if it's an initial invite and route it accordingly. Regards, Ali Pey On Mon, Dec 16, 2013 at 10:26 AM, Nick Cameo sym...@gmail.com wrote: Are we able to use

Re: [OpenSIPS-Users] Checking for Initial INVITE in branch route

Examine to the to tag. If there is a to tag, that's not a initial invite: has_totag() Regards, Ali Pey On Fri, Dec 13, 2013 at 4:08 PM, Nick Cameo sym...@gmail.com wrote: Hello Everyone, I am changing the TO Header once in the branch route, and I seeing the following errors

Re: [OpenSIPS-Users] Nat_keepalive help

is more expensive in terms of resources but that is also a possible way to do. Regards, Ali Pey On Thu, Dec 12, 2013 at 7:16 AM, Miha m...@softnet.si wrote: HI, I need a little help with nat. UAC register's ok and it is reachable but after a while it become unreachable due to nat issue. If I

Re: [OpenSIPS-Users] Nat_keepalive help

BTW. the error you are getting is probably for the fact that your client doesn't like the Notify pinging. Change it to Options message and it will work. Regards, Ali Pey On Thu, Dec 12, 2013 at 11:19 AM, Ali Pey ali...@gmail.com wrote: Hello Miha, The best way to handle nat keep alive

Re: [OpenSIPS-Users] Nat_keepalive help

will be around 20k users registration time 30s is not possible:) On Thu, 12 Dec 2013 12:22:00 -0500 Ali Pey ali...@gmail.com wrote: The udp packets should come from inside the firewall for nat binding to stay open so it should come from the client side. Opensips can send Options

Re: [OpenSIPS-Users] Nat_keepalive help

Yes, the best thing is to do it on the client side and most (almost all) phones support many nat pinging features. You should be able to add it to the phones' config file and push it out. Regards, Ali Pey On Thu, Dec 12, 2013 at 4:35 PM, Miha m...@softnet.si wrote: Ignor this question:) Ok

Re: [OpenSIPS-Users] convert UPDATE into re-INVITE

need to look for ways to provide the same functionality and design with different means and modules in opensips. Regards, Ali Pey On Thu, Dec 5, 2013 at 3:52 AM, matrix testmail4...@gmail.com wrote: We have opensips as a voip switch. We are getting UPDATE sip message but some of our cpe

Re: [OpenSIPS-Users] convert UPDATE into re-INVITE

Can you elaborate more on what you are trying to do? Who is sending who an update and for what purpose? You can't convert a message to another message. They are different nature, there is no such a thing and sip message converter. Regards, Ali Pey On Tue, Dec 3, 2013 at 4:36 AM, M.D.Patel

Re: [OpenSIPS-Users] Implementing routing and failure using perl

Hello, When you use perl, you would have to implement your own logic and you can't use the dynamic routing or load balancer module from inside your perl script. You must be thinking of something similar to ESL in freeswitch or AGI in asterisk and that's not here. Regards, Ali On Tue, Dec 3,

Re: [OpenSIPS-Users] Implementing routing and failure using perl

-setFlag( 7 ); return 1; } } $m-resetFlag( 7 ); return 1; } Regards, Ali Pey On Tue, Dec 3, 2013 at 12:45 PM, Nick Cameo sym...@gmail.com wrote: Hello Ali, Thank your for your response. Yes I was inquiring about a OpenSIPS perl API (use OpenSIPS). The only thing is I cannot find any API

Re: [OpenSIPS-Users] Users Digest, Vol 64, Issue 70

Please always respond on a proper thread with a proper subject line. No one can tell what you are talking about and to what thread you are responding when you reply to a Users Digest email. Regards, Ali Pey On Tue, Dec 3, 2013 at 4:47 AM, Chandra Prakash chandraprak...@virtualemployee.com

Re: [OpenSIPS-Users] opensips for route traffic only

or assign resources. 3- For outbound, you could use drouting module to define rules and have alternate routes. 4- You could also use the dial plan module for additional manipulation or rules. Regards, Ali Pey On Thu, Nov 28, 2013 at 4:36 AM, driver dri...@op.pl wrote: Hello, I have

Re: [OpenSIPS-Users] Opensips as presence server + OpenIMS = 407 Status

Hello Estefania, Who sends the 407 message? OpenSIPS? In opensips.cfg, you can examine the source IP and port and if it is from you OpenIMSCore, then don't not authenticate it. RTPProxy wouldn't have anything to do with this. Regards, Ali Pey On Wed, Nov 27, 2013 at 2:05 PM, Estefania

Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

You also examining isbflagset(10). Isn't that set? Regards, Ali Pey On Mon, Nov 25, 2013 at 3:09 AM, dpa denis7...@mail.ru wrote: Hello I understand but in onreply route I make a test: nat_uac_test(55) and only if it successful I make “fix_nated_contact()”. In my case nat_uac_test(55

Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

the source IP or subnet to decide if this message is from internal or external and then apply different logic to it - or whatever else that is specific to your environment. What is it that you are trying to do? Regards, Ali Pey On Mon, Nov 25, 2013 at 10:52 PM, dpa denis7...@mail.ru wrote: OK

Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

Why don't you use nat_uac_test() when you receive a request to examine NAT? Using nat_uac_tes() you can tell if the message is coming from a client behind the nat or not. With lookup and the nat_bflg, you can tell if the destination is behind a nat. They are two different things. Regards, Ali Pey

Re: [OpenSIPS-Users] Opensips 1.9.1 and NAT

Hello, The question is not quite clear. In your opensips.cfg you call fix_nated_contact() on both route and reply route. that's why it changes the route. There is no such a thing as direction in opensips unless it's implemented in your logic. A message goes through your route or reply route

Re: [OpenSIPS-Users] OpenSIPS + ingress profit protectioin

Yes, I would also recommend drouting module as well. It's quite powerful for handling scenarios like this. On the egress, try the load balancer module. Regards, Ali Pey On Tue, Nov 19, 2013 at 10:40 PM, Nick Cameo sym...@gmail.com wrote: I sent that email too fast from the handheld. I am

Re: [OpenSIPS-Users] Opensips Users Registartion Problem

Hello Vishnu, You can look at the opensips logs or network traces to see what's happening. No one can tell you what's wrong with your system because we don't know what you have done. You need to do your debugging and post specific question that can be answered. Regards, Ali Pey On Thu, Nov 14

Re: [OpenSIPS-Users] perl script abnormal termination handling

Don't do die in the perl script. Try return 0 or 1. Regards, Ali Pey On Fri, Nov 8, 2013 at 10:32 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hello, I run a perl script from an Opensips 1.6 config. I'm having trouble handling a perl die condition in the Opensips script. For example

Re: [OpenSIPS-Users] TCP to UDP proxy

Add this in your route: subst_uri('/(sip:.*);transport=tcp/\1/'); To remove transport=tcp from request-uri. This should fix it. Regards, Ali Pey On Wed, Nov 6, 2013 at 2:15 PM, Simon Quigley squig...@versature.comwrote: Hi, I've done some more testing, and it seems the issue I'm having

Re: [OpenSIPS-Users] Request For Opensips Information

understand something. Regards, Ali Pey On Thu, Nov 7, 2013 at 2:46 PM, Vishnu Vardhan vishnuv...@gmail.com wrote: Hi, I am new to opensips.Opensips will support Centos 6.4 and asterisk 11/12.And i am getting confusion how to install openSips and asterisk is that are both to install

Re: [OpenSIPS-Users] Dialog module sql server 2008 r2 database backend

You need to use ODBC and freetds. You can't directly connect to MS SQL. It works quite well with ODBC and freetds. Regards, Ali Pey On Thu, Nov 7, 2013 at 3:38 PM, David J da...@styleflare.com wrote: I hope this does not sound silly but what driver are you using to connect to mssql

Re: [OpenSIPS-Users] uac_registrant, how to force registration port?

the same port as opensips and will keep it as long as it's up. So, just do some research on your router. Regards, Ali Pey On Thu, Nov 7, 2013 at 2:48 PM, Max E. Reyes Vera J. navai...@gmail.comwrote: Hi, I was testing the uac_registrant module to register opensips with my sip provider, but I

Re: [OpenSIPS-Users] Dialog module sql server 2008 r2 database backend

it. Regards, Ali Pey On Thu, Nov 7, 2013 at 5:14 PM, bluerain frank21...@yahoo.com wrote: As I stated in my reply to dave, yes I have that setup, I have all module running on sql server 2008 r2 via freetds, the only module does not load correctly is the dialog module. That is why I am wondering what

[OpenSIPS-Users] Astricon Discount Code

a discount code for this event. Regards, Ali Pey ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [OpenSIPS-Devel] OpenSIPS Tracker Migration on GitHub

I would vote for the second option. It would take a bit to clean up but it would be cleaner and much less work. Usually when you move things, there are errors, mistakes, etc. It would be cleaner and less work with option 2. Regards, Ali On Tue, Jun 4, 2013 at 9:18 AM, Răzvan Crainea

Re: [OpenSIPS-Users] OpenSIPS Documentation re-Factory

Hello, The new documentation lay out is much much better and it is much easier to find the info you need faster. Thanks for the great work. I found a broken link though: Database schema Overall, well done!! Regards, Ali Pey On Mon, Jun 3, 2013 at 5:01 AM, Bogdan-Andrei Iancu bog

[OpenSIPS-Users] Send a SIP Notify message to a registered SIP Phone (config resync)

Hello, What's the best way to generate and send a custom Notify SIP message to a registered sip phone (for configuration resync)? Thanks, Ali Pey ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RFC] New Release Policy for OpenSIPS project

I agree with this. Not every release needs to be supported for a full 2 years. Regards, Ali Pey On Fri, Nov 30, 2012 at 12:24 PM, Ryan Bullock rrb3...@gmail.com wrote: On Mon, Nov 26, 2012 at 1:57 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: ** Hi Ali, Thanks for feedback

Re: [OpenSIPS-Users] OpenSIPS Control Panel limitation with table dialplan for more than 40k rows.

I think this was fixed in opensips-cp last week (or the week before). Make sure you get the latest fixes from trunk and try again. Regards, Ali Pey 2012/11/21 Muhammad Shahzad shaherya...@gmail.com Its most like your web server issue, you need to increase output buffer size. If using Apache

Re: [OpenSIPS-Users] [RFC] New Release Policy for OpenSIPS project

. For instance when 1.9 comes out, we probably will upgrade to 1.8.2. Again, this is a great step forward for opensips development and thank you for the great work. Regards, Ali Pey On Fri, Nov 23, 2012 at 4:36 AM, Saúl Ibarra Corretgé s...@ag-projects.comwrote: Hi, The problem I see

Re: [OpenSIPS-Users] how to Register with OpenSIPS?

take over. All the registration info would be in a database such as mysql. Regards, Ali Pey On Thu, Nov 15, 2012 at 3:13 AM, Christian Cambier c...@voxtron.com wrote: Hi. ** ** I'm new to using SIP proxies (OpenSIPS) so maybe it is a basic question but anyway... ** ** I have

Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

based on the registration server. Basically opensips would be your proxy server. It keeps and digests all the registration info and will handle the routing between your sip clients/trunks and your pbxs. You need some reading and some help. This is the general idea. Regards, Ali Pey On Thu, Nov 15

Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

, Ali Pey On Thu, Nov 15, 2012 at 9:46 AM, Christian Cambier c...@voxtron.com wrote: Hello. Your sip phones only register to the opensips servers. Your pbx dosen't* *** need to sip registrations. But what do you do then with the account-settings that were created on the PBX

Re: [OpenSIPS-Users] 404 Not Found error with Sipp UAC

with no lookup. Hope this is clear enough. Regards, Ali Pey On Thu, Nov 8, 2012 at 8:05 PM, Steve Mitchell swmitch...@gmail.com wrote: Hi, I'm trying to get a simple scenario working to generate CDRs in bulk and am using a basic configuration (generated with osipconfig) with the sipp UAC. However

Re: [OpenSIPS-Users] Feature request for OpenSIPS 1.9: support for macro definitions

I second this as well. Named flags would make debugging and scripting quite simpler. Regards, Ali On Thu, Nov 8, 2012 at 11:23 AM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: Hi Michael, You can already use names for the route, not only numerical IDs (without the need of defining).

Re: [OpenSIPS-Users] Errors in OpenSIPS

I think it's telling you it can't add another dialog to db. I am almost certain that's what's happening. Regards, Ali Pey On Thu, Nov 8, 2012 at 4:22 AM, Jorge Ortea dar...@hotmail.com wrote: Hi all, I am getting the following errors: Nov 7 11:47:19 hgt-tero45 /usr/local/opensips/sbin

Re: [OpenSIPS-Users] Feature request for OpenSIPS 1.9: support for macro definitions

-solutions.com On 11/08/2012 07:06 PM, Ali Pey wrote: I second this as well. Named flags would make debugging and scripting quite simpler. Regards, Ali On Thu, Nov 8, 2012 at 11:23 AM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: Hi Michael, You can already use names for the route

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

can also be used when you are using resources of one of your gateways but outside the load balancer module such as routing a test number to a particular gateway and so on. Regards, Ali Pey ___ Users mailing list Users@lists.opensips.org http

Re: [OpenSIPS-Users] OPTIONs flood with large numbers of registered users

Another way to tackle this is to enable pinging from your sip clients. Makes things quite simpler. Regards, Ali Pey On Sat, Nov 3, 2012 at 3:47 AM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: Hi Brett, If you are using nathelper module, use multipe timer processes for pinging

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

, Ali Pey On Wed, Oct 31, 2012 at 8:21 AM, Saúl Ibarra Corretgé s...@ag-projects.comwrote: On Oct 31, 2012, at 12:52 PM, Bogdan-Andrei Iancu wrote: Hi Saul, OK, aside the TCP part (which anyhow is scheduled for fixing) and some extra parsing, does supporting WebRTC imply something more

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

Which one sounds simpler? Having a new layer of proxies and extra hardware on different software packages with their own set of configurations, limitations and bugs than having WebSocket enabled on opensips and control your routing logic all in one place off of same DB. Regards, Ali Pey On Wed

Re: [OpenSIPS-Users] sip proxy and nat traversal

configuration on both asterisk and opensips to negotiate with proper public IP addresses for both media and sip to your far end devices. You need to know NAT and SIP to really be able to set this up. Regards, Ali Pey On Sat, Oct 27, 2012 at 6:53 PM, spencer sal1...@yahoo.com wrote: Hi, I am

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

for a sip proxy server to support it. Regards, Ali Pey On Fri, Oct 26, 2012 at 11:54 AM, Saúl Ibarra Corretgé s...@ag-projects.com wrote: On Oct 26, 2012, at 5:46 PM, Duane Larson wrote: Is there any roadmap for SIP over Websocket? I know there is now OverSIP but wasn't sure if OpenSIPS had

Re: [OpenSIPS-Users] ACK looping issue with end-point co-located with OpenSIPS

What do you have for alias in your opensips.cfg? Opensips seems to think 192.168.117.4:5070 is itself. Regards, Ali Pey On Mon, Oct 15, 2012 at 6:33 PM, Daniel Eiland daniel.eil...@gmail.comwrote: Hi folks, I've got an issue with ACK messages being looped when they are sent to an endpoint

Re: [OpenSIPS-Users] multiple headers

Did you try sipmsgops module? You can maybe remove the header and the add the ones you'd like: http://www.opensips.org/html/docs/modules/1.8.x/sipmsgops.html You can also check out the textops module. There are some utilities there as well. Regards, Ali Pey On Thu, Oct 11, 2012 at 1:26 PM, DM

Re: [OpenSIPS-Users] duplicate information in sip_trace table

Did you try to add logs to make sure sip_trace() is not called twice? Regards, Ali Pey On Wed, Oct 10, 2012 at 10:29 AM, Dragomir Haralambiev goup2...@gmail.comwrote: Hi, Thanks for your replay. The problem is not in IF operator. When use sip_trace() Opnesips make two records in sip_trace

Re: [OpenSIPS-Users] Proxying - Hiding Caller From Callee

Hi David, As I take it you need to manipulate your from header. There are a few easy ways you can achieve this. Check out the textops module: http://www.opensips.org/html/docs/modules/1.8.x/textops.html Regards, Ali Pey On Tue, Oct 9, 2012 at 1:24 PM, David Wylie dwy

Re: [OpenSIPS-Users] Proxying - Hiding Caller From Callee

Also check out these to places. You can use the uac module or directly change the from uri username: http://www.opensips.org/html/docs/modules/devel/uac.html http://www.opensips.org/Resources/DocsCoreVar18#toc45 Regards, Ali Pey On Tue, Oct 9, 2012 at 6:56 PM, Ali Pey ali...@gmail.com wrote

Re: [OpenSIPS-Users] bash shell variable not kept when run in opensips.cfg

Hello Xavier, Try the avops module. This is your best option in this scenario: http://www.opensips.org/html/docs/modules/1.8.x/avpops.html#id292750 Regards, Ali Pey On Sat, Oct 6, 2012 at 7:38 PM, Xavier Herlindo xherli...@yahoo.com.mxwrote: Hello all, this is my first post, so please bare

Re: [OpenSIPS-Users] Max TCP connections

Thanks Muhammad. These are great information. Thank you sharing it with us. Two follow up questions: 1- What tool did you use to send registers over TCP and keep connections open? 2- What did you set tcp_max_connections to in opensips? Regards, Ali Pey On Sat, Sep 29, 2012 at 10:59 AM

Re: [OpenSIPS-Users] High Volume MediaProxy

, Ali Pey On Mon, Oct 1, 2012 at 3:19 AM, afshin afzali a.afzali2...@gmail.comwrote: Hi Guys, I'm looking for a solution for traversing 10,000 concurrent sessions (no encryption). Is it possible this on a single modern box by MediaProxy ? BEST, -- afshin

Re: [OpenSIPS-Users] High Volume MediaProxy

. Thank you. On Mon, Oct 1, 2012 at 6:47 PM, Ali Pey ali...@gmail.com wrote: Hi Afshin, I take it that you need to proxy both the sip signalling and media. I don't think you can do that on one server. One opensips server can do the signalling but you would need additional servers for media

Re: [OpenSIPS-Users] Problem with 408 Request Timeout

So you don't want to send 408 request timeout??? What do you expect to happen here? What is your desired behaviour? Look at your routing script: /etc/opensips/opensips.cfg The behaviour is configured there and can be change to your liking. Regards, Ali Pey On Thu, Sep 27, 2012 at 4:16 AM

Re: [OpenSIPS-Users] Max TCP connections

I am also interested in this. Any responds/updates? Regards, Ali Pey On Wed, Sep 12, 2012 at 9:34 AM, John Quick john.qu...@smartvox.co.ukwrote: Does anyone know what the practical limit is for the maximum number of TCP connections to OpenSIPS? It is a question that often comes up

Re: [OpenSIPS-Users] authentication of authorized user agents

Hi Dave, First of all, this is opensips mailing list not Kamailio. Secondly, you can pass the users sip credentials (sip username and password) through your http secure connection and pass it to your sip stack (user agent) to register with that credentials. Regards, Ali Pey On Tue, Sep 18

Re: [OpenSIPS-Users] Can load balancer show total number of call for a gateway

a gateway has Incoming/Outgoing from/to a particular gateway. The real deal is to communicate the populated incoming calls variable to the load-balancer module. (Dynamically increment/decrement a g/w capacity based on the $avp(IC_GW1) ) BR Sammy On Fri, Sep 7, 2012 at 1:58 AM, Ali Pey ali

Re: [OpenSIPS-Users] Can load balancer show total number of call for a gateway

Yes, that's almost the topology. Thank you for clarifying it. Yes, I would want to be able to account for both legs. It's not always two legs though, sometimes it can only terminate on the gateway, e.g. voice mail, etc. Regards, Ali Pey On Fri, Sep 7, 2012 at 11:30 AM, Muhammad Shahzad

Re: [OpenSIPS-Users] Can load balancer show total number of call for a gateway

Hi Sammy, I rather to be able to see the through number of active calls on a server than re-sizing the capacity of each server twice for each call. That will create a mess. Regards, Ali Pey On Fri, Sep 7, 2012 at 12:02 PM, SamyGo govoi...@gmail.com wrote: Hi Ali, No , I said it dialog

Re: [OpenSIPS-Users] opensips Cache

Hello Nick, Do you have domains enabled? Maybe you need an alias for 202.55.233.169? Make sure you have these: alias=202.55.233.169 modparam(usrloc, use_domain, 1) Regards, Ali Pey On Wed, Sep 5, 2012 at 9:10 PM, Nick Chang nick.ch...@kland.com.tw wrote: Hello Ali ** ** I change

Re: [OpenSIPS-Users] opensips Cache

Re-evaluate your db mode setting: http://www.opensips.org/html/docs/modules/1.8.x/usrloc.html#id293049 Regards, Ali Pey On Wed, Sep 5, 2012 at 3:44 AM, Nick Chang nick.ch...@kland.com.tw wrote: Hello ** ** I have two opensips server. ** ** User A àregisterà Opensips

Re: [OpenSIPS-Users] About L4 Swtich + 2 * opensips

Hello Nick, Set your db_mode to 3. http://www.opensips.org/html/docs/modules/1.8.x/usrloc.html#id293049 Regards, Ali Pey On Tue, Sep 4, 2012 at 10:57 PM, Nick Chang nick.ch...@kland.com.tw wrote: Hello Ali ** ** Debug is on ** ** I saw log Sep 5 10:53:07 Sharesip1

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