Re: [OpenSIPS-Users] Opensips behind a firewall with internal IP

Daniel, I am calling fix_nated_register() on REGISTER and on INVITE fix_nated_contact() On 5/21/10 1:46 PM, Daniel Goepp wrote: Ah yes, are you also doing NAT detection and fixing contact? -dg On Fri, May 21, 2010 at 10:40 AM, David J. <mailto:da...@styleflare.com>> wrote:

Re: [OpenSIPS-Users] Opensips behind a firewall with internal IP

you are. -dg On Fri, May 21, 2010 at 9:05 AM, David J. <mailto:da...@styleflare.com>> wrote: Daniel, You said you got opensips/rtp-proxy working on EC2, when testing it, I see the VIA headers have the EC2 internal ip and the BYE messages not getting routed correctly, a

Re: [OpenSIPS-Users] Opensips behind a firewall with internal IP

Daniel, You said you got opensips/rtp-proxy working on EC2, when testing it, I see the VIA headers have the EC2 internal ip and the BYE messages not getting routed correctly, any advice on how to solve that? Thanks On 5/21/10 12:02 PM, Daniel Goepp wrote: You need to be careful though becau

Re: [OpenSIPS-Users] OpenSIPS > announcement > pstn

Stefan, I had a similar question, we don't play ads to the caller, but we do locate the callee while the caller is waiting or "on hold", so we could use the B2BUA functionality. Is there an example available of this functionality in the SEMS examples? (Sorry to post a SEMS question on the Open

Re: [OpenSIPS-Users] OpenSIPS Perl module branching and logging

Bogdan, Thank you for your help again! We found the source of the problem. On Thu, May 13, 2010 at 7:05 PM, Bogdan-Andrei Iancu wrote: > Sorry David, > > never used the perl stuff and have no idea how it works :(... > > Regards, > Bogdan > > David Cunningham

Re: [OpenSIPS-Users] OpenSIPS defunct processes

Hi Bogdan, Just to let you know, we traced the problem to the Perl code. Thank you for your help! On Mon, May 3, 2010 at 8:48 AM, Bogdan-Andrei Iancu wrote: > Hi David, > > Based on the "ps" output, it seams that the zombies processes were > forked by opensips worker pro

Re: [OpenSIPS-Users] OpenSIPS Perl module branching and logging

dan-Andrei Iancu wrote: > Hi David, > > looking at the message, I see that the ACK is wrongly generated. The > RURI of the ACK must be the contact received in 200 OK (as received by > the phone). The 200 OK shows ast02, but the ACK goes to ast01 , which is > bogus. > > Also

Re: [OpenSIPS-Users] OpenSIPS Perl module branching and logging

ACK > corresponding to the winning branch (which sent the 200 OK). > > Regards, > Bogdan > > David Cunningham wrote: >> Hi Bogdan, >> >> That's correct, the ACK for the 200OK. >> >> On Wed, May 12, 2010 at 8:46 AM, Bogdan-Andrei Iancu >> wrote

Re: [OpenSIPS-Users] OpenSIPS Perl module branching and logging

Hi Bogdan, That's correct, the ACK for the 200OK. On Wed, May 12, 2010 at 8:46 AM, Bogdan-Andrei Iancu wrote: > Hi David, > > David Cunningham wrote: >> Hello, >> >> We are using OpenSIPS with a Perl module which needs to branch SIP >> messages to multiple

[OpenSIPS-Users] OpenSIPS Perl module branching and logging

;append_branch()? Thanks for your help! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/c

Re: [OpenSIPS-Users] opensips and asterisk

Sorry, The way I recommend doing this was assuming the user on the Asterisk box needed to be publicly reachable from anywhere. I think that approach makes sense when using DID's and inbound routing that does need authentication. On 5/4/10 12:55 PM, Olle E. Johansson wrote: > 4 maj 2010 kl. 18

Re: [OpenSIPS-Users] opensips and asterisk

Check in the SIP.conf where you send all unauthenticated calls. On 5/4/10 11:45 AM, wüber wrote: > The problem seems to be not only in the extensions.conf file, but also in the > sip.conf file. > I still get this forbidden message! > ___ Users ma

Re: [OpenSIPS-Users] opensips and asterisk

You need to add a route in your extensions.conf in the context where you send all un-authenticated calls. Maybe its your default context? [default] exten = 1001,1,Dial(SIP/1001,10,tTr); On 5/4/10 9:02 AM, wüber wrote: > Hi Bogdan, > > connecting Opensips with Asterisk I can see that if a cl

Re: [OpenSIPS-Users] OpenSIPS defunct processes

quot;ping_nated_only", 1 ) modparam( "nathelper", "sipping_bflag", 1 ) modparam( "nathelper", "sipping_from", "sip:keepal...@foo" ) modparam( "nathelper|registrar", "received_avp", "$avp(i:42)" ) modparam( "perl"

Re: [OpenSIPS-Users] uac_replace_from

Thanks help much apreciated modparam("rr", "append_fromtag", 1) modparam("uac","restore_mode","auto") xlog("$fu \n"); uac_replace_from(2101573...@10.10.1.100 ); xlog("$fu \n"); Output sip:02086847...@10.10.1.50 sip:02086847...@10.10.1.50 I cant seem to

Re: [OpenSIPS-Users] OpenSIPS defunct processes

s:: ID=18 PID=5849 Type=TCP receiver Process:: ID=19 PID=5850 Type=TCP receiver Process:: ID=20 PID=5851 Type=TCP main On Mon, Apr 26, 2010 at 12:22 PM, Bogdan-Andrei Iancu wrote: > Hi David, > > Let's try and see what's the parent process of the zombie procs -> check >

Re: [OpenSIPS-Users] cdrtool: looking in the wrong location for radacct table when mysql on the same IP address?

Paul, Please re-read the documentation again. I've posted my global.inc in another mail. I'd appreciate some more experienced eyes on this. Thanks. On 4/27/10 10:05 PM, Paul Wise wrote: On Tue, 2010-04-27 at 10:41 +0200, Adrian Georgescu wrote: I have no clue how you arrive at this

Re: [OpenSIPS-Users] Vars Webinar.

I appreciate it. Thanks so much. On 4/27/10 11:39 AM, Bogdan-Andrei Iancu wrote: > Hi David, > > sorry for not answering you sooner ...I had no answer for you :)...But I > found the raw recording of the webinar - I will have to process it a bit > and I will upload it i

[OpenSIPS-Users] Vars Webinar.

I dont mean to keep asking, but is there an upload from the last webinar? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS defunct processes

if you're confident it will solve our problems. Much appreciate your help. On Thu, Apr 22, 2010 at 6:07 PM, Bogdan-Andrei Iancu wrote: > Hi David, > > David Cunningham wrote: >> Hello, >> >> Thank you for the reply! >> >> The log doesn't say anything

Re: [OpenSIPS-Users] OpenSIPS defunct processes

t main.c:867 Can anyone point me where to go from here - maybe advice on what gdb commands would help? I should have mentioned that this is on version 1.4.5-notls. We originally saw it on 1.4.3-notls and upgraded to try and fix it. Thanks in advance! On Wed, Apr 21, 2010 at 8:55 AM, Bogdan-

[OpenSIPS-Users] OpenSIPS defunct processes

-P /var/run/user/opensips.pid user 7834 7811 0 03:19 ?00:00:00 \_ /sbin/opensips -m 256 -P /var/run/user/opensips.pid Cheers, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180

[OpenSIPS-Users] Max Contacts Flag Documentation

I was looking at the registrar documentation and not sure if this is in-consistent documentation. /flags/ (optional)- string of the following flags: /'cnn' (max Contacts)/ - this flag can be used to limit the number of contacts for this AOR (Address of Record) in the user location database. V

[OpenSIPS-Users] Last weeks webinar.

Any idea about when last weeks webinar will be posted? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] cdrtool rate engine on centos 4

Hi, Can anyone point me in the right direction. I have two centos 4 servers. the cdrtool rate engine starts on one server but not the other. I get the following on the non starting server. Apr 5 20:40:43 th-db02 cdrtool[10497]: Starting CDRTool Rating Engine... Apr 5 20:40:43 th-db02 cdrto

Re: [OpenSIPS-Users] can OpenSIPS-cp and SerMyAdmin co-exist?

Just curious, Whats the sense of this? Christian Vo wrote: > Trying to manually install SerMyAdmin (step by step , since install > script uses apt-get) on my OpenSUSE 11 setup.. > Which already has OpenSIPs-CP installed… > I’m noticing that the sql script has several conflicts… anyone try > this

Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS

Matt, I am for sure probably wrong, but I think you would need Asterisk or Variant to Determine that it is a Fax Call, I dont think UAC's send T38 information without negotiating with the other side who request that it is capable, then it brings you to Jeff's answer. See above. Matthew S. Cr

Re: [OpenSIPS-Users] Errors while setting up Opensips with Berkeley DB

ex: opensipsctl db show subscriber Jan Rozhon wrote: > Definitely sureit's the only one I have. > > Dne 5.3.2010 16:44, David J. napsal(a): > >> And your sure your checking the same database your script is pointing to? >> >> >> >> Jan Rozhon

Re: [OpenSIPS-Users] Errors while setting up Opensips with Berkeley DB

Yikes! Can you run any of the other opensipsctl commands on that database? Or do they fail as well? Jan Rozhon wrote: > Definitely sureit's the only one I have. > > Dne 5.3.2010 16:44, David J. napsal(a): > >> And your sure your checking the same database you

Re: [OpenSIPS-Users] Errors while setting up Opensips with Berkeley DB

And your sure your checking the same database your script is pointing to? Jan Rozhon wrote: > No, user in database - checked twice. Problem is, that I cannot add any > user > > > Dne 5.3.2010 16:39, David J. napsal(a): > >> Where is the error? >> >> C

Re: [OpenSIPS-Users] Errors while setting up Opensips with Berkeley DB

Where is the error? Check your database and make sure you dont have an entry for that user. Jan Rozhon wrote: > Hi list, > > I am trying to set up and run opensips with berkeley module. After > configuration, and creating of database I get this error: > > chieftec:~# opensipsctl add 1234 1234 >

Re: [OpenSIPS-Users] [RFC] What repo to use for 2.0 ?

Brett, From a user perspective then its not so difficult to learn a few extra commands. Most users I would imagine want to download the code and compile. Although, I agree that SVN is by far more adopted. I think going forward, companies like GitHub and BitBucket offer great code hosting opti

Re: [OpenSIPS-Users] [New] Probing for drouting module

nge this you will have to update your tables > > David J. wrote: > >> Sounds good, Do we need to update the DB Schema or if the column is >> missing it fails gracefully? >> >> Thanks. >> >> >> Andrei Dragus wrote: >> >> >

Re: [OpenSIPS-Users] [New] Probing for drouting module

Sounds good, Do we need to update the DB Schema or if the column is missing it fails gracefully? Thanks. Andrei Dragus wrote: > Hi all, > > A new feature has been added to the drouting module that allows the > probing of gateways in case they become unreachable. > > Typical usages:

[OpenSIPS-Users] Remote Party ID

Where is the best place to use the append_rpid_hf()? It seems I add it once, but subsequent messages dont include the header. Perhaps I have to add it for each request? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/c

[OpenSIPS-Users] Record Route Natted IP

what is responsible for using the external ip for the record route? Right now cancels are not being routed properly. It seems from the SIP trace that we are using the natted ip rather than the public. However the box does not have a public ip, we are port forwarding from our router to our opens

Re: [OpenSIPS-Users] Fw: opensips as a transcoder

RTP Proxy wont help either, It is also just a media relay. wüber wrote: > Perhaps you can try RTPproxy ... I'm not so sure but I guess it should > support transcoding. Regards > ___ Users mailing list Users@lists.opensips.org http://lists.opensips

Re: [OpenSIPS-Users] Fw: opensips as a transcoder

MediaProxy is a separate module for Media Relay also has nothing to do with transcoding. Transcoding is a conversion process from one codec to another, maybe you can try using asterisk if it supports the codecs your are trying to transcode. OpenSIPS is a sip proxy, media proxy is a media relay

Re: [OpenSIPS-Users] opensips as a transcoder

I am not the expert, but I dont think OpenSIPS can be used as a transcoder. It acts only as a signaling proxy, media is not passed through opensips, just the SDP information. But I am not the expert. Live School wrote: > Hi, > > is it possible to use openSIPS as transcoder only ? > if yes,

Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)

Pardon my ignorance, but what exactly is the problem? Your customer does not need to register to you, however, then you need to have a static ip address to route to. and vice versa. When a customer sends an INVITE to your opensips, you need to authenticate by IP address. The sipXecs does not ne

Re: [OpenSIPS-Users] Permissions Module Fails to Load with DB Support - Error in Address Reload.

Changing the mask does resolve the problem. Thank you. Bogdan-Andrei Iancu wrote: > The mask is not correct - it must be less than 32 , the max len of an > IPv4 address. > > Regards, > Bogdan > > David J. wrote: >

Re: [OpenSIPS-Users] Permissions Module Fails to Load with DB Support - Error in Address Reload.

Will try to update, However, it was added through opensipsctl address add I didnt manually insert the data. Bogdan-Andrei Iancu wrote: > The mask is not correct - it must be less than 32 , the max len of an > IPv4 address. > > Regards, > Bogdan > > David J. wrote: >

Re: [OpenSIPS-Users] Permissions Module Fails to Load with DB Support - Error in Address Reload.

ase post here > the content of the address table (do a select * ). > > Regards, > Bogdan > > David J. wrote: > >> MySQL version 5.0.51a >> >> >> >> >> Bogdan-Andrei Iancu wrote: >> >> >>> Hi David, >>

Re: [OpenSIPS-Users] Permissions Module Fails to Load with DB Support - Error in Address Reload.

MySQL version 5.0.51a Bogdan-Andrei Iancu wrote: > Hi David, > > what db backend are you using for permissions module? > > Regards, > Bogdan > > David J. wrote: > >> I am running SVN Trunk and getting the following Error. >> >> /sbin/opensips[

[OpenSIPS-Users] Permissions Module Fails to Load with DB Support - Error in Address Reload.

I am running SVN Trunk and getting the following Error. /sbin/opensips[1923]: ERROR:permissions:reload_address_table: database problem /sbin/opensips[1923]: CRITICAL:permissions:init_address: reload of address table failed /sbin/opensips[1923]: ERROR:permissions:mod_init: failed to initialize t

Re: [OpenSIPS-Users] Opensips ( Monit Cant! Connect )

module, I have check the boxes config and monit config both have identical passwords and usernames. hmmm. When i do a netstat -tal, I dont see anything running on port 2812 or 8080. On Thu, Dec 10, 2009 at 9:05 PM, Indiver wrote: > > Hi David, > I worked on Monit tool and succeeded i

Re: [OpenSIPS-Users] OpenSIPS Modules

Yes lulia, thanks On Wed, Dec 9, 2009 at 2:47 AM, Iulia Bublea wrote: > Hi, > > Have you changed the the mysql login credential and DB name in > config/db.inc.php? > > > Iulia > > David Perrin wrote: > > php installed, pear, MDB2 and MDB2 mysql driver, all ins

[OpenSIPS-Users] Opensips ( Monit Cant! Connect )

a netstat -tal I dont see anything on the monit port or 8080, could someone that set this up on centos please give me a little push in the right direction. Thanks kindly, David P/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org

Re: [OpenSIPS-Users] OpenSIPS Modules

php installed, pear, MDB2 and MDB2 mysql driver, all installed On Tue, Dec 8, 2009 at 10:06 PM, ram wrote: > > > On Tue, Dec 8, 2009 at 4:16 PM, David Perrin wrote: > >> Hello everyone, >> >> I have just recently setup opensips and I am using the control panel,

[OpenSIPS-Users] OpenSIPS Modules

modules during the opensips installation in the opensips.cfg, for example loadmodule "siptrace.so" to get some of the modules to work properly, >?. I just need to set up the basics for trunking only. Thanks David P ___ Users maili

Re: [OpenSIPS-Users] drouting module

stop asking so much! ;) D El 25/08/2009, a las 18:05, "Sebastian Sastre" escribió: Hi, When applying the do_routing function without specifying the group id, what are the matching patterns that it will look for? In other words If I want ALL the users behind domain test.com how

Re: [OpenSIPS-Users] presence_dialoginfo context

want to do the same thing for presence_dialog info, how can I rewrite > the subscribe, presence and and notify messages to append the > appropriate prefix ? > > Thanks, > > David > > ___ > Users mailing li

Re: [OpenSIPS-Users] [Kamailio-Users] dialog info + Grandstreams = freezing

is making the full trip. Does this answer your question? David Daniel-Constantin Mierla wrote: > Hello, > > I am using the module with snom 370 and works ok. The BLF is on for > Polycoms but they do not implement the call pickup properly (they have > direct call pickup as far a

Re: [OpenSIPS-Users] mailing list reply-to : list or sender

Hey, The 'reply-to-all' button in Thunderbird will cause it to be sent to the list and the person who wrote the message. David Iñaki Baz Castillo wrote: > 2009/8/11 Victor Gamov : > >> I read many list many years and OpenSER-based software lists (OpenSER, >> Op

[OpenSIPS-Users] dialog info + Grandstreams = freezing

only freeze for about 30 seconds. Obviously I am doing something wrong, but despite having searched google for endless hours, and poured over documentation, I can not seem to find what I did wrong. I would really appreciate if someone could shed light on my problem. I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS 1.5.2-tls compilde on Ubuntu 8.04 Thanks, David ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] load_balance and 483 error.

On Wed, Jul 15, 2009 at 4:09 PM, Iñaki Baz Castillo wrote: > > The ACK for a 200 is an *in-dialog* request so it must be setnt to the proxy > (since it did loose-routing by adding Record-Route header) and the RURI must > be the final destination. > > SIPp doesn't handle corretly this case, I alread

Re: [OpenSIPS-Users] load_balance and 483 error.

es. Contact: . Content-Type: application/sdp. Content-Length: 182. . v=0. o=root 1496 1496 IN IP4 asterisk-ip. s=session. c=IN IP4 asterisk-ip. t=0 0. m=audio 19044 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. On Wed, Jul 15, 2009 at 3:44 PM, Iñaki Baz Castillo wr

Re: [OpenSIPS-Users] load_balance and 483 error.

I've been staring at nget output for a while now, comparing the working transaction (rewritehost at the top of route logic) against the rewrite after load_balance. They look nearly identical, the change comes after the client (sipp) sends an ACK; In the working capture, sipp sends an ACK, and then

Re: [OpenSIPS-Users] load_balance and 483 error.

On Wed, Jul 15, 2009 at 1:29 PM, Iñaki Baz Castillo wrote: > El Miércoles, 15 de Julio de 2009, David escribió: > > I think your config is correct. However I've never used load_balancer module. > It would help if you describe what "483 looping error" means. opensip

Re: [OpenSIPS-Users] load_balance and 483 error.

Hmm.. I'm still getting the 483 looping error. I've set $ru to "sip:" + $tU + "@" + $dd + ":" + $dp, but this should just do the same as setting rewritehost to $du, correct? Putting rewritehost(hostname) at the top of the route puts the call through how I want, but I don't have the $dd, $du, etc.

[OpenSIPS-Users] load_balance and 483 error.

Hello, I've been trying to configure opensips as a load balancer, but keep getting a "483 looping" error. It seems the address is not being rewritten in the INVITE. Here's the setup: client (sipp) > opensips (1.2.3.4) > asterisk (5.6.7.8) I'm using the same config as in the load balancing tutori

Re: [OpenSIPS-Users] H323

Hello, I got to about 100 simultaneous calls, without touching rtps. David On Thu, Jun 4, 2009 at 3:48 PM, Brett Nemeroff wrote: > David,What capacities are you able to achieve with yate and h323? roughly? > > > On Thu, Jun 4, 2009 at 7:15 AM, David Villasmil < >

[OpenSIPS-Users] 302 Moved Temporarily routing

are that a 302 reply was received by OpenSIPS. Is there any way to drop the message so that OpenSIPS do not relay it to the UAC? In the documentation, it seems that only a provisionnal response 1xx can be dropped. Thanks a lot. David ___ Use

[OpenSIPS-Users] H323

Hello, yate is great for h.323<->SIP, it even has a config on its web to do it. I've used it a lot. have fun! On Mon, Jun 1, 2009 at 12:55 PM, Bogdan-Andrei Iancu wrote: > Wayne Scholar wrote: > > A bit more info. > > > > We use opensips as a gateway to know about all our asterisk serve

[OpenSIPS-Users] t_relay() for Tel-URI

contains any domain part, how OpenSIPS will be able to forward the message to the correct destination? Many thanks. /David ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Removing a VIA line required by Telco.

Bogdan-Andrei Iancu wrote: > Hi Davis, > > David Gilbert wrote: >> Jeff Pyle wrote: >> >>> David, >>> >>> Opensips is a proxy. A proxy's role is dictated by RFC3261. It's >>> fast, >>> lightweight, very scalable,

Re: [OpenSIPS-Users] Removing a VIA line required by Telco.

Jeff Pyle wrote: > David, > > Opensips is a proxy. A proxy's role is dictated by RFC3261. It's fast, > lightweight, very scalable, and has some very nice features you're already > taking advantages of. But, it's still only a proxy. A proxy needs Via > li

Re: [OpenSIPS-Users] Removing a VIA line required by Telco.

Jeff Pyle wrote: > David, > > As I understand it, a SIP proxy needs the Via lines to properly route. It > sounds like in your case you need to move Asterisk (or any B2BUA) forward in > your diagram to be the entity that communicates with your telco. No Via > lines, no proxy.

[OpenSIPS-Users] Removing a VIA line required by Telco.

I've read a considerable number of posts. It doesn't seem like anyone has had to do this. My Telco (no choice in this matter) doesn't "support" using proxies. My call must come from a specific IP (running the proxy) but it must not contain VIA lines with anything other than the IP. Currently, I

Re: [OpenSIPS-Users] OpenSIPS 1.5.0 is released.

Hello Bogdan, Very nice indeed! Congrats! David 2009/3/23 Brett Nemeroff > Woo! Thanks for all your hard work. This looks like a really fantastic > release!!-Brett > > > On Mon, Mar 23, 2009 at 2:25 PM, Bogdan-Andrei Iancu < > bog...@voice-system.ro> wrote: > &

Re: [OpenSIPS-Users] osipsconsole

Thanks Iulia, At the end I got it to work modifying the scrip to hardcode mysql's info and the location of opensipsctlrc (its in /etc/opensips). thanks a lot On Tue, Nov 18, 2008 at 11:35 AM, Iulia Bublea <[EMAIL PROTECTED]>wrote: > David Villasmil wrote: > >> Ha

[OpenSIPS-Users] osipsconsole

Has anyone gotten to work osipsconsole? any info is appreciated! David ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] drounting

... failover was not being done. I'm sure I've done something wrong. I have no failover routes as on the doc it says it is done automatically... Is this right? thanks David On Mon, Nov 17, 2008 at 9:49 AM, Bogdan-Andrei Iancu <[EMAIL PROTECTED] > wrote: > Hi David, > &

Re: [OpenSIPS-Users] drounting

THanks for answering, what do you mean an AVP? THe caller is not registered on opensips, so I want to authorized based on IP. david On Fri, Nov 14, 2008 at 8:22 PM, Jesus Rodriguez <[EMAIL PROTECTED]> wrote: > Hello, > > > > I've been testing drouting, and as alw

[OpenSIPS-Users] Fwd: drounting

sorry to repeat myself, but anyone on this? d -- Forwarded message -- From: David Villasmil <[EMAIL PROTECTED]> Date: Fri, Nov 14, 2008 at 12:29 AM Subject: drounting To: "users@lists.opensips.org" Hello, I've been testing drouting, and as always w

[OpenSIPS-Users] drounting

cted to that domain will be routed using the group id. Now, lets say I want to route based on a source IP instead of username? Say the source IP is NOT registered... what then? Cheers to all David ___ Users mailing list Users@lists.opensips.org h

Re: [OpenSIPS-Users] [Fwd: [Serdev] the sip router project]

YOU THE MAN I KNEW YOU'D GET THAT OUT! hehe! D On Thu, Nov 6, 2008 at 5:06 PM, Bogdan-Andrei Iancu <[EMAIL PROTECTED]>wrote: > Well, if you need more in that area, then stay tune :) > we already have the automatic call ending (from proxy) on timeout ;) > > Regards, > Bogdan > > Brett Nemerof

[OpenSIPS-Users] presence not starting

? Thanks in advance! cheers David ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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