Re: [OpenSIPS-Users] How to convert BYE to CANCEL( status code 480)

What you are asking to do breaks the protocol rules of SIP. Neither Opensips nor any other SIP proxy will do it. I am not sure which you actually wanted as the result as CANCEL and 480 are not the same thing, but you cannot convert a BYE into either one. A CANCEL has to refer to an in-progress

Re: [OpenSIPS-Users] register phone in same local network as opensips

It doesn’t sound like it has anything to do with the registration. It sounds like your router has some sort of SIP Helper application that is trying to assist by re-writing the ports in the INVITE. Many modern routers come with this functionality enabled by default, even though in my experience

Re: [OpenSIPS-Users] s.index madness and my waning grip on sanity

g example just so I have something else to stare at? And if anyone else has any bright ideas, I'm definitely all ears (or eyes); Thanks very much; - Jock On Fri, Jan 27, 2017 at 12:17 PM, Newlin, Ben <ben.new...@inin.com<mailto:ben.new...@inin.com>> wrote: That is not correct. OpenSIP

Re: [OpenSIPS-Users] s.index madness and my waning grip on sanity

That is not correct. OpenSIPS 1.11 definitely contains the index transformation [1]. I’m using it in my code! I think the problem is the quoting. For the example that uses a hard-coded search string it should work if you leave off the quotes. This is how I’m using it. For the variable

Re: [OpenSIPS-Users] Opensips 2.2.2 and top hiding

You said you are doing load balancing as well. Are you doing load balancing on the ACK? What module are you using (dispatcher, loadbalancer, etc.)? Load balancing functions can change the R-URI. Ben Newlin From: on behalf of Denis

Re: [OpenSIPS-Users] usage of setdsturi

I would recommend just using $du. [1] $du = “sip:” + $var(Fqdn) + “:5060”; [1] http://www.opensips.org/Documentation/Script-CoreVar-2-2#toc35 Ben Newlin From: on behalf of "Ramachandran, Agalya (Contractor)" Reply-To:

Re: [OpenSIPS-Users] multiple retransmissions on 4XX after ACK [Solved]

3eb34e5355e54575f8ec391740d3@192.168.36.68:5060<mailto:7f643eb34e5355e54575f8ec391740d3@192.168.36.68:5060> CSeq: 103 ACK User-Agent: FPBX-12.0.76.4(11.5.1) Content-Length: 0 Regards, Richard On 10/10/2016 17:52, Newlin, Ben wrote: It’s impossible to say whether the ACK or the 4XX re

Re: [OpenSIPS-Users] Remove to-tag from 1XX provisional responses

s! On Fri, Oct 21, 2016 at 11:10 PM, Alex Balashov <abalas...@evaristesys.com<mailto:abalas...@evaristesys.com>> wrote: On 10/21/2016 06:36 PM, Newlin, Ben wrote: Not only that, but provisional responses (except 100 Trying) are required to have a To tag [1]. So you would likely r

Re: [OpenSIPS-Users] Remove to-tag from 1XX provisional responses

Not only that, but provisional responses (except 100 Trying) are required to have a To tag [1]. So you would likely run into issues with UAs if you start returning messages without them. You could just drop the provisional replies, but you may run into issues with timeouts in that case as user

Re: [OpenSIPS-Users] multiple retransmissions on 4XX after ACK

4XX after ACK The ack is from the cp. opensips is sending the 4xx. I'll try the same on another provider. Thanks for the info On 7 Oct 2016 18:27, "Newlin, Ben" <ben.new...@inin.com<mailto:ben.new...@inin.com>> wrote: The most likely cause is that there is something wrong in t

Re: [OpenSIPS-Users] multiple retransmissions on 4XX after ACK

t four and it still sends subsequebt ones Regards Richard On 07/10/2016 19:46, Richard Robson wrote: The ack is from the cp. opensips is sending the 4xx. I'll try the same on another provider. Thanks for the info On 7 Oct 2016 18:27, "Newlin, Ben" <ben.new...@inin.com<mailt

Re: [OpenSIPS-Users] multiple retransmissions on 4XX after ACK

The most likely cause is that there is something wrong in the format of the ACK which is causing the far end to not recognize it as being the ACK for that 4XX response. So the far end will continue to retransmit the 4XX response. On the OpenSIPS side, these are recognized as retransmissions so

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

No problem. Thanks, but I am not using 2.2 and not using the topology_hiding module. I am using the Dialog module with the topology_hiding function in 1.11. Ben Newlin From: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Friday, September 30, 2016 at 4:39 AM To: "Newlin, Be

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

in the logs. Ben Newlin From: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Tuesday, October 4, 2016 at 5:28 AM To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topo

Re: [OpenSIPS-Users] sipmsgops and codec_move_up

The priority of the codecs is given by their ordering on the “m=audio” line. PCMU is “0” and you can see in the after SDP that “0” has been moved to the front of the list. The ordering of the “a=rtpmap” lines is not important and does not indicate the preferred codec order. Ben Newlin From:

Re: [OpenSIPS-Users] mulitple instance for different rules

not sure I would recommend it. Ben Newlin From: ‪H Yavari ‪ <hyav...@rocketmail.com> Reply-To: ‪H Yavari ‪ <hyav...@rocketmail.com> Date: Wednesday, September 14, 2016 at 2:10 AM To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list <users@lists.opens

Re: [OpenSIPS-Users] mulitple instance for different rules

<hyav...@rocketmail.com> Date: Wednesday, September 7, 2016 at 9:26 AM To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] mulitple instance for different rules Hi, I want to have physical

Re: [OpenSIPS-Users] mulitple instance for different rules

I would recommend using Docker for this to ensure the 2 instances of OpenSIPS are isolated from each other. We are running OpenSIPS in Docker with multiple containers on a single machine with great success. It also makes local testing very simple, especially with a SIPp container as well.

Re: [OpenSIPS-Users] reject 3xx code

I see now that status is a global parameter that should also be the reply status. I would still recommend using t_check_status(). Ben Newlin From: <users-boun...@lists.opensips.org> on behalf of "Newlin, Ben" <ben.new...@inin.com> Reply-To: OpenSIPS user

Re: [OpenSIPS-Users] reject 3xx code

The comma in your regex is unnecessary; it allows a match for “30,”. It should just be “30[12]”. Also, where is the status variable being set from? There are two places to get the return code: $rs and $T_reply_code. Or you could use t_check_status() like so: if (t_check_status(“30[12]”)) {

Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

take a while for me to revive an older configuration that allows me to enable the memory debugger. Ben Newlin From: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Tuesday, August 2, 2016 at 3:47 AM To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list <

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

is not applied and $du is set to the IP from the incoming message’s Route header, which is my server. Ben Newlin From: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Monday, August 1, 2016 at 7:13 AM To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list <

Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

. Ben Newlin From: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Monday, August 1, 2016 at 10:57 AM To: OpenSIPS users mailling list <users@lists.opensips.org>, "Newlin, Ben" <ben.new...@inin.com> Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

-Andrei Iancu <bog...@opensips.org> Date: Monday, August 1, 2016 at 10:01 AM To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding() Hi Ben, 1) I'm

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

of the incompatibility between these modules would be beneficial as well, as it would have saved me many hours attempting to make this work and may save others that time as well. Ben Newlin From: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Monday, August 1, 2016 at 7:13 AM To: "

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

topology. Ben Newlin From: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Friday, July 29, 2016 at 8:40 AM To: "Newlin, Ben" <ben.new...@inin.com>, OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

This is 1.11.6, running on CentOS 7. Ben Newlin From: <users-boun...@lists.opensips.org> on behalf of Bogdan-Andrei Iancu <bog...@opensips.org> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Friday, July 29, 2016 at 8:50 AM To: "Newlin, Be

Re: [OpenSIPS-Users] How to tell if uac_auth() is successful

g list <users@lists.opensips.org>, "Newlin, Ben" <ben.new...@inin.com> Subject: Re: [OpenSIPS-Users] How to tell if uac_auth() is successful Hi Ben The uac_auth() should return true if the authentication was successfully performed. Are you sure you correctly perform the test

Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

their behavior. Ideally the Record-Route headers from previous replies could be used in this case to allow the call to succeed, but I don’t know if that is possible. Thanks, Ben Newlin From: "Newlin, Ben" <ben.new...@inin.com> Date: Wednesday, July 27, 2016 at 9:44 AM To: Bogdan-A

Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

:57 AM To: OpenSIPS users mailling list <users@lists.opensips.org>, "Newlin, Ben" <ben.new...@inin.com> Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes Hi Ben, First, if you use TH, makes no sense to do Record-Routing - there are 2 SIP concepts that overlaps. Yo

Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

so there will be no Route/RR headers at all. So no need to do loose_route or so. You just do TH matching for the sequential requests and nothing more. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 22.07.2016 16:48, Newlin, Ben wrote: Hi,

[OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

I have had 3 OpenSIPS server crashes in the last week. All were due to segmentation faults. I was not able to capture core dumps; I am configuring that now to catch the next crash. My logs leading up to the crash are full of errors from fix_route_dialog() complaining about invalid URIs for

[OpenSIPS-Users] How to tell if uac_auth() is successful

I am using the UAC and UAC_AUTH modules to perform trunk authentication. The uac_auth() function will only use credentials when the realm matches one in the 401/407 challenge. Is there any way to tell whether there was a successful authentication match? The function doesn’t provide any return

Re: [OpenSIPS-Users] Response handling for Re-INVITE and BYE

Failure routes, reply routes, and branch routes are all transaction specific and will only be triggered for the specific transaction for which they have been armed; in this case, the initial INVITE. If you want replies to sequential requests to also hit the reply route, you must add

[OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi, I am using the Dialog module with topology_hiding() in my server and I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. I have found what I believe is an inconsistency in the handling of Record-Route within the Dialog

Re: [OpenSIPS-Users] Issue with ACK and rtpproxy, setID

AVPs are tied to a transaction, so the transaction must be matched before they will be available. You should use t_check_trans() to do this. However, I think this will not work for you because ACKs are their own transactions and I don’t believe they will have access to the AVPs from the INVITE

Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header

; Date: Tuesday, June 28, 2016 at 5:45 AM To: "Newlin, Ben" <ben.new...@inin.com>, "users@lists.opensips.org" <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header Hi Ben, You mean you used the re

Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header

to perform any tests or get tracing/logs. Thanks! [1] https://github.com/OpenSIPS/opensips/issues/917 Ben Newlin From: "Newlin, Ben" <ben.new...@inin.com> Date: Monday, June 27, 2016 at 11:06 AM To: Bogdan-Andrei Iancu <bog...@opensips.org>, "users@lists.opensips.org" &l

Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header

: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Monday, June 27, 2016 at 10:41 AM To: "Newlin, Ben" <ben.new...@inin.com>, "users@lists.opensips.org" <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong addr

Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header

appears to be working, but the local generated ACK is not using that address. Ben Newlin From: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Monday, June 27, 2016 at 5:37 AM To: "users@lists.opensips.org" <users@lists.opensips.org>, "Newlin, Ben" <ben.new...@ini

[OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header

I have run into the same problem that was described in this previous post [1], however it doesn’t appear it was ever solved at the time. I am using the dispatcher module to route calls to external carriers and I am using set_advertised_address to set the outgoing public address prior to

Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

It also seems like AVPOPS module [1] may be a good solution here as it has functions to pull data from a database into AVPs based by user. [1] http://www.opensips.org/html/docs/modules/2.2.x/avpops.html Ben Newlin From: on behalf of sevpal

Re: [OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

Bogdan, Thanks. I assume the parameter will be available in 1.11.9? Ben Newlin From: Bogdan-Andrei Iancu <bog...@opensips.org> Date: Tuesday, June 21, 2016 at 9:42 AM To: OpenSIPS users mailling list <users@lists.opensips.org>, "Newlin, Ben" <ben.new...@inin.com

[OpenSIPS-Users] TM flag 0x01 not working in 1.11.7

In my script I send all “100 Trying” responses manually* and use the 0x01 flag when calling t_relay so that it will not send its own “100 Giving a try” response, as per the documentation [1]. Unfortunately, in 1.11.7 and 1.11.8 this flag seems to have stopped working which results in multiple

[OpenSIPS-Users] Bad 1.11.7 tarball on open sips.org

It looks like a new tarball for OpenSIPS 1.11.7 was uploaded earlier today, but it appears to have a bad format. I cannot extract it with any of my usual archive utilities. Can anyone else confirm? It was working fine yesterday with the previous tarball.

Re: [OpenSIPS-Users] M4 config generation issue

You could also have wrapped the DB_PASS in an extra set of m4-style quotes: define( `DB_PASS',``some#password'') Ben Newlin From: > on behalf of John Nash

Re: [OpenSIPS-Users] opensips 2.1.2 not accepting transformation in if statement and t_relay()

Stefan, $var type variable have process-level scope [1]. This means they are only valid in the same script process that initialized them. It is very likely in your case that the startup_route is running in a different child process than the message processing routes and that is causing the

Re: [OpenSIPS-Users] cache_fetch on failure route issue

In on_reply_route all pseudo-variables are evaluated by default in the context of the reply, but in failure_route they are evaluated by default in the context of the request. I think you are probably getting a different value for $si than you are expecting in the failure_route case and there is

Re: [OpenSIPS-Users] question about avp DB querry

In the case where you perform the SQL query I don’t see where you are actually setting the result into the Request-URI. You seem to only be storing it in the cache. I think you need to add: $rd = $avp(PBX); to the else leg. Ben Newlin From:

Re: [OpenSIPS-Users] presence xpidf

I know I have used presence extensively with Polycom phones using pidf+xml. I know they support it. Maybe there is some setting in your model specifying the remote server type? If that is set to Microsoft Lync the Polycom may be sending xpidf for compatibility. But Polycom phones absolutely

Re: [OpenSIPS-Users] uac_replace_from multiple times?

If you use uac_replace_from in a branch route then the changes are specific to that branch. If the call fails the header will be reverted and can be modified again in another branch. Ben Newlin From: > on behalf of

Re: [OpenSIPS-Users] CARRIERROUTE module

users mailling list Date: Wednesday, August 12, 2015 at 2:40 AM To: OpenSIPS users mailling list, Newlin, Ben Subject: Re: [OpenSIPS-Users] CARRIERROUTE module Hi Ben, The carrierroute module is not maintained (for like 6 years). Still, if you want to do a routing based on both caller and caller

[OpenSIPS-Users] CARRIERROUTE module

Hi all, We have some interest in possibly using the CARRIERROUTE module instead of DROUTING as it allows prefix matching of the originating number as well as the destination, whereas DROUTING must match the originating user exactly. However, we noticed that the CARRIERROUTE module is labeled

Re: [OpenSIPS-Users] Compact/Full Headers

/expanded headers. Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, August 4, 2015 at 12:25 PM To: OpenSIPS users mailling list, Newlin, Ben Subject: Re: [OpenSIPS-Users] Compact/Full Headers Hi Ben, The OpenSIPS 2.1 has the compression module that is able to do that for you automatically

Re: [OpenSIPS-Users] Compact/Full Headers

that for any given header? Ben Newlin From: Newlin, Ben Date: Friday, July 10, 2015 at 5:54 PM To: users@lists.opensips.orgmailto:users@lists.opensips.org Subject: Compact/Full Headers Is there a simple way to convert Compact headers to their full form and vice versa while processing a message? I

[OpenSIPS-Users] CANCEL not including rport parameter in VIA

We are using the topology_hiding functionality of the Dialog module in OpenSIPS 1.11. We have 2 OpenSIPs servers set up in tandem, with one acting as the load-balancer and the other as the proxy. Only the load-balancer has a public IP address, so on the proxy we use

[OpenSIPS-Users] Compact/Full Headers

Is there a simple way to convert Compact headers to their full form and vice versa while processing a message? I haven’t found much mention of compact headers except that they are supported. I also saw the new compression module in 2.1, but we are using 1.11. Ben Newlin

Re: [OpenSIPS-Users] Route_to_gw not pushing gw_id to the first index of avp

to the first index of avp On Jul 1, 2015 3:44 PM, Newlin, Ben ben.new...@inin.commailto:ben.new...@inin.com wrote: Sorry, actually the documentation confused me. You are correct in the way that it should work if the do_routing() and route_to_gw() are really pushing the functions on the end

Re: [OpenSIPS-Users] Route_to_gw not pushing gw_id to the first index of avp

access the avp, we can either get explicitly the specific index, or we get the last index which is `gw2`. So if I'm not mistaken, then I don't understand what the problem is here? Correct me if this is wrong. On Wed, Jul 1, 2015 at 11:46 AM, Newlin, Ben ben.new...@inin.commailto:ben.new

Re: [OpenSIPS-Users] Route_to_gw not pushing gw_id to the first index of avp

is `gw2`. So if I'm not mistaken, then I don't understand what the problem is here? Correct me if this is wrong. On Wed, Jul 1, 2015 at 11:46 AM, Newlin, Ben ben.new...@inin.commailto:ben.new...@inin.com wrote: I found similar behavior in my implementation, except that I use dr_routing() instead

Re: [OpenSIPS-Users] Route_to_gw not pushing gw_id to the first index of avp

I found similar behavior in my implementation, except that I use dr_routing() instead of route_to_gw(). When the AVP is already populated, these functions simply push on the end. This may be an implementation decision, but it is certainly not intuitive or what I expected. I am currently using

Re: [OpenSIPS-Users] Double via in MI generated NOTIFY

NOTIFY to the phone. The only thing is that the packet has 2 Via headers with the same IP and port of OpenSIPS server. As I said, it works, but looks weird. On Wed, Jun 3, 2015 at 7:00 PM, Newlin, Ben ben.new...@inin.commailto:ben.new...@inin.com wrote: It sounds like you may be sending

Re: [OpenSIPS-Users] Double via in MI generated NOTIFY

It sounds like you may be sending the NOTIFY to yourself when you use the domain name instead of the IP. Have you verified the address that the domain resolves to? Is it the same as the OpenSIPS instance? Ben Newlin From: Stas Kobzar Reply-To: OpenSIPS users mailling list Date: Wednesday, June

Re: [OpenSIPS-Users] Residential Script - noob question

James, Only SIP Requests are processed by the route function. That is why it makes sense for the INVITEs, REGISTERs, and OPTIONs, BYEs, ACKs, etc. The 486 is a reply and is handled by OpenSIPS according to normal SIP rules. It will not cause the script to run at all unless you are using the

Re: [OpenSIPS-Users] Dialog start event over EVI

The usage of $dlg_val is described in the module documentation [1]. You set and read them just like any other pseudo-variable: $dlg_val(foo) = “bar”; The only thing I don’t know is whether these variables can be accessed from the E_DLG_STATE_CHANGED event route, as it is not clear whether the

Re: [OpenSIPS-Users] Dialog start event over EVI

You can add variables to the Dialog using $dlg_var. Ben Newlin From: DanB Reply-To: OpenSIPS users mailling list Date: Monday, May 11, 2015 at 8:11 AM To: Bogdan-Andrei Iancu, OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Dialog start event over EVI For now no emergency since we

Re: [OpenSIPS-Users] ANIs check for source number

There is also a function in the URI module for checking if the user portion of a URI is an E.164 number. http://www.opensips.org/html/docs/modules/1.11.x/uri.html#id294513 From: Podrigal, Aron Reply-To: OpenSIPS users mailling list Date: Thursday, April 30, 2015 at 1:20 PM To: OpenSIPS users

Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client

Something in between is manipulating the addressing in the SIP message. You said that ALG was disabled in the router, but either that is incorrect or there is some other piece of equipment changing the message. This is clear because the contact is not the only header that is changed. The

Re: [OpenSIPS-Users] URGENT: contact header changed between opensips and client

do ALG? On Thu, Apr 9, 2015 at 10:39 AM, Newlin, Ben ben.new...@inin.commailto:ben.new...@inin.com wrote: Something in between is manipulating the addressing in the SIP message. You said that ALG was disabled in the router, but either that is incorrect or there is some other piece of equipment

Re: [OpenSIPS-Users] Regex on R-URI question

“$rU@$rd:$rp” $rU – R-URI Username $rd – R-URI Domain $rp – R-URI Port http://www.opensips.org/Documentation/Script-CoreVar-1-11 Ben Newlin From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Satish Patel Sent: Tuesday, April 07, 2015 10:23 AM To:

Re: [OpenSIPS-Users] set_advertised_address suspected bug 1.9

Terrance, Yes, I am sure it works. I have changed the values in onreply_route and see them change in the message. I believe Razvan’s comment applies only when operating in normal proxy mode. The usage of the Dialog module and topology_hiding() is what makes it work. Ben Newlin From:

Re: [OpenSIPS-Users] set_advertised_address suspected bug 1.9

Yes, topology_hiding(). I am actually doing exactly what you have described wanting to do. I am using set_advertised_address() in request route and then in onreply_route to change the address in each direction. When topology_hiding() is being used, the module is already going to replace the IP

Re: [OpenSIPS-Users] set_advertised_address suspected bug 1.9

Terrance, I am doing something similar and have found the “topology hiding” feature of the dialog module will do this. The feature’s purpose is to re-write all IP information in SIP headers to hide network topology, but in the process it will also perform the type of translation you are

Re: [OpenSIPS-Users] DROUTING module changing Carrier IDs

-Andrei Iancu [mailto:bog...@opensips.org] Sent: Thursday, March 26, 2015 10:02 AM To: OpenSIPS users mailling list; Newlin, Ben Subject: Re: [OpenSIPS-Users] DROUTING module changing Carrier IDs Hi Ben, If you run the dr_carrier_status MI command, do you see the same #012 in the ID of the carrier

Re: [OpenSIPS-Users] DROUTING module changing Carrier IDs

places in the logs, which also do not appear when logging to stderr. Sorry for the false alarm, but I appreciate the help to point me in the right direction. Ben Newlin From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Newlin, Ben Sent: Thursday, March 26

[OpenSIPS-Users] DROUTING module changing Carrier IDs

I am using OpenSIPS 1.11 and I am recently getting some weird behavior from the DROUTING module. The module is corrupting the Carrier ID that is being returned via AVP. This behavior is occurring only with the Carrier ID, not the gateway or rule IDs, and only when it is returned from the

Re: [OpenSIPS-Users] B2BUA Proxy Interaction

| ben.new...@inin.com Interactive Intelligence Deliberately Innovative www.inin.comhttp://www.inin.com/ From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Friday, December 26, 2014 2:59 AM To: OpenSIPS users mailling list; Newlin, Ben Subject: Re: [OpenSIPS-Users] B2BUA Proxy Interaction Hi

[OpenSIPS-Users] B2BUA Proxy Interaction

I have been reading through the documentation for the B2BUA functionality. I want to use a B2BUA while also having many of the proxy functionalities like drouting, accounting, etc. However, it is indicated that they do not work easily together. The B2BUA Tutorial says this: A solution is to