Re: [OpenSIPS-Users] Managing the implicit subscription and its NOTIFYs within B2BUA REFER handling

the subscription > prematurely by unsubscribing or refreshing the subscription. > > [1] https://opensips.org/docs/modules/3.2.x/b2b_logic.html#func_b2b_bridge > > Regards, > > -- > Vlad Patrascu > OpenSIPS Core Developerhttp://www.opensips-solutions.com > > On 24.01.2022 17:

[OpenSIPS-Users] Managing the implicit subscription and its NOTIFYs within B2BUA REFER handling

OpenSIPS 3.2 supports some slick in-script B2B functions, documented here [1] <[1] https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2#toc10> with an example for handling REFER. Does this approach include support for the subscription and the subsequent NOTIFYs required by RFC 3515 2.4.4 [2]

[OpenSIPS-Users] TCP-related errors

Hello, I have two similarly configured systems running a recent OpenSIPS 3.2 with may errors like this: CRITICAL:core:io_watch_add: >>> fd_array idx 1 (fd=193) points to bogus map (fd=-1,type=0,flags=2000,data=(nil)) It seems you have hit a programming bug. Please help us make OpenSIPS bette

Re: [OpenSIPS-Users] double sdp.

Johan, To avoid problems like this, I call rtpengine_offer() in branch_route on initial invites, and make sure to call rtpengine_delete() in any failure route to remove any session from a failed offer that was never used. Perhaps these will help in your situation as well. - Jeff On Mon, Oct 18

[OpenSIPS-Users] rtpengine weight configuration

Hello, This is on OpenSIPS 2.4, although I suspect the answer will be the same for any modern-ish OpenSIPS version. I have a collection of rtpengine entries loaded into the rtpengine table. All works well. The "rtpengine_show" fifo command lists them all with weight=1. If I'd like to load some

Re: [OpenSIPS-Users] rtpengine sets - load balancing proxy weighting

This may be a good place to ask this, or perhaps not. Is it possible to define the DB rows in such a way that causes a priority-based behavior? Prefer this one, move to the next if the first isn't available, etc. As I think through it, it may be possible to do that manually in the script by call

Re: [OpenSIPS-Users] strange behavior with TCP reply port

this might break? - Jeff On Mon, Aug 16, 2021 at 4:06 PM Liviu Chircu wrote: > On 11.08.2021 22:01, Jeff Pyle wrote: > > the sl_send_reply() function opens a new TCP socket to the UAC on the > IP:port listed in the original message's Contact, rather than sending the > 100 on

[OpenSIPS-Users] strange behavior with TCP reply port

Hello, This is on the 3.1.3~20210731~b333a222f-1 from the Debian 3.1-nightly repo. Typically I run with modparam("tm", "auto_100trying", 0) so I can manually send a 100 with sl_send_reply(100, "Trying"); earlier in the script, before blocking processes like DB lookups and such. No problem

Re: [OpenSIPS-Users] Delayed Bye?

Eek. This sounds like a recipe for future problems with this carrier. If you really want to try, you could do a sleep() or usleep() on BYE requests in the section of your script that processes sequential requeststhe has_totag() and loose_route() part (assuming no topology_hiding). That's goi

Re: [OpenSIPS-Users] Remove previously added header

Add it in a branch_route. That way if you have to route advance, it'll already be gone because you'll be on a new branch. - Jeff On Mon, Jun 14, 2021 at 5:52 PM David Villasmil < david.villasmil.w...@gmail.com> wrote: > Hello guys, > > So, I'm appending a header (append_hf("header")) to a for

Re: [OpenSIPS-Users] Getting header from 302

DIRECT will come in here. > onreply_route[reply_from_REDIRECT] > { > xlog("L_ERR", "[$ci][$rm]: I'm in onreply_route[reply_from_REDIRECT]"); > if (t_was_cancelled()) { > exit; > } > > # detect redirect, store the header and send

Re: [OpenSIPS-Users] Getting header from 302

If I arm both t_on_failure() and t_on_reply(), do a t_relay(), and a 302 comes back, I have access to the reply in the onreply_route, then the failure_route. From a SIP perspective, a 302 is a failure since it's not 2xx-series, no? I don't do a drop() in the onreply_route. It just naturally foll

Re: [OpenSIPS-Users] MS teams, reinvite after ACK

_relay() also seq Invites, after this >> everything works ok. >> >> Just on question, regarding transfers, i see that MS Teams send REFER in >> which trafter is defined. How do you deal with this? You do not allow REFER >> from MS teams and hope that MS teams will send new

Re: [OpenSIPS-Users] Getting header from 302

@gmail.com > phone: +34669448337 > > > On Tue, Jun 1, 2021 at 6:46 PM Jeff Pyle wrote: > >> In my experience you don't need drop() in the reply route. Just store >> the AVP and move on. Something like this: >> >> onreply_route[collect_identity] { >&g

Re: [OpenSIPS-Users] Getting header from 302

and ignore the 302 and send to the invite route again to >> actually send the call out via do_routing. >> >> What I'm trying to do is: >> >> - On receiving an invite: forward to an endpoint. >> >> - This endpoint will simply reply with 302 including a header

Re: [OpenSIPS-Users] Getting header from 302

gt; Thanks again Jeff! > > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > > > On Tue, Jun 1, 2021 at 12:37 PM Jeff Pyle wrote: > >> In which route are you trying to use if (is_present_hf("Identity"))? >> Si

Re: [OpenSIPS-Users] MS teams, reinvite after ACK

Miha, First, do you need to use "mtsbc.test.com:5060" in the first record_route_preset() param? Can you use the IP address of your proxy instead? FQDNs are legal of course, but outside of MS Teams' implementation, they're rarely required. It's just another thing to go wrong. Especially while t

Re: [OpenSIPS-Users] Getting header from 302

In which route are you trying to use if (is_present_hf("Identity"))? Since the 302 is both a reply and a "failure", I suggest seeing if it appears in either the armed onreply_route or failure_route. I think From is available because it was present in the original request route. - Jeff On Tue,

Re: [OpenSIPS-Users] Ignore redirect?

David, You catch the 302 in the failure_route, decide if it's what you want, and if not just send to another route block for do_routing and such? - Jeff On Tue, May 25, 2021 at 5:12 PM David Villasmil < david.villasmil.w...@gmail.com> wrote: > Hello guys, > > I'm receiving an INVITE which I fo

Re: [OpenSIPS-Users] Tips for troubleshooting OpenSIPS as a Teams SBC

John, This has worked brilliantly -- the domain module took care of teaching OpenSIPS what's local. As far as a single versus double RR, the problem turned out to be with a downstream SBC that couldn't resolve the domain name properly. Once fixed, the single RR works as expected. Regards, Jeff

Re: [OpenSIPS-Users] MS team issue

For inbound from Teams, I use the permissions module with the following: MariaDB [opensips_teamsproxy]> select * from address where grp=1; ++-+-+--+--+---+-+--+ | id | grp | ip | mask | port | proto | pattern | context_info | ++-

Re: [OpenSIPS-Users] Tips for troubleshooting OpenSIPS as a Teams SBC

John, This is great, especially along side Alexey's blog post. I'm having a problem related to multi-tenant configuration. I suspect it's the result of something basic I've overlooked since I don't see it covered in your article. An ancillary issue that I have is where I must use the FQDN *and*

Re: [OpenSIPS-Users] Dialogs with fix_nated_contact() have wrong RURI domain on sequential requests

John, Certainly. - Jeff On Thu, Feb 18, 2021 at 10:47 AM John Quick wrote: > Jeff, > > Would it be okay if I included this as an interesting 'special case' to my > recently published article? > The article is all about fixing Contact headers - when and how to do it, > when not to do it, etc.

Re: [OpenSIPS-Users] where to catch CANCELs in script for branches canceled by tm module

A week ago I asked the question where the CANCELs appear in the script when tm sends then to cancel losing branches after a winning one presents itself. No takers so far. Any thoughts? - Jeff On Fri, Feb 5, 2021 at 1:01 PM Jeff Pyle wrote: > Hello, > > This is on OpenSIPS 3.1.1 (

Re: [OpenSIPS-Users] Dialogs with fix_nated_contact() have wrong RURI domain on sequential requests

And voilà, $du preserves the routing destination while $ru contains the original contact before NAT mangling. Excellent. - Jeff On Fri, Feb 5, 2021 at 2:58 PM Jeff Pyle wrote: > This makes sense. I've been able to store the values, but I'm still > experimenting with th

Re: [OpenSIPS-Users] (no subject)

Elaine, You can use the avp_db_query() function to run queries to a database. This is to manually query a database table that OpenSIPS doesn't use automatically with one of its modules. The blacklist module loads its da

Re: [OpenSIPS-Users] Dialogs with fix_nated_contact() have wrong RURI domain on sequential requests

contact); > } > > Hope this helps, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 1/26/21 6:36 PM, Jeff Pyle wrote: > > Hi Johan, > > > > There typically isn't loose_route() in my script because there >

[OpenSIPS-Users] where to catch CANCELs in script for branches canceled by tm module

Hello, This is on OpenSIPS 3.1.1 (abe7ab7c7). I use t_relay() after lookup() to relay INVITEs to all registered contacts for an AOR. I use rtpengine_offer() per-branch, and I keep track of the branches with rtpengine's extra_id_pv modparam and via-branches=extra. To do this right I should use r

Re: [OpenSIPS-Users] Dialogs with fix_nated_contact() have wrong RURI domain on sequential requests

nge the loose route part to fix route dialog ? > > Op di 26 jan. 2021 om 16:39 schreef Jeff Pyle : > >> Hello, >> >> This is on OpenSIPS nightly 3.1.1~20210125~8bab0da7b-1. >> >> I have a registrar configured with basic call routing between the >> regis

[OpenSIPS-Users] Dialogs with fix_nated_contact() have wrong RURI domain on sequential requests

Hello, This is on OpenSIPS nightly 3.1.1~20210125~8bab0da7b-1. I have a registrar configured with basic call routing between the registered AORs. I use topology_hiding("D") to create the dialog on calls and normal stuff like has_totag() and topology_hiding_match() for sequential request handling

Re: [OpenSIPS-Users] interaction between fix_nated_contact(), topology_hiding() and serial forking

for the second > branch? Are you calling t_newtran() anywhere in your script? > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 10/28/20 8:30 PM, Jeff Pyle wrote: > > Liviu, > > > > It looks like the fix

Re: [OpenSIPS-Users] interaction between fix_nated_contact(), topology_hiding() and serial forking

Liviu, It looks like the fixed/update contact is lost only when topology_hiding() is involved. Would you prefer a separate issue, or shall I append the issue you referenced before? - Jeff On Wed, Oct 28, 2020 at 2:15 PM Jeff Pyle wrote: > Hey Liviu, > > fix_nated_contact

Re: [OpenSIPS-Users] interaction between fix_nated_contact(), topology_hiding() and serial forking

Liviu Chircu wrote: > Hi! > > On 28.10.2020 18:49, Jeff Pyle wrote: > > First, I lose the updated Contact from fix_nated_contact() after a serial > fork. Is this expected? > > I would assume the `fix_nated_contact()` lump changes get backed up into > shared memory

Re: [OpenSIPS-Users] interaction between fix_nated_contact(), topology_hiding() and serial forking

ot;second attempt to change URI Contact" error when running fix_nated_contact() in the branch_route[]. This feels like a bug. - Jeff - Jeff On Tue, Oct 27, 2020 at 8:31 PM Jeff Pyle wrote: > Hello, > > This is on OpenSIPS 2.4.8. Early in the script: > >

[OpenSIPS-Users] interaction between fix_nated_contact(), topology_hiding() and serial forking

Hello, This is on OpenSIPS 2.4.8. Early in the script: if (nat_uac_test("34")) { # 2, 32 setflag(NAT_FLAG); fix_nated_contact(); force_rport(); } Later in the script I run topology_hiding("CD"), and then t_relay() an

Re: [OpenSIPS-Users] can't get is_contact_registered() to return true

ircu wrote: > On 08.10.2020 16:40, Jeff Pyle wrote: > > > > This loads the registered contacts as branches, which allows me to > > iterate through each and compare 1) the base URI from the inbound > > INVITE with the base URI of the registered contact, 2) the received

Re: [OpenSIPS-Users] can't get is_contact_registered() to return true

eceived port. Assuming one matches, that's enough to satisfy me that we've received a message from a registered endpoint. I'm calling this route on initial INVITEs and so far, so good. There are optimizations that could occur, such as exiting the while loop once we've found a mat

Re: [OpenSIPS-Users] can't get is_contact_registered() to return true

INVITE's Contact is " sip:+12162454107@192.168.201.123:4135". is_contact_registered() fails. Is that caused by the lack of the rinstance param, or something else? - Jeff On Wed, Oct 7, 2020 at 3:11 PM Liviu Chircu wrote: > On 07.10.2020 21:15, Jeff Pyle wrote: > > At

Re: [OpenSIPS-Users] can't get is_contact_registered() to return true

ct is that $ct is wrapped in <>. I wrote a nasty regex to strip the <> characters and it works with is_contact_registered(), at least from this phone. - Jeff On Wed, Oct 7, 2020 at 10:38 AM Liviu Chircu wrote: > On 07.10.2020 17:18, Jeff Pyle wrote: > > Am I missing

[OpenSIPS-Users] can't get is_contact_registered() to return true

Hello, This is on OpenSIPS 3.1.0 git revision 499831928. I'm attempting to verify an inbound message is coming from a registered source. is_registered("location") returns true. is_ip_registered("location", , $si) also returns true. is_contact_registered("location"), however, does not. I've pl

Re: [OpenSIPS-Users] learning the realm from authentication challenges

I am not route-advancing in a typical way, so my application of credentials is a bit different perhaps. The environment I'm in has a variety of customer-facing platforms, over a dozen at last count. Some are for trunking, some hosted, some hybrid. The platform I'm writing on OpenSIPS is a testin

Re: [OpenSIPS-Users] learning the realm from authentication challenges

Johan, I will definitely try that. Thank you! Ben, The problem is I have multiple destinations with the same realm. In my case, several different Broadworks app servers. I haven't checked them exhaustively but I think they all reply with realm="BroadWorks" in their authentication headers.

Re: [OpenSIPS-Users] learning the realm from authentication challenges

Good catch on Proxy-Authorization vs Proxy-Authenticate. I think I've been looking at this too long. I checked the module and that's exactly what it is. My hope was to load the uac_auth user/pass AVPs ahead of time from a DB based on where I knew I was sending the call, load the realm one in the

[OpenSIPS-Users] learning the realm from authentication challenges

I'm trying to recover the realm of an auth challenge to OpenSIPS so I can respond to it with the uac_auth() function, and that requires knowing the realm. The docs say that $ar should provide that, perhaps written like $(ar) to get i

Re: [OpenSIPS-Users] opensips-cli connection problems

ons? That's weird to me. Looking back now I'm not sure where I found the "url" type. If I come across it again, I'll let you know. "http" is working great! - Jeff On Thu, Sep 24, 2020 at 10:28 AM Liviu Chircu wrote: > On 24.09.2020 17:15, Jeff Pyle wrot

[OpenSIPS-Users] opensips-cli connection problems

Hello, I'm having a rough time getting opensips-cli to connect to my OpenSIPS instance. If I try fifo, I get a permissions problem... PermissionError: [Errno 13] Permission denied: '/tmp/opensips-class4_fifo' ...even though: # ls -lh /tmp/opensips-class4_fifo prw-rw-rw- 1 opensips opensips 0 S

[OpenSIPS-Users] Command cannot be used in the block

Hello, I'm using 3.1.0~20200919~c5ef382ac-1 from the 3.1-nightly APT repo. I'm porting over a config from 2.4 that works well. The problem I'm having is with the gflags module, specifically the is_gflag() function. I get an error when I try to use it in onreply_route, failure_route or branch_ro

Re: [OpenSIPS-Users] opensips-cli permissions problem on database creation

No customizations to any opensips-cli.cfg anywhere. It doesn't exist on this system. MariaDB [(none)]> create user `opensips`@`localhost` identified by 'opensipsrw'; Query OK, 0 rows affected (0.001 sec) MariaDB [(none)]> grant all privileges on opensips.* to `opensips`@`localhost`; Query OK, 0

[OpenSIPS-Users] opensips-cli permissions problem on database creation

Hello, I'm attempting to use opensips-cli to create the initial databases for OpenSIPS 3.1 with opensips-cli 0.1~20200901~524c020-1 installed from 3.1-nightly. The database is MariaDB 10.3.23-0+deb10u1 installed from the 'default-mysql-server' package. The system is Debian 10 on a container crea

Re: [OpenSIPS-Users] drouting probe_mode in active/passive cluster

Hi Alexey, I see the "dr_enable_probing" MI command; that's great! Is there an equivalent modparam I can add to default to 0, so that it will only probe when we know it has the IP (when enabled by MI)? It will be some time until I'm able to cherry-pick and test. The two proxies are currently ru

[OpenSIPS-Users] drouting probe_mode in active/passive cluster

Hello, On v2.4.7 I have a two-node cluster configured as active/standby with keepalived managing which side has the IP. I want to use drouting probe_mode=2, but I don't see how to prevent the passive side from trying to ping the gateways even when it doesn't have the IP. And, of course, it fails

[OpenSIPS-Users] drouting gateway status with route_to_gw()

Hello, On v2.4.7 I'm using route_to_gw() to load routing for a particular gateway. It works well. It also works when the gateway is in an "Inactive" status, which surprises me. I would expect the function to return with an error code in that scenario, or at least have the option to achieve that

Re: [OpenSIPS-Users] ratelimit network algorithm traffic limit

2 Mar 2020 at 18:24, Ovidiu Sas wrote: >> >> Take a look at the pipe parameter: >> https://kamailio.org/docs/modules/5.4.x/modules/ratelimit.html#idp49883756 >> >> # define pipe 4 with a limit of 1 pending bytes in the rx_queue >> # using NETWORK algorith

Re: [OpenSIPS-Users] ratelimit network algorithm traffic limit

uot;, "Server Unavailable"); > exit; > }; > > > > On Mon, 2 Mar 2020 at 16:19, Jeff Pyle wrote: > >> This doesn't appear to have anything to do with a any type of limit or >> the network algorithm. In fact, it says, "...and

Re: [OpenSIPS-Users] ratelimit network algorithm traffic limit

; > https://opensips.org/html/docs/modules/2.4.x/ratelimit.html#param_limit_per_interval > > > On Mon, 2 Mar 2020 at 15:54, Jeff Pyle wrote: > >> Hello, >> >> The ratelimit doc page (v2.4) section 1.3.4 says the following: "If the >> returned amount excee

[OpenSIPS-Users] ratelimit network algorithm traffic limit

Hello, The ratelimit doc page (v2.4) section 1.3.4 says the following: "If the returned amount exceeds the limit specified in the modparam, rl_check returns an error." The problem is I don't see a modparam to define the limit. - Jeff ___ Users maili

Re: [OpenSIPS-Users] HA ongoing call support in 2.4 - transaction replication

ndaries because of the timing and need to happen quickly for > synchronization. There are plenty of infrastructure components with this > guideline for similar reasons. > > > > On Thu, Feb 20, 2020 at 10:48 AM Jeff Pyle wrote: > >> Alexei, >> >> I see the arti

Re: [OpenSIPS-Users] HA ongoing call support in 2.4 - transaction replication

Alexei, I see the article. In summary, transactions are too complicated to synchronize between nodes of a cluster because of their short timing intervals and complex structures. Instead the approach is to get the messages of a transaction back to the individual node that owns the transaction so

[OpenSIPS-Users] HA ongoing call support in 2.4 - transaction replication

Hello, I'm attempting to implement an active/backup pair of OpenSIPS v2.4 proxies. I'm using keepalived to move the IP between the two instances, and clusterer module to allow the two proxies to talk to each other. The dialog module is replicating its information, and keepalived successfully set

Re: [OpenSIPS-Users] rtpengine + manual SDP manipulations

or more of: >>> >>>- >>> >>>origin >>> >>>Replace the address found in the *origin* (o=) line of the SDP body. >>>Corresponds to *rtpproxy* o flag. >>>- >>> >>>session connection or ses

Re: [OpenSIPS-Users] rtpengine + manual SDP manipulations

session-level connection* (c=) line >of the SDP body. Corresponds to *rtpproxy* c flag. > > > Have you tried this? > > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > > > On Sat, Jan 18, 2020 at 3:59 PM Je

[OpenSIPS-Users] rtpengine + manual SDP manipulations

Hello, I'm running OpenSIPS 2.4 and rtpengine 7.0. I have the following commands in my script: route[sanitize_sdp] { subst_body('/^o=\S+ /o=- /'); subst_body('/^s=.*/s=-\r/'); } This works fine, unless I'm also using rtpengine, in which case these subst_body() changes are lost.

[OpenSIPS-Users] mid_registrar mode=0 expires problem

Hello, This is on OpenSIPS 2.4.5 from the apt.opensips.org repo. I'm running mid_registrar in mode=0 with all parameters default. The REGISTER message comes from the client towards OpenSIPS with an expires=3600 in the Contact. I run mid_registrar_save("location") and relay the REGISTER upstream

[OpenSIPS-Users] combining topology_hiding and mid_registrar

Hello, My v2.4 implementation is a type of SBC suitable for the edge of a customer network. I have topology_hiding working for INVITEs. I have mid_registrar working for REGISTERs. I do not have them working together. I'd like to completely hide the topology of the inside network when sending t

Re: [OpenSIPS-Users] benchmark bm_start_timer / bm_log_timer across async functions

Never mind. It helps if you bm_start_timer() and bm_stop_timer() with the same name! - Jeff On Tue, Sep 18, 2018 at 5:54 PM Jeff Pyle wrote: > Hello, > > On 2.4.2 I'm running bm_start_timer("db_query_time") right before I call > async(avp_db_query()), and bm_log_t

[OpenSIPS-Users] benchmark bm_start_timer / bm_log_timer across async functions

Hello, On 2.4.2 I'm running bm_start_timer("db_query_time") right before I call async(avp_db_query()), and bm_log_timer("db_query_time") first thing in the resume route. The results are strange: benchmark (timer db_query_time [1]): 1537307242867591 [ msgs/total/min/max/avg - LR: 1/15373072428675

[OpenSIPS-Users] Add Contact parameters

Hello, Is it possible to add a parameter to the Contact header as a message passes through the proxy? Specifically I need to add at least one parameter to the Contact user. I realize this isn't as simple as it seems on the surface since it would probably have to be managed on every message passi

Re: [OpenSIPS-Users] codec_delete_except_re() has no effect

some text replaces[2] on it, then >> replace the body "manually". >> >> [1] http://www.opensips.org/html/docs/modules/2.3.x/rtpengine.ht >> ml#rtpengine.f.rtpengine_offer >> [2] http://www.opensips.org/html/docs/modules/2.3.x/textops#idp5907728 >> >&

[OpenSIPS-Users] codec_delete_except_re() has no effect

Hello, This is on OpenSIPS 2.3, downloaded from git and compiled today. An INVITE arrives over TLS with the following SDP: v=0 o=- 1492528621 1492528621 IN IP4 172.22.202.191 s=Polycom IP Phone c=IN IP4 172.22.202.191 t=0 0 m=audio 16852 RTP/SAVP 115 9 0 8 110 18 127 a=rtpmap:115 G7221/32000 a=f

Re: [OpenSIPS-Users] unixodbc connection problem - gnutls /dev/urandom error

...@lists.opensips.org] on behalf of Jeff Pyle Sent: Friday, March 31, 2017 09:29 To: users@lists.opensips.org Subject: [OpenSIPS-Users] unixodbc connection problem - gnutls /dev/urandom error Hello, I'm trying to connect OpenSIPS 2.2.3 to a MS SQL database for avpops through unixodbc +

[OpenSIPS-Users] unixodbc connection problem - gnutls /dev/urandom error

Hello, I'm trying to connect OpenSIPS 2.2.3 to a MS SQL database for avpops through unixodbc + freetds on Debian 8. I see the following OpenSIPS errors: DBG:db_unixodbc:db_unixodbc_new_connection: opening connection: unixodbc://:@localhost/SQLMaster ERROR:db_unixodbc:db_unixodbc_new_co

Re: [OpenSIPS-Users] siptrace picks incorrect source socket

ncoming IP of the request ?? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 02/08/2017 11:12 PM, Jeff Pyle wrote: > > Bogdan and team, > > This does appear to fix the source interface issue ... at least for IP

Re: [OpenSIPS-Users] siptrace picks incorrect source socket

then in sip_trace you have to declare a trace_id like: > > * modparam("siptrace", "trace_id", "[hep_id]uri=hep:heppy")* > > The docs will be updated soon. > > Regards, > > Ionut Ionita > OpenSIPS Developer > > On 02/07/2017 04:27 AM, Je

Re: [OpenSIPS-Users] siptrace picks incorrect source socket

/github.com/OpenSIPS/opensips/commit/ > b30af734cdb84991e1f906e3920a94e87c33ea04 > > If you confirm the fix, I will do the backporting to 2.2 too. > > Thanks and Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On

[OpenSIPS-Users] siptrace picks incorrect source socket

Hello, I recently enabled siptrace on an OpenSIPS 2.2.2 system acting as a registrar and a proxy. It has one IPv4 address with several ports for UDP, TCP and TLS. In a case where the INVITE is relayed from TLS to UDP, the replies to the UAC are incorrectly being reported as coming from the UDP s

[OpenSIPS-Users] IPv6 causing problems with HEP in siptrace

Hello, I have a simple dual-stack OpenSIPS 2.2.2 configuration. It includes a registrar and basic routing between locally registered users. These users have a mix of IPv4 and IPv6 contacts. All is well. I recently added Homer to the mix. The siptrace module is acting as a capture agent, dupli

Re: [OpenSIPS-Users] 2.2 crash with async, db_virtual and unixodbc

pen a feature request on the issues page[1] to support async queries for unixodbc. [1] https://github.com/OpenSIPS/opensips/issues Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com<http://www.opensips-solutions.com> On 09/07/2016 04:16 AM, Jeff Pyle wrote: Hello,

[OpenSIPS-Users] 2.2 crash with async, db_virtual and unixodbc

Hello, I'm working from the 2.2 nightly build repo on Debian Jessie, 64-bit, specifically, 2.2.1~20160830~7261cf0-1. I have a simple test script that runs a stored procedure on a Microsoft SQL 2014 server and xlogs the returned AVPs. This works fine. When I break it up into an async() fu

Re: [OpenSIPS-Users] return code from perl_exec() versus perl_exec_simple()

I think the difference is a mis. Could you please test the attached patch. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 07.07.2016 04:42, Jeff Pyle wrote: > > Hello, > > I'm using 2.1 cloned from git earli

[OpenSIPS-Users] return code from perl_exec() versus perl_exec_simple()

Hello, I'm using 2.1 cloned from git earlier today. I notice when I use perl_exec_simple(), the return code is always 1, regardless of the return code at the end of the perl function. With perl_exec(), however, the return code matches the return code in the function. Is this difference intentio

[OpenSIPS-Users] How to append header to stateless reply

Hello, How can I add a header to a stateless reply? Or, if that's not possible, to a transactional reply? I'm trying to do something like this: route { ... append_hf("P-Custom-Header: $avp(value)\r\n"); sl_send_reply("123", "Custom Response"); exit; } where P-Custom-Header appears in t

Re: [OpenSIPS-Users] Ubuntu 16.04 can no longer access trusty packages at apt.opensips.org

Thank you. - Jeff On Thu, May 12, 2016 at 2:50 AM, Nick Altmann wrote: > Hi, Jeff! > > I’ve resigned the repository. Please, try again. > > -- > Nick > > 2016-05-11 16:26 GMT+02:00 Jeff Pyle : > >> Thanks, Nick. That's great. >> >> I think I

Re: [OpenSIPS-Users] Ubuntu 16.04 can no longer access trusty packages at apt.opensips.org

wrote: > Hi, Jeff > > I’ve changed GPG key to more secure and also I’ve added xenial builds. > Please, test. > We are not watching for new releases of ubuntu, but always glad to add > them by request. > > -- > Nick > > 2016-05-01 2:23 GMT+02:00 Jeff Pyle : > >

[OpenSIPS-Users] Ubuntu 16.04 can no longer access trusty packages at apt.opensips.org

Hello, First off, thanks to everyone who makes this repo possible. Ubuntu 15.10 works just fine with the trust repo at apt.opensips.org. 16.04, however, disables support for SHA1: W: http://apt.opensips.org/dists/trusty/Release.gpg: Signature by key 33FED9119AC17EB465F51BF001839BE75F2FBB7C uses

Re: [OpenSIPS-Users] Can I use loop back when there is no network link ?

Rodrigo, I've run OpenSIPS on the loopback interface to talk to other SIP-speaking entities on the same system. I haven't had any problems. If your co-workers' project has OpenSIPS and a softphone, with no network connectivity, who are they going to call? - Jeff On Fri, Apr 29, 2016 at 2:3

Re: [OpenSIPS-Users] Processing calling-name(CNAM) from PRI

nd from Cisco > documentation, gateway seems to act that way when using H323 but, sadly, > not for SIP. I will push Cisco about this. > > Assuming the worst, Is there any thing I can try on the proxy side to get > the desired results? > > Regards, > -- > Zahid > > >

Re: [OpenSIPS-Users] Processing calling-name(CNAM) from PRI

In Adtran TA900 series gateways (very Cisco-like) I'm able to configure the PRI interface to wait for the FACILITY message before sending the initial INVITE. When the INVITE does leave the gateway towards the proxy, it has full caller name information. Perhaps something like this is available on

Re: [OpenSIPS-Users] How to view bitrate, sampling rate and frame size during a SIP call?

flx raddr >> 10.53.232.161 rport 21001 >> a=candidate:2 2 udp 16777214 162.212.130.252 57171 typ relay raddr >> 188.29.165.133 rport 49191 > > > > On 8 December 2015 at 14:21, Jeff Pyle > wrote: > >> OpenSIPS doesn't handle media so it has no knowledge

Re: [OpenSIPS-Users] Cachedb_local question

Travis, You don't really have to "tell" OpenSIPS if a client is unregistered. In the simple case, you'll do a lookup() within your script, and it will return [1] with an error code of -1 if the contact is not registered. If you want to run your logic when a client goes unregistered, rather than

Re: [OpenSIPS-Users] How to view bitrate, sampling rate and frame size during a SIP call?

OpenSIPS doesn't handle media so it has no knowledge of these things. You could glean some of this information by inspecting the offer and answer SDPs as they pass through. For example, here is an answer SDP that passed through reply_route attached to a 200 OK: v=0. o=FreeSWITCH 1449573019 14495

Re: [OpenSIPS-Users] hide to_tag from upstream provider

The to_tag is going to appear for any message beyond the initial request. Short answer, no, you can't hide the to_tag. It's very important to the SIP message. Still, if you want to change the to_tag to something different, you could use B2BUA-based topology hiding[1]. You will need a separate Op

Re: [OpenSIPS-Users] Knowing when a call was diverted

If all endpoints in your SIP network supports REFER and INVITE with Replaces correctly, all you have to do with OpenSIPS is route the SIP messages between the endpoints for transfer to work correctly. It's up to the endpoints to understand what to do with the dialogs, media sessions, etc. In my e

[OpenSIPS-Users] OpenSIPS control panel user modify PHP error

On the most recent version of the control panel, downloaded yesterday, editing a user requires an email address to pass the PHP check and save any changes. Razvan saw this at the Austin training -- just documenting it here. - Jeff ___ Users mailing lis

Re: [OpenSIPS-Users] OpenSIPS + Ekiga + Linphone: Could not send message: Forbidden

an point me to it, I'd be very grateful. Otherwise, I guess I > better off with another softphone client anyway. > > Kind regards, > Alexander > > On 02.11.2015 21:54, Jeff Pyle wrote: > >> Alexander, >> >> First, let me say I do not use Opensips for instan

Re: [OpenSIPS-Users] OpenSIPS + Ekiga + Linphone: Could not send message: Forbidden

Alexander, First, let me say I do not use Opensips for instant message or related operations. Looking at your capture, however, I may have a thought. Let's look at the headers from Ekiga's authenticated MESSAGE: U 2015/10/31 13:02:54.844064 192.168.2.109:5060 -> 127.0.0.1:5060 MESSAGE sip:5678@

Re: [OpenSIPS-Users] Flexible daemon name for logging

Perfect! Thanks, Aron. - Jeff On Thu, Oct 1, 2015 at 1:43 PM, Podrigal, Aron wrote: > Yes have a look here > http://www.opensips.org/Documentation/Script-CoreParameters-2-1#toc64 > Hello, > > I will have multiple Opensips instances running on one box. Is it > possible to indicate the daemon

[OpenSIPS-Users] Flexible daemon name for logging

Hello, I will have multiple Opensips instances running on one box. Is it possible to indicate the daemon name Opensips uses when logging? Can I specify something other than, or in addition to, the default 'opensips' ? Regards, Jeff ___ Users mailing

Re: [OpenSIPS-Users] OpenSIPS Data Base tables in a MySQL Cluster

Short answer, yes. I've used Opensips in a MySQL Cluster environment for years with much success. I did have to modify the table creation scripts to change the type from MyISAM to NDB where appropriate. - Jeff On Thu, Jul 30, 2015 at 9:15 AM, Victor Medina wrote: > Thanks Rik! > > 2015-07-3

[OpenSIPS-Users] OpenSIPS 2.x LTS ?

Hello, What is to be the first LTS version in the 2.x series? - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Manually self generate failure

Since the failure route is triggered by tm, I think you have to t_relay() the request for the failure path to be taken. Maybe you could t_relay() it to another proxy with custom headers to indicate how you wanted it to fail, or even t_relay() it to the same proxy with care to handle it as you see

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