Looking forward to the MS Teams Direct Routing SBC config ;)
On Tue, Oct 29, 2024 at 8:55 AM Bogdan-Andrei Iancu
wrote:
>
> The OpenSIPS Community Editions are complete, fully functional, easy to
> roll Open Source SIP Platforms to power a quick demo, a Proof of Concept
> or a startup production
tty(reply/tags/$ft/medias[0]/streams[0]/local
port);
$var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
address);
append_to_reply("Contact: wrote:
> Looks like if I put t_newtran(); in the main route this created the
> transaction and allowed the ACK to be r
Looks like if I put t_newtran(); in the main route this created the
transaction and allowed the ACK to be recognized. Now How do I force
Opensips to send a BYE.
Thank you.
On Fri, Nov 10, 2023 at 11:44 AM Kevin Kennedy
wrote:
>
>
>
>>>>>>>> I was able to get
>>> I was able to get audio, The problem I was having is the Originator
>>> string in the SDP. However, I am still having the same issue with
>>> accepting the ACK from the Originator and not resending the 200OK. Can
>>> someone please help with this issue?
>>>
>>> Thank
quot;call-id=$ci from-tag=$ft
file=/etc/rtpengine/unk_num.wav");
async(sleep(10), after_media);
Here is the sngrep showing that first ACK is coming in with CSEQ 1 for
example, Reinvite, 200OK and ack for CSEQ 2, then 200OK and ACK for CSEQ 1
then 200OK and ACK for CSEQ 2. until
: SIP/2.0 1...,
shmem=0x7f920581dfa8: SIP/2.0 1
Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670]
DBG:tm:_reply_light: finished
Thank you.
Kevin
On Mon, Nov 6, 2023 at 2:58 PM Kevin Kennedy wrote:
> I would like to clarify the issue in case its not 100% clear.
> * Caller s
h new CSEQ from Re-INVITE with no SDP
* 200OK Loop created
* Opensips send 200 OK with old CSEQ
* B2BUA sends ACK with old CSEQ
* Call times out.
No audio sent
Thank you
Kevin.
*
On Mon, Nov 6, 2023 at 12:54 PM Kevin Kennedy wrote:
> I tried updating from Opensips 3.2 to Opensips 3.
I tried updating from Opensips 3.2 to Opensips 3.4.2 as I saw that there
was some re-invite fixes. Still doesn't seem to resolve this issue. What
am I missing to handle this correctly?
Thank you.
Kevin
On Fri, Nov 3, 2023 at 12:51 PM Kevin Kennedy wrote:
> Dmitry,
> Thank y
make adjustments for this?
Thank you.
Kevin
On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov wrote:
> It turns out that this is no early_media, there were simply successful
> attempts with 183 Session Progress, which is why there was such a
> misunderstanding, I’ll attach the snippet c
om-tag=$ft");
#t_reply(603, "Decline");
exit();
What do I need to add to handle this scenario correctly?
Note: I was able to get this to work with Early Media (183
reply_with_body, and send t_reply(603, "Decline")), but we have customers
using late media inv
exit;
Still having problems with late media invite on this, but trying with 200
OK instead of 183 but not able to absorb the ACK coming back.
On Wed, Nov 1, 2023 at 8:02 PM Kevin Kennedy wrote:
> Devang,
> Curious if you have found a solution to this problem, as I am having the
> sa
Devang,
Curious if you have found a solution to this problem, as I am having the
same problem with Opensips 3.3 and rtpengine.
Thank you.
Kevin
On Mon, Aug 14, 2023, 3:54 AM Devang Dhandhalya via Users <
users@lists.opensips.org> wrote:
> Hello All
>
> I am facing the problem t
Anyone have some ideas?
Thank you
Kevin
On Mon, Nov 21, 2022, 12:20 PM Kevin Kennedy wrote:
> I am testing opensips 3.2 as a loadbalancer for a group of SBCs. I am
> using Mid-Registrar and Dispatcher to load balance to the SBCs. I am able
> to get Registrations and Calls working
I am testing opensips 3.2 as a loadbalancer for a group of SBCs. I am
using Mid-Registrar and Dispatcher to load balance to the SBCs. I am able
to get Registrations and Calls working, but the issue I am having is with
the Subscribe messages not receiving the NOTIFY response. Since it is set
as a
lse {
xlog("did not find $avp(str) in
$avp(contacturi)\n");
$avp(contacthash) = $(avp(contacturi));
}
ds_select_dst(1, 7, , "default", 1);
The issue I am trying to run down now is the 200 OK Contact sen
l. Is there an easier way to do this?
Thank you.
Kevin
On Tue, Nov 8, 2022 at 2:57 AM Răzvan Crainea wrote:
> Hi, Kevin!
>
> It would be simpler if you would have used the uri transformations:
> https://www.opensips.org/Documentation/Script-Tran-3-2#toc32
>
> Simply grab the
tcher:ds_select_dst: using destination [0]
DBG:dispatcher:ds_select_dst: selected [7-3/0]
I am expecting to see the hash as
*tgrp=1098765432;trunk-context=mydomain.com@192.168.1.122:5076
<http://mydomain.com@192.168.1.122:5076>!*
that way it matches no matter what number is sent in the User
Thank you. That fixed the ERROR I was seeing.
INVITEs are still not matching the hash and sending out via the Request-URI
DNS record instead of the hashed record from Dispatcher.
Thank you.
Kevin
On Wed, Nov 2, 2022 at 1:34 PM M S wrote:
> you dont need $ru = (ds_select_dst(3, 7, , "
- $ru = (ds_select_dst(3, 7, , "default", 1));
Not sure if you could help with what is causing this.
Thank you
Kevin
On Wed, Nov 2, 2022 at 12:50 PM Kevin Kennedy wrote:
> I did find setting this, works for contact
> modparam("dispatcher", "hash_pvar", "
oking at the hash and is still sending out based on
FQDN resolution resolved in the R-URI. Not sure how to resolve this to
force it to match the hash.
Thank you.
Kevin
On Wed, Nov 2, 2022 at 11:00 AM Kevin Kennedy wrote:
> I am not seeing an option, but Is it possible to hash on the Contact
looking at the
DNS record of the R-URI domain and routing on that.
$ru = (ds_select_dst(1, 1, "u", "default", 1));
How do I get it to not use the domain in the R-URI to route on.
Thank you.
Kevin
On Tue, Nov 1, 2022 at 12:51 AM Bogdan-Andrei Iancu
wrote:
> Hi,
>
>
ncer would not handle Registrations/Subscribes, etc.
[image: image.png]
Hopefully that can explain what I am trying to do better and help you
understand what I am trying to accomplish.
Thank you
Kevin
On Mon, Oct 31, 2022 at 6:07 AM Giovanni Maruzzelli
wrote:
> On Mon, Oct 31, 2022 at 1:5
identifier for
looking up the registration cache is the CONTACT header. As this will have
the TGRP and Trunk-Context in it.
On Mon, Oct 31, 2022 at 1:00 PM Kevin Kennedy wrote:
> I am pretty new to Opensips, so maybe I am not understanding what you mean
> by hashing. The only plac
swer is inside you! And it's wrong :) (famous
> quote from an Italian comedian)
>
> answered from mobile, please pardon terseness and typos,
> -giovanni
>
> On Mon, Oct 31, 2022, 20:41 Kevin Kennedy wrote:
>
>> Thank you all for your responses. Maybe I am going ab
rforming a match on the Registration destination.
Am I missing something with this?
Is there a way to pull information out of the usrloc contact kv-store. The
information that is needed is in the hop field of the json string.
Thank you.
On Fri, Oct 28, 2022 at 4:03 AM Bogdan-Andrei Iancu
wrote:
>
>
> чт, 18 авг. 2022 г. в 18:37, Kevin Kennedy :
>
>> I am looking for a configuration to be able to front end multiple SBC's
>> to load balance Registrations and keep track of them to send INVITE's to
>> the corresponding SBC.
>>
>> I am thinking
y thoughts?
Thank you.
Kevin
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shorter? Perhaps
this is some parameter length limit.
Kevin
> On Apr 29, 2022, at 8:23 AM, Sasmita Panda wrote:
>
> Hi All ,
>
>
> I was exploring fullsharing-cachedb-cluster in opensips 3.2 . I have tested
> this with single stand alone mongo db instance . Its w
AVP is for sure supported on 3.1.x and up. We use the following:
uac_replace_from($avp(caller_cnam),"”);
> On Mar 8, 2022, at 8:12 AM, Alain Bieuzent wrote:
>
> Hmm , not sure $avp is supported can you try with $var
>
> $var(ds)="abc";
> uac_replace_from($var(ds),"");
>
>
> De :
Hi Murat,
Assuming you are using 3.x.x version you would set it in opensips.cfg as
follows:
socket=udp:192.168.0.1:5080
socket=tcp:192.168.0.1:5080
Obviously you would replace the 192.168.0.1 with your IP and the 5080 with your
desired port.
Hope this helps,
Kevin
> On Jun 2, 2021, at 8
I would start with double checking that the permissions module is actually
loaded on the server. That is the most common reason to get a reload error.
Kevin
> On Jul 20, 2021, at 11:26 AM, Jeff Wilkie wrote:
>
> Opensips 3.1.2
> CP 8.3.1
> Debian 10
>
> When adding permi
and automatically re-anchors the
media, what is the timing to detect and re-anchor and can the timing be tuned?
Thanks!
Kevin V.
Original Message
From: raz...@opensips.org
Sent: June 9, 2021 5:58 a.m.
To: de...@lists.opensips.org; users@lists.opensips.org;
n
Hi Social,I'll try my luck with Google Translate if you want to share your tutorial.Thanks, Ke
like to be able to fix it
completely…but maybe that is not currently possible.
Thanks,
Kevin
> On Jan 19, 2021, at 9:31 AM, Social Boh via Users
> wrote:
>
> To switch calls from one server to another you have to use redis and
> rptengine using HA with pacemaker y corosync.
>
That seems to be how most are setup…maybe I’m making it harder than it should
be :-)
Out of curiosity what did/do you use to monitor OpenSIPS as up for your
failover or did you just rely on the IP (keepalived, etc.) reachability?
Thanks,
Kevin
> On Jan 19, 2021, at 9:08 AM, Andy Dier
-cli -x mi dlg_send_sequential
callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge body=outbound” but
that either seems to terminate the call immediately or say the dialog wasn’t
found.
Thanks,
Kevin
> On Jan 19, 2021, at 8:46 AM, Andy Dierlam wrote:
>
> With dialog writing to db
o tag, same tag, different tags
but haven’t had any luck. I’m guessing that I’m missing something to trigger
the remaining node to send re-invites. Has anyone attempted this type of setup
and have any ideas?
Thanks,
Kevin
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incoming TLS connectionsIf you need one cert per domain, maybe it implies that you need to have the domain as the CN instead of a SAN?Kevin
https://fossies.org/linux/opensips/modules/tls_mgm/README Kevin From: farm...@gmail.comSent: November 13, 2020 9:49 a.m.To: us
The call IDs don't match. Your ACK has an extra :5060 appended.
Just a thought...
Kevin V.
+15195726354
Original Message
From: santi.an...@quarea.com
Sent: April 29, 2020 4:13 a.m.
To: users@lists.opensips.org
Reply to: users@lists.opensips.org
Subject: Re: [Ope
(401) for that invite.
I was wondering is there a mechanism do identify duplicate CSeq numbers in a
session so we can drop them in opensips?
Yes I am trying to get the source of the duplication identified but it is
currently breaking authentication and call setup.
Kevin
Kevin Stewart | S
After some stracing of opensips I found that the reinvites where not making it
to opensips then tracked the issue down to fail2ban :/
solution:
aptitude remove ufw fail2ban
I am going back to hand crafted iptables rules [😊]
thanks for the help.
Kevin
}else{
xlog("L_NOTICE","got invite $sp");
xlog("L_INFO","[$mi] not from 5061\n");
}
exit;
}else{
xlog("L_ERR","got request method $rm fr
ld explain to me where i am wrong and/or maybe redirect me to a
tutorial for dummies (wasn't able to find anything i understand in
googling...maybe lake of good keyword....) ? Thank you in advance.
Kevin
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cess_920.txt?dl=0
https://www.dropbox.com/s/uim49o6zvj6zvu3/memdump_process_923.txt?dl=0
https://www.dropbox.com/s/c6tksw4j4lgk5d3/memdump_process_937.txt?dl=0
So now, I'm sure that the memdumps I've taken concern processes impacted by
the lack of pkg mem...
Thanks for your help,
Kevin
Hi Bogdan,
Got the same errors on my second opensips server (second node of the
cluster). Here is the memdump :
https://www.dropbox.com/s/d2gacgrxik3xs1t/memdump_server2_1.txt?dl=0
Thanks for your help,
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-09-09 14
things, I didn't have restarted the
service yet, so I'm able to get other memdumps.
Thanks for your help,
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-09-09 13:06 GMT+02:00 Bogdan-Andrei Iancu :
> Hi Kevin,
>
> yes, that's the righ
Hi Bogdan,
Unfortunately, no... I'll try to get that next time it'll happen ! I can do
that with "kill -SIGUSR1 OPENSIPS_PID", that's right ?
Thanks,
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-09-04 20:10 GMT+02:00 Bogdan-Andrei Ianc
So, I think it's a memory leak, no ? Now, what can I do to solve this ?
Thanks for your help,
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-08-20 17:22 GMT+02:00 Bogdan-Andrei Iancu :
> Hi Kevin,
>
> So it is more a mem leak than a crash.
ensips[25620]:
> ERROR:signaling:sig_send_reply_mod: failed to send reply with sl module
> Aug 12 11:48:34 asbc2 /usr/local/sbin/opensips[25620]:
> ERROR:options:opt_reply: failed to send 200 via send_reply
Unfortunately, we don't have the memory dump :-s ...
So, I don't know if
ere a way in this case to get some backtrace, even if it didn't
crashed ? Maybe something generated at startup ?
Thanks for your help,
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-08-19 20:34 GMT+02:00 Bogdan-Andrei Iancu :
> Hi Kevin,
>
> Try
/opensips[25620]:
> ERROR:signaling:sig_send_reply_mod: failed to send reply with sl module
> Aug 12 11:48:34 asbc2 /usr/local/sbin/opensips[25620]:
> ERROR:options:opt_reply: failed to send 200 via send_reply
Unfortunately, we don't have the memory dump :-s ...
So, I don't kno
4.7
Now we'll keep an eye on our server to check if everything is OK, and if
the memory error still occur; as we upgraded from 1.9.1 to 1.9.2...
I'll get back to you with some logs if needed ;-)
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-07
compiled on 11:15:37 Jun 20 2014 with gcc 4.7
So I think I'll have to re-compile opensips with QM_DBG_MALLOC, and try
again to export the memdump log...
I'll get back to you when done.
Thanks a lot for your help !
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur
Cacti...
I'll come back to you with logs; thanks for all !
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-06-30 11:54 GMT+02:00 Bogdan-Andrei Iancu :
> Hi Kevin,
>
> There is no need to send your email three times ;). One time is enough.
>
&g
around 700M, before
the traffic comes back.
So, opensips seems to well free the memory, isn't it ?
Thanks for your help,
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-06-27 10:38 GMT+02:00 Bogdan-Andrei Iancu :
> Hi Kevin,
>
> There is no need
Hi Bogdan,
I've set given options, and now I'm waiting for a new crash of the
service... Where the memdump will be located ? In another logfile than
opensips.log, or in the same ?
Thanks
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-06-26 18:32 GMT+02
Yes, I'll set those options, and wait 'till the trafic on the server
reduces a lot (tonight I think) to restart opensips.
Unless if you have another method to reload the config without losing calls
?
Thanks a lot,
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
or your help,
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
2014-06-26 16:46 GMT+02:00 Bogdan-Andrei Iancu :
> Hi Kevin,
>
> For debugging potential mem leaks, please look into:
> http://www.opensips.org/Documentation/TroubleShooting-OutOfMem
> If y
it's a memory-freeing
problem, it'll not solve our issue (I think).
My questions are : How can we calculate the appropriate value for S_MEMORY
and P_MEMORY ? And how can we solve our "out of pkg memory" problem ?
If you need further informations, or anything else, feel free to
ained in the opensips's memory cache.
Also, if it can handle an argument to filter (like a flag tag for example)
the results, if would be perfect :-)
Thanks,
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP
--
___
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}
> exit;
> }
> xlog("L_INFO","$ci ENUM not found\n");
> sl_send_reply("404", "Not Found");
> }
Please note : I've intentionally hidden the IP's in both scripts. And the
script may not be perfect, as it's only t
s
to be perfect, I prefer if I can avoid the perl method, and only use
Opensips's integrated methods ;-)
Thanks,
Best regards,
Kevin
Le jeudi 27 mars 2014, Bogdan-Andrei Iancu a écrit :
> Hi Kevin,
>
> Nothing is normal here, everything is strange, so. :)
>
> You c
after an ENUM query, with the informations inserted in the
location table ?
I've got an idea on how to make it with a perl script, but I prefer to not
use this method, and to use only existant modules of methods.
Thanks for your ideas
Kevin
*Bien cordialement, Best Regard
Hi Chen-Che-,
Finally it's what I've done : there's no more loadbalancing between my two
servers, but a failover mechanism based on SRV DNS entries.
Now it's OK.
However, thanks for your advice ;-)
Kevin
*Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoI
uth_hf: 'WWW-Authenticate: Digest realm="REDIRECT",
> nonce="52c57e280012d49d9ee05dd12af13f29ed28bacffb06", stale=true^M '
It seems that the disable check nonce function doesn't completely disable
the nonce checking, as there's still an inspection wha
");
> }
>
> exit;
> }
So, as you can see, I configured the auth module with "disable_nonce_check"
parameter, because of my "loadbalanced" architecture as it's said in the
documentation (
http://www.opensips.org/html/docs/modules/1.9.x
048 bytes, with SDP body of 1024 bytes, and we have to adapt our OpenSIPS
configuration...
Thanks a lot,
*
Bien cordialement,
Best Regards,
**Kevin MATHY*
*
*
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Out of curiosity, what might be the use cases where a registering client
needs/wants to know all of the others with the same binding? I'm probably
blinded by the functions of our platform and can't imagine a use for that.
Thanks!
-Original Message-
From: users-boun...@lists.opensips.o
Thanks for the reply, Ovidiu.
Unfortunately, I can't modify the client to use TCP.
Because of UDP fragmentation, it seems there should be a way to limit the size
of the Contact header in the reply. Would you agree?
Is it uncommon for the list of bindings for an AOR to be this large?
L8r
-
I'm replacing an existing proxy with OpenSIPs and I have one known problem yet
to overcome.
Several hundred clients register with the proxy so that the proxy can load
balance calls between them (based on q-value). The clients use one of a set
of different AORs based on their type/capabilities
I'm replacing an existing proxy with OpenSIPs and I have one known problem yet
to overcome.
Several hundred clients register with the proxy so that the proxy can load
balance calls between them (based on q-value). The clients use one of a set
of different AORs based on their type/capabilities
Is there a way to change the From header’s display name in a reply to a
client device?
I have a client device that requires a particular string in the From header
in the response from opensips to a REGISTER request. I’ve tried
uac_replace_from() and subst() but neither of them changed the heade
3 ==> Route to provider A only
prefix 44 ==> Route to provider B only
prefix 72 ==> Route to A and B, with loadbalancing, 50/50
Thanks,
Bien cordialement,
Best regards,
*Kevin MATHY*
*HEXANET* | Opérateur Télécom et fournisseur de services internet
Tél. +33 (0) 3 26 79 30 05
Web. h
based on R-URI domain or
username, but I'd like to do it based on FROM domain...
Have you got any idea, suggestions ?
Thanks a lot,
Regards,
Kevin
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rmally (by calling save()).
This should do the trick.
Regards,
Ovidiu Sas
--
VoIP Embedded, Inc.
http://www.voipembedded.com
On Mon, Jan 28, 2013 at 11:41 AM, Pauba, Kevin L wrote:
> Is there a way to change the From header's display name in a reply to
> a client device?
>
>
Is there a way to change the From header's display name in a reply to a client
device?
I have a client device that requires a particular string in the From header in
the response from opensips to a REGISTER request. I've tried
uac_replace_from() and subst() but neither of them changed the head
Hi list,
Is there a possibility now to download 1.9 beta version ? I tried with svn
links provided on opensips.org, but, I just can get 1.8.2, no more...
Thanks a lot,
*
Bien cordialement,
Best Regards,
**Kevin MATHY*
*HEXANET*
*
--
*
Phone : +33 (0) 3 26 79 30 05
Tech Support : +33 (0) 3 51
Hi list,
Does anyone have an idea to solve my problem ?
Thanks a lot,
*
Bien cordialement,
Best Regards,
**Kevin MATHY*
*HEXANET*
*
--
*
Phone : +33 (0) 3 26 79 30 05
Tech Support : +33 (0) 3 51 08 42 07
Web : www.hexanet.fr
Twitter : http://twitter.com/Hexanet
2012/11/27 Brett Nemeroff
ange those parameters :
modparam("dialog", "dlg_match_mode", 0) and
modparam("dialog", "dlg_match_mode", 2)
But I always have ACK loop...
Thanks a lot,
*
Bien cordialement,
Best Regards,
**Kevin MATHY*
*HEXANET*
*
--
*
Phone : +33 (0) 3 26 79 30 05
Tech Su
Then the other providers are doing something similar to the prior suggestion -
replacing $rU with $tU before relaying.
One registration = one contact URI. So, multiple registrations, rewrite the
user, or ditch the registrations and route to them by IP or host name.
--
kevin sandy, dcap, mcp
We resolved a similar issue by having Asterisk register once for each DID.
Alternatively, you could modify their dial plan to take the To header into
account, but I'd go the multiple registrations route if possible.
--
kevin sandy, dcap, mcp
On Oct 17, 2012, at 3:44 AM, Mike O'Con
through provider 1 or 2... My question is : is there any possibility to
send this information, with a SIP header, set with informations contained
in drouting tables ?
Thanks a lot,
If you have further questions, feel free to ask me !
Bien cordialement,
Best Regards,
**Kevin MATHY*
*HEXANET*
*
ed.
Further, we are sure that DNS resolution of CUSTOMER_SIP_DOMAIN returns
exactly CUSTOMER_DEVICE_IP, so, it doesn't seems to be a DNS resolution
problem...
If it can help you !
Thanks a lot,
*Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr
Pour tout
advance,
If you need further informations, feel free to ask us.
Regards,
*Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr
Pour toute demande de support, merci de contacter le *03.51.08.42.07*, ou
bien d'adresser un e-mail à *supp...@hexanet.fr*
2012/8/27 Bogda
ols, but no
one covering as much as SipSchool's one (according that everything written
in Sipschool outline is really introduced...)
So, I'll continue searching about Sipschool, and wait for someone else's
answer.
Thanks,
Regards,
*
**Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26
ning ? Does the SSCA really have a
"value" in SIP world ?
Then, if you know some other trainings, could you tell me what they are ?
Thanks a lot,
Regards,
*Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr
Pour toute demande de support, merci de contacter le
it have to support MPLS.
If you have some other questions, feel free to ask me !
Thanks a lot for your help ! Hope you will have some answer for us...
Regards,
Kevin
*Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr
Pour toute demande de support, merci
Here's one way:
$ m4 <<'EOF' > foo.cfg
> define(`myIP',1.2.3.4)dnl
>
> route {
> if (dst_ip == myIP) {
> ...
> }
> }
> EOF
$ cat foo.cfg
route {
if (dst_ip == 1.2.3.4) {
...
}
}
From: users-boun...@lists.opensips.org
[mailto:users
The GNU M4 Manual gives lots of simple examples. Reading the two (very short)
sections http://www.gnu.org/software/m4/manual/m4.html#Ifdef and
http://www.gnu.org/software/m4/manual/m4.html#Include, you can do something
like this:
define(`foo',1)
ifdef(`foo',include(`foo.m4'))
But it's really
My opinion is to stick with M4. Where's the added value in re-implementing a
small subset of M4's capability into OpenSIPs? I would rather that development
effort be directed toward enhancing OpenSIPs even more.
Leaning m4 can make the management of multiple configurations a breeze -- I
supp
Hi Friends
I'm so sorry to trouble you.
Who can tell me how to configure the msilo module to store all of
the off line messages? Or give me a configure example?
Thank you very much
Regards,Kevin
___
Users mailing list
Users@lists.opensip
I use m4 for our OpenSIPs configuration file -- it allows me to maintain one
file for 18 different OpenSIPs servers.
Works great!
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Friday, October 28
when i try and log in through the main page or using the password recovery
link i get this error
sorry -- cannot connect to xmlrpc server
111 - Connection refused
p.s. i just made my account but it has been confirmed...
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Users mailing list
Users@list
to:d...@goepp.net>> wrote:
Unfortunately the hostname is not what we are using, but rather a public IP
address which is on the other side of NAT, so the proxies are not aware of what
it is. I believe the m4 solution posted earlier will meet our need though for
now, but thanks
proxies are not aware of what
it is. I believe the m4 solution posted earlier will meet our need though for
now, but thanks for the feedback, this is a creative solution ;)
-dg
On Thu, Sep 30, 2010 at 1:00 PM, Pauba, Kevin L
mailto:klpa...@west.com>> wrote:
define(`_OPENSER_HOST',`re
now, but thanks for the feedback, this is a creative solution ;)
-dg
On Thu, Sep 30, 2010 at 1:00 PM, Pauba, Kevin L
mailto:klpa...@west.com>> wrote:
define(`_OPENSER_HOST',`regexp(esyscmd(`hostname -f'),`\<.+\>',\&)')dnl
...
alias="_OPENSER_HOST"
Take s
define(`_OPENSER_HOST',`regexp(esyscmd(`hostname -f'),`\<.+\>',\&)')dnl
...
alias="_OPENSER_HOST"
Take special note of the backquote (`).
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Brett Nemeroff
Sent: Thursday, September 30, 2010 12:22 PM
To: O
e. When I
changed the code in my custom module to use a module function built
specifically for updating the contact information, the lookup() function works
as expected.
Thanks go to Bogdan for pointing me in the right direction.
-Original Message-
From: Pauba, Kevin L
Sent: Thursd
: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Registrar's lookup() function not returning
contact with greatest q-value
Pauba,
Strange...especially that the UL SHOW also shows the wrong orderDo
you have the "desc_time_order" param set in USRLOC ?
Regards,
Bogda
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