Hi,
Don't want to sound like a captain obvious, but what I can say after a
quick look at your traces is that both client and server are behind NAT
and either some entity drops big packets silently or somebody drops ICMP
(fragmentation needed) which doesn't allow TCP to adjust max MSS. The
Hi,
But the fact that packet sent from a server to a client looks
differently captured from both points says that somebody tries to help
you (read SIP ALG). So I would join to Daniel's suggestion to try with TLS.
On 24/06/15 17:30, Andrey Utkin wrote:
Thanks for all the suggestions.
Have
Hi,
I would also add that if you see partial packets you can try to remove
any transport protocol (udp/tcp) and port filters. It will help if you
are dealing with IP fragmentation. Otherwise sniffer won't catch IP
fragments since they don't have transport level headers.
Best Regards,
What would happen in this case? If I understand correctly, the proxy
will always forward the request but the last hop (before the UAS) will
forward the request only if a connection is active?
correct.
If the last hop doesn't forward the request because no TCP connection
exists, what
*set_forward_no_connect()* called after lookup() prevents kamailio from
making a new connection to a phone if there is no active connection.
This is useful when you are trying to call to a UA which kamailio still
keeps registration record for, but TCP/TLS connect to this phone is
already gone
don't match your
criteria and in the end a call will be routed to the cheapest gateway
with proper capabilities.
*From:*sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
Behalf Of *Vitaliy Aleksandrov
*Sent:* Wednesday, April 29, 2015 10:36 AM
*To:* sr-users@lists.sip-router.org
What about configuring two LCR instances with different lcr_id.
The first one can use only gateways with requested capabilities and the
second one all gateways.
Then you can make a decision about which instance to use during call
routing process providing this lcr_id to load_gws() function.
The Tag column size is 64. If I make this larger in the database, will
it be truncated once it is loaded into memory?
According to modules source code Tag's max size is hardcoded and will be
truncated. But this is not a bit problem. You can keep capabilities list
in htable and only put a
Hi,
You can take a look at this option:
http://kamailio.org/docs/modules/4.2.x/modules/usrloc.html#usrloc.p.handle_lost_tcp
It's a bit limited and doesn't work for:
- usrloc db only mode
- if you have separated edge proxy and registrar
Thank you for information..
What can Kamailio do about
According to your description BYE was sent using the information from
R-URI which had no 5080 port.
Asterisk should have added port 5080 to the outgoing Invite contact so
that it could be used for in-dialog routing.
Can you show a full trace with sip traffic between kamailio and
asterisk. To
We use TLS for SIP. The Wireshark pcap would be encrypted. I’ll try to
get a pcap anyway. Wonder if there is a way to dump pcap from inside
kamailio.
Wireshark can decrypt SIP signalling sent over TLS connections if you
provide server's private key to it.
All the requests within dialog
You shouldn't feel any performance issues after increasing tcp_max_conn
to 4096. Connections hash table size is pretty high (1024) so it's not a
problem at all. Of course if you want to handle twice bigger number of
simultaneous clients you need to check if you current hardware can
handle it
Hi,
What is the practical limit to the number of async worker processes?
With SIP child processes, it seems to be about the number of available
CPUs in /proc/cpuinfo. After that--at least, per my testing--one
begins to hit the point of diminishing returns, presumably due to SHM
IPC and
On 22.08.14 03:26, Muhammad Shahzad wrote:
Sorry for putting this question on both dev and user mailing lists, as
it is a rather theoretical question and i hope some SIP guru on either
mail list will answer.
For non-WS endpoints which use TCP or UDP for SIP transport, each
upstream request
.
On Mon, Aug 25, 2014 at 3:24 PM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
On 22.08.14 03:26, Muhammad Shahzad wrote:
Sorry for putting this question on both dev and user mailing
lists, as it is a rather theoretical question and i
changes I did in master for
a bit more optimizations for memory manager.
Cheers,
Daniel
On 18/07/14 18:22, Vitaliy Aleksandrov wrote:
Hi list,
I have a problem with pkg.stats for tcp.main process on kamailio
4.1.2 73ea61-dirty.
kamailio has 64M of private memory for each process (-M 64
Hi list,
I have a problem with pkg.stats for tcp.main process on kamailio 4.1.2
73ea61-dirty.
kamailio has 64M of private memory for each process (-M 64).
pkg.stats for tcp.main process shows that is uses 75M and still has 62M
free:
{
entry:36
pid:10032
Hello list,
Can anybody suggest the way to remove a line/attribute from SDP that
matches some pattern (regexp), some kind of grep -v for SDP ?
I'm trying to remove unnecessary ICE candidates from the SDP body and
only thing I know is the IP address of the candidate I need to remove.
I was
I absolutely forgot about the possibility of embedding scripting
languages to configuration.
Lua did the trick.
Thanks for the right direction and example.
I can share my solution, using app_python. I'm using this in
production environment during 2 years.
loadmodule app_python.so
Hi,
What are the requirements for connecting with tls/wss.
I have not come across any information or example for this.
My config is working when the client uses ws. However if I change this to use
wss, (this is it only paramter I change) it does not work.
I understand Kamailio does not
Just add listen=tcp:_kamailio_server_ip:5060 to the kamailio.cfg if
don't have one and change sip settings of your SIP application to
disable tls.
You can have both clear tcp and tls at the same time and application
will choose which type of connection to use.
Hi Support,
I have recently
On 10/01/2013 12:54 PM, Peter Dunkley wrote:
Done.
The patch is in master and the 4.0 branch.
Thanks,
Peter
Thanks for applying the patch. It looks like websockets in master are
stable now. No memory leaks are detected for the last two days.
pks.stats still shows some strange numbers, but
Hello,
Thank you for the explanation.
Could somebody review the patch in the attachment ? I tried to fix the
problem with a growing tcpconn-refcnt for websocket connections.
On 30 September 2013 17:14, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote
I found one place where tcpconn_put() never called after tcpconn_get():
--- a/msg_translator.c
+++ b/msg_translator.c
@@ -2509,9 +2509,11 @@ char* via_builder( unsigned int *len,
} else if (con-rcv.proto==PROTO_WSS) {
memcpy(line_buf+MY_VIA_LEN-4, WSS , 4);
At first thanks for trying to help.
It's my fault that I messed up top to this story, just wanted to show
that while my system is working just fine:
1. used and real_used fields of a process (tcp receiver) is bigger
that I set in -M
2. free hasn't changed from the last restart.
root@proxy:~#
Yes, I found you commit. That's why now I'm using latest master.
tcl_list shows 200+ tcp connections and only a few of them have
ref_count bigger that 1. netstat shows the same number of established
connections.
If lost tcp_conn structures are not shown in tcp_list how can I check if
it is my
I switched to the latest master branch and it seems it works better, but
unfortunately I can't understand how much PKG memory kamailio really
uses to know it still has problems with PKG.
For instance kamcmd pkg.stats always shows that tcp_main process has
free: 32627984 (started with -M 32),
On 09/23/2013 11:23 PM, Andreas Granig wrote:
Hi,
On 09/13/2013 11:27 AM, Daniel-Constantin Mierla wrote:
thanks, patch was commited and pushed to remote repository.
The patch only handles the case where a tcp connection is directly
made to the registrar, as no event route is fired, right?
module from the master branch
with a 4.0 verion ?
Hello,
can you get the type of the process with 'kamctl ps'?
Cheers,
Daniel
On 9/20/13 6:51 PM, Vitaliy Aleksandrov wrote:
Didn't check master branch before writing previous email. There were
some commits about memory leaks in websocket
Didn't check master branch before writing previous email. There were
some commits about memory leaks in websocket module. Will try master.
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fails to apply because it
doesn't match any longer the context.
Can you resend in it raw format with 'git diff ../filename.patch'?
Cheers,
Daniel
On Fri, Sep 6, 2013 at 12:46 AM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
Make the patch for master
to add the documentation for the new usrloc parameter, in xml
files from modules/usrloc/doc/ - send over the patch and I will commit
it as well.
Cheers,
Daniel
On 8/29/13 12:47 PM, Vitaliy Aleksandrov wrote:
Thanks community for all replies.
I did the second try. Result is in the attachment
Make the patch for master branch, 4.0 doesn't take any new feature and
I applied the patches to master branch anyhow.
Also, do not forget the patch for xml documentation.
Cheers,
Daniel
Patch for master branch is in the attachment.
Changelog:
- documentation is updated
- tcpconn_id
As written many time in this mailing list it depends on what services
you want to provide using such voice servers. If they are transparent
media relays then rtpproxy or mediaproxy(-ng) can help. Also you can
integrate it with your billing to authorize calls and limit their duration.
Though
Thanks community for all replies.
I did the second try. Result is in the attachment.
This time a did it in the way proposed by Daniel-Constantin Mierla.
All job is done in timer process when it iterates through all
registrations looking for expired ones.
It works for all database modes
On 08/28/2013 06:45 PM, Juha Heinanen wrote:
Vitaliy Aleksandrov writes:
If anybody else except me need this It would be great to fix known
problems and add it to kamailio.
i don't know if this come already up, but why not use this in branch
failure route:
unregister(location, , $T_reply_rid
On 08/27/2013 10:22 AM, Olle E. Johansson wrote:
Let's work on a description on the logic needed and see if Vitally's
code is close:
1. Connection dies (tcp, tls, sctp)
2. Event-route activates
3. Check if there's a outbound flow with reg-id associated with the flow
4. If not, is there a
On 08/27/2013 11:57 AM, Daniel-Constantin Mierla wrote:
Hello,
thanks for the patch. Some remarks.
As you highlighted that the solution is not very efficient (due to no
direct mapping between records and tcp connection), it should have a
way to disable it (maybe via mod param for usrloc).
Also, if I didn't get it wrong, it works only with in-memory records,
if the usrloc is configured in db-only mode, it does not work with
current patch, right?
To remove old contacts i've used delete_ucontact from usrloc api.
This function checks db_mode and removes contact from db in case
Hello,
I've made a patch to kamailio-4.0.3 which removes stale registration
when kamailio looses an incoming tcp connection.
Of course this patch needs more work.
Since the are no direct references between user location contacts and
tcp connections callback function uses linear search
,
Sammy
On Sat, Aug 24, 2013 at 3:10 PM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
How about next config snippet:
/route[GET_NEXT_HOP]//
//{//
//$var(next_hop) = $null;//
//if (is_request()) {//
//$var
On 08/14/2013 07:32 PM, Roberto Fichera wrote:
On 08/14/2013 04:36 PM, Vitaliy Aleksandrov wrote:
If you won't be able to disable SIP ALG on your router you can fill
$avp(received) manually before calling save():
$avp(received) = sip: + $si + : + $sp + ;transport= + $proto;
In this case
On 08/01/2013 03:30 PM, Eugene Prokopiev wrote:
Hi,
Is it possible to configure two Kamailio servers with shared
registration database? Any SIP UA can be registered on any server but
it must transparently call another SIP UA which can be registered on
the same or on another Kamailio server.
Is
Hello.
I'm trying to configure kamailio as a gateway between Websocket and
TCP/TLS transports.
When I call record_route() for an initial INVITE that comes via WS and
will be forwarded via TCP to a registered UA kamailio inserts only a one
record-route header with its IP and transport=ws
suggest trying git master rather
than the 4.0 branch.
Regards,
Peter
On 31 July 2013 10:53, Vitaliy Aleksandrov vitalik.v...@gmail.com
mailto:vitalik.v...@gmail.com wrote:
Hello.
I'm trying to configure kamailio as a gateway between Websocket
and TCP/TLS transports.
When I call
Hello,
does it add two record route headers when calling to an UDP
destination? IIRC, the condition for double route is that incoming
socket is different than outgoing socket and I am not sure if Peter
updated it to detect a different sub-protocol type (as ws/wss are on
top of tcp/tls).
It works. Thanks a lot.
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On 07/31/2013 04:18 PM, Vitaliy Aleksandrov wrote:
It works. Thanks a lot.
I'm going to test this with the latest master branch as Peter Dunkley
proposed.
Should I add this issue to Jira if this behaviour is not fixed in the
master ?
___
SIP
For example you can add dns records for
customer{one,two,...}.whatever.com pointing to your proxy.
Then proxy can forward requests from outside world to asterisks based on
uri domain part (customerone - 192.168.0.5, two- .10).
You can keep table of uri-ip conversions either in the kamailio.cfg
Hello,
Is this option still not available ?
I'm asking because there was a message from, if I recall right, Olle
Johansson that removing stale registrations is on the todo list of someone
from the dev team.
So maybe someone already has this option in its devel branch ?
On Fri, Apr 5, 2013 at
Indeed, this is a very good suggestion.
I can propose a little plan for the first step:
1. Learn how sip registration works
2. Configure kamailio as a sip registrar server.
3. Try to route calls between registered phones.
4. Read about in-dialog requests routing (record_route(), loose_route())
Kamailio can do this but it doesn't by default.
http://kamailio.org/docs/modules/4.0.x/modules/registrar.html#idp110136
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loadmodule pua_usrloc.so
#!endif
And of course:
loadmodule dialog.so
Attached are the relevant sip traces.
On Wed, May 29, 2013 at 1:00 AM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
You
.
On Wed, May 29, 2013 at 1:00 AM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com
wrote:
You didn't receive any NOTIFY requests because you
didn't have a PUA client. rfc3903 describes what a
PUA client
You didn't receive any NOTIFY requests because you didn't have a PUA
client. rfc3903 describes what a PUA client is.
To make BLF work you can try to use the next modules:
1. dialog - to enable dialog awareness
2. pua + pua_dialoginfo - to enable a PUA client based on active dialogs
information
I missed that $branch(..) variable is r/w. I like this way more than
playing with AVPs.
According to documentation $branch(..) variable gives access to
additional branches.
Is it possible to change Q value for the first branch ?
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Hello.
As I understand t_load_contacts() / t_next_contacts() can help only in
case when branches have different q values.
The only way I see is to get all contacts via reg_fetch_contacts(), save
the list to an AVP and then iterate through the list in the way
described in tm module docs.
On 04/29/2013 03:52 PM, Victor V. Kustov wrote:
В Mon, 29 Apr 2013 14:46:34 +0200
Olle E. Johansson o...@edvina.net пишет:
All functions of this module load AVPs from SIP-AVP reply items
received from RADIUS upon a successful request. Value of the SIP-AVP
reply item must be a string of form:
radius_load_caller_avps() from misc_radius module can help you.
Sorry can't find an example, but as I remember it's quite simple to use
this function.
Radius can insert next-hop to its Access-Accept reply and then you can
use it to send an INVITE to right destination:
$rd = $avp(s:next-hop);
Save() function of the registrar module always saves Contact and can
save real source to the received field of a location record.
lookup() function uses Contact to form R-URI and puts ip:port from the
received field to the $du.
If $du is present t_relay() uses it to forward a request to a
On 03/25/2013 02:36 PM, Daniel-Constantin Mierla wrote:
Hello,
On 3/24/13 4:20 PM, Vitaliy Aleksandrov wrote:
On 03/21/2013 12:59 AM, Daniel-Constantin Mierla wrote:
Hello,
On 3/18/13 7:09 PM, Omar wrote:
Kamailio Team.
Was this feature added already or we still need to restart kamailio
On 03/21/2013 12:59 AM, Daniel-Constantin Mierla wrote:
Hello,
On 3/18/13 7:09 PM, Omar wrote:
Kamailio Team.
Was this feature added already or we still need to restart kamailio
after we modify the Kamailio.cfg.
it is still required to restart after changing the config file. But
some core
On 03/18/2013 01:14 PM, Victor V. Kustov wrote:
В Mon, 18 Mar 2013 11:40:24 +0100
davy van de moere davy.van.de.mo...@gmail.com write:
funny, but not usable :)
debug=9
loaded debugger.so but in syslog no sip debugging info
What exactly you want to debug ?
Wireshark produces nice sip flows
Hi, all
Have a question about failure_routes.
As I remember if kamailio statefuly sent a request and didn't receive a
reply script writer can handle this situation at a failure_route[] block.
But if kamailio can't resolve a destination from R-URI it sends back 478
reply and finishes current
On 02/04/2013 06:07 PM, Klaus Darilion wrote:
On 04.02.2013 11:26, Vitaliy Aleksandrov wrote:
The problem was caused by memlog and memdbg options.
So, how the problem is triggered and how did you solved it?
regards
Klaus
Both memlog and memdbg were removed from a kamailio config as I don't
The problem was caused by memlog and memdbg options.
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I had the same problem with CANCEL requests for transactions created by
an INVITE that was rewritten in a branch_route.
I sent a small patch to dev list in October and didn't receive any
comment. You can try to find it in sr-dev archive for October 2012.
Yes, we know it's wrong.
But if we
Hi list.
I have a strange problem with syslog.
Kamailio doesn't send any messages to syslog from UDP workers that are
bound to sock=127.0.0.1:5060.
Neither LM_DBG() from a source code nor xlog(any level, ) works.
I use kamailio-3.2.4 with this opions:
log_stderror=no
debug=3.
With
On 02/01/2013 11:38 AM, Vitaliy Aleksandrov wrote:
Hi list.
I have a strange problem with syslog.
Kamailio doesn't send any messages to syslog from UDP workers that are
bound to sock=127.0.0.1:5060.
Neither LM_DBG() from a source code nor xlog(any level, ) works.
I use kamailio-3.2.4
On 01/11/2013 05:03 PM, Fabian Borot wrote:
Hi, we need to implement TLS in our current setup and have some questions about
it:
1- We have a farm of Asterisk servers behind our kamailio proxy. The call flow is
Customers/Providers -- Proxy -- Asterisks.
Is it possible to have TLS only
You are right after reboot registrar module takes registrations from a db.
If someone send UnRegister during outage, kamailio will keep active
registration until occurence of expiration time that was calculated
during registration.
To speed up user location cleaning you can decrease the
On 11/27/2012 06:30 AM, Hemanshu Patel wrote:
Hi vitaliy,
thank you very much for your reply.
Here I am not talking about kamailio as Presence server, I am asking about BLF
handling
For SUBSCRIBE+NOTIFY+PUBLISH in case of presence, everything is working fine
In case of BFL, once the
The NOTIFY without an XML body you received just after a subscription is
correct.
rfc3265 says:
This SUBSCRIBE request will be confirmed with a final response.
200-class responses indicate that the subscription has been accepted,
and that a NOTIFY will be sent immediately.
You can
||I thought t_any_replied() can be used for that purpose.
For example in failure route I used:
if (t_any_replied() t_check_status(408) ) {
xlog(L_DBG, 408 reply received\n);
}
or
if ( !(t_any_replied()) t_check_status(408)) {
xlog(L_DBG, Local timeout\n);
}
But I carefully read tm
Hello,
Have you found the way to run SELECT COUNT(*) ?
I use kamailio 3.2.4 with sqlite 3.7.5-1 and it crashes on such queries too.
As a workaround I use $dbr(res=rows) with SELECT some_field FROM
table query, but the bug is really exists.
Hello Timo,
We've done a basic test and it seems
for checking the to URI for
a particular string (in this case the domain), changing it,
and then forwarding it?
Thanks,
Ed
On Sat, Oct 20, 2012 at 7:23 PM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote
to add the gateway in any other settings besides the one
you outlined?
Thanks,
Ed
On Fri, Oct 19, 2012 at 6:17 AM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
If I understood you right you just need
branch.
Vitaliy Aleksandrov writes:
db_mode is 0 (DB_MEMORY_ONLY) and according to comments in the
presence.h kamailio holds subscription in memory and periodically
updates to db, but retrieves from db only at startup.
And it's true, I see that kamailio tries to update the active_watchers
table
Hi all,
I'm facing problems with active subscription restoration after sipproxy
restart.
db_mode is 0 (DB_MEMORY_ONLY) and according to comments in the
presence.h kamailio holds subscription in memory and periodically
updates to db, but retrieves from db only at startup.
And it's true, I
Hi all.
Is there any easy method to find if somebody with $si:$sp is registered
(usrloc-received.ip == $si, usrloc-received.port == $sp) ?
reg_fetch_contacts uses uri as search key, but in my case I don't have it.
The only way I know is to write a module that will use
get_all_ucontacts
Hello,
On 9/6/12 6:33 PM, Vitaliy Aleksandrov wrote:
Migrating kamailio v1.5.x to v.3.0.0 says that test operator no
longer exists.
But it is mentioned in Core Cookbook 3.2 (and 3.3) and can confuse
someone.
Can somebody remove it from the documentation ?
Indeed the operator was removed
Migrating kamailio v1.5.x to v.3.0.0 says that test operator no
longer exists.
But it is mentioned in Core Cookbook 3.2 (and 3.3) and can confuse someone.
Can somebody remove it from the documentation ?
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Below is the example from http://kamailio.org/docs/modules/3.2.x/modules/tm.htm
:
failure_route[0]{
if (t_check_status(5[0-9][0-9])){
# I do not like the 5xx responses,
# so I give another chance to foobar.com,
# and I drop all the replies
Hi all.
I'm working with kamailio for about 1 year and still don't understand
difference between integer and string IDs of AVPs.
Is it just a question of usability ?
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On 08/17/2012 03:42 PM, JR Richardson wrote:
Hi All,
I'm considering running Kamailio as a virtual machine, with such low
utilization, it doesn't seem to make sense to keep running it on a
physical host server.
I've been virtualizing Asterisk PBX's for years and run a host of
other
Have you specified interfaces with listen command ?
I had a problem as you described and have fixed it by moving a listen
directive with a floating ip to the top of the list.
So you can try to specify interfaces you will use for SIP and set a
virtual ip at the top of that list.
Kamailio
Hi all,
I use $sht() pseudo-variabled provided by htable module for caching
various information about sip customers. I thought that htable has some
locking mechanism inside it and today while i was reading core-cookbook
i payed attention to one example with the lock/unlock functions and
I have lost in Route and Record-Route headers.
But now it's clear for me. Huge thanks.
On 25.06.2012 14:44, Vitaliy Aleksandrov wrote:
Hi All,
After reading default kamailio configuration i can't understand why does
kamailio remove preloaded route headers from the incoming initial INVITE
Hi All,
After reading default kamailio configuration i can't understand why does
kamailio remove preloaded route headers from the incoming initial INVITE
before calling record_route().
remove_hf(Route);
if (is_method(INVITE|SUBSCRIBE))
record_route();
What will happen if an
Hi all,
Does kamailio allow to configure more than one branch_route for an
initial INVITE ?
I have tried to add:
t_on_branch(branch_r1);
t_on_branch(branch_r2);
before t_relay(). Mentioned branch routes write debug to the
kamailio.log and it looks like that only the last
Hi all,
I'm trying to find a way to get a number of the registered user
locations and if it is bigger that some limit then kamailio will not
save new registrations.
reg_fetch_contacts() from registrar module doesn't help as it can fetch
contacts only for one user.
max_contacts parameter of
On 05/26/2012 05:39 PM, Aft nix wrote:
Hi,
I'm trying to create a ruri from contact header like following :
In request route I've saved contact header in a hashtable using htable module.
request_route{
-
-
#!ifdef WITH_HASH
if
at the functions mentioned above.
Cheers,
Vitaliy Aleksandrov
Hello,
On 5/23/12 5:22 PM, Vitaliy Aleksandrov wrote:
Hi all,
I have a question about CANCEL message processing.
My call sceraio:
When an INVITE request comes I need to rewrite a domain part of the
several headers(to, from, contact, SDP ip
contact can be saved, i think htable will be
the nice one.
Please pay attention to autoexpire parameter of htable module. You
should take care about the staled records to avoid memory usage problems.
Cheers,
Vitaliy Aleksandrov
___
SIP Express Router
Hi all,
I have a question about CANCEL message processing.
My call sceraio:
When an INVITE request comes I need to rewrite a domain part of the
several headers(to, from, contact, SDP ip) according to the outgoing
interface.
I can do it before t_relay(), but when destination user has more then
though if you can append_hf there.. Give it a try.
FYI, for using a function in event_route (seems not possible for
append_hf):
event_route[tm:local-request] {
...
append_hf(...);
...
}
Reda
On 26 avr. 2012, at 18:10, Vitaliy Aleksandrov vitalik.v...@gmail.com
mailto:vitalik.v...@gmail.com
with a
500 which is created from original request (build_res_buf_from_sip_req).
I have tried append_hf from textops module, but it's not working with a
locally generated replies.
How can i copy Reason header from 503 to 500 ?
I will appreciate your help with this situation.
Best regards,
Vitaliy
3.2) and didn't find any changes
which can help me.
Hi,
In which route did you try append_hf ?
event_route[tm:local-request] is the route executed for locally
generated requests, I think you should be trying to modify it there.
Reda
On Thu, Apr 26, 2012 at 16:10, Vitaliy Aleksandrov
Yes, i tried to call t_set_retr from route block before t_relay.
On 10/27/11 6:15 PM, Vitaliy Aleksandrov wrote:
Thank you for your quick reply. t_set_retr doesn't affects
transaction created by uac_req_send.
The second way proposed by you works great (tm settings for uac,
t_set_retr
Thank you for your quick reply. t_set_retr doesn't affects transaction
created by uac_req_send.
The second way proposed by you works great (tm settings for uac,
t_set_retr for t_relay).
Hello,
On 10/26/11 12:10 PM, Vitaliy wrote:
Hello,
How can i change T1 timeout for transaction created
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