[SR-Users] LCR from_uri help
Geoffrey Mina writes:
The LCR module is behaving in a way that is inconsistent with our old 1.5
versions. We have a customer that passes a certain caller-id. That
from_uri should be routing over a specific trunk. Here are the relevant
bits of the LCR data from kamcmd.
I would expect gw_index: 7 to match before gw_index: 6 here, but that is
not the case. Any help would be greatly appreciated. Please note, i Have
stripped out a bunch of other routes that are not in play here... hence the
reason gw_index starts at 6.
you didn't show how from uri of the call looks like. if it matches
from_uri: 2089328663, it will also match from_uri: and either rule can
get selected.
-- juha
kamcmd lcr.dump_rules
{
lcr_id: 1
rule_id: 31
prefix: +1
from_uri: 2089328663
request_uri:
stopper: 0
}
{
gw_index: 6
priority: 1
weight: 1
}
{
lcr_id: 1
rule_id: 5
prefix: +1
from_uri:
request_uri:
stopper: 0
}
{
gw_index: 7
priority: 1
weight: 1
}
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Re: [SR-Users] Planning release of v4.1.3
Hello, short reminder about upcoming release of v4.1.3. If there are patches left to be backported or issues that you are aware, report them. Cheers, Daniel On 16/04/14 12:07, Daniel-Constantin Mierla wrote: Hello, I am considering releasing v4.1.3 by mid of next week, on Wednesday or Thursday (April 23 or 24). If there are issues you are aware of and not reported to the bug tracker, add them there asap to investigate them. Also, if you noticed some fixes in the master branch not backported yet, report them to the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Planning release of v4.1.3
Hi One point from my side https://sip-router.org/tracker/index.php?do=detailstask_id=353 I did not hear anything from the developers ... KR, Oli -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Daniel-Constantin Mierla Gesendet: Dienstag, 22. April 2014 09:00 An: sr-dev; Kamailio (SER) - Users Mailing List Betreff: Re: [SR-Users] Planning release of v4.1.3 Hello, short reminder about upcoming release of v4.1.3. If there are patches left to be backported or issues that you are aware, report them. Cheers, Daniel On 16/04/14 12:07, Daniel-Constantin Mierla wrote: Hello, I am considering releasing v4.1.3 by mid of next week, on Wednesday or Thursday (April 23 or 24). If there are issues you are aware of and not reported to the bug tracker, add them there asap to investigate them. Also, if you noticed some fixes in the master branch not backported yet, report them to the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Planning release of v4.1.3
Hello, On 22/04/14 09:19, Oliver Roth wrote: Hi One point from my side https://sip-router.org/tracker/index.php?do=detailstask_id=353 I did not hear anything from the developers ... the patch related to the issue was applied by Carsten on master branch. He pointed to a apt repository he maintains, I cannot comment more about it. Change in address table version was due to version 4.1 for kamailio compared with 4.0 what you used before. Cheers, Daniel KR, Oli -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Daniel-Constantin Mierla Gesendet: Dienstag, 22. April 2014 09:00 An: sr-dev; Kamailio (SER) - Users Mailing List Betreff: Re: [SR-Users] Planning release of v4.1.3 Hello, short reminder about upcoming release of v4.1.3. If there are patches left to be backported or issues that you are aware, report them. Cheers, Daniel On 16/04/14 12:07, Daniel-Constantin Mierla wrote: Hello, I am considering releasing v4.1.3 by mid of next week, on Wednesday or Thursday (April 23 or 24). If there are issues you are aware of and not reported to the bug tracker, add them there asap to investigate them. Also, if you noticed some fixes in the master branch not backported yet, report them to the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] SIREMIS - SQL-based CDR rating engine
Hi, I'm looking for any documentation on SIREMIS especially related to the rating engine. Where can I find it ? Or at least which module(s) are related to the rating engine. Any help is greatly appreciated. Thanks. -- -Dea ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Planning release of v4.1.3
Hi, i did not cherry-pick the change for 4.1 yet, as i wanted to wait, if any of the original developers of Carrierroute have any comments on it. Obviously not, i will cherry-pick and commit for 4.1... Kind regards, Carsten 2014-04-22 9:33 GMT+02:00 Daniel-Constantin Mierla mico...@gmail.com: Hello, On 22/04/14 09:19, Oliver Roth wrote: Hi One point from my side https://sip-router.org/tracker/index.php?do=detailstask_id=353 I did not hear anything from the developers ... the patch related to the issue was applied by Carsten on master branch. He pointed to a apt repository he maintains, I cannot comment more about it. Change in address table version was due to version 4.1 for kamailio compared with 4.0 what you used before. Cheers, Daniel KR, Oli -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Daniel-Constantin Mierla Gesendet: Dienstag, 22. April 2014 09:00 An: sr-dev; Kamailio (SER) - Users Mailing List Betreff: Re: [SR-Users] Planning release of v4.1.3 Hello, short reminder about upcoming release of v4.1.3. If there are patches left to be backported or issues that you are aware, report them. Cheers, Daniel On 16/04/14 12:07, Daniel-Constantin Mierla wrote: Hello, I am considering releasing v4.1.3 by mid of next week, on Wednesday or Thursday (April 23 or 24). If there are issues you are aware of and not reported to the bug tracker, add them there asap to investigate them. Also, if you noticed some fixes in the master branch not backported yet, report them to the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:cars...@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SIREMIS - SQL-based CDR rating engine
Hello, documentation about accounting services done by siremis is at: - http://kb.asipto.com/siremis:install40x:accounting The store procedures are created by installation wizard when you check all boxes for databases creation and update. You can find them inside the sql file from ser module of siremis. Cheers, Daniel On 22/04/14 09:37, Dea Putera wrote: Hi, I'm looking for any documentation on SIREMIS especially related to the rating engine. Where can I find it ? Or at least which module(s) are related to the rating engine. Any help is greatly appreciated. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kam 4.1 Crash: HELP PLZ!!!!
Hello, wondering if you got new log messages with qm_free: freeing already freed pointer. I tried to simulate a similar situation when t_send_branch() fails, but couldn't reproduce the crash. It would be good to sort out the reason and be sure that there are no side effects, so I am trying to troubleshoot it further. Cheers, Daniel On 10/04/14 19:50, Samuel Ware wrote: Daniel, I was unable to locate them. We made the mem_safety change and put the machine back into Production. It has been running for 5 hours without crashing. The only error that seems different than before is the following Apr 10 17:36:29 tel-vc-fs03 /usr/local/sbin/kamailio[31365]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 10 17:36:29 tel-vc-fs03 /usr/local/sbin/kamailio[31365]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Apr 10 17:37:02 tel-vc-fs03 /usr/local/sbin/kamailio[31368]: : core [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f8c19c22f10), called from tm: h_table.c: free_cell(186), first free tm: h_table .c: free_cell(186) - aborting On Apr 9, 2014, at 12:34 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, the logs show that the core file was generated: Apr 3 13:14:55 tel-vc-fs03 /usr/local/sbin/kamailio[7553]: ALERT: core [main.c:778]: handle_sigs(): core was generated Unless you deleted them, then you should have them somewhere on the file system. mem_safety won't affect any kind of processing - it is just a protection for a double call of free() function. Actually, it is turned on with f_malloc memory manager which was the default in the past. Now we use q_malloc as default which is kind of f_malloc with defrag option. We should make this back on by default, but catching double free() is also good to solve anyhow. Cheers, Daniel On 09/04/14 17:15, Samuel Ware wrote: Daniel, Here is the version information. r...@tel-vc-fs03.telariscom.com kamailio -v version: kamailio 4.1.2 (x86_64/linux) 73ea61 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 73ea61 compiled on 20:41:21 Apr 1 2014 with gcc 4.4.7 We don’t see this happen in the lab environment. It only happens in production so we are little concern about it failing. Unfortunately, we don’t have any core dumps of this occurring; we didn’t have that feature enabled when this happened. By enabling “men_safety”, will this affect the processing of any of other signaling messages when an error occurs or will it only affect the transaction in which the error occurred. There isn’t much detail in the description of this flag in the documentation to layout how it affects the overall system in an otherwise “abort” situation. Your assistance is appreciated and I hope that it doesn’t sound like we are being difficult but this element is setting in front of our entire network and failure causes us to block all calls. We were running close to 800 CPS through it when it failed. We don’t see any load on the machine that would indicate that the issue isn’t software based. If the “men_safety” will not affect the rest of the traffic, we can add this as well as enabling core dumps and try it again when we can closely monitor the system. Please provide the additional details about “men_safety” so we can assess the situation. Is there any other information that I can provide to assist your input like the part of the config file or what else? Sam On Apr 9, 2014, at 2:27 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Can you get the output of kamailio -v? Also, send the backtrace for the corefiles taken with gdb - the logs show that the core were generated (check / or the value of -w parameter): gdb /usr/local/sbin/kamailio /path/to/core bt full To protect against such cases, you can use in config file: mem_safety=1 Cheers, Daniel On 08/04/14 17:34, Samuel Ware wrote: Here are the logs from the first crash Apr 3 13:13:59 tel-vc-fs03 /usr/local/sbin/kamailio[7602]: ERROR: tm [../../forward.h:199]: msg_send(): msg_send: ERROR: udp_send failed Apr 3 13:13:59 tel-vc-fs03 /usr/local/sbin/kamailio[7602]: ERROR: tm [t_fwd.c:1609]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Apr 3 13:13:59 tel-vc-fs03 /usr/local/sbin/kamailio[7602]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used:
Re: [SR-Users] LDAP and Kerberos backend for authentification / Or PAM / SASL ?
Hello, the constraint to have access to text password or HA1 format (md5 over username, password, realm) comes from WWW Authentication mechanism which is used by SIP. Writing something different in kamailio would be possible (it is open source), but you don't have phones able to do it and provide the adequate details. The only alternative now is using tls/ssl client certificates that attest who is the user/phone. Cheers, Daniel On 21/04/14 13:56, Yoann Gini wrote: Hello, My users are managed on a common LDAP / Kerberos setup. I don’t have access in any ways to the user password in clear text mode. I’ve start to follow this tutorial http://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth and check this documenation http://kamailio.org/docs/modules/4.1.x/modules/ldap.html But it seems LDAP module provided in Kamailio don’t handle this scenario, it except to have user password in clear text mode in the LDAP database (wrong in so many ways…). Do we have a way with existing Kamailio plugin to use a authentication backend who don’t provide access to clear text password? I provide different authentication « adapter » who can help any service not compatible with Kerberos to handle authentication: - ldap bind - sasl - pam Do we’ve a way to use one of this backend as an authentication service (and not just a users database). Best regards, Yoann ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio-Asterisk - route FROMASTERISK not found
Hello, if you search for FROMASTERISK, do you find it on kamailio.cfg? Check also that you have the line: #!define WITH_ASTERISK somewhere at the top of kamailio.cfg. Cheers, Daniel On 20/04/14 08:38, Patrik Kristel wrote: Hello all, I'm trying to implement Kamailio 4.1 with Asterisk 12.1.0 regarging this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb And when I try to compile kamaili.cfg, I still got this error: ERROR route FROMASTERISK not found I have loaded all modules like in tutorial. I tried to find some solve this issue, but with no result. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Issues with Kamailio Asterisk Integracion
Hello, can you get the sip trace with ngrep of such call on kamailio server port 5060? Cheers, Daniel On 16/04/14 19:50, William Baylón Huerta wrote: Hi, i see this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb, I've done step by step as described in the tutorial, but i have a problem, when A extension dial to B extension, B extension doesn't ringing... I send my sip.conf and kamailio.cfg Addittional, asterisk and kamailio are installed on same server, which have one private IP (192.168.50.217) and public IP 200.41.110.94 -- *Kevin W. Baylón Huerta* *Departamento Sistemas - Sitatel. * *Teléfono**: (51) 17073501 Anexo 208* *Celular: (51) 989715598* ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
Olli Heiskanen ohjelmistoarkkite...@gmail.com: Thanks for the help, here's what I dug up: The users are visible in Kamailio, output of kamcmd ul.dump: (here 1.1.1.1 is the public ip of my Kamailio+Asterisk server and 2.2.2.2 is the public ip of my home network) Looks like problem is not in Kamailio or SIP message flow. At least I can't spot any problems from registration dance or usrloc sample. -- Mikko Lehto ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] LCR from_uri help
OK. WE have made a change so that the priority is 0 for the route that has
the from_uri. We are still not seeing this route get matched properly.
Here is what the INVITE looks like:
INVITE sip:+19137274555@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK5b52e524;rport
From: 2089328663 sip:2089328...@blah.com;tag=as37fe796f
To: sip:+19137274555@1.2.3.4:5060
Contact: sip:2089328663@1.2.3.4
Call-ID: 09ed895a452fac516a0d11e237bd1...@blah.com
CSeq: 102 INVITE
User-Agent: Kamailio
Max-Forwards: 70
Date: Tue, 22 Apr 2014 14:42:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
P-Account-ID: 12680001
P-Charge-Info: 4019921313 sip:4019921...@blah.com
Content-Type: application/sdp
Content-Length: 236
Here is what lcr.dump_rules looks like:
kamcmd lcr.dump_rules
{
lcr_id: 1
rule_id: 31
prefix: +1
from_uri: 2089328663
request_uri:
stopper: 0
}
{
gw_index: 6
priority: 0
weight: 1
}
{
lcr_id: 1
rule_id: 5
prefix: +1
from_uri:
request_uri:
stopper: 0
}
{
gw_index: 7
priority: 1
weight: 1
}
On Tue, Apr 22, 2014 at 12:56 AM, Juha Heinanen j...@tutpro.com wrote:
Geoffrey Mina writes:
The LCR module is behaving in a way that is inconsistent with our old 1.5
versions. We have a customer that passes a certain caller-id. That
from_uri should be routing over a specific trunk. Here are the relevant
bits of the LCR data from kamcmd.
I would expect gw_index: 7 to match before gw_index: 6 here, but that is
not the case. Any help would be greatly appreciated. Please note, i
Have
stripped out a bunch of other routes that are not in play here... hence
the
reason gw_index starts at 6.
you didn't show how from uri of the call looks like. if it matches
from_uri: 2089328663, it will also match from_uri: and either rule can
get selected.
-- juha
kamcmd lcr.dump_rules
{
lcr_id: 1
rule_id: 31
prefix: +1
from_uri: 2089328663
request_uri:
stopper: 0
}
{
gw_index: 6
priority: 1
weight: 1
}
{
lcr_id: 1
rule_id: 5
prefix: +1
from_uri:
request_uri:
stopper: 0
}
{
gw_index: 7
priority: 1
weight: 1
}
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Re: [SR-Users] LCR from_uri help
Geoffrey Mina writes: OK. WE have made a change so that the priority is 0 for the route that has the from_uri. We are still not seeing this route get matched properly. i tried to say that from uri plays no role in your lcr setup, because +1 matches both rules 31 and 5. if you want it match only rule 31, you have to replace empty from pattern in rule 5 with one that does NOT match to 2089328663 or define another lcr instance for this caller. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Translate calling/called party based on username to telephone number base id.
Hi all.
I want to thank you all for the great info and help that you provide in this
mailing list. I am just starting to work with kamailio and still learning.
I would like some help to point me on the right direction on our new
challenge:
We have a kamailio server acting basically as location server and call
proxy. We are connected to a PSTN gw to reach the outside world. It is
working fine but now the client requested some changes that may imply we
work with username base subscribers (user1. Characters based) at kamailio
side but we still need to convert these usernames to numbers when sending
the call to the PSTN gw. Username-number should be fix and can be stored in
a table on the db.
We would need to lookup userpart of all the header (from, to, contact.) and
substitute incoming username with the peer number for calls from the app to
pstn. And translate numbers to usernames on the calls from pstn to the apps.
I am reading uac module and the sqlops module to search in the db before
the substitution but I am not sure if that is the best approach to the
problem.
Could anyone point me in the right direction or suggest a better way to do
that? Have anyone already implemented this type of modifications? I am a bit
worried for the dialog consistency although I've read the uac module should
take care of it
Thanks in advanced
Helena
From: Helena Garcia-Nieto [mailto:helena.gni...@morodo.co.uk]
Sent: martes, 10 de diciembre de 2013 16:19
To: 'Kamailio (SER) - Users Mailing List'
Cc: helena.gni...@morodo.co.uk
Subject: 500 I'm terribly sorry error
Hello,
Thanks in advanced for the help. I am almost new with kamailio and still
struggling through silly problems so please forgive me if the solution is so
obvious.
I have a network like
Xlitle -- Kamailio -- GW
The GW is more or less out of my reach for changing the behaivour.
As devices I have xlitle
Kamailio is on version 4.0.2
I've changed only few things from the default config file. Add mysql
support, auth, userlocdb, pstngw.
For this part, gw routing , I've defined gw ip and port inside the PSTN
definition like:
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = 10.0.0.101 desc My PSTN GW Address
#
# - by default is empty to avoid misrouting
pstn.gw_ip = desc PSTN GW Address
pstn.gw_port = desc PSTN GW Port
iskratel.gw_ip = 10.XX.XX.XX
iskratel.gw_port = 5060
#!endif
I route the calls with:
route(ISKRATEL);
And defined a routing function
route[ISKRATEL] {
#!ifdef WITH_PSTN
# check if ISKRATEL GW IP is defined
if (strempty($sel(cfg_get.iskratel.gw_ip))) {
xlog(SCRIPT: PSTN rotuing enabled but iskratel.gw_ip not
defined\n);
return;
}
# only local users allowed to call
if(from_uri!=myself) {
sl_send_reply(403, Not Allowed);
exit;
}
if (strempty($sel(cfg_get.iskratel.gw_port))) {
$ru = sip: + $rU + @ + $sel(cfg_get.iskratel.gw_ip);
} else {
$ru = sip: + $rU + @ + $sel(cfg_get.iskratel.gw_ip) +
:
+ $sel(cfg_get.iskratel.gw_port);
}
# Add profix to ISKRATEL: A99901
subst_uri('/^sip:(.*)/sip:A99901\1/i'); # add A99901
route(RELAY);
exit;
#!endif
return;
}
I can see that the code goes all the way to the route(relay) but all I can
see is the 500 I'm terribly sorry, server error occurred (7/SL) and a second
response 500 I'm terribly sorry, server error occurred (7/TM)
I've tried defining the IP as the given pstn.gw_ip and route(pstn) without
changing anything on the pst default routing but the response of the server
is the same.
I cannot see any special error on the logs.
I have the exact same config for this part, in another test server and the
calls go to the gw without reporting any error.
I'll appreciate any help from you! Thanks in advanced for your time
Helena
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[SR-Users] Drouting
Hi, I am trying to use the drouting module and was wondering if there is a wildcard character for the dr_groups table for the user? My situation is that I want to look at any user from a certain IP range (i.e. the carrier). Any ideas? Thanks Keith ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] LCR from_uri help
OK, has this behavior changed since Kamailio 1.5 because this previously worked. Right now, I would think that with our Priority of 0 on the route with the from_URI match, the algorithm should always look at that one first and route the calls accordingly. Since we have the priority correctly set to evaluate in the order we want, what next can we look to? On Tue, Apr 22, 2014 at 9:27 AM, Juha Heinanen j...@tutpro.com wrote: Geoffrey Mina writes: OK. WE have made a change so that the priority is 0 for the route that has the from_uri. We are still not seeing this route get matched properly. i tried to say that from uri plays no role in your lcr setup, because +1 matches both rules 31 and 5. if you want it match only rule 31, you have to replace empty from pattern in rule 5 with one that does NOT match to 2089328663 or define another lcr instance for this caller. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] DRouting, routeid is not triggered
Hello,
I am working on version: kamailio 4.1.2 (x86_64/linux) and heaving troubles
with drouting module.
The problem i am facing is that kamailio cannot enter routeid that is
provided inside dr_rule table.
My last try was to use standard script file with placing following data
before t_relay
if (!do_routing(10)) {
sl_send_reply(403, No route for You);
exit;
}
additionaly i have added route definition
route[1] {
xlog(L_INFO,[INFO] Default route --- fn-$fn, fu-$fu, fU-$fU,
ru-$ru, rU-$rU, sp-$sp, si-$si, tu-$tu, tU-$tU);
uac_replace_from(sip:$fU@10.10.10.5);
}
My DB contains following data:
++-++-+--+-++--+
| ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
description |
++-++-+--+-++--+
| 1 | 10 | 43 | |0 | 1 | 1 | test
rule |
++-++-+--+-++--+
+--+--+---+---++---+-+
| gwid | type | address | strip | pri_prefix | attrs | description |
+--+--+---+---++---+-+
|1 | 10 | 10.10.10.9 | 0 | NULL | NULL | FirstGW|
+--+--+---+---++---+-+
The result is that script is omitting route[1] without any purpose (xlog is
not shown, $fU is not modified).
Could somebody help me to understand whats going on?
Thanks in advanced,
Mac
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Re: [SR-Users] Drouting
Hello, On 22/04/14 18:26, Keith wrote: Hi, I am trying to use the drouting module and was wondering if there is a wildcard character for the dr_groups table for the user? My situation is that I want to look at any user from a certain IP range (i.e. the carrier). Any ideas? Do you want to have a list of users routed by a specific group id without inserting all associated usernames? Maybe you can do the mapping of the username drouting group id with another module, like mtree where you can match on prefix or htable, matching on source ip. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] DRouting, routeid is not triggered
Hello,
can you set debug=3 in kamailio.cfg and then send the syslog messages
for routing a call matching the drouting rule?
Cheers,
Daniel
On 22/04/14 19:12, Maciej Bylica wrote:
Hello,
I am working on version: kamailio 4.1.2 (x86_64/linux) and heaving
troubles with drouting module.
The problem i am facing is that kamailio cannot enter routeid that is
provided inside dr_rule table.
My last try was to use standard script file with placing following
data before t_relay
if (!do_routing(10)) {
sl_send_reply(403, No route for You);
exit;
}
additionaly i have added route definition
route[1] {
xlog(L_INFO,[INFO] Default route --- fn-$fn, fu-$fu,
fU-$fU, ru-$ru, rU-$rU, sp-$sp, si-$si, tu-$tu, tU-$tU);
uac_replace_from(sip:$fU@10.10.10.5 mailto:fU@10.10.10.5);
}
My DB contains following data:
++-++-+--+-++--+
| ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
description |
++-++-+--+-++--+
| 1 | 10 | 43 | |0 | 1 | 1 | test
rule |
++-++-+--+-++--+
+--+--+---+---++---+-+
| gwid | type | address | strip | pri_prefix | attrs | description |
+--+--+---+---++---+-+
|1 | 10 | 10.10.10.9 | 0 | NULL | NULL | FirstGW|
+--+--+---+---++---+-+
The result is that script is omitting route[1] without any purpose
(xlog is not shown, $fU is not modified).
Could somebody help me to understand whats going on?
Thanks in advanced,
Mac
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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?
Don't forget to include peer type (friend), and The callbacknumber In The
table.
It happened to me and asterisk/kamailio behavior was wayyy to weird until
made sure both parameters were there.
-
In this setup I have SIP peers in an asterisk table added like this:
INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
');
--
El abr 19, 2014 1:17 PM, Olli Heiskanen ohjelmistoarkkite...@gmail.com
escribió:
Hello,
One of the tests I've been working with is Asterisk realtime integration
according to Daniel's guide here:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
Weird thing is the client looks registered but I'm not sure if it really
is registered. If I'm not mistaken I should see the peers when I issue 'sip
show peers' on asterisk cli. Instead I get this:
*CLI sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port
Status Description Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
offline]
Also, calling between clients will fail; in Asterisk cli I get:
*CLI
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [661@default:1] NoOp(SIP/660-, Testing:
Dialed 661) in new stack
-- Executing [661@default:2] Dial(SIP/660-,
SIP/661,3600,rt) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/661
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [661@default:3] Hangup(SIP/660-, ) in new
stack
== Spawn extension (default, 661, 3) exited non-zero on
'SIP/660-'
In this setup I have SIP peers in an asterisk table added like this:
INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
testers.com');
I have Kamailio and Asterisk on the same machine where Kamailio listens
port 5060 and Asterisk listens 5070. Things that differ from the guide are
Kamailio and Asterisk versions, which in my case are newer. Also, for
another testing case I have MULTIDOMAIN enabled in Kamailio, does this
interfere with the realtime integration? I'm using only one domain though.
Please let me know if any configs or traces I can provide will help figure
out what's going on.
cheers,
Olli
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