[OpenSIPS-Users] Getting error using Dispatcher module
Hi,
I'm getting error using Dispatcher module as listing below;
Dec 11 07:22:57 newtest /usr/local/sbin/opensips[15487]:
ERROR:core:parse_uri: bad uri, state 0 parsed: 77.6 (4) /
77.66.x.x:5060 (16)
Dec 11 07:22:57 newtest /usr/local/sbin/opensips[15487]:
ERROR:dispatcher:add_dest2list: bad uri [77.66.x.x:5060]
Dec 11 07:22:57 newtest /usr/local/sbin/opensips[15487]:
ERROR:dispatcher:mod_init: could not initiate a connect to the database
Dec 11 07:22:57 newtest /usr/local/sbin/opensips[15487]:
ERROR:core:init_mod: failed to initialize module dispatcher
Dec 11 07:22:57 newtest /usr/local/sbin/opensips[15487]: ERROR:core:main:
error while initializing modules
Even when I configured OpenSIPs version 1.5.2, dispatcher module was easily
configured but now I using same configuration and applied on version 1.6
getting the error as I listed above.
Using command 'opensipsctl dispatcher show' its showing me the fields of
table dispatcher as listing below;
++---+--+---++---+--+
| id | setid | destination | flags | weight | attrs | description |
++---+--+---++---+--+
| 1 | 1 | 77.66.x.x:5060 | 0 | 1 | | |
| 2 | 1 | 77.66.x.x:5060 | 0 | 1 | | |
++---+--+---++---+--+
The settings I done in opensips.cfg file is listed below;
loadmodule dispatcher.so
modparam(dispatcher, db_url,mysql://opensips:opensip...@localhost
/opensips)
if (is_method(INVITE)) {
ds_select_dst(1, 4);
forward();
route(1);
setflag(1); # do accounting
}
Further added I can even login to mysql using opensips credentials as well.
Kindly advise me how to resolve this issue.
--
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Ahmed Munir
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[OpenSIPS-Users] How to send Registration Request to Asterisk via OpenSips
Hi,
Currently I'm working on case i.e. OpenSips and Asterisk, where I'm using
OpenSIps as a Proxy server using dispatcher module and dispatcher list
contains Asterisk machines IP as destination address.
The configuration I've done in opensIps.cfg is listed down below;
if (is_method(INVITE)) {
ds_select_dst(1, 4);
forward();
route(1);
setflag(1); # do accounting
}
My UAC IP is xx.xx.xx.xx, OpenSips IP: yy.yy.yy.yy and Asterisk IP:
zz.zz.zz.zz.
When I make a call I'm getting code error 603, Decline. Even though the
settings I've set on UAC as outbound proxy using valid credentials as used
on my Asterisk machine.
Kindly advise me how can I send Registration request OpenSips - Asterisk.
Please give me some sample to resolve this issue. At the end I'm listing few
traces;
U zz.zz.zz.zz:5060 - yy.yy.yy.yy:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-;received=yy.yy.yy.yy.
Via: SIP/2.0/UDP 192.168.0.168:5060
;received=203.215.176.22;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183.
From: 322025sip:322...@yy.yy.yy.yy.;tag=9d7c6756.
To: 322025sip:322...@yy.yy.yy.yy.;tag=as0f0e0e90.
Call-ID: b115ce088a57d010.
CSeq: 8160 REGISTER.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
WWW-Authenticate: Digest algorithm=MD5, realm=rtsip.vopium.com,
nonce=0a26e4a7, stale=true.
Content-Length: 0.
.
U yy.yy.yy.yy.:5060 - xx.xx.xx.xx:46183
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.0.168:5060
;received=xx.xx.xx.xx;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183.
From: 322025sip:322...@yy.yy.yy.yy.;tag=9d7c6756.
To: 322025sip:322...@yy.yy.yy.yy.;tag=as0f0e0e90.
Call-ID: b115ce088a57d010.
CSeq: 8160 REGISTER.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
WWW-Authenticate: Digest algorithm=MD5, realm=rtsip.vopium.com,
nonce=0a26e4a7, stale=true.
Content-Length: 0.
U zz.zz.zz.zz:5060 - yy.yy.yy.yy:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-;received=77.66.2.137.
Via: SIP/2.0/UDP 77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-.
Via: SIP/2.0/UDP 192.168.0.168:5060
;received=203.215.176.22;received=203.215.176.22;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183;rport=46183.
From: 322025sip:322...@yy.yy.yy.yy;tag=9d7c6756.
To: 322025sip:322...@yy.yy.yy.yy.
Call-ID: b115ce088a57d010.
CSeq: 8160 REGISTER.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: sip:322...@zz.zz.zz.zz.
Content-Length: 0.
.
U zz.zz.zz.zz:5060 - yy.yy.yy.yy:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-;received=77.66.2.137.
Via: SIP/2.0/UDP 77.66.2.137;branch=z9hG4bK-d87543-928337242-1--d87543-.
Via: SIP/2.0/UDP 192.168.0.168:5060
;received=203.215.176.22;received=203.215.176.22;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183;rport=46183.
From: 322025sip:322...@yy.yy.yy.yy;tag=9d7c6756.
To: 322025sip:322...@yy.yy.yy.yy;tag=as0f0e0e90.
Call-ID: b115ce088a57d010.
CSeq: 8160 REGISTER.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
WWW-Authenticate: Digest algorithm=MD5, realm=rtsip.vopium.com,
nonce=2c6529c9, stale=true.
Content-Length: 0.
.
U yy.yy.yy.yy:5060 - xx.xx.xx.xx:46183
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.0.168:5060
;received=203.215.176.22;received=203.215.176.22;branch=z9hG4bK-d87543-928337242-1--d87543-;rport=46183;rport=46183.
From: 322025sip:322...@yy.yy.yy.yy;tag=9d7c6756.
To: 322025sip:322...@yy.yy.yy.yy.
Call-ID: b115ce088a57d010.
CSeq: 8160 REGISTER.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: sip:322...@zz.zz.zz.zz.zz.
Content-Length: 0.
.
--
Regards,
Ahmed Munir
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[OpenSIPS-Users] How to send SIP header 302 registration request to Asterisk
Hi, I'm using OpenSIPs version 1.6, the module I'm using is dispatcher using mysql. My question is how can I send SIP header 302 registration request to Asterisk? Because Asterisk is sending me unAuthorized message to OpenSIPs. Even the credentials I'm using for Asterisk is the same as I'm using for OpenSIPs. Kindly advise me to resolve this issue. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to send SIP header 302 registration request to Asterisk
Hi, Thanks for your reply, I'm attaching the file named trace. Where OpenSIPs IP is: yy.yy.yy.yy, Asterisk IP: zz.zz.zz.zz and my UAC IP: xx.xx.xx.xx. As you can see I'm getting unauthourized error from Asterisk side, even using these credentials I can get registered on my Asterisk machine Previously I forgot to ask, how can I set SIP header 302 on registeration section in OpenSIPs? Kindly advise me. Date: Thu, 17 Dec 2009 22:40:15 -0800 From: Jai Rangi jpra...@gmail.com Subject: Re: [OpenSIPS-Users] How to send SIP header 302 registration request to Asterisk To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: eb007ec0912172240hc779cbflbf3460b835f1f...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Getting some ngrep traces will help some other to help you. Unauthorized message is for useraccount or for opensip. -Jai On Thu, Dec 17, 2009 at 10:25 PM, Ahmed Munir ahmedmunir...@gmail.com wrote: Hi, I'm using OpenSIPs version 1.6, the module I'm using is dispatcher using mysql. My question is how can I send SIP header 302 registration request to Asterisk? Because Asterisk is sending me unAuthorized message to OpenSIPs. Even the credentials I'm using for Asterisk is the same as I'm using for OpenSIPs. Kindly advise me to resolve this issue. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- next part -- An HTML attachment was scrubbed... URL: http://lists.opensips.org/pipermail/users/attachments/20091217/2c123ff7/attachment-0001.htm -- -- Regards, Ahmed Munir trace Description: Binary data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Bypass UAC IP
Hi,
I want to know how can I bypass UAC IP through OpenSIPS. Like my UAC IP is
xx.xx.xx.xx, my OpenSIPS IP is yy.yy.yy.yy and Asterisk IP is zz.zz.zz.zz,
where I'm using OpenSIPs as redirect server, when I make a call I want UAC
IP displayed on Asterisk machine not OpenSIPs IP.
Kindly advise me which fuction do I require for it. The configuration I've
done is listed down below;
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if (is_method(INVITE)) {
log(INVITE###);
ds_select_domain(1,4);
sl_send_reply(300,Redirect);
route(1);
log(#END);
exit;
}
if (is_method(REGISTER)) {
log(REGISTER###);
ds_select_dst(1,4);
#t_replicate(77.66.2.136);
#sl_send_reply(200, ok);
forward();
log(#END);
#exit;
return;
}
}
--
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Ahmed Munir
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[OpenSIPS-Users] Sharing Database
Hi, I'm Currently running Asterisk in real time environment. Is it possible that OpenSIPs can read/share the same database that Asterisk is using? Like I want OpenSips to share sip_buddies table from database which is used by Asterisk machine as well. Kindly advise me which parameters do I require for doing it. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Need help to forward Access number
Hi, I want to forward an Access Number from OpenSIPS to Asterisk machine. Kindly advise how can i do that? Which modules/functions are use to forward by INVITE section? -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Users Digest, Vol 18, Issue 2
--d87543-;received=xx.xx.2.137. Via: SIP/2.0/UDP yy.yy.176.22:9782;received=yy.yy.176.22;branch=z9hG4bK-d87543-934705935-1--d87543-;rport=9782. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. To: sip:3214426...@xx.xx.2.137;tag=as70d2441c. Call-ID: b95db141291c3838. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Length: 0. . U xx.xx.2.137:5060 - yy.yy.176.22:9782 SIP/2.0 491 Request Pending. Via: SIP/2.0/UDP yy.yy.176.22:9782;received=yy.yy.176.22;branch=z9hG4bK-d87543-934705935-1--d87543-;rport=9782. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. To: sip:3214426...@xx.xx.2.137;tag=as70d2441c. Call-ID: b95db141291c3838. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Length: 0. . U yy.yy.179.54:5060 - xx.xx.2.137:5060 SIP/2.0 503 Server error. Via: SIP/2.0/UDP xx.xx.2.137;branch=z9hG4bK2d48.760e5071.0;received=xx.xx.2.137. Via: SIP/2.0/UDP yy.yy.176.22:9782;received=yy.yy.176.22;branch=z9hG4bK-d87543-388987748-1--d87543-;rport=9782. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. To: sip:3214426...@xx.xx.2.137;tag=as70d2441c. Call-ID: b95db141291c3838. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: sip:3214426...@yy.yy.179.54. Content-Length: 0. . U xx.xx.2.137:5060 - yy.yy.179.54:5060 ACK sip:3214426...@yy.yy.179.54:5060 SIP/2.0. Via: SIP/2.0/UDP xx.xx.2.137;branch=z9hG4bK2d48.760e5071.0. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. Call-ID: b95db141291c3838. To: sip:3214426...@xx.xx.2.137;tag=as70d2441c. CSeq: 1 ACK. Max-Forwards: 70. User-Agent: OpenSIPS (1.6.0-notls (i386/linux)). Content-Length: 0. . U xx.xx.2.137:5060 - yy.yy.176.22:9782 SIP/2.0 500 Server error. Via: SIP/2.0/UDP yy.yy.176.22:9782;received=yy.yy.176.22;branch=z9hG4bK-d87543-388987748-1--d87543-;rport=9782. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. To: sip:3214426...@xx.xx.2.137;tag=as70d2441c. Call-ID: b95db141291c3838. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: sip:3214426...@yy.yy.179.54. Content-Length: 0. . U yy.yy.176.22:9782 - xx.xx.2.137:5060 ACK sip:3214426...@xx.xx.2.137 SIP/2.0. To: sip:3214426...@xx.xx.2.137;tag=as70d2441c. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. Via: SIP/2.0/UDP yy.yy.176.22:9782;branch=z9hG4bK-d87543-934705935-1--d87543-;rport. Call-ID: b95db141291c3838. CSeq: 2 ACK. Content-Length: 0. . Kindly help me to resolve this problem. Date: Fri, 1 Jan 2010 19:15:00 +0530 From: ram talk2...@gmail.com Subject: Re: [OpenSIPS-Users] Need help for Call to another network To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: b74751491001010545y73756771jc1ef52fb4d205...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 check any Firewall involved or use ngrep to track the packet Ram On Fri, Jan 1, 2010 at 6:17 PM, Ahmed Munir ahmedmunir...@gmail.com wrote: Hi, I've configured OpenSIPs + Asterisk machines, where OpenSIPs forwards calls to Asterisk using dispatcher module. Call is made when OpenSIPs and Asterisk are configured on same network i.e. xx.xx.2.137 IP of OpenSIPs and xx.xx.2.136 IP of Asterisk. But when I mention different network IP of Asterisk i.e. yy.yy.179.54 to OpenSIPs, I'm getting error of Request Pending. Even same configuration is set on yy.yy.179.137 Asterisk machine A as on xx.xx.2.136 Asterisk machine B. Kindly advise me how can I resolve this issue/which part I need to configure. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- next part -- An HTML attachment was scrubbed... URL: http://lists.opensips.org/pipermail/users/attachments/20100101/622a1c74/attachment-0001.htm -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 18, Issue 2 -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to implement access control list on opensips
Hi, I want to implement ACL on OpenSIPs to accept the call on behalf of source URI + IP address. Can anyone tell me which modules and functions are required for it? Also kindly share some example template with it. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to implement access control list on opensips
Hi, I want to implement ACL on OpenSIPs to accept the call on behalf of source URI + IP address. Can anyone tell me which modules and functions are required for it also which tables will involve in it? Also kindly share some example template with it. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to increase cache in opensips tables
Hi,
I'm using permission module's function check_source_address(), the problem
I'm facing is that I can not add not than more 8 IPs in address table, but I
want to permit more than 100 IPs. I only want to use these IPs on group 0
what I am using. When I enter more than 8 IPs in address table and make a
call, I observe a message i.e. not found in hash table.My opensips.cfg
configuration for check_source_address() is listed below;
if (is_method(INVITE) check_source_address(0)) {
log(INVITE###);
ds_select_domain(1,4);
forward();
route(1);
log(#END);
setflag(1);
}
Kindly advise me how to increase cache of OpenSIPs database tables so I can
reslove my case. Further added, how can I enter domain name in 'ip' column
section of address table i.e. abc.com can't be used and gives me an error,
kindly advise this well.
--
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Ahmed Munir
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[OpenSIPS-Users] One Way Audio
Hi Irina,
Thanks for reply. After looking in forums I observed that on opensips
version 1.6 has a bug and its bug fix is uploaded on svn. I recompile svn
version 1.6 and test it and working ok.
But now I'm facing weird problem, while using non-svn version 1.6 I was able
to call to my asterisk boxes and media was passing on both ways. But when I
recompile svn version 1.6 and make a call there is only one way voice from
eyebeam to twinkle i.e.
eyebeam - opensips -- asterisk -- twinkle
twinkle can hear from eyebeam side
---
eyebeam can't hear from twinkle side
Opensips and Asterisk both hosted on Public IPs and UAC are located at
private network. Firewall is permitted on both servers and I'm using stun
for my UAC.
Kindly advise to sort this problem? But I don't understand why was media is
passing both ways when using non-svn version?
Further added, I am also using module dispatcher.
Date: Wed, 06 Jan 2010 16:36:44 +0200
From: Irina Stanescu istane...@opensips.org
Subject: Re: [OpenSIPS-Users] How to increase cache in opensips tables
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID: 4b449ffc.5050...@opensips.org
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello Ahmed,
Firstly, I need to see the log so I could understand better the error
you get. I don't think the problem is that the cache is too small.
Also, you cannot use 0 for the group id, the documentation says:
group_id
This argument represents the group id to be matched. It can be an
integer string or a string pvar. If the group_id argument is 0, the
query can match any group in the cached address table
Secondly, as the name suggests, the ip column is reserved for IPs only.
You cannot add domain name addresses to this column.
Regards,
Irina Stanescu
Ahmed Munir wrote:
Hi,
I'm using permission module's function check_source_address(), the
problem I'm facing is that I can not add not than more 8 IPs in
address table, but I want to permit more than 100 IPs. I only want to
use these IPs on group 0 what I am using. When I enter more than 8 IPs
in address table and make a call, I observe a message i.e. not found
in hash table.My opensips.cfg configuration for check_source_address()
is listed below;
if (is_method(INVITE) check_source_address(0)) {
log(INVITE###);
ds_select_domain(1,4);
forward();
route(1);
log(#END);
setflag(1);
}
Kindly advise me how to increase cache of OpenSIPs database tables so
I can reslove my case. Further added, how can I enter domain name in
'ip' column section of address table i.e. abc.com http://abc.com
can't be used and gives me an error, kindly advise this well.
--
Regards,
Ahmed Munir
--
Regards,
Ahmed Munir
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[OpenSIPS-Users] Getting Error When Configuring OpenSIPS + FreeRadius`
Hi, I've configured the OpenSIPS version 1.6 (svn version) with FreeRadius (latest now a days.) and clientradius_ng (latest now a days). The connectivity with radius server and mysql was successful, as I follow the steps as mentioned in this link; http://voiprookie.blogspot.com/2009/04/freeradius-and-mysql.html and book 'building telephony system with openser', (with minor changes like modules naming convention in opensips v 1.6), Opensips services starts and stops successfully. A partial opensips.cfg is listed below; loadmodule acc.so loadmodule aaa_radius.so #Settings For Radius- #modparam(auth_diameter, diameter_client_host, localhost) modparam(aaa_radius, radius_config,/usr/etc/radiusclient-ng/radiusclient.conf) modparam(acc, aaa_url, radius:/usr/etc/radiusclient-ng/radiusclient.conf) modparam(acc, aaa_flag, 2) modparam(acc, aaa_missed_flag, 3) modparam(acc, aaa_extra,User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)) The problem I'm facing is when I register my phones (which they registered successfully) and make a successful call between them, but when I check log messages I'm getting only these errors as listed below; Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]: ERROR:aaa_radius:rad_avp_add: failure Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]: ERROR:acc:acc_aaa_request: failed to add Source-IP, 13 And I also check table radacct in mysql database, no records are inserted into it. Kindly advise this issue. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Users Digest, Vol 19, Issue 80
Hi, Thanks for your reply, I've copied dictionary.opensips to /etc/freeradius directory and added few lines in it after that when I make the call I'm getting this same error as mentioned in my previous mail earlier, I even copied the dictionary.freeradius to /etc/clientradius_ng directory as well. Lines for dictionary.opensips are listed below; Attributes ### ATTRIBUTE Sip-Uri-User 208 string # Proprietary, auth_radius ATTRIBUTE Sip-Group211 string # Proprietary, group_radius ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius ATTRIBUTE SIP-AVP 225 string # Proprietary, avp_radius ATTRIBUTE Sip-Method 101 integer ### Service-Type Values ### VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 VALUE Service-Type SIP-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type SIP-Callee-AVPs 31 # Proprietary, avp_radius ### Sip-Method Values ### VALUE Sip-Method Undefined 0 VALUE Sip-Method Invite 1 VALUE Sip-Method Cancel 2 VALUE Sip-Method Ack4 VALUE Sip-Method Bye8 VALUE Sip-Method Info 16 VALUE Sip-Method Options32 VALUE Sip-Method Update 64 VALUE Sip-Method Register 128 VALUE Sip-Method Message256 VALUE Sip-Method Subscribe 512 VALUE Sip-Method Notify 1024 VALUE Sip-Method Prack 2048 VALUE Sip-Method Refer 4096 VALUE Sip-Method Other 8192 And my dictionary settings are listed below; dictionary(freeradius) $INCLUDE/opt/freeradius/share/freeradius/dictionary $INCLUDE/opt/freeradius/etc/raddb/dictionary.opensips dictionary(radiusclient) ##At the end of the line $INCLUDE /opt/freeradius/etc/raddb/dictionary.opensips Kindly tell me from which section I'm facing this problem. Waiting for your reply. Date: Fri, 26 Feb 2010 15:21:03 +0200 From: Andrew Pogrebennyk andrew.pogreben...@portaone.com Subject: Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS + FreeRadius` To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: 4b87cabf.9000...@portaone.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 26.02.2010 14:33, Ahmed Munir wrote: Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]: ERROR:aaa_radius:rad_avp_add: failure Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]: ERROR:acc:acc_aaa_request: failed to add Source-IP, 13 And I also check table radacct in mysql database, no records are inserted into it. I think this means an incorrect RADIUS dictionary. You should verify that the extra attributes you have defined are present there. -- Sincerely, Andrew Pogrebennyk -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS + FreeRadius
Hi, Thanks for your reply, I've copied dictionary.opensips to /etc/freeradius directory and added few lines in it after that when I make the call I'm getting this same error as mentioned in my previous mail earlier, I even copied the dictionary.freeradius to /etc/clientradius_ng directory as well. Lines for dictionary.opensips are listed below; Attributes ### ATTRIBUTE Sip-Uri-User 208 string # Proprietary, auth_radius ATTRIBUTE Sip-Group211 string # Proprietary, group_radius ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius ATTRIBUTE SIP-AVP 225 string # Proprietary, avp_radius ATTRIBUTE Sip-Method 101 integer ### Service-Type Values ### VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 VALUE Service-Type SIP-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type SIP-Callee-AVPs 31 # Proprietary, avp_radius ### Sip-Method Values ### VALUE Sip-Method Undefined 0 VALUE Sip-Method Invite 1 VALUE Sip-Method Cancel 2 VALUE Sip-Method Ack4 VALUE Sip-Method Bye8 VALUE Sip-Method Info 16 VALUE Sip-Method Options32 VALUE Sip-Method Update 64 VALUE Sip-Method Register 128 VALUE Sip-Method Message256 VALUE Sip-Method Subscribe 512 VALUE Sip-Method Notify 1024 VALUE Sip-Method Prack 2048 VALUE Sip-Method Refer 4096 VALUE Sip-Method Other 8192 And my dictionary settings are listed below; dictionary(freeradius) $INCLUDE/opt/freeradius/share/freeradius/dictionary $INCLUDE/opt/freeradius/etc/raddb/dictionary.opensips dictionary(radiusclient) ##At the end of the line $INCLUDE /opt/freeradius/etc/raddb/dictionary.opensips Kindly tell me from which section I'm facing this problem. Waiting for your reply. Date: Fri, 26 Feb 2010 15:21:03 +0200 From: Andrew Pogrebennyk andrew.pogreben...@portaone.com Subject: Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS + FreeRadius` To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: 4b87cabf.9000...@portaone.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 26.02.2010 14:33, Ahmed Munir wrote: Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]: ERROR:aaa_radius:rad_avp_add: failure Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]: ERROR:acc:acc_aaa_request: failed to add Source-IP, 13 And I also check table radacct in mysql database, no records are inserted into it. I think this means an incorrect RADIUS dictionary. You should verify that the extra attributes you have defined are present there. -- Sincerely, Andrew Pogrebennyk -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS + FreeRadius
Hi, Thanks for replying, Norman I've added the line in dictionary.opensips i.e. ATTRIBUTE Source-IP 214 string, and start freeradius service and make a call and still getting same error as I mentioned on my previous mail. Kindly assist me to resolve this problem because its quite too long that I'm this problem. Date: Thu, 04 Mar 2010 12:52:49 -0500 From: Norman Brandinger n...@goes.com Subject: Re: [OpenSIPS-Users] Getting Error When ConfiguringOpenSIPS + FreeRadius To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: 4b8ff371.6000...@goes.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed I've got the following (among other attributes) defined in a local.dictionary file: ATTRIBUTE Source-IP 214 string If it's added to opensips.dictionary, then it no longer needs to be in my local.dictionary. Regards, Norm Bogdan-Andrei Iancu wrote: Hi Ahmed, Do you see in any RADIUS dictionary the Source-IP AVP ? if yes, please post here its definition. Regards, Bogdan Ahmed Munir wrote: Hi, Thanks for your reply, I've copied dictionary.opensips to /etc/freeradius directory and added few lines in it after that when I make the call I'm getting this same error as mentioned in my previous mail earlier, I even copied the dictionary.freeradius to /etc/clientradius_ng directory as well. Lines for dictionary.opensips are listed below; Attributes ### ATTRIBUTE Sip-Uri-User 208 string # Proprietary, auth_radius ATTRIBUTE Sip-Group211 string # Proprietary, group_radius ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius ATTRIBUTE SIP-AVP 225 string # Proprietary, avp_radius ATTRIBUTE Sip-Method 101 integer ### Service-Type Values ### VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 VALUE Service-Type SIP-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type SIP-Callee-AVPs 31 # Proprietary, avp_radius ### Sip-Method Values ### VALUE Sip-Method Undefined 0 VALUE Sip-Method Invite 1 VALUE Sip-Method Cancel 2 VALUE Sip-Method Ack4 VALUE Sip-Method Bye8 VALUE Sip-Method Info 16 VALUE Sip-Method Options32 VALUE Sip-Method Update 64 VALUE Sip-Method Register 128 VALUE Sip-Method Message256 VALUE Sip-Method Subscribe 512 VALUE Sip-Method Notify 1024 VALUE Sip-Method Prack 2048 VALUE Sip-Method Refer 4096 VALUE Sip-Method Other 8192 And my dictionary settings are listed below; dictionary(freeradius) $INCLUDE/opt/freeradius/share/freeradius/dictionary $INCLUDE/opt/freeradius/etc/raddb/dictionary.opensips dictionary(radiusclient) ##At the end of the line $INCLUDE /opt/freeradius/etc/raddb/dictionary.opensips Kindly tell me from which section I'm facing this problem. Waiting for your reply. Date: Fri, 26 Feb 2010 15:21:03 +0200 From: Andrew Pogrebennyk andrew.pogreben...@portaone.com mailto:andrew.pogreben...@portaone.com Subject: Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS + FreeRadius` To: OpenSIPS users mailling list users@lists.opensips.org mailto:users@lists.opensips.org Message-ID: 4b87cabf.9000...@portaone.com mailto:4b87cabf.9000...@portaone.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 26.02.2010 14:33, Ahmed Munir wrote: Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]: ERROR:aaa_radius:rad_avp_add: failure Feb 26 14:47:47 rose /usr/local/sbin/opensips[25988]: ERROR:acc:acc_aaa_request: failed to add Source-IP, 13 And I also check table radacct in mysql database, no records are inserted into it. I think this means an incorrect RADIUS dictionary. You should verify that the extra attributes you have defined are present there. -- Sincerely, Andrew Pogrebennyk -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Users Digest, Vol 20, Issue 18
Hi,
Thanks for your reply Norman, I figure it out the problem. The problem is
that many of the attributes weren't defined in dictionary.opensips, after
adding the attributes, accounting with radius is working now. Here is the
list of dictionary.opensips attributes I want to share;
### Lines Added
ATTRIBUTE Sip-Method101 integer
ATTRIBUTE Sip-Response-Code 102 integer# Schulzrinne, acc
ATTRIBUTE Sip-To-Tag104 string # Schulzrinne, acc
ATTRIBUTE Sip-From-Tag 105 string # Schulzrinne, acc
ATTRIBUTE Sip-Translated-Request-URI107 string # Proprietary, acc
ATTRIBUTE Source-IP 214 string
ATTRIBUTE Source-Port 215 string
ATTRIBUTE Sip-Src-IP108 string # Proprietary, acc
ATTRIBUTE Sip-Src-Port 109 string # Proprietary, acc
ATTRIBUTE Digest-Response 206 string # Sterman,
auth_radius
ATTRIBUTE Sip-Uri-User 208 string # Proprietary,
auth_radius
ATTRIBUTE Sip-Group 211 string # Proprietary,
group_radius
ATTRIBUTE Sip-Rpid 213 string # Proprietary,
auth_radius
ATTRIBUTE SIP-AVP 225 string # Proprietary,
avp_radius
ATTRIBUTE Digest-Realm 1063 string# Sterman,
auth_radius
ATTRIBUTE Digest-Nonce 1064 string# Sterman,
auth_radius
ATTRIBUTE Digest-Method 1065 string# Sterman,
auth_radius
ATTRIBUTE Digest-URI1066 string# Sterman,
auth_radius
ATTRIBUTE Digest-QOP1067 string# Sterman,
auth_radius
ATTRIBUTE Digest-Algorithm 1068 string# Sterman,
auth_radius
ATTRIBUTE Digest-Body-Digest1069 string# Sterman,
auth_radius
ATTRIBUTE Digest-CNonce 1070 string# Sterman,
auth_radius
ATTRIBUTE Digest-Nonce-Count1071 string# Sterman,
auth_radius
ATTRIBUTE Digest-User-Name 1072 string# Sterman,
auth_radius
ATTRIBUTE Contact 1073 integer
###
Well I say it again thanks for helping me out.
Date: Fri, 05 Mar 2010 11:11:16 -0500
From: Norman Brandinger n...@goes.com
Subject: Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS
+ FreeRadius
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID: 4b912d24.40...@goes.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Please post the current error that you're receiving so that someone on
the list might be able to point you in the correct direction.
Regards,
Norm
Ahmed Munir wrote:
Hi,
Thanks for replying, Norman I've added the line in dictionary.opensips
i.e. ATTRIBUTE Source-IP 214 string, and start
freeradius service and make a call and still getting same error as I
mentioned on my previous mail.
Kindly assist me to resolve this problem because its quite too long
that I'm this problem.
Date: Thu, 04 Mar 2010 12:52:49 -0500
From: Norman Brandinger n...@goes.com mailto:n...@goes.com
Subject: Re: [OpenSIPS-Users] Getting Error When Configuring
OpenSIPS
+ FreeRadius
To: OpenSIPS users mailling list users@lists.opensips.org
mailto:users@lists.opensips.org
Message-ID: 4b8ff371.6000...@goes.com
mailto:4b8ff371.6000...@goes.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I've got the following (among other attributes) defined in a
local.dictionary file:
ATTRIBUTE Source-IP 214 string
If it's added to opensips.dictionary, then it no longer needs to be
in
my local.dictionary.
Regards,
Norm
Bogdan-Andrei Iancu wrote:
Hi Ahmed,
Do you see in any RADIUS dictionary the Source-IP AVP ? if yes,
please
post here its definition.
Regards,
Bogdan
Ahmed Munir wrote:
Hi,
Thanks for your reply, I've copied dictionary.opensips to
/etc/freeradius directory and added few lines in it after that
when I
make the call I'm getting this same error as mentioned in my
previous
mail earlier, I even copied the dictionary.freeradius to
/etc/clientradius_ng directory as well.
Lines for dictionary.opensips are listed below;
Attributes ###
ATTRIBUTE Sip-Uri-User 208 string # Proprietary,
auth_radius
ATTRIBUTE Sip-Group211 string # Proprietary,
group_radius
ATTRIBUTE Sip-Rpid 213 string # Proprietary,
auth_radius
ATTRIBUTE SIP-AVP 225 string # Proprietary,
avp_radius
ATTRIBUTE Sip-Method 101 integer
### Service
Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS + FreeRadius
Hi,
Thanks for your reply Norman, I figure it out the problem. The problem is
that many of the attributes weren't defined in dictionary.opensips, after
adding the attributes, accounting with radius is working now. Here is the
list of dictionary.opensips attributes I want to share;
### Lines Added
ATTRIBUTE Sip-Method101 integer
ATTRIBUTE Sip-Response-Code 102 integer# Schulzrinne, acc
ATTRIBUTE Sip-To-Tag104 string # Schulzrinne, acc
ATTRIBUTE Sip-From-Tag 105 string # Schulzrinne, acc
ATTRIBUTE Sip-Translated-Request-URI107 string # Proprietary, acc
ATTRIBUTE Source-IP 214 string
ATTRIBUTE Source-Port 215 string
ATTRIBUTE Sip-Src-IP108 string # Proprietary, acc
ATTRIBUTE Sip-Src-Port 109 string # Proprietary, acc
ATTRIBUTE Digest-Response 206 string # Sterman,
auth_radius
ATTRIBUTE Sip-Uri-User 208 string # Proprietary,
auth_radius
ATTRIBUTE Sip-Group 211 string # Proprietary,
group_radius
ATTRIBUTE Sip-Rpid 213 string # Proprietary,
auth_radius
ATTRIBUTE SIP-AVP 225 string # Proprietary,
avp_radius
ATTRIBUTE Digest-Realm 1063 string# Sterman,
auth_radius
ATTRIBUTE Digest-Nonce 1064 string# Sterman,
auth_radius
ATTRIBUTE Digest-Method 1065 string# Sterman,
auth_radius
ATTRIBUTE Digest-URI1066 string# Sterman,
auth_radius
ATTRIBUTE Digest-QOP1067 string# Sterman,
auth_radius
ATTRIBUTE Digest-Algorithm 1068 string# Sterman,
auth_radius
ATTRIBUTE Digest-Body-Digest1069 string# Sterman,
auth_radius
ATTRIBUTE Digest-CNonce 1070 string# Sterman,
auth_radius
ATTRIBUTE Digest-Nonce-Count1071 string# Sterman,
auth_radius
ATTRIBUTE Digest-User-Name 1072 string# Sterman,
auth_radius
ATTRIBUTE Contact 1073 integer
###
Well I say it again thanks for helping me out.
- Hide quoted text -
Date: Fri, 05 Mar 2010 11:11:16 -0500
From: Norman Brandinger n...@goes.com
Subject: Re: [OpenSIPS-Users] Getting Error When Configuring OpenSIPS
+ FreeRadius
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID: 4b912d24.40...@goes.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Please post the current error that you're receiving so that someone on
the list might be able to point you in the correct direction.
Regards,
Norm
Ahmed Munir wrote:
Hi,
Thanks for replying, Norman I've added the line in dictionary.opensips
i.e. ATTRIBUTE Source-IP 214 string, and start
freeradius service and make a call and still getting same error as I
mentioned on my previous mail.
Kindly assist me to resolve this problem because its quite too long
that I'm this problem.
Date: Thu, 04 Mar 2010 12:52:49 -0500
From: Norman Brandinger n...@goes.com mailto:n...@goes.com
Subject: Re: [OpenSIPS-Users] Getting Error When Configuring
OpenSIPS
+ FreeRadius
To: OpenSIPS users mailling list users@lists.opensips.org
mailto:users@lists.opensips.org
Message-ID: 4b8ff371.6000...@goes.com
mailto:4b8ff371.6000...@goes.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I've got the following (among other attributes) defined in a
local.dictionary file:
ATTRIBUTE Source-IP 214 string
If it's added to opensips.dictionary, then it no longer needs to be
in
my local.dictionary.
Regards,
Norm
Bogdan-Andrei Iancu wrote:
Hi Ahmed,
Do you see in any RADIUS dictionary the Source-IP AVP ? if yes,
please
post here its definition.
Regards,
Bogdan
Ahmed Munir wrote:
Hi,
Thanks for your reply, I've copied dictionary.opensips to
/etc/freeradius directory and added few lines in it after that
when I
make the call I'm getting this same error as mentioned in my
previous
mail earlier, I even copied the dictionary.freeradius to
/etc/clientradius_ng directory as well.
Lines for dictionary.opensips are listed below;
Attributes ###
ATTRIBUTE Sip-Uri-User 208 string # Proprietary,
auth_radius
ATTRIBUTE Sip-Group211 string # Proprietary,
group_radius
ATTRIBUTE Sip-Rpid 213 string # Proprietary,
auth_radius
ATTRIBUTE SIP-AVP 225 string # Proprietary,
avp_radius
ATTRIBUTE Sip-Method 101 integer
[OpenSIPS-Users] OpenSIPS + FreeRadius Accounting and Authentication
Hi, I've configured OpenSIPS + FreeRadius Accounting and Authentication setup, which was implemented success full. Using Authentication via Freeradius can anybody tell me how can I populate data on radius database tables? Mean what sort of values do I required for its tables so I can authenticate and register my softphone? Like in tables radreply, radgroupcheck, radgroureply, realms, etc. Kindly put the light on it and assist me with some sample data. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Call Distinguish in OpenSIPS
Hi, Current I'm working on OpenSIPS + FreeRadius, where FreeRadius is for AAA. Accounting and Authentication are working well i.e. SIP phones get authenticated and can make calls between them. I want to know how can I distinguish calls, the flow is listed down below; User A's number is 1234 and User B's number is 1235. Both users' phone registered on UAS (OpenSIPS+FreeRadius), can make SIP-SIP (on-net) calls. If User A do not registered his number, he can make call to User B where User B is registered on UAS like PSTN-SIP call. If User A is registered on UAS and make a call to User B who is not registered on UAS but located on PSTN, SIP-PSTN. In summary I need to know how can I configure SIP-SIP, SIP-PSTN and PSTN-SIP peers and how can I distribute their routes? Further added, which modules, modparam and function requires for it? -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Check Live Peers on OpenSIPS
Hi, I want to know how can I check the peers of source and destination phones? Like if both phones are located (registered) on one UAS(OpenSIPS) can call SIP-SIP, if any one phone is registered on UAS and other is on PSTN, call will be re-routed to SIP-PSTN. In case of SIP-SIP, lookup(location) function works and I need to know how can I forward call to SIP-PSTN ? Kindly advise me the method/ function can used for it. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
Hi Bogdan, Thanks for reply. I forgot to mention earlier that for I'm using OpenSIPS + FreeRadius, where radius is doing accounting and authentication. I used aaa_does_uri_exist() function as well, but seems not working or making mistake while implementing it. On other hand using lookup(location,m) function, on retcode = -1, I redirected the INVITE to GW, using Dispatcher. But though thanks for your suggestion and I'll consider it. Few things I want to ask you, as I listed below; 1-How can I forward SIP INVITE request to other SIP machine in state full manner ? 2- While accounting using radius, when user A (registered on OpenSIPS) calls the user B who is located at GW side, accounting doesn't take place. On the other hand when user B (from GW) calls user A (to OpenSIPS), accounting take place. I want to know its cause? Because I want its accounting on both sides. Kindly advise me at your earliest. -- Message: 6 Date: Thu, 18 Mar 2010 10:23:27 +0200 From: Bogdan-Andrei Iancu bog...@voice-system.ro Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: 4ba1e2ff.3060...@voice-system.ro Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi Ahmed, if the destination number (called number) is not a local subscriber (a SIP user), you simply route the call to a PSTN GW (you do this re-route from the script) To check if a user is a local subscriber, you can either check a pattern (like all my local users are alphanumeric, or all starts with 3345*, etc), either simply check if the user does exists in the subscriber table (see the URI module, the db_does_uri_exists() function: http://www.opensips.org/html/docs/modules/1.6.x/uri.html#id271131 Regards, Bogdan Ahmed Munir wrote: Hi, I want to know how can I check the peers of source and destination phones? Like if both phones are located (registered) on one UAS(OpenSIPS) can call SIP-SIP, if any one phone is registered on UAS and other is on PSTN, call will be re-routed to SIP-PSTN. In case of SIP-SIP, lookup(location) function works and I need to know how can I forward call to SIP-PSTN ? Kindly advise me the method/ function can used for it. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
))
{
route(2);
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
# apply DB based aliases (uncomment to enable)
##alias_db_lookup(dbaliases);
# do lookup with method filtering
if (!lookup(location,m)) {
switch ($retcode) {
case -1:
log(# LOOKUP LOCATION FLAG -1
PASS ###);
setflag(2);
rewritehostport(11.22.33.44:5060);
log(### CALL ROUTING TO ROUTE 1
###);
route(1);
exit;
case -3:
log(# LOOKUP LOCATION FLAG -3
PASS ###);
t_newtran();
t_reply(404, Not Found);
exit;
case -2:
log(# LOOKUP LOCATION FLAG -2
PASS ###);
sl_send_reply(405, Method Not Allowed);
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
log( LOOKUP LOCATION FLAG 1 PASS );
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
#if (is_method(INVITE) check_source_address(0)) {
if (is_method(INVITE)) {
log(INVITE ROUTE 1
Function);
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
#ds_select_dst(1,4);
#forward();
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[2]
{
log(## AAA-REGISTRATION #);
if (!aaa_www_authorize(rose.abc.com))
{
www_challenge(rose.abc.com, 1);
return;
}
if (!save(location))
sl_reply_error();
exit;
}
branch_route[2] {
xlog(new branch at $ru\n);
}
onreply_route[2] {
xlog(incoming reply\n);
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
}
Kindly assist me, how can I permit or deny user from source IP ? Because on
machine A, check_source_address() function is working perfectly but I
haven't integrated FreeRadius with OpenSIPs. Please sort out my problem as
your earliest.
Date: Thu, 18 Mar 2010 18:38:29 +0200
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID: 4ba25705.10...@voice-system.ro
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi Ahmed,
Ahmed Munir wrote:
Hi Bogdan,
Thanks for reply. I forgot to mention earlier that for I'm using
OpenSIPS + FreeRadius, where radius is doing accounting and
authentication. I used aaa_does_uri_exist() function as well, but
seems not working or making mistake while implementing it. On other
hand using lookup(location,m) function, on retcode = -1, I
redirected the INVITE to GW, using Dispatcher. But though thanks for
your suggestion and I'll consider it.
Few things I want to ask you, as I listed below;
1-How can I forward SIP INVITE request to other SIP machine in state
full manner ?
simply do:
# set new destination in RURI
$rd= 11.22.33.44;
# send it out in stateful mode
t_relay();
exit;
2- While accounting using radius, when user A (registered on OpenSIPS)
calls the user B who is located at GW side, accounting doesn't take
place. On the other hand when user B (from GW) calls user A (to
OpenSIPS), accounting take place. I want to know its cause? Because I
want its accounting on both sides.
take care and check where you set in script the acc flag - maybe you are
setting it only if lookup is successful.
Regards,
Bogdan
Kindly advise me at your earliest.
--
Message: 6
Date: Thu, 18 Mar 2010 10:23:27 +0200
From: Bogdan-Andrei Iancu bog...@voice-system.ro
mailto:bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
To: OpenSIPS users mailling list users@lists.opensips.org
mailto:users@lists.opensips.org
Message-ID: 4ba1e2ff.3060...@voice-system.ro
mailto:4ba1e2ff.3060...@voice-system.ro
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi Ahmed,
if the destination number (called number) is not a local subscriber
(a
SIP user), you simply route
Re: [OpenSIPS-Users] Users Digest, Vol 20, Issue 85
Hi Bogdan,
Thanks for your reply. As you suggested about check_source_address()
function, I get its return value using $avp(i:checksrc) as listed down
below;
$avp(s:checksrc) = check_source_address(0);
log(#\n);
xlog(Check Source Address from Address TABLE Where Value 1 is Equal
to True: $(avp(s:checksrc))\n);
log(#\n);
if($avp(s:checksrc)!=1)
{
if(is_method(INVITE))
{
log( CHECK SOURCE ADDRESS
##);
route(1);
setflag(1);
}
}
else
{
t_reply(403,Forbidden);
exit;
}
But the problem I'm facing is when I enlist IP in address table i.e.
11.22.33.44, call is rejected when else condition is used, when else
condition is commented call is made. But on other hand when I remove the IP
as mentioned from address table, it should reject the call (commenting else
condition), unfortunately the call is made.
Kindly assist me how can I permit or deny calls on IP bases, when user is
not registered from OpenSIPS but sending calls from GW to OpenSIPs?
Date: Mon, 22 Mar 2010 00:09:43 +0200
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID: 4ba69927.2050...@voice-system.ro
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi Ahmed
Ahmed Munir wrote:
Hi Bogdan,
Thanks for your suggestion, few things I want to ask from you;
1- Can I use rewritehostport(); function instead of $rd='11.22.33.44'
and append it to t_relay()? Like;
setflag(2);
rewritehostport(11.22.33.55:5060 http://203.215.179.34:5060);
t_relay();
route(1);
exit;
Yes, that is correct.
2- When using check_source_address() function of permissions module,
I'm facing weird problem. On machine A I've installed OpenSIPS ver
1.6.1 svn one, I used this function to permitted certain source IPs as
I listed in address table. On machine B (currently working on it using
Radius) I've installed same version of OpenSIPS as on machine A, when
I call its check_source_address() function in INVITE section, it is
working as it worked on machine A. Machine A settings are listed below;
if(is_method(INVITE) check_source_address(0))
{
log( CHECK SOURCE ADDRESS
##);
route(1);
setflag(1);
}
Machine B description I'm mentioning below;
2-1- If user registered him/her self on SIP phone their source IP not
going to be checked, and make calls to each other.
2-2- If user A is on GW calls user B who is located and Registered on
OpenSIPS, user A GW's source IP must be checked by OpenSIPs, if the
IP exists on address table, call is permitted if not deny the call.
Problems;
When I user A and user B registered on OpenSIPs (using Radius) they
can call each other, but if a user A calling from GW to user B who is
registered on OpenSIPs, calls is made even the address is not listed
on address table. And also in logs I see that that permissions module
shows that it doesn't find any IP enlisted in its hash table, but
still permitting it.
The function just checks if the source IP is in the table, but does not
take any action - you need to so this manually from the script, based on
the return code (true or false) of the function.
Regards,
Bogdan
The configuration of machine B is listed below;
[]
Kindly assist me, how can I permit or deny user from source IP ?
Because on machine A, check_source_address() function is working
perfectly but I haven't integrated FreeRadius with OpenSIPs. Please
sort out my problem as your earliest.
Date: Thu, 18 Mar 2010 18:38:29 +0200
From: Bogdan-Andrei Iancu bog...@voice-system.ro
mailto:bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
To: OpenSIPS users mailling list users@lists.opensips.org
mailto:users@lists.opensips.org
Message-ID: 4ba25705.10...@voice-system.ro
mailto:4ba25705.10...@voice-system.ro
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi Ahmed,
Ahmed Munir wrote:
Hi Bogdan,
Thanks for reply. I forgot to mention earlier that for I'm using
OpenSIPS + FreeRadius, where radius is doing accounting and
authentication. I used aaa_does_uri_exist() function as well, but
seems not working or making mistake while implementing it. On other
hand using lookup(location,m) function, on retcode = -1, I
redirected the INVITE to GW, using Dispatcher. But though
[OpenSIPS-Users] Getting Error when using NATHELPER module
] {
# for INVITEs enable some additional helper routes
if (is_method(INVITE)) {
log( INVITE ROUTE 1
Function ##);
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
}
if (subst_uri('/(sip:.*);nat=yes/\1/')){
log( IF SUBSTR CONTAINS
NAT=YES );
setbflag(6);
};
if (isflagset(5)||isbflagset(6)) {
log( CHECK FLAGSET AND
ROUTE TO 4 ###);
route(4);
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[2]
{
log( AAA-REGISTRATION
###);
if (!aaa_www_authorize(rose.vopium.com))
{
www_challenge(rose.vopium.com, 1);
return;
##exit;
}
if(isflagset(5))
{
log(### IF FLAG SET IS 5
##);
# set branch flag -- when someone will call this user
# the INVITE will have branch flag 6 set after
lookup(location)
setbflag(6);
# if you want OPTIONS natpings uncomment next
# setbflag(7);
}
if (!save(location))
sl_reply_error();
exit;
}
route[3]
{
log( FUNCTION ROUTE 3 NAT
DETECTION );
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
};
setflag(5);
};
}
route[4]
{
log( FUNCTION ROUTE 4 RTP PROXY
);
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy();
#t_on_failure(2);
t_on_failure(3);
};
if (isflagset(5))
search_append('Contact:.*sip:[^[:cntrl:]]*', ';nat=yes');
#t_on_reply(1);
t_on_reply(3);
}
branch_route[2] {
xlog(new branch at $ru\n);
}
onreply_route[2] {
xlog(incoming reply\n);
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
}
failure_route[3] {
log( FAILURE ROUTE 3 FUNCTION
);
if (isbflagset(6) || isflagset(5)) {
unforce_rtp_proxy();
}
}
onreply_route[3] {
log( ONREPLY ROUTE 3 FUNCTION
);
if ((isflagset(5) || isbflagset(6))
status=~(183)|(2[0-9][0-9])) {
force_rtp_proxy();
}
search_append('Contact:.*sip:[^[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) {
fix_nated_contact();
}
exit;
}
Kindly state, how can I resolve this error in my above configuration. Please
advise.
--
Regards,
Ahmed Munir
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Re: [OpenSIPS-Users] Users Digest, Vol 21, Issue 4
Hi Bogdan, Thanks for reply. I fill up; modparam(nathelper,rtpproxy_sock,udp: 127.0.0.1:7890), but still getting errors as listed below; Apr 2 11:43:49 rose /usr/local/sbin/opensips[16309]: ERROR:nathelper:force_rtp_proxy_body: no available proxies Apr 2 11:43:52 rose /usr/local/sbin/opensips[16310]: ERROR:nathelper:unforce_rtp_proxy_f: no available proxies Apr 2 11:43:52 rose /usr/local/sbin/opensips[16311]: ERROR:nathelper:force_rtp_proxy: Unable to parse body Please advise to overcome this problem. From: Bogdan-Andrei Iancu bog...@voice-system.ro Subject: Re: [OpenSIPS-Users] Getting Error when using NATHELPER module To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: 4bb471de.8070...@voice-system.ro Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello Ahmed, you script does not configure any rtpproxy to be used - the rtpproxy_sock parameter is empty: modparam(nathelper,rtpproxy_sock,) You need to set a valid link to a running rtpproxy : http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id228332 Regards, Bogdan Ahmed Munir wrote: Hi, I've configured OpenSIPs with Radius and now working to configure NAT on OpenSIPs using module mod_nathelper. After configuring, I'm getting following errors as listed down below; Apr 1 11:53:31 rose /usr/local/sbin/opensips[11386]: ERROR:nathelper:select_rtpp_node: script error -no valid set selected Apr 1 11:53:31 rose /usr/local/sbin/opensips[11386]: ERROR:nathelper:force_rtp_proxy_body: no available proxies Apr 1 11:53:46 rose /usr/local/sbin/opensips[11382]: ERROR:nathelper:select_rtpp_node: script error -no valid set selected Apr 1 11:53:46 rose /usr/local/sbin/opensips[11382]: ERROR:nathelper:unforce_rtp_proxy_f: no available proxies Apr 1 11:53:46 rose /usr/local/sbin/opensips[11386]: ERROR:nathelper:force_rtp_proxy: Unable to parse body And the configuration of OpenSIPs is listed below; [...] Kindly state, how can I resolve this error in my above configuration. Please advise. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] NAT Problem using Nat helper
( FUNCTION ROUTE 4 RTP PROXY
);
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy();
#t_on_failure(2);
t_on_failure(3);
};
if (isflagset(5))
search_append('Contact:.*sip:[^[:cntrl:]]*', ';nat=yes');
#t_on_reply(1);
t_on_reply(3);
}
branch_route[2] {
xlog(new branch at $ru\n);
}
onreply_route[2] {
xlog(incoming reply\n);
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status(3[0-9][0-9])) {
##t_reply(404,Not found);
## exit;
##}
# uncomment the following lines if you want to redirect the failed
# calls to a different new destination
##if (t_check_status(486|408)) {
## sethostport(192.168.2.100:5060);
## # do not set the missed call flag again
## t_relay();
##}
}
failure_route[3] {
log( FAILURE ROUTE 3 FUNCTION
);
if (isbflagset(6) || isflagset(5)) {
unforce_rtp_proxy();
}
}
onreply_route[3] {
log( ONREPLY ROUTE 3 FUNCTION
);
if ((isflagset(5) || isbflagset(6)) status=~(183)|(2[0-9][0-9])) {
force_rtp_proxy();
}
search_append('Contact:.*sip:[^[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) {
fix_nated_contact();
}
exit;
}
Kindly help me out with this problem, in which other section Natting is
required?(or am I missing something in the configuration?) Please assist me
on it.
--
Regards,
Ahmed Munir
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Re: [OpenSIPS-Users] NAT Problem using Nat helper
-Length: 130. . v=0. o=- 2 2 IN IP4 192.168.0.168. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.0.168. t=0 0. m=audio 1876 RTP/AVP 8 0. a=sendrecv. U 81.201.82.45:5060 - 11.22.33.44:5060 ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes SIP/2.0. Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45. CSeq: 102 ACK. From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com ;tag=43772. To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e. Via: SIP/2.0/UDP 81.201.82.45:5060 ;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd. Max-Forwards: 69. Contact: sip:4572727...@81.201.82.45:5060;transport=udp. Route: sip:11.22.33.44;lr. User-Agent: Vox Callcontrol. Content-Length: 0. . U 11.22.33.44:5060 - 203.215.176.22:55134 ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0. Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45. CSeq: 102 ACK. From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com ;tag=43772. To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e. Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.2. Via: SIP/2.0/UDP 81.201.82.45:5060 ;rport=5060;received=81.201.82.45;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd. Max-Forwards: 68. Contact: sip:4572727...@81.201.82.45:5060;transport=udp. User-Agent: Vox Callcontrol. Content-Length: 0. . U 11.22.33.44:5060 - 203.215.176.22:55134 U 203.215.176.22:55134 - 11.22.33.44:5060 . . .. U 203.215.176.22:55134 - 11.22.33.44:5060 BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.0.168:55134 ;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport. Max-Forwards: 70. Route: sip:11.22.33.44;lr. Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26. To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com ;tag=43772. From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e. Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45. CSeq: 2 BYE. User-Agent: X-Lite release 1104o stamp 56125. Reason: SIP;description=User Hung Up. Content-Length: 0. . U 11.22.33.44:5060 - 81.201.82.45:5060 BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0. Via: SIP/2.0/UDP 192.168.0.168:55134 ;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134. Max-Forwards: 69. Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes. To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com ;tag=43772. From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e. Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45. CSeq: 2 BYE. User-Agent: X-Lite release 1104o stamp 56125. Reason: SIP;description=User Hung Up. Content-Length: 0. . U 81.201.82.45:5060 - 11.22.33.44:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0,SIP/2.0/UDP 192.168.0.168:55134 ;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134. To: 4572727220 sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com ;tag=43772. From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e. Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45. CSeq: 2 BYE. Content-Length: 0. . U 11.22.33.44:5060 - 203.215.176.22:55134 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.0.168:55134 ;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134. To: 4572727220 sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com ;tag=43772. From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e. Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45. CSeq: 2 BYE. Content-Length: 0. . Date: Thu, 29 Apr 2010 19:34:16 -0300 From: Antonio Anderson Souza anto...@voicetechnology.com.br Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: s2o285c24cc1004291534m1deec8c4zb6c4ddb003311...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Ahmed, Could you send an wireshark trace to the list? It will be easier to check what's going wrong. Besta regards, Antonio Anderson M. Souza Voice Technology http://www.antonioams.com Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu: Hi, I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm using is 1.6.1 and FreeRadius verison is update 2 date. When I register 2 sofphone, they got authenticated and authorized by radius and got registered sucessfully. Even I made calls between two softphone sucessfully(Can hear one another). The UAS configured on different network means hosted with public IP and my softphones are registered other and NATed network. I mapped a DID on UAS and mapped it on my one of my softphone. The problem I'm facing is when call coming from DID and ring my phone the caller can hear me but I can't hear the caller(one way calling issue). But not facing the problem
[OpenSIPS-Users] Getting Error when using NATHELPER module
);
# if you want OPTIONS natpings uncomment next
# setbflag(7);
}
if (!save(location))
sl_reply_error();
exit;
}
route[3]
{
log( FUNCTION ROUTE 3 NAT
DETECTION );
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
};
setflag(5);
};
}
route[4]
{
log( FUNCTION ROUTE 4 RTP PROXY
);
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy();
#t_on_failure(2);
t_on_failure(3);
};
if (isflagset(5))
search_append('Contact:.*sip:[^[:cntrl:]]*', ';nat=yes');
#t_on_reply(1);
t_on_reply(3);
}
branch_route[2] {
xlog(new branch at $ru\n);
}
onreply_route[2] {
xlog(incoming reply\n);
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
failure_route[3] {
log( FAILURE ROUTE 3 FUNCTION
);
if (isbflagset(6) || isflagset(5)) {
unforce_rtp_proxy();
}
}
onreply_route[3] {
log( ONREPLY ROUTE 3 FUNCTION
);
if ((isflagset(5) || isbflagset(6))
status=~(183)|(2[0-9][0-9])) {
force_rtp_proxy();
}
search_append('Contact:.*sip:[^[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) {
fix_nated_contact();
}
exit;
}
Further more to add when I call within UAS means call between 2 registered
softphones on OpenSIPs 2 way audio is heard, but when calling from DID one
way audio is passing through, caller can hears the UAC which is registered
on OpenSIPs but UAC can't hears the caller. Note: OpenSIPs is hosted on
public IP and UAC are located on different network behind the Nat.
Please assist me to resolve this problem. Waiting for your reply.
--
Regards,
Ahmed Munir
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Re: [OpenSIPS-Users] NAT Problem using Nat helper
Hi, Thanks for replying. Can you please check my configuration of OpenSIPs what I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146. Please point out in which section do I required to add force_rtp_proxy(), because I already configured Nat on it. kindly advise me soon. On Fri, Apr 30, 2010 at 11:35 AM, users-requ...@lists.opensips.org wrote: Send Users mailing list submissions to users@lists.opensips.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.opensips.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-requ...@lists.opensips.org You can reach the person managing the list at users-ow...@lists.opensips.org When replying, please edit your Subject line so it is more specific than Re: Contents of Users digest... Today's Topics: 1. Re: NAT Problem using Nat helper (Laszlo) -- Message: 1 Date: Fri, 30 Apr 2010 08:35:00 +0200 From: Laszlo las...@voipfreak.net Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Ahmed, As you can see, the other party gets local ip in SDP c=IN IP4 192.168.0.168. You can try to play with flags: http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028 -Laszlo 2010/4/30 Ahmed Munir ahmedmunir...@gmail.com Hi. Thanks for your reply, the traces are metioned below; U 203.215.176.22:55134 - 11.22.33.44:5060 . . .. U 81.201.82.45:5060 - 11.22.33.44:5060 INVITE sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44 SIP/2.0. Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45. CSeq: 102 INVITE. From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com sip%3a4572727...@voxbone.com sip%253a4572727...@voxbone.com ;tag=43772. To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44. Via: SIP/2.0/UDP 81.201.82.45:5060 ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0. Max-Forwards: 69. Content-Type: application/sdp. Contact: sip:4572727...@81.201.82.45:5060;transport=udp. User-Agent: Vox Callcontrol. Content-Length: 210. . v=0. o=root 13293 13293 IN IP4 81.201.82.146. s=session. c=IN IP4 81.201.82.146. t=0 0. m=audio 11458 RTP/AVP 8 0. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 11.22.33.44:5060 - 81.201.82.45:5060 SIP/2.0 100 Giving a try. Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45. CSeq: 102 INVITE. From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com sip%3a4572727...@voxbone.com sip%253a4572727...@voxbone.com ;tag=43772. To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44. Via: SIP/2.0/UDP 81.201.82.45:5060 ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0;rport=5060. Server: OpenSIPS (1.6.1-notls (i386/linux)). Content-Length: 0. . U 11.22.33.44:5060 - 203.215.176.22:55134 INVITE sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0. Record-Route: sip:11.22.33.44;lr=on. Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45. CSeq: 102 INVITE. From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com sip%3a4572727...@voxbone.com sip%253a4572727...@voxbone.com ;tag=43772. To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44. Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0. Via: SIP/2.0/UDP 81.201.82.45:5060 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0. Max-Forwards: 68. Content-Type: application/sdp. Contact: sip:4572727...@81.201.82.45:5060;transport=udp. User-Agent: Vox Callcontrol. Content-Length: 210. P-hint: usrloc applied. . v=0. o=root 13293 13293 IN IP4 81.201.82.146. s=session. c=IN IP4 81.201.82.146. t=0 0. m=audio 11458 RTP/AVP 8 0. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 203.215.176.22:55134 - 11.22.33.44:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0. Via: SIP/2.0/UDP 81.201.82.45:5060 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0. Record-Route: sip:11.22.33.44;lr. Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26. To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44;tag=611cee1e. From: 4572727220sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com sip%3a4572727
Re: [OpenSIPS-Users] NAT Problem using Nat helper
Hi, Thanks for supporting me, really appreciated your help. Date: Mon, 03 May 2010 12:39:55 +0300 From: Bogdan-Andrei Iancu bog...@voice-system.ro Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: 4bde99eb.9090...@voice-system.ro Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi Ahmed, as a hint, probably you do not handle correctly the case when only the callee is nated (caller is public) - for such cases, to see if rtpproxy is needed, after the lookup(location) the nat_bflag will will automatically set if the callee location is nated - you can use that flag to detect the nated callee and to do the nat fixups - force rtpp and fix the 200 ok from the callee (SDP and contact). Regards, Bogdan Ahmed Munir wrote: Hi, Thanks for replying. Can you please check my configuration of OpenSIPs what I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146. Please point out in which section do I required to add force_rtp_proxy(), because I already configured Nat on it. kindly advise me soon. On Fri, Apr 30, 2010 at 11:35 AM, users-requ...@lists.opensips.org mailto:users-requ...@lists.opensips.org wrote: Send Users mailing list submissions to users@lists.opensips.org mailto:users@lists.opensips.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.opensips.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-requ...@lists.opensips.org mailto:users-requ...@lists.opensips.org You can reach the person managing the list at users-ow...@lists.opensips.org mailto:users-ow...@lists.opensips.org When replying, please edit your Subject line so it is more specific than Re: Contents of Users digest... Today's Topics: 1. Re: NAT Problem using Nat helper (Laszlo) -- Message: 1 Date: Fri, 30 Apr 2010 08:35:00 +0200 From: Laszlo las...@voipfreak.net mailto:las...@voipfreak.net Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper To: OpenSIPS users mailling list users@lists.opensips.org mailto:users@lists.opensips.org Message-ID: r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com mailto: r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Ahmed, As you can see, the other party gets local ip in SDP c=IN IP4 192.168.0.168. You can try to play with flags: http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028 -Laszlo -- Bogdan-Andrei Iancu www.voice-system.ro -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 22, Issue 13 * -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Nat Problem
;
##exit;
}
#else
#{
# t_reply(405,UnAuhorized);
# exit();
#}
if(isflagset(5))
{
log(### IF FLAG SET IS 5
##);
# set branch flag -- when someone will call this user
# the INVITE will have branch flag 6 set after
lookup(location)
setbflag(6);
# if you want OPTIONS natpings uncomment next
# setbflag(7);
}
if (!save(location))
sl_reply_error();
exit;
}
route[3]
{
log( FUNCTION ROUTE 3 NAT
DETECTION );
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
};
setflag(5);
};
}
route[4]
{
log( FUNCTION ROUTE 4 RTP PROXY
);
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy();
#t_on_failure(2);
t_on_failure(3);
};
if (isflagset(5))
search_append('Contact:.*sip:[^[:cntrl:]]*', ';nat=yes');
#t_on_reply(1);
t_on_reply(3);
}
branch_route[2] {
xlog(new branch at $ru\n);
}
onreply_route[2] {
xlog(incoming reply\n);
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
}
failure_route[3] {
log( FAILURE ROUTE 3 FUNCTION
);
if (isbflagset(6) || isflagset(5)) {
unforce_rtp_proxy();
}
}
onreply_route[3] {
log( ONREPLY ROUTE 3 FUNCTION
);
if ((isflagset(5) || isbflagset(6))
status=~(183)|(2[0-9][0-9])) {
force_rtp_proxy();
}
search_append('Contact:.*sip:[^[:cntrl:]]*', ';nat=yes');
if (isbflagset(6)) {
fix_nated_contact();
}
exit;
}
Kindly assist me in my script to sort out this problem, (please point out
what other changes or addition function do I required for it). Note My
OpenSIPs is hosted on public IP and on different network and my UAC is at
private IP. Please advise.
--
Regards,
Ahmed Munir
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[OpenSIPS-Users] Getting Error when using STUN
Hi, I'm getting and error when I configure STUN in OpenSIPs. As following documentation of OpenSIPs, I enabled stun as listed below; listen=udp:198.65.166.165:5060 listen=udp:75.101.138.128:5060 loadmodule stun.so # Stun modparam(stun,primary_ip,198.65.166.165) modparam(stun,primary_port,5060) modparam(stun,alternate_ip,75.101.138.128) modparam(stun,alternate_port,5060) Where the IP of OpenSIPs which is hosted on public IP i.e. 11.22.33.44. And the error I'm getting after restarting the OpenSIPs is listed below; May 17 05:37:13 newtest /usr/local/sbin/opensips[31199]: ERROR:core:udp_init: bind(5, 0x81b7374, 16) on 77.66.16.35: Cannot assign requested address When I commented out the listen=udp:IP:5060 the error I'm getting is; May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if host==us: 11==9 [77.66.16.35] == [127.0.0.1] May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if host==us: 11==11 [77.66.16.35] == [77.66.2.137] May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:stun:stun_mod_init: grep_sock_in()1 failed May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: ERROR:core:init_mod: failed to initialize module stun May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: ERROR:core:main: error while initializing modules Kindly assist me how can I resolve this problem. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] SNMP MIB for OpenSIPS
Hi all, I would like to know which SNMP MIB(s) for OpenSIP can be used for checking current number of active calls, channels available and connection attempts? Further added, after configuring OpenSIPs with SNMP, which shell command shall I used to verify my requirements? Please advice at earliest as I haven't worked on SNMP. -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPs + SNMP
Hi all, As I looked in to the document page for OpenSIPs + SNMP configuration; http://www.opensips.org/html/docs/modules/devel/snmpstats.html Some points which are not cleared to me, like to ask. As far as the configuration in OpenSIPs part, I only need to call following options in the configuration? loadmodule snmpstats.so modparam(snmpstats, sipEntityType, proxyServer) modparam(snmpstats, snmpgetPath, /usr/bin/) The thing is I need to check the number of current active calls which is provided by 'opensipDialogTable'. How can I run this option using snampwalk command to check current active calls? Do I need to some more lines in OpenSIPs configuration? Please advice. -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs + SNMP
Anyone, please assist me out at earliest. Date: Fri, 1 Jun 2012 11:08:28 -0400 From: Ahmed Munir ahmedmunir...@gmail.com Subject: [OpenSIPS-Users] OpenSIPs + SNMP To: OpenSIPs Users users@lists.opensips.org Message-ID: CAGMN=Jfaa2_YoAbo6T+ZUnbQD2euBUK2XYM9W4TU=y97tuo...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi all, As I looked in to the document page for OpenSIPs + SNMP configuration; http://www.opensips.org/html/docs/modules/devel/snmpstats.html Some points which are not cleared to me, like to ask. As far as the configuration in OpenSIPs part, I only need to call following options in the configuration? loadmodule snmpstats.so modparam(snmpstats, sipEntityType, proxyServer) modparam(snmpstats, snmpgetPath, /usr/bin/) The thing is I need to check the number of current active calls which is provided by 'opensipDialogTable'. How can I run this option using snampwalk command to check current active calls? Do I need to some more lines in OpenSIPs configuration? Please advice. -- Regards, Ahmed Munir Chohan -- next part -- An HTML attachment was scrubbed... URL: http://lists.opensips.org/pipermail/users/attachments/20120601/e6180c3c/attachment.html -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Connecting pbx to Opensips
Hi Schneur, In my opinion, you are missing to configure inbound route in OpenSIPs. Date: Sun, 3 Jun 2012 09:42:53 +0300 From: Schneur Rosenberg rosenberg11...@gmail.com Subject: [OpenSIPS-Users] Connecting pbx to Opensips To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: canvjr0v-c6kj_crshwvpooz52pbgjxu6qtx7mhqt3oza-kx...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 I'm using OpenSIPS to load balance multiple Asterisk servers, all phones are registered to OpenSIPS, and Asterisk shares the subscriber table, every INVITE gets sent to asterisk and when Asterisk sees the invite it recognizes the user and sends call accordingly, (call plan, caller id etc). Everything worked fine until I tried connecting a FreePbx system as a client, problem is that the FreePbx sends the invite to OpenSIPS with the internal username and therefore Asterisk and OpenSIPS have no idea what to do with it, for example, I set up on my OpenSIPS/Asterisk system a user freepbx1 so that OpenSIPS can authenticate it, on the FreePbx there are 50 users 100-150, FreePbx sends the INVITE as from 101@freepbx. When connecting the FreePbx directly to Asterisk it works fine, how can I fix this? -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Getting opensips: Unknown Object Identifier (Sub-id not found: (top) - opensips)
Hi, After configuring SNMP + OpenSIPS as described in the document; http://www.opensips.org/html/docs/modules/devel/snmpstats.html#id250252 and http://www.kamailio.org/dokuwiki/doku.php/utils:kamailio-and-snmp. When I try to run the 'snmapwalk localhost openser/opensips' on shell, I'm getting this message as listed below; opensips: Unknown Object Identifier (Sub-id not found: (top) - opensips) Please assist me at earliest, to resolve this issue. -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Set GW from probing to active mode in dispatcher module
Hi all, I would like to know, what configuration do I need to for setting up GW from probing to active mode automatically while using dispatcher module? As currently, OpenSIPs automatically able to set failed GW to probing mode after I called following modparam listed below; modparam(dispatcher, ds_ping_method, INFO) modparam(dispatcher, ds_ping_interval, 10) modparam(dispatcher, ds_probing_threshhold, 3) modparam(dispatcher, ds_probing_mode, 1) Further added, as a note; I don't want to reload dispatcher once failed GW in active state. Please advise. -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Async DB statement
Thanks, have been working on this and it is working.
Btw, I would like to know, is there a way to resume route while using async
avp_db_query? As currently setting/declaring another async route for DB
query, looking for resume route in main routing script.
On Tue, Jan 24, 2017 at 6:45 AM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:
> Hi Ahmed,
>
> Note the $rc holds the return code of the LAST executed
> statement/instruction/function in the script. In the first case you do it
> right by saving the ret code of the avp_db_query into a separate variable,
> so you can use it even later.
>
> In the sync script, the $rc, when entering the resume route, it will hold
> the return code of the avp_db_query() function. But the $rc will be changed
> when doing the xlog(), the if(), etc...So when you do the last xlog(), the
> $rc will have nothing to do with the avp_db_query(). If you need it later
> in the script, better save it, as you do in the first example.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01/20/2017 01:31 AM, Ahmed Munir wrote:
>
> Hi,
>
> Currently I'm trying to use async fucntion for avp_db_query. The issue I'm
> facing while using it as not retrieving or returning correct return code
> and not execute later part of the routing script. See old & new DB queries;
>
> Without Async:
> --
> route[1]{
> ...
>
> if($var(Outpluseflag) == 0) {
> avp_db_query("SELECT Outpulse_number,setid FROM Prefix_data
> where Program_prefix = '$var(pg_prefix)'", "$avp(outpluse), $avp(trunkid)");
> $var(res) = $retcode; # or you can just use $retcode!
> xlog("- OB Route 1-1 DB fetched value outpluse ->
> $avp(outpluse) | trunkid -> $avp(trunkid) | Return Code: $rc | Var Res:
> $var(res)---");
> if ($var(res) > 0) {
>cache_store("local", "DID_$tU",
> "$avp(outpluse)", 60);
>cache_store("local", "Trunk_$tU",
> "$avp(trunkid)", 60);
> }
> #xlog("DB fetched value outpluse -> $avp(outpluse) |
> trunkid -> $avp(trunkid) | Return Code -> $var(res)");
> xlog("- OB Route 1-2 DB fetched value outpluse ->
> $avp(outpluse) | trunkid -> $avp(trunkid) | Return Code: $rc | Var Res:
> $var(res)---");
> }
> }
>
> With Async:
> ---
> route[1]{
>
> ...
>
> if($var(Outpluseflag) == 0) {
> async(avp_db_query("SELECT Outpulse_number,setid FROM Prefix_data
> where Program_prefix = '$var(pg_prefix)'", "$avp(outpluse),
> $avp(trunkid)"),ob_route_1);
> }
> }
>
> route[ob_route_1]{
> xlog("- OB Route 1-1 DB fetched value outpluse ->
> $avp(outpluse) | trunkid -> $avp(trunkid) | Return Code: $rc---");
>
> if ($rc > 0) {
>cache_store("local", "DID_$tU", "$avp(outpluse)", 60);
>cache_store("local", "Trunk_$tU", "$avp(trunkid)", 60);
> }
>xlog("- OB Route 1-2 DB fetched value outpluse ->
> $avp(outpluse) | trunkid -> $avp(trunkid) | Return Code: $rc---");
>
> }
>
>
> The records in xlog I'm getting without using async;
>
> Jan 19 18:05:39 qorblpsisprxyd1 /usr/sbin/opensips[14040]: - OB
> Route 1-1 DB fetched value outpluse -> 609902 <(609)%20902-> |
> trunkid -> 117 | Return Code: 1 | Var Res: 1---
> Jan 19 18:05:39 qorblpsisprxyd1 /usr/sbin/opensips[14040]: - OB
> Route 1-2 DB fetched value outpluse -> 609902 <(609)%20902-> |
> trunkid -> 117 | Return Code: 1 | Var Res: 1---
>
> Whereas, records in xlog I'm getting using async;
>
> Jan 19 18:10:07 qorblpsisprxyd1 /usr/sbin/opensips[14109]: - OB
> Route 1-1 DB fetched value outpluse -> 609902 <(609)%20902-0000> |
> trunkid -> 117 | Return Code: 1---
> Jan 19 18:10:07 qorblpsisprxyd1 /usr/sbin/opensips[14109]: - OB
> Route 1-2 DB fetched value outpluse -> 609902 <(609)%20902-> |
> trunkid -> 117 | Return Code: 0---
>
> Is there is way to properly retain the $retcode/$rc in version 2.2.2?
> Seems like using async return code(s) are not properly set or the avp
> variables are not setting up correct using async statement.
>
> Please advise, if the above async db statement is correct as shared in
> sample above.
>
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
>
>
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>
>
>
--
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Ahmed Munir Chohan
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Re: [OpenSIPS-Users] OpenSIPs 2.2.2 warnings during start service
Yes, using startup route in my routing script and running DB query. This
kind of warning I didn't face using 1.6 and 1.8 opensips versions but 2.2.2.
Will you recommend async statement for my current routing (see below) for
the startup?
startup_route
{
$var(res) = 1;
$avp(tmp) = "1";
# $var(x) = 0;
while($var(res) > 0)
{
$var(res) = avp_db_query("SELECT Distinct One800, dnis FROM
DNIS_Mapping where One800 > $avp(tmp) order by One800;", "$avp(One800),
$avp(dnis)");
if($var(res) >= 0)
{
$var(i) = 0;
while($(avp(One800)[$var(i)]) != "NULL")
{
cache_store("local", "DNIS_$(avp(dnis)[$var(i)])",
"$(avp(One800)[$var(i)])");
$avp(tmp) = $(avp(One800)[$var(i)]);
# $var(x) = $var(x) + 1;
$(avp(One800)[$var(i)]) = "NULL";
$var(i) = $var(i) + 1;
# xlog("$var(x) : $(avp(s:dnis)[$var(i)])");
}
}
}
}
On Tue, Jan 24, 2017 at 6:38 AM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:
> Hi Ahmed,
>
> So, the warnings pop up ONLY during startup sequence. Do you use startup
> route or any module performing mem caching of some DB table (drouting,
> permission, etc) ? Usually, the first UDP child is doing some heavy lifting
> during startup.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01/19/2017 06:01 PM, Ahmed Munir wrote:
>
> Hi Razvan,
>
> During starting up the opensips service, I see the first opensips child
> process (pid"11172) consumes CPU process to 70-80% and later drop downs to
> 0.3 - 0.0 % CPU per core. See below;
>
> [root@qorblpsisprxyd1 ~]# top -c -u opensips
> top - 10:49:54 up 76 days, 23:31, 5 users, load average: 0.00, 0.00, 0.00
> Tasks: 229 total, 1 running, 228 sleeping, 0 stopped, 0 zombie
> Cpu(s): 0.0%us, 0.1%sy, 0.0%ni, 99.9%id, 0.0%wa, 0.0%hi, 0.0%si,
> 0.0%st
> Mem: 65964364k total, 2568124k used, 63396240k free, 180220k buffers
> Swap: 1023996k total,0k used, 1023996k free, 1226104k cached
>
> PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
> 11177 opensips 20 0 165m 5696 4528 S 0.3 0.0 0:00.23
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11187 opensips 20 0 165m 5628 4460 S 0.3 0.0 0:00.41
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11166 opensips 20 0 165m 6892 5752 S 0.0 0.0 0:00.23
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11168 opensips 20 0 165m 1980 840 S 0.0 0.0 0:00.00
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11169 opensips 20 0 165m 1464 328 S 0.0 0.0 0:00.53
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11171 opensips 20 0 165m 1640 504 S 0.0 0.0 0:00.15
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11172 opensips 20 0 166m 40m 38m S 0.0 0.1 0:02.61
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11174 opensips 20 0 165m 6304 5136 S 0.0 0.0 0:00.24
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11175 opensips 20 0 165m 5884 4716 S 0.0 0.0 0:00.22
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11179 opensips 20 0 165m 7660 6492 S 0.0 0.0 0:00.27
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11181 opensips 20 0 165m 7756 6588 S 0.0 0.0 0:00.33
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11183 opensips 20 0 165m 5520 4352 S 0.0 0.0 0:00.34
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11185 opensips 20 0 165m 7336 6168 S 0.0 0.0 0:00.36
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11189 opensips 20 0 165m 7320 6152 S 0.0 0.0 0:00.36
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
> 11190 opensips 20 0 165m 4688 3528 S 0.0 0.0 0:00.30
> /usr/sbin/opensips -P /var/run/opensips.pid -m 64 -u opensips -g opensips
>
> [root@qorblpsisprxyd1 ~]# opensipsctl fifo ps
> Process:: ID=0 PID=11166 Type=attendant
> Process:: ID=1 PID=11168 Type=MI FIFO
> Process:: ID=2 PID=11169 Type=time_keeper
> Process:: ID=3 PID=11171 Type=timer
> Process:: ID=4 PID=11172 Type=SIP receiver udp:10.3.120.94:5060
> Process:: ID=5 PID=11174 Type=SIP receiver udp:10
Re: [OpenSIPS-Users] OpenSIPs crashed
Is there updates on this? Date: Mon, 16 Jan 2017 09:51:13 -0500 > From: Ahmed Munir <ahmedmunir...@gmail.com> > To: OpenSIPs Users <users@lists.opensips.org> > Subject: Re: [OpenSIPS-Users] OpenSIPs crashed > Message-ID: >
[OpenSIPS-Users] Issues with avp_db_query in opensips 2.2.2
Hi,
I've currently upgraded opensips from 1.8.8 to 2.2.2. The issue currently
facing after upgrade as getting error messages and failed to start to
opensips service when using avp_db_query () function like;
$var(res) = avp_db_query("SELECT Outpulse_number,setid FROM
Prefix_data where Program_prefix = $var(pg_prefix)", "$avp(outpluse),
$avp(trunkid)");
Errors below;
ERROR:avpops:__fixup_db_query_avp: no db url defined to be used by
this function
ERROR:core:fix_actions: fixing failed (code=-6) at
//etc/opensips/opensips.cfg:207
CRITICAL:core:fix_expr: fix_actions error
ERROR:core:main: failed to fix configuration with err code -6
Whereas, variable $var(res) is storing return code after executing DB query.
If I add this line: avp_db_query("SELECT 1"); above to my $var(res) db
query, opensips service starts successfully and don't see the errors.
Please advise the steps do I need to take to fix above issues.
BTW, declared avpops 'db_url' in module parameters.
modparam("dispatcher|avpops","db_url","mysql://opensips:opensipsrw@localhost
/opensips")
--
Regards,
Ahmed Munir Chohan
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Re: [OpenSIPS-Users] OpenSIPs crashed
See details below;
[root@qorblpsisprxyd1 ~]# opensips -V
version: opensips 2.2.2 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
main.c compiled on 11:09:47 Jan 13 2017 with gcc 4.4.7
[root@qorblpsisprxyd1 ~]# opensipsctl ps
Process:: ID=0 PID=8269 Type=attendant
Process:: ID=1 PID=8271 Type=MI FIFO
Process:: ID=2 PID=8272 Type=time_keeper
Process:: ID=3 PID=8274 Type=timer
Process:: ID=4 PID=8275 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=5 PID=8278 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=6 PID=8279 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=7 PID=8281 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=8 PID=8283 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=9 PID=8285 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=10 PID=8287 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=11 PID=8289 Type=SIP receiver udp:10.3.120.94:5060
Process:: ID=12 PID=8291 Type=Timer handler
[root@qorblpsisprxyd1 ~]# lscpu
Architecture: x86_64
CPU op-mode(s):32-bit, 64-bit
Byte Order:Little Endian
CPU(s):8
On-line CPU(s) list: 0-7
Thread(s) per core:1
Core(s) per socket:4
Socket(s): 2
NUMA node(s): 2
Vendor ID: GenuineIntel
CPU family:6
Model: 44
Stepping: 2
CPU MHz: 1197.000
BogoMIPS: 4266.58
Virtualization:VT-x
L1d cache: 32K
L1i cache: 32K
L2 cache: 256K
L3 cache: 8192K
NUMA node0 CPU(s): 0-3
NUMA node1 CPU(s): 4-7
In /etc/default/opensips config, declaring shared and pkg memory as server
memory is 64 GB;
# Amount of shared memory to allocate for the running OpenSIPS server (in
Mb)
S_MEMORY=256
# Amount of pkg memory to allocate for the running OpenSIPS server (in Mb)
P_MEMORY=32
Let me know any other info needed from my end.
Date: Mon, 16 Jan 2017 10:49:37 +0200
> From: Răzvan Crainea <raz...@opensips.org>
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] OpenSIPs crashed
> Message-ID: <1584806e-a154-a5ad-a464-4eef60915...@opensips.org>
> Content-Type: text/plain; charset="utf-8"; Format="flowed"
>
> Hi, Ahmed!
>
> Can you tell us exactly what revision of OpenSIPS you are using? Please
> provide the output of the following commands:
> opensips -V
> opensipsctl ps
>
> Also, during startup, is there a process who's "eating" a lot of CPU? If
> so, can you pinpoint the PID to see what type of process is that?
>
> Regarding the avp_db_query() issue, did you define a db_url parameter
> for it? Also I am not sure you can do something like $var(res) =
> avp_db_query(...). But anyways, this is something completely different,
> so please open a different topic for it.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
> On 01/14/2017 12:24 AM, Ahmed Munir wrote:
> > Hi,
> >
> > I've just installed new version of opensips 2.2.2 on the test box and
> > updated by routing script, the issue currently I'm seeing alot warning
> > messages while starting opensips service below;
> >
> > /usr/sbin/opensips[6902]: WARNING:core:handle_timer_job: utimer job
> > has a 283 us delay in execution
> >
> > Number of children running on that server is 8 as it is 8 core processor.
> >
> > I would like to know what steps do I need to take to fix this issue.
> > Btw, warnings only occurred during the time of starting opensips
> > service but not during calls.
> >
> >
> > Further added, a issue I face using avp_db_query () function i.e. when
> > using it as
> >
> > $var(res) = avp_db_query("SELECT Outpulse_number,setid FROM
> > Prefix_data where Program_prefix = $var(pg_prefix)", "$avp(outpluse),
> > $avp(trunkid)");
> >
> > failed to start opensips service due to errors below;
> >
> > ERROR:avpops:__fixup_db_query_avp: no db url defined to be used by
> > this function
> > ERROR:core:fix_actions: fixing failed (code=-6) at
> > //etc/opensips/opensips.cfg:207
> > CRITICAL:core:fix_expr: fix_actions error
> > ERROR:core:main: failed to fix configuration with err code -6
> >
> >
> > If I add this line: avp_db_query("SELECT 1"); above to my $var(res) db
> > query, opensips service starts successfully.
> >
> > Please advise the steps do I need to take to fix above issues.
> >
> >
> >
>
>
--
Regards,
Ahmed Munir Chohan
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[OpenSIPS-Users] OpenSIPs 2.2.2 warnings during start service
Hi, I'm currently seeing the warnings when I start opensips service; Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: WARNING:core:handle_timer_job: timer job has a 150 us delay in execution Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: WARNING:core:handle_timer_job: timer job has a 150 us delay in execution Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: WARNING:core:handle_timer_job: timer job has a 150 us delay in execution Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: WARNING:core:handle_timer_job: utimer job has a 229 us delay in execution Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: INFO:core:do_action: max while loops are encountered Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3088]: WARNING:core:utimer_ticker: utimer task already scheduled for 190 ms (now 2470 ms), it may over lap.. I've tried to update the source code for timer.c (line#: 190) ref: https://github.com/OpenSIPS/opensips/commit/fd8f6ec442b4365da9d274af6939954246ece865?diff=split, but didn't work at all. Currently running 8 child processors, see below; [root@qorblpsisprxyd1 ]# opensips -V version: opensips 2.2.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on 12:39:45 Jan 18 2017 with gcc 4.4.7 [root@qorblpsisprxyd1 ]# opensipsctl fifo ps Process:: ID=0 PID=3083 Type=attendant Process:: ID=1 PID=3085 Type=MI FIFO Process:: ID=2 PID=3086 Type=time_keeper Process:: ID=3 PID=3088 Type=timer Process:: ID=4 PID=3089 Type=SIP receiver udp:10.3.120.94:5060 Process:: ID=5 PID=3091 Type=SIP receiver udp:10.3.120.94:5060 Process:: ID=6 PID=3092 Type=SIP receiver udp:10.3.120.94:5060 Process:: ID=7 PID=3094 Type=SIP receiver udp:10.3.120.94:5060 Process:: ID=8 PID=3096 Type=SIP receiver udp:10.3.120.94:5060 Process:: ID=9 PID=3098 Type=SIP receiver udp:10.3.120.94:5060 Process:: ID=10 PID=3100 Type=SIP receiver udp:10.3.120.94:5060 Process:: ID=11 PID=3102 Type=SIP receiver udp:10.3.120.94:5060 Process:: ID=12 PID=3104 Type=Timer handler I would like to know what changes required to fix this change? Please advise. -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Async DB statement
Hi,
Currently I'm trying to use async fucntion for avp_db_query. The issue I'm
facing while using it as not retrieving or returning correct return code
and not execute later part of the routing script. See old & new DB queries;
Without Async:
--
route[1]{
...
if($var(Outpluseflag) == 0) {
avp_db_query("SELECT Outpulse_number,setid FROM Prefix_data
where Program_prefix = '$var(pg_prefix)'", "$avp(outpluse), $avp(trunkid)");
$var(res) = $retcode; # or you can just use $retcode!
xlog("- OB Route 1-1 DB fetched value outpluse ->
$avp(outpluse) | trunkid -> $avp(trunkid) | Return Code: $rc | Var Res:
$var(res)---");
if ($var(res) > 0) {
cache_store("local", "DID_$tU",
"$avp(outpluse)", 60);
cache_store("local", "Trunk_$tU",
"$avp(trunkid)", 60);
}
#xlog("DB fetched value outpluse -> $avp(outpluse) |
trunkid -> $avp(trunkid) | Return Code -> $var(res)");
xlog("- OB Route 1-2 DB fetched value outpluse ->
$avp(outpluse) | trunkid -> $avp(trunkid) | Return Code: $rc | Var Res:
$var(res)---");
}
}
With Async:
---
route[1]{
...
if($var(Outpluseflag) == 0) {
async(avp_db_query("SELECT Outpulse_number,setid FROM Prefix_data
where Program_prefix = '$var(pg_prefix)'", "$avp(outpluse),
$avp(trunkid)"),ob_route_1);
}
}
route[ob_route_1]{
xlog("- OB Route 1-1 DB fetched value outpluse ->
$avp(outpluse) | trunkid -> $avp(trunkid) | Return Code: $rc---");
if ($rc > 0) {
cache_store("local", "DID_$tU", "$avp(outpluse)", 60);
cache_store("local", "Trunk_$tU", "$avp(trunkid)", 60);
}
xlog("- OB Route 1-2 DB fetched value outpluse ->
$avp(outpluse) | trunkid -> $avp(trunkid) | Return Code: $rc---");
}
The records in xlog I'm getting without using async;
Jan 19 18:05:39 qorblpsisprxyd1 /usr/sbin/opensips[14040]: - OB
Route 1-1 DB fetched value outpluse -> 609902 | trunkid -> 117 | Return
Code: 1 | Var Res: 1---
Jan 19 18:05:39 qorblpsisprxyd1 /usr/sbin/opensips[14040]: - OB
Route 1-2 DB fetched value outpluse -> 609902 | trunkid -> 117 | Return
Code: 1 | Var Res: 1---
Whereas, records in xlog I'm getting using async;
Jan 19 18:10:07 qorblpsisprxyd1 /usr/sbin/opensips[14109]: - OB
Route 1-1 DB fetched value outpluse -> 609902 | trunkid -> 117 | Return
Code: 1---
Jan 19 18:10:07 qorblpsisprxyd1 /usr/sbin/opensips[14109]: - OB
Route 1-2 DB fetched value outpluse -> 609902 | trunkid -> 117 | Return
Code: 0---
Is there is way to properly retain the $retcode/$rc in version 2.2.2? Seems
like using async return code(s) are not properly set or the avp variables
are not setting up correct using async statement.
Please advise, if the above async db statement is correct as shared in
sample above.
--
Regards,
Ahmed Munir Chohan
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Re: [OpenSIPS-Users] Users Digest, Vol 102, Issue 62
These warnings appears during opensips startup service. Date: Thu, 19 Jan 2017 14:55:01 + > From: "Ramachandran, Agalya (Contractor)" > <agalya_ramachand...@comcast.com> > To: OpenSIPS users mailling list <users@lists.opensips.org> > Subject: Re: [OpenSIPS-Users] OpenSIPs 2.2.2 warnings during start > service > Message-ID: > <21840442533f445397309d17eec6b...@copdcex28.cable.comcast.com> > Content-Type: text/plain; charset="utf-8" > > Hi Razvan, > > I didn’t see any process that is using more than 80% of a core. OpenSIPS > is simply being idle and these warnings come periodically. > > Like once in couple of hours. I didn’t track the exact time line, how > frequent am getting this warnings. > > Ahmed, > Do you notice these warnings only when you start OpenSIPS or could see it > in regular intervals? > > Regards, > Agalya > > From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan > Crainea > Sent: Thursday, January 19, 2017 3:33 AM > To: users@lists.opensips.org > Subject: Re: [OpenSIPS-Users] OpenSIPs 2.2.2 warnings during start service > > When starting opensips, is there any opensips process that is using more > than 80% of a core? If so, can you pinpoint the PID in the opensipsctl ps > command? > > Best regards, > > > Răzvan Crainea > > OpenSIPS Solutions > > www.opensips-solutions.com<http://www.opensips-solutions.com> > On 01/18/2017 11:55 PM, Ramachandran, Agalya (Contractor) wrote: > Same with my case too. > > Regards, > Agalya > > From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Ahmed > Munir > Sent: Wednesday, January 18, 2017 1:31 PM > To: OpenSIPs Users <users@lists.opensips.org><mailto:users@lists.opensips. > org> > Subject: [OpenSIPS-Users] OpenSIPs 2.2.2 warnings during start service > > Hi, > I'm currently seeing the warnings when I start opensips service; > > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > WARNING:core:handle_timer_job: timer job has a 150 us > delay in execution > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > WARNING:core:handle_timer_job: timer job has a 150 us delay > in execution > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > WARNING:core:handle_timer_job: timer job has a 150 us delay > in execution > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > WARNING:core:handle_timer_job: utimer job has a 229 us > delay in execution > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > INFO:core:do_action: max while loops are encountered > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3088]: > WARNING:core:utimer_ticker: utimer task already scheduled for > 190 ms (now 2470 ms), it may over > lap.. > > > I've tried to update the source code for timer.c (line#: 190) ref: > https://github.com/OpenSIPS/opensips/commit/fd8f6ec442b4365da9d274af693995 > 4246ece865?diff=split, but didn't work at all. > Currently running 8 child processors, see below; > > [root@qorblpsisprxyd1 ]# opensips -V > version: opensips 2.2.2 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > main.c compiled on 12:39:45 Jan 18 2017 with gcc 4.4.7 > > > [root@qorblpsisprxyd1 ]# opensipsctl fifo ps > Process:: ID=0 PID=3083 Type=attendant > Process:: ID=1 PID=3085 Type=MI FIFO > Process:: ID=2 PID=3086 Type=time_keeper > Process:: ID=3 PID=3088 Type=timer > Process:: ID=4 PID=3089 Type=SIP receiver udp:10.3.120.94:5060<http:// > 10.3.120.94:5060> > Process:: ID=5 PID=3091 Type=SIP receiver udp:10.3.120.94:5060<http:// > 10.3.120.94:5060> > Process:: ID=6 PID=3092 Type=SIP receiver udp:10.3.120.94:5060<http:// > 10.3.120.94:5060> > Process:: ID=7 PID=3094 Type=SIP receiver udp:10.3.120.94:5060<http:// > 10.3.120.94:5060> > Process:: ID=8 PID=3096 Type=SIP receiver udp:10.3.120.94:5060<http:// > 10.3.120.94:5060> > Process:: ID=9 PID=3098 Type=SIP receiver udp:10.3.120.94:5060<http:// > 10.3.120.94:5060> > Process:: ID=10 PID=3100 Type=SIP receiver udp:10.3.120.94:5060<http:// > 10.3.120.94:5060> > Process:: ID=11 PID=3102 Type=SIP receiver udp:10.3.120.94:5060<http:// > 10.3.120.94:5060> > Process:: ID=12 PID=3104 Type=Timer handler > I would like to know what changes required to fix this change? Please > advise. > > -- > Regards, > > Ahmed Munir Chohan > > -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs 2.2.2 warnings during start service
ge-ID: <2d785128-affa-c955-e779-1d4305ec1...@opensips.org> > Content-Type: text/plain; charset="utf-8"; Format="flowed" > > When starting opensips, is there any opensips process that is using more > than 80% of a core? If so, can you pinpoint the PID in the opensipsctl > ps command? > > Best regards, > > Răzvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 01/18/2017 11:55 PM, Ramachandran, Agalya (Contractor) wrote: > > > > Same with my case too. > > > > Regards, > > Agalya > > > > *From:*Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of > > *Ahmed Munir > > *Sent:* Wednesday, January 18, 2017 1:31 PM > > *To:* OpenSIPs Users <users@lists.opensips.org> > > *Subject:* [OpenSIPS-Users] OpenSIPs 2.2.2 warnings during start service > > > > Hi, > > > > I'm currently seeing the warnings when I start opensips service; > > > > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > > WARNING:core:handle_timer_job: timer job has a 150 > > us delay in execution > > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > > WARNING:core:handle_timer_job: timer job has a 150 us > > delay in execution > > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > > WARNING:core:handle_timer_job: timer job has a 150 us > > delay in execution > > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > > WARNING:core:handle_timer_job: utimer job has a 229 us > > delay in execution > > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3089]: > > INFO:core:do_action: max while loops are encountered > > Jan 18 13:04:35 qorblpsisprxyd1 /usr/sbin/opensips[3088]: > > WARNING:core:utimer_ticker: utimer task already scheduled > > for 190 ms (now 2470 ms), it may over > > lap.. > > > > I've tried to update the source code for timer.c (line#: 190) ref: > > https://github.com/OpenSIPS/opensips/commit/ > fd8f6ec442b4365da9d274af6939954246ece865?diff=split, > > but didn't work at all. > > > > Currently running 8 child processors, see below; > > > > [root@qorblpsisprxyd1 ]# opensips -V > > version: opensips 2.2.2 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > main.c compiled on 12:39:45 Jan 18 2017 with gcc 4.4.7 > > > > > > [root@qorblpsisprxyd1 ]# opensipsctl fifo ps > > Process:: ID=0 PID=3083 Type=attendant > > Process:: ID=1 PID=3085 Type=MI FIFO > > Process:: ID=2 PID=3086 Type=time_keeper > > Process:: ID=3 PID=3088 Type=timer > > Process:: ID=4 PID=3089 Type=SIP receiver udp:10.3.120.94:5060 > > <http://10.3.120.94:5060> > > Process:: ID=5 PID=3091 Type=SIP receiver udp:10.3.120.94:5060 > > <http://10.3.120.94:5060> > > Process:: ID=6 PID=3092 Type=SIP receiver udp:10.3.120.94:5060 > > <http://10.3.120.94:5060> > > Process:: ID=7 PID=3094 Type=SIP receiver udp:10.3.120.94:5060 > > <http://10.3.120.94:5060> > > Process:: ID=8 PID=3096 Type=SIP receiver udp:10.3.120.94:5060 > > <http://10.3.120.94:5060> > > Process:: ID=9 PID=3098 Type=SIP receiver udp:10.3.120.94:5060 > > <http://10.3.120.94:5060> > > Process:: ID=10 PID=3100 Type=SIP receiver udp:10.3.120.94:5060 > > <http://10.3.120.94:5060> > > Process:: ID=11 PID=3102 Type=SIP receiver udp:10.3.120.94:5060 > > <http://10.3.120.94:5060> > > Process:: ID=12 PID=3104 Type=Timer handler > > > > I would like to know what changes required to fix this change? Please > > advise. > > > > > > -- > > > > Regards, > > > > Ahmed Munir Chohan > > > > > > > > ___ > > Users mailing list > > Users@lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- next part -- > An HTML attachment was scrubbed... > URL: <http://lists.opensips.org/pipermail/users/attachments/ > 20170119/81b8d547/attachment-0001.html> > > > -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dispatcher issues.
Hi,
I've currently migrated from opensips 1.6.3 to 1.8.9 and upgraded the
routing script and DB attributes. The issue currently I'm facing on 1.8.9
is the dispatcher module i.e. showing most of the nodes in passive mode
rather than active. However, the nodes are showing passive mode in 1.8.9
were actually showing active state in 1.6.3. My question is, is there any
changes in dispatcher module in 1.8.9? As wen through the documentation of
1.8 opensips, don't see any difference in change.
Below are the module parameter I've set for dispatcher both in 1.8.9 and
1.6.3;
modparam("dispatcher|avpops","db_url","mysql://opensips:opensipsrw@localhost
/opensips")
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_ping_interval", 10)
modparam("dispatcher", "ds_probing_threshhold", 3)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "options_reply_codes", "501, 403")
modparam("dispatcher", "ds_ping_from", "sip:pr...@proxy.com")
Setting 'Flag' value to 0 and 'weight' to 1 for each node in dispatcher
table.
Please advise, if I missed out any config.
--
Regards,
Ahmed Munir Chohan
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs crashed
Hi,
I've just installed new version of opensips 2.2.2 on the test box and
updated by routing script, the issue currently I'm seeing alot warning
messages while starting opensips service below;
/usr/sbin/opensips[6902]: WARNING:core:handle_timer_job: utimer job
has a 283 us delay in execution
Number of children running on that server is 8 as it is 8 core processor.
I would like to know what steps do I need to take to fix this issue. Btw,
warnings only occurred during the time of starting opensips service but not
during calls.
Further added, a issue I face using avp_db_query () function i.e. when
using it as
$var(res) = avp_db_query("SELECT Outpulse_number,setid FROM Prefix_data
where Program_prefix = $var(pg_prefix)", "$avp(outpluse), $avp(trunkid)");
failed to start opensips service due to errors below;
ERROR:avpops:__fixup_db_query_avp: no db url defined to be used by this
function
ERROR:core:fix_actions: fixing failed (code=-6) at
//etc/opensips/opensips.cfg:207
CRITICAL:core:fix_expr: fix_actions error
ERROR:core:main: failed to fix configuration with err code -6
If I add this line: avp_db_query("SELECT 1"); above to my $var(res) db
query, opensips service starts successfully.
Please advise the steps do I need to take to fix above issues.
> From: Răzvan Crainea <raz...@opensips.org>
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] OpenSIPs crashed
> Message-ID: <40f6dada-e121-a2da-b283-69dff891c...@opensips.org>
> Content-Type: text/plain; charset="utf-8"; Format="flowed"
>
> Hi, Ahmed!
>
> OpenSIPS 1.6.3 is no longer supported (since 2013), so there's not much
> we can do right now. Try upgrading your install to the latest 1.6.4
> version and see if your problem is solved. Otherwise, upgrade to a
> newer, supported version, preferably the latest stable release, 2.2.2.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
> On 01/12/2017 11:55 PM, Ahmed Munir wrote:
> > Found coredump on one of the server, see some partial message below
> > while taking the back trace;
> >
> >
> >
> > Core was generated by `/usr/sbin/opensips -P /var/run/opensips.pid -m
> > 64 -u opensips -g opensips'.
> >
> > Program terminated with signal 11, Segmentation fault.
> >
> > #0 0x7f650687a069 in sip_msg_cloner () from
> > /usr/lib64/opensips/modules/tm.so
> >
> > Missing separate debuginfos, use: debuginfo-install
> > opensips-1.6.3-notls.x86_64
> >
> >
> >
> > Please advise what might be the reason causing opensips to crash.
> >
> > --
> > Regards,
> >
> > Ahmed Munir Chohan
> >
> >
> >
> >
> > --
> > Regards,
> >
> > Ahmed Munir Chohan
> >
>
--
Regards,
Ahmed Munir Chohan
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPs crashed
Hi, Our OpenSIPs service crashed with below error; Jan 11 12:16:19 QORBLPSIPROXY05 abrtd: Directory 'ccpp-2017-01-11-12:16:19-2807' creation detected Jan 11 12:16:19 QORBLPSIPROXY05 abrt[65402]: Saved core dump of pid 2807 (/usr/sbin/opensips) to /var/spool/abrt/ccpp-2017-01-11-12:16:19-2807 (70225920 bytes) Jan 11 12:16:19 QORBLPSIPROXY05 abrt[65402]: Can't open 'core.2807': Permission denied Jan 11 12:16:19 QORBLPSIPROXY05 kernel: opensips[2807]: segfault at 29 ip 004bae7a sp 7fffdb7734d0 error 4 in opensips[40+13a000] We would like to know, what might be the reason for the crash. Further added, there is another server we are running OpenSIPs, the opensips child processes utilizing 100% of CPU and the system load average reach around 'load average: 20.01, 18.03, 24.00' as normally it is below 1 (load average). After looking into logs, unable to find the info what might causing the CPU to spike. Please advise what useful steps to take for narrowing down this issue. -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs crashed
The version currently running is 1.6.3. Today again we got the opensips crashed issues i.e. 5 out of 8 were crashed due to below message (common on all 5); Jan 12 10:07:36 QORCLPSIPROXY02 kernel: opensips[2820]: segfault at 0 ip 004a2936 sp 7fff5cfaa430 error 6 in opensips[40+13a000] Jan 12 10:07:36 QORCLPSIPROXY02 abrt[39302]: Can't open 'core.2820': Permission denied Jan 12 10:07:36 QORCLPSIPROXY02 abrt[39302]: Saved core dump of pid 2820 (/usr/sbin/opensips) to /var/spool/abrt/ccpp-2017-01-12-10:07:36-2820 (70246400 bytes) Jan 12 10:07:36 QORCLPSIPROXY02 abrtd: Directory 'ccpp-2017-01-12-10:07:36-2820' creation detected Jan 12 10:07:38 QORCLPSIPROXY02 kernel: Bridge firewalling registered Jan 12 10:08:46 QORCLPSIPROXY02 abrtd: Sending an email... Jan 12 10:08:46 QORCLPSIPROXY02 abrtd: /usr/sbin/sendmail: No such file or directory Jan 12 10:08:46 QORCLPSIPROXY02 abrtd: . . . message not sent. Jan 12 10:08:46 QORCLPSIPROXY02 abrtd: Error running '/bin/mailx' Jan 12 10:08:46 QORCLPSIPROXY02 abrtd: 'post-create' on '/var/spool/abrt/ccpp-2017-01-12-10:07:36-2820' exited with 1 Jan 12 10:08:46 QORCLPSIPROXY02 abrtd: Deleting problem directory '/var/spool/abrt/ccpp-2017-01-12-10:07:36-2820' We would like to know why we are getting this segmentation fault? Is there a way to backtrace the issue as don't have the core dump? Further added, did someone faced this similar issue(s) in past and got this fixed without upgrade? Note: we are running opensips for quite a while didn't faced this kind of issue and there is no changes made on opensips routing script. > From: Răzvan Crainea <raz...@opensips.org> >> To: users@lists.opensips.org >> Subject: Re: [OpenSIPS-Users] OpenSIPs crashed >> Message-ID: <2b35e2dc-6e11-1787-b87e-33bd29a32...@opensips.org> >> Content-Type: text/plain; charset="utf-8"; Format="flowed" >> >> Hi, Ahmed! >> >> Make sure OpenSIPS is run as root and it is allowed to write in the >> /var/spool/abrt/ directory, otherwise it is unable to write the core >> dump, therefore we can't inspect it to say what is happening. If this >> does not work, make OpenSIPS write the core dump in a writeble directory >> by changing the /proc/sys/kernel/core_pattern settings. >> Also, please let us know the version of OpenSIPS you are running. >> >> Best regards, >> >> Răzvan Crainea >> OpenSIPS Solutions >> www.opensips-solutions.com >> >> >> On 01/11/2017 11:10 PM, Ahmed Munir wrote: >> > Hi, >> > >> > Our OpenSIPs service crashed with below error; >> > >> > Jan 11 12:16:19 QORBLPSIPROXY05 abrtd: Directory >> > 'ccpp-2017-01-11-12:16:19-2807' creation detected >> > Jan 11 12:16:19 QORBLPSIPROXY05 abrt[65402]: Saved core dump of pid >> > 2807 (/usr/sbin/opensips) to >> > /var/spool/abrt/ccpp-2017-01-11-12:16:19-2807 (70225920 bytes) >> > Jan 11 12:16:19 QORBLPSIPROXY05 abrt[65402]: Can't open 'core.2807': >> > Permission denied >> > Jan 11 12:16:19 QORBLPSIPROXY05 kernel: opensips[2807]: segfault at 29 >> > ip 004bae7a sp 7fffdb7734d0 error 4 in >> opensips[40+13a000] >> > >> > >> > We would like to know, what might be the reason for the crash. >> > >> > Further added, there is another server we are running OpenSIPs, the >> > opensips child processes utilizing 100% of CPU and the system load >> > average reach around 'load average: 20.01, 18.03, 24.00' as normally >> > it is below 1 (load average). >> > >> > After looking into logs, unable to find the info what might causing >> > the CPU to spike. >> > >> > Please advise what useful steps to take for narrowing down this issue. >> > >> > >> > -- >> > Regards, >> > >> > Ahmed Munir Chohan >> > >> >> > > > -- > Regards, > > Ahmed Munir Chohan > > -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs crashed
The version currently running is 1.6.3. Will try to enable core dump and share the info if run into the issues again. From: Răzvan Crainea <raz...@opensips.org> > To: users@lists.opensips.org > Subject: Re: [OpenSIPS-Users] OpenSIPs crashed > Message-ID: <2b35e2dc-6e11-1787-b87e-33bd29a32...@opensips.org> > Content-Type: text/plain; charset="utf-8"; Format="flowed" > > Hi, Ahmed! > > Make sure OpenSIPS is run as root and it is allowed to write in the > /var/spool/abrt/ directory, otherwise it is unable to write the core > dump, therefore we can't inspect it to say what is happening. If this > does not work, make OpenSIPS write the core dump in a writeble directory > by changing the /proc/sys/kernel/core_pattern settings. > Also, please let us know the version of OpenSIPS you are running. > > Best regards, > > Răzvan Crainea > OpenSIPS Solutions > www.opensips-solutions.com > > On 01/11/2017 11:10 PM, Ahmed Munir wrote: > > Hi, > > > > Our OpenSIPs service crashed with below error; > > > > Jan 11 12:16:19 QORBLPSIPROXY05 abrtd: Directory > > 'ccpp-2017-01-11-12:16:19-2807' creation detected > > Jan 11 12:16:19 QORBLPSIPROXY05 abrt[65402]: Saved core dump of pid > > 2807 (/usr/sbin/opensips) to > > /var/spool/abrt/ccpp-2017-01-11-12:16:19-2807 (70225920 bytes) > > Jan 11 12:16:19 QORBLPSIPROXY05 abrt[65402]: Can't open 'core.2807': > > Permission denied > > Jan 11 12:16:19 QORBLPSIPROXY05 kernel: opensips[2807]: segfault at 29 > > ip 004bae7a sp 7fffdb7734d0 error 4 in > opensips[40+13a000] > > > > > > We would like to know, what might be the reason for the crash. > > > > Further added, there is another server we are running OpenSIPs, the > > opensips child processes utilizing 100% of CPU and the system load > > average reach around 'load average: 20.01, 18.03, 24.00' as normally > > it is below 1 (load average). > > > > After looking into logs, unable to find the info what might causing > > the CPU to spike. > > > > Please advise what useful steps to take for narrowing down this issue. > > > > > > -- > > Regards, > > > > Ahmed Munir Chohan > > > > -- Regards, Ahmed Munir Chohan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
