[OpenSIPS-Users] acc record outbound response to originator

I'm wanting to record the response sent back to the call originator, not just the last winning response selected by tm module. There are cases where opensips translates like a received 503 to a 500 or like if you don't trust the response you received from a vendor that is not using sip codes proper

Re: [OpenSIPS-Users] Forward() question

Luis, forward() does not accept a variable for an argument. instead use like: $du = "sip:" + src_ip + ":5070"; I think it needs URI format without the user portion. see: http://www.opensips.org/Resources/DocsCoreVar17#toc34 There are a bunch of older functions that do not accept variables as a

Re: [OpenSIPS-Users] need step by step guide lines for media proxy installation

Have you checked if relay and dispatcher are actually running? Have you run them in the foreground with debug turned on to see why use_media_proxy returned an error? I'm betting Brett is right that radius is at least primarily the problem. I would disable it's use in opensips.cfg and mediaproxy/con

Re: [OpenSIPS-Users] avp set in branch_route

avps are per transaction. A branch is just part of a transaction. Until the transaction is terminated the avp stays active. Be aware that a call's avp is write-able in all branches and assigning a value to an existing avp pushes the new value on the top of the avp array. Calls do not go back to req

Re: [OpenSIPS-Users] Preparing new major release 1.7.0 - UPDATE

Speaking of suggestions... (understandable probably won't make it in 1.7, more likely 2.0) I've been thinking about variables in the script and a couple other things. 1. New variables for direct read/write access to value currently only available by function (very beneficial for example with cdrs)

Re: [OpenSIPS-Users] /etc/init.d/opensips restart returns OK when opensips is down...

Toyima, What I've done to deal with errors on a restart is to copy the config to a temp, replace a couple things like fifo file and ports, start it with foreground option, piping output to check for errors. The group is backgrounded with & and the init script waits for 2 sec. If bk process is stil

Re: [OpenSIPS-Users] multiple subscribers per alias

Erik, Just typed this up so probably has some mistakes. :-) This form pulls all alias db results into an avp which are looped though to do a lookup for each that only a user alias. You could do the append branch mode for alias_db but then lookup would have to be in branch route and Im not sure if

Re: [OpenSIPS-Users] Rewrite Username in URI when using userloc

Spencer, Looks like you just need to move $rU=$oU; into branch_route[2] Note that in failure route it will be reset to what it was before changes done in branch route so you may want to do it in main route and in branch route. Dave On Thu, May 19, 2011 at 4:02 PM, Spencer Thomason wrote: > Hello

Re: [OpenSIPS-Users] adding 1 alias to more than one subscriber

d not the ip of > the location user 1000, what's happening? seems to be a loop on the second > branch (192.168.131.129:5060 -> 192.168.131.129:5060)...as i can see, the DB > alias is correctly fetching both users from DB, but only the first one is > resolved via lookup(location

Re: [OpenSIPS-Users] adding 1 alias to more than one subscriber

Toyima, check: http://www.opensips.org/html/docs/modules/1.6.x/alias_db.html#id250030 Dave On Tue, May 17, 2011 at 6:44 AM, Toyima Dias wrote: > > Is it possible to add the same alias to more than one subscriber? for > example, if an incoming call to the number 14882736524 is assigned to > su

Re: [OpenSIPS-Users] opensips+rtpproxy strange locks

Nick, Have you verified there is no firewall on the opensips server with iptables -L I believe the dump pulls the packets before they go through iptables. So what you capture may be just dropped by iptables and not reaching opensips. Though really that shouldn't be a problem since iptables shou

Re: [OpenSIPS-Users] FIFO very fragile

, it works great. > Really, the concern I have it doing things that cause the fifo to "die" and > become unavailable. I've seen this with the opensipsctl script as well, where > the fifo just stops responding until you restart. > -Brett > > On Mon, May 9, 2011 at 4:

Re: [OpenSIPS-Users] Pjsip dialler with openIMS core

Looks like you sent your request to the wrong list. On Mon, May 9, 2011 at 12:24 AM, abid khan wrote: > Hello everybody, > I want to know main criteria or parameters which > i have to fulfill if i want to register* pjsip dialer* with openIMS core. > When i try to regist

Re: [OpenSIPS-Users] FIFO very fragile

n extra carriage return spoils the fun for > everyone and I can't fix it without restarting opensips. > > Do you believe SELinux could cause an issue like that? > Thanks! > -Brett > > > On Mon, May 9, 2011 at 3:39 PM, Dave Singer wrote: > >> Brett, >> I beli

Re: [OpenSIPS-Users] FIFO very fragile

Brett, I believe you need the full path to the reply fifo. Try: echo -e ":address_dump:/tmp/my_fifo\n\n" > /tmp/opensips_fifo you may also have a permissions ( chmod a+w /tmp/my_fifo ) and/or SELinux issue (very likely if SELinux is enabled, I posted a howto for SELinux in the list a number of mon

Re: [OpenSIPS-Users] Validate To Uri

Jan, I'm assuming you actually meant $tU which is the user portion of the To URI. If all you are trying to do is test if they are numeric then regex is the best way to test it. Something like: if ( ! $tU =~ "^[0-9]*$" ) { ... deal with non numeric user ... } Dave On Wed, May 4, 2011 at 2:07 A

Re: [OpenSIPS-Users] Mediaproxy remove unknown session errors

Jeff, Sounds like the bye is not getting matched with the dialog. There are some settings for dialog module for matching fallback that you probably want to look at. Also there are a number of threads that have discussed matching problems that may be helpful. Dave On Tue, May 3, 2011 at 2:46 PM,

Re: [OpenSIPS-Users] high process count

Bret, try: ps aux --forest | grep opensips that will show you parent and child processes in tree diagram and if any opensips master process is actually responsible for so many of them. I agree that it is probably being spawned many times with a cron job, loop in the init script or a service like m

Re: [OpenSIPS-Users] Opensips geographic redundancy

Marcello, Might check into getting an IP with BGP (Border Gateway Protocol) or some other IP routing protocol. Not all ISPs offer these services and it is usually offered on a subnet of addresses not a single address. If your providers/contacts can't do this or don't have good answers you might ch

Re: [OpenSIPS-Users] need OpenSIPS installation

Jeff, I would be interested in working with you. I've been integrating OpenSIPS in hosted VOIP environment for 4 years. Unfortunately I will be laid off the end of this week, Apr 22 (I'm the last one to go). You can look at my resume on Linkedin ( http://www.linkedin.com/in/davesingercando ) or Mo

[OpenSIPS-Users] Problem with Diversion when added in failure route and RURI uses DNS SRV

I have a server running 1.6.3. In failure route I add a Diversion and the RURI is a DNS SRV record and sometimes one of the servers times out and opensips auto tries the second server from the SRV records but without the diversion header in the new message Is this proper/expected behavior or a bug?

Re: [OpenSIPS-Users] Need ideas to tamper with CSeq

Cinthia, The CSeq needs to be changed before it hits opensips. So you could do something where if the cancel is not matched, lookup the right cseq using a memcache lookup (that you stored during the accepted invite) and send it out to it self with the modified CSeq. I wouldn't be surprised if this

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

ram("mi_fifo", "fifo_name", "/tmp/opensips_fifo") > " > But error is still occured :-( > My "acc table" has "cdr_id filed", not the "cdr field" > additionally, i have "Cdrs" table separately. > Both 2 tables abov

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

4.122/cp/tools/system/domains/apply_changes.php > Cannot connect to OpenSIPS Server via Management Interface ()" > I come to /cp/tools/system/domains but find no "apply_changes.php" file :-( > Please help me to find out how i can connect to Opensips Server and making > chang

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

nnect to my Opensips server on > 5060 port. I remember that i have not configured CP listening port before, it > is default :-( > (CP & Opensips server were installed in the same machine) > Checking these files "db.inc.php" ; local.inc.php; globals.php  in the CP >

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

ta found" > (While checking by using opensipsctl is ok) >  opensipsctl online > database engine 'MYSQL' loaded > Control engine 'FIFO' loaded > 1001 > > Please tell me how to check my CP and its connection to Opensips server? > Thanks ! > > > > &

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

s and Best Regards. > > > > Date: Mon, 21 Mar 2011 08:45:03 -0700 > From: Dave Singer > Subject: Re: [OpenSIPS-Users] Opensips Control Pannel does not take >        effect into Opensips Server > To: OpenSIPS users mailling list > Message-ID: >         > Content-Type:

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

Duong, Some modules, like drouting, you have to click a reload module button in the control pannel. However, I think your problem is that you haven't implemented the needed stuff in opensips.cfg script. CP only provides an interface to managing/accessing the databases tables. You still have to imp

Re: [OpenSIPS-Users] Asynchronous DB queries in OpenSIPS 1.x

Vlad, Would it be possible extend DB_VITRUAL kind of a combination of your two options and Brett's suggestion. I'm just guessing how things might work and possibilities. So for what it is worth... Since it is a middle layer, it could either use the async capabilities of the native driver or put it

Re: [OpenSIPS-Users] t_relay(tcp:)

Jeff, I like using the core rewritable variables to control things. They allow you to easily change just one part of the request and many times the equivalent function does not accept variables as arguments. $rP RURI protocol: http://www.opensips.org/Resources/DocsCoreVar16#toc65 $fs Forced socke

Re: [OpenSIPS-Users] Stopping a T.38 reinvite with Opensips loose_route()

JP, The two end points have to agree on an RTP protocol. You may look at the SDP in the initial INVITEs to see if T.38 is offered and if it is, remove it. Also if the re-INVITE contains G711 along with the T.38 you should be able to just remove the T.38. That should accomplish the same effect as t

Re: [OpenSIPS-Users] How much A records are retained in IP comparison

Igor, I'm pretty sure that won't work at all. You would need to use the transformation ip.resolve. I don't know if it returns multiple IPs but if it does you would want to assign it to an avp that can handle multiple. An example to try: $var(fqdn) = "some.server.com"; # Not sure if you can do a tr

Re: [OpenSIPS-Users] route based on previous load_balance dialog

stefano, You might look at the dialog module. Check if what you need is stored in the db for a dialog. You would need to have the db mode set to realtime. The nice thing if you can use dialog module is that it is built in to write the info to the db and clean it up. Otherwise you will need to writ

Re: [OpenSIPS-Users] running opensips under another user

w it seems that pid file isn’t erased when openSIPS failed to start. > > > > > > > > *From:* users-boun...@lists.opensips.org [mailto: > users-boun...@lists.opensips.org] *On Behalf Of *Dave Singer > *Sent:* Friday, February 25, 2011 4:14 AM > *To:* OpenSIPS users maillin

Re: [OpenSIPS-Users] running opensips under another user

Anton, Have you checked the selinux logs? Search the list about selinux. A couple months back I posted what I ran into and how to solve it fairly simply. Dave. On Thu, Feb 24, 2011 at 6:23 AM, Duane Larson wrote: > And you're positive that the /var/run/opensips directory is owned by the > opensi

Re: [OpenSIPS-Users] {ip.resolve} script transformation trouble

ion. > Am I correct? And so, from which version "ip" is supported? > Or may be is there another method to get resolved ip by string? > > Igor. > > > On Thu, Feb 24, 2011 at 11:06 PM, Dave Singer > wrote: > >> Igor, >> >> Very close. Notice

Re: [OpenSIPS-Users] {ip.resolve} script transformation trouble

Igor, Very close. Notice the ( between $ and var and ) after var name: $var(fs_addr) = $(var(fs_addr_s){ip.resolve}{ip.ntop}); see syntax near top of http://www.opensips.org/Resources/DocsCoreVar16 Dave On Thu, Feb 24, 2011 at 10:36 AM, Igor Solovyov wrote: > Hi All, > > I try to execute in

Re: [OpenSIPS-Users] pseudo-variable problem with increasing cps

Ronald, The only time I've seen it be null in failure route is if there was no reply received, you might add a check to see if it was a local timeout: if ( t_local_replied("all") ) { xlog("did not get any response"); } else { xlog("$(ci): $C(rx)failure route: $(rs) $(rr)$C(xx)\n"); } Dave

Re: [OpenSIPS-Users] Modify a URI

Or just: $rU = "12345#" + $rU; Dave On Wed, Feb 23, 2011 at 10:36 AM, Max Mühlbronner wrote: > Hello, > > > > if you just want to add a prefix, you could use prefix() which is very easy > to use. J > > > > Regards > > > > Max M. > > > > *Von:* users-boun...@lists.opensips.org [mailto: > users

[OpenSIPS-Users] Dialog module app callback hooks not saved to DB

I tested setting up acc module to use the cdr_flag with dialog module. Works nicely until opensips is restarted while there is an open call. After an opensips restart, calls that were started before the restart do not get an entry in the DB. Not even an old style BYE record. I verified that the dia

Re: [OpenSIPS-Users] non-transactional 4XX messages

Jason, That is very strange behavior. If there is no matching transaction I think there is no way to catch the message as reply and failure routes are triggered by tm module. I'm not sure since I haven't setup a stateless config. I'm curious too if there is something that can be done here. Dave

Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

'm working on Red Hat; have found the same problem > on google > (http://lists.opensips.org/pipermail/users/2010-November/015448.html) to > another person and seems that he never solve the situation. Any help will be > appreciated. > > 2011/2/18 Dave Singer >> >

Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

t; 2011/2/17 Dave Singer > >> Toyima, >> >> Have you configured it to connect to your database? >> > > the file opensipsctlrc is already configured to connect to my database > where i'm storing domains, locations, subscribers, etc etc...the DB_PATH > para

Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

:50 AM, Toyima Dias wrote: > > Hello Dave, > > 2011/2/17 Dave Singer >> >> Toyima, >> >> On Wed, Feb 16, 2011 at 5:55 AM, Toyima Dias wrote: >> > >> > >> > 2011/2/16 Dave Singer >> >> >> >> Toyima, >> &g

Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

Toyima, On Wed, Feb 16, 2011 at 5:55 AM, Toyima Dias wrote: > > > 2011/2/16 Dave Singer >> >> Toyima, >> This is the right list. You just have to take into account that no one >> is getting paid to answer your questions in the list. And sometimes >> th

Re: [OpenSIPS-Users] Interproxy Authentication

Juri, If you are looking to get opensips1 itself to respond to a proxy_challange of opensips2 your looking at a headache. Any way you go you'll have to change the way things work to make a special case for opensips1 on opensips2. My suggestion would be to, where you are about to do the proxy_autho

Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

Toyima, This is the right list. You just have to take into account that no one is getting paid to answer your questions in the list. And sometimes that means waiting a couple days. Best tactic I've seen to keep putting it in front of people is to respond to your own thread with a little update of w

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

         # send a ACK upstream for cdr purposes. >                sl_send_reply("200","OK"); >                exit; >        } > > } > > Regards > Juri Nysschen > -Original Message- > From: users-boun...@lists.opensips.org > [mailto:us

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

sorts of combinations for $branch but the vales are always > NULL > > > Regards > Juri Nysschen > > -Original Message- > From: users-boun...@lists.opensips.org > [mailto:users-boun...@lists.opensips.org] On Behalf Of Dave Singer > Sent: Monday, February 14, 2011

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

n't choose anything else specifically, will be sent back to the originator. You will want to verify it with packet captures and see how it affects your cdrs. Let me know the outcome. Dave On Mon, Feb 14, 2011 at 11:05 AM, Dave Singer wrote: > I was half expecting that kind of result. >

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

keeps ringing. > The key is finding the reason why the transaction id disappears or at least > be able to put it back when delivering the CANCEL downstream. > > Regards > Juri Nysschen > > -Original Message- > From: users-boun...@lists.opensips.org > [mailto:use

Re: [OpenSIPS-Users] Weird behaviour

t; [mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Vasile > Sent: Saturday, February 12, 2011 9:26 AM > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Weird behaviour > > That's why I dropped the ideea of having numbered usernames. > > On Fe

Re: [OpenSIPS-Users] Basic doubt of sip routing

Toyima, The client does NOT have to use NAPTR or SRV. It can use regular A or CNAM records as well or even just an IP. (unless it is a half baked client) SRV and NAPTR are special DNS records that can be used tell the client what services are available and where and how to connect to them. If your

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

Juri, You say it only happens after a do_routing(). To be clear, you mean that you used do_routing on the invite and not that you already tried do_routing for the cancel earlier in the script. (I'm not sure it would make any difference) The hack to store the destination in a variable is one you w

Re: [OpenSIPS-Users] Weird behaviour

only REGISTER requests from anywhere and > the rest check the source ip. > Great ideea. > > I will implement it as soon as possible. > > Thanks, > Adrian Vasile > y...@opennet.ro > > > On Feb 10, 2011, at 10:41 PM, Dave Singer wrote: > >> Adrian, >> >&g

Re: [OpenSIPS-Users] Weird behaviour

opensips reboots and other situations where the cache is lost or unavailable. Like a memcached server fails. Advantage to using external memcached vs local cache would be that cache would not be cleared on opensips restart. Dave On Thu, Feb 10, 2011 at 11:16 AM, Dave Singer wrote: > I'

Re: [OpenSIPS-Users] Weird behaviour

her than Pike combined with > fail2ban would you advise? > > > And I finally found the culprit. "Auth INVITE": > "When enabled, authorization is required for initial incoming INVITE > requests from the SIP proxy." > On Feb 10, 2011, at 6:57 PM, Dave Singer wro

Re: [OpenSIPS-Users] Weird behaviour

Adrian, There are lots of people out there with servers doing sip scans to see if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then they will either try to send register and/or invites for all sorts of numbers trying to get a hit. Of course the invites are not actual calls so if t

Re: [OpenSIPS-Users] b2b top-hiding and SDP origin line

Jeff, I checked nathelper code in opensips 1.6.4 and looks like adding the "oldip" header to the SDP is not there anymore. So you shouldn't need to modify the source like I had to previously to accomplish complete topology hiding. Also it doesn't really matter if you use rtp_proxy or media proxy c

Re: [OpenSIPS-Users] every user is registered

mato848, The default/sample config has a section for registration. you need to basically use that. just copy it to your script. Look for www_authorize. Registrations are stored with the usrloc module and auth is done by auth and auth_db modules so they need to be configured as well like the sample

Re: [OpenSIPS-Users] Control Panel Dialog config for multiple servers

Cinthia, I'm fairly new to OCP myself, just got it running about a month ago. Since no one with more experience responded, I'll help if I can. Sounds like you have it all setup and working as designed. Looking at my system that I have connected to two servers, it looks like the dialog module secti

Re: [OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6

/opensips.cfg, depending on compile time options. Failing that I would scrub out opensips installed pieces and recompile/reinstall with a different --prefix= Dave On Tue, Feb 8, 2011 at 11:51 AM, Chris Stone wrote: > Dave, > > On Tue, Feb 8, 2011 at 12:02 AM, Dave Singer > wrote: &

Re: [OpenSIPS-Users] b2b top-hiding and SDP origin line

Jeff, I ran into this too. I was not using opensips to do the SIP top hiding but was for the SDP using media_proxy. Media proxy doesn't touch "o" unfortunately. So I used fix_nated_sdp : fix_nated_sdp("8","$Ri"); #the recieved on IP (my IP) That fixed the "o" header but it created a "oldip" or "

Re: [OpenSIPS-Users] Notify Message

Kyle, use t_replicate("sip:"); Look at the tm module docs for a more complete picture. Then modify the message. One branch will go out with the message the way it was before the t_replicate and one will have the changes after the t_replicate. If you need more replicated destinations, use append_br

Re: [OpenSIPS-Users] How to stop script execution?

Anton, My bet is that since the first one was dropped, a transaction was never created and thus the transaction cleanup doesn't happen either. If you want it to it do the automatic cleanup it might work to create a transaction prior to the reply, then use t_reply and finally drop. Dave On Tue, F

Re: [OpenSIPS-Users] How to modify the 100 Trying packet

quot;Server");" but alas... > I've told them this is a no-go for now.  Seems that we would have to modify > the OpenSIPs core to make it happen.  Not worth the time or effort for this > reason. > > > On Tue, Feb 8, 2011 at 3:10 AM, Dave Singer > wrote: >> >

Re: [OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6

On Mon, Feb 7, 2011 at 7:05 PM, Ovidiu Sas wrote: > On Mon, Feb 7, 2011 at 9:14 PM, Chris Stone wrote: >> Sorry all for the last message - too quick on the Send button. >> >> On Mon, Feb 7, 2011 at 6:48 PM, Henk Hesselink >> wrote: >>> Hi Chris, >>> >>> That config should't touch the Contac

Re: [OpenSIPS-Users] trying to rewriteuri to the value of a variable

te: >> that seems to be working, thanks alot ! clearly I need to re-read the docs >> >> On Sun, Feb 6, 2011 at 6:48 PM, Dave Singer >> wrote: >> >>> Isn't >>> $var(my_uri) = "5...@example.com"; >>> $ru = "sip:" + $var(my

Re: [OpenSIPS-Users] How to test if a message is from myself

return(-1); } } On Thu, Feb 3, 2011 at 4:32 AM, Stefano Pisani wrote: > Hi Dave > you could try > > if ($si == $hdr(X-src-ip)){...} > > > Il 03/02/2011 12:59, Bogdan-Andrei Iancu ha scritto: >> >> Hi Dave, >> >> Unfortunately

Re: [OpenSIPS-Users] How to modify the 100 Trying packet

The 100 Trying is sent out to the originating server/user automatically when you send the invite off to the next destination with t_relay and exit in the main routing script. The 100 Trying you receive in the on_reply is just to let you know that the next hop is not dead and it is working on it. Yo

Re: [OpenSIPS-Users] Help with is_present_hf

Tyler, Not sure but I believe custom headers are supposed to start with "X-" and it may be that opensips is rejecting putting it in because it is not a standard SIP header and it does not start with "X-". Turning up the debug would probably show why. Dave On Mon, Feb 7, 2011 at 6:02 AM, Tyler Me

Re: [OpenSIPS-Users] trying to rewriteuri to the value of a variable

Isn't $var(my_uri) = "5...@example.com"; $ru = "sip:" + $var(my_ruri); the same as rewriteuri("sip:5...@example.com"); just that using $ru you can use it just like assigning to other vars like you are doing. On Sun, Feb 6, 2011 at 6:36 PM, Duane Larson wrote: > Avpops > > Sent from Droi

Re: [OpenSIPS-Users] OpenSIPS 1.6 on Ubuntu

Tyler, Just went through the OpenSIPS default script webminar => http://www.opensips.org/html/docs/video/webinar005/ And while the audio at the beginning is bad (and very end), it is only just a little bit and it is because it was coming through a bad connection to the seminar where the webinar wa

Re: [OpenSIPS-Users] OpenSIPS 1.6 on Ubuntu

The best place to start is http://www.opensips.org/ In the left column of the web page there is a section titled Resources with links to many very helpful resources. Your using the mailing list so you probably already have seen them to get here. So. Where are you getting stuck? We need specifics in

Re: [OpenSIPS-Users] How to test if a message is from myself

1 at 4:37 AM, Bogdan-Andrei Iancu wrote: > Hi Dave, > > do :  if (src_ip==myself) {} > > Regards, > Bogdan > > Dave Singer wrote: >> >> Is there any way to check if the source IP/port is one that opensips >> is listening on or one ? something like if (&q

[OpenSIPS-Users] How to test if a message is from myself

Is there any way to check if the source IP/port is one that opensips is listening on or one ? something like if ("sip:$si:$sp" == myself) { ...bla; bla;} Thanks Dave ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailm

Re: [OpenSIPS-Users] Installing OpenSIPS on Red Hat

Toyima, I posted in the list last week on the thread "multiple use_media_proxy() calls" my notes on getting media proxy installed on centos 5.5. To compile opensips just do the yum installs mentioned in that thread, get the opensips source, unpack, compile and install according to docs on install

Re: [OpenSIPS-Users] OpenSIPS handling B2B features

Toyima, Asterisk is a pain for making conference work because of its dependence on the dahdi kernel driver for timing. I like the FreeSwitch implementation. FreeSwitch has a very active irc at irc.freenode.net. Dave On Thu, Jan 27, 2011 at 12:21 AM, Toyima Dias wrote: > > Anca, > > What confere

Re: [OpenSIPS-Users] multiple use_media_proxy() calls

I have 2.4.4 running on centos 5.5. Here are my notes for getting it working using mostly rpms. Hope this helps you out and others too. If anyone wanted to put this in an official doc for installing mediaproxy, that would be cool or point me to where I could do that. :) install stuff for opensips

Re: [OpenSIPS-Users] CDR Accounting

Thanks for that clarification! So just to be sure I'm clear on this. $Ts rounds down ( truncates ) the current second. So using $avp(s:start_time) = $Ts.$Tsm; would give something like "12343253.543233" and always be accurate? Further (standard ACC [without dialog]) you could just put $Tsm in the

Re: [OpenSIPS-Users] Handling messages to more than one final destination

Anca, When using t_replicate, is opensips then expecting a response from the server it is replicated to? I'm guessing not and also therefore will not send retries when/if no response is received. Then on the recording server you can just use tcpdump to save the messages. Further you would need to

Re: [OpenSIPS-Users] CDR Accounting

Be careful when using $Tsm. I read a thread the other day that was just talking about it and that it returns milliseconds since midnight, not epoch. So if that is true you will need to somehow handle calls that cross midnight for duration at least and start/answer/end of call if including the preci

Re: [OpenSIPS-Users] Some strange things with starting opensips

Adjust STARTOPTIONS to include "-p /var/run/opensips.pid" ( replace with the path to where you want the pid file ) There are other options you may want to adjust for starting opensips. Note that some options are overridden if set in the config file like listen address. Here is the list of options

[OpenSIPS-Users] solution to opensips control pannel MI connection using fifo on CentOS 5 - selinux audit2allow

Banged my head for a while with why I couldn't get fifo working for the Control Panel I was getting "sorry -- cannot open write fifo". Hope this can help other and maybe even make it into the docs. Found two problems. 1. Apache process couldn't use /tmp/opensips_fifo because the permissions were

Re: [OpenSIPS-Users] [NEW] Media timeout detection and call termination

It seems VAD would also be a potential problem with NAT and state full packet inspection on firewalls. It would see a better practice for the UA to send keep alive RTP packets. That would solve both dead call detection and potential NAT issues. That would be nice RFC change. In my situation I have

Re: [OpenSIPS-Users] Freeswitch vs Asterisk

We have both asterisk and Freeswitch in production. The primary place where we have * installed is as a pbx for our business customers (where we started doing business and didn't know any better). We are still using * for them for two reasons: migration time and voicemail app I feel is still better

Re: [OpenSIPS-Users] mediaproxy 2.4.2

I replaced the 5060 with 5065, recompiled (make clean; make) and renamed the binary to pcapsipdump5065 so I would know which one I was using. Worked great. Dave On Fri, Dec 3, 2010 at 8:28 AM, Saúl Ibarra Corretgé wrote: > On 12/03/2010 04:50 PM, Iñaki Baz Castillo wrote: > >> 2010/12/3 Saúl Iba

Re: [OpenSIPS-Users] Handle end of a transaction/dialog

catch a BYE request when a established sip session is ending. What > about other cases? > > > > On Fri, 19 Nov 2010 09:27:38 -0800 > Dave Singer wrote: > >> You can use a combination of the following dialog profiles (grouping >> calls) >> functions for this:

Re: [OpenSIPS-Users] Handle end of a transaction/dialog

You can use a combination of the following dialog profiles (grouping calls) functions for this: get_profile_size(profile,[value],size) Check if a prifile is at limit. set_dlg_profile(profile,[value])

Re: [OpenSIPS-Users] New to OpenSIPS

Learning opensips is not like learning a programming language like C, ruby, puthon, etc. It is very important to get an understanding of how sip flows and opensips works (get the big picture) then get into the details of how to manipulate things. Checkout the past webinars on the Webinars page

Re: [OpenSIPS-Users] Timer Based Failover Question

I ran into the same problem with one of our carriers. The way I did it, with advice from bogdan, was to set the fr_inv_timer_avp to 6 sec (so long because of some calls to cell phone systems have long delays) when sending to the particular carrier then in a reply route special to that carrier, rese

Re: [OpenSIPS-Users] Mediaproxy broken

aproxy log compressed from 272MB to 11MB and is at this link<http://viper.wideideas.net/mediaproxy.log.gz> . Thanks for any help. Dave On Wed, Oct 20, 2010 at 11:43 PM, Saúl Ibarra Corretgé wrote: > Hi, > > On 10/20/2010 11:14 PM, Dave Singer wrote: > > I have one server where mediap

[OpenSIPS-Users] opensips coredump on stop/restart

On my production servers, both 1.6.2 and 1.6.3 versions of opensips, almost every time I restart opensips it creates a core dump. Since I'm giving it 2GB shared mem, it takes a little while to write the core to disk and start running again. So a couple questions. 1. Am I giving it more memory than

[OpenSIPS-Users] Mediaproxy broken

I have one server where mediaproxy is having serious problems. In the logs I have a lot of: media-dispatcher[6067]: error: failed to end dialog: 500 Operation failed and a lot of of the following but much less than the above: media-dispatcher[6067]: error: Got `remove' command from OpenSIPS f

Re: [OpenSIPS-Users] any reason why Opensips would replace 503 with 500 ?

dule) > > Regards, > Bogdan > > Dave Singer wrote: > > Julien, > > > > I have been catching it in failure_route and sending it on up with this: > > if (t_check_status("^503$")) { > > t_reply("503", "Ser

Re: [OpenSIPS-Users] any reason why Opensips would replace 503 with 500 ?

Julien, I have been catching it in failure_route and sending it on up with this: if (t_check_status("^503$")) { t_reply("503", "Service Unavailable"); exit; } I think you could use t_reply("503", $(rr)); (note the use of to indicate the reply con

Re: [OpenSIPS-Users] get error with opensip-cp on centos 5

Setting setenforce 0 is only active for the current running session of the server. The problem will be back after the reboot. While that is fine to temporarily do that to see if SELinux is the thing blocking, it is generally very bad to use that to solve the problem and you definitely wouldn't want

Re: [OpenSIPS-Users] Out of mem all of the sudden

I ran into out of memory and adjusted the shared memory opensips uses with the -m option. It tells opensips how much shared memory to use in MB. eg: .../sbin/opensips -m 500 -l eth0 gives it 500MB of shared memory and listen on eth0 I believe by default it uses 100MB if -m is not specified. Dave

Re: [OpenSIPS-Users] Monit error under OpenSIPS 1.6.3 and Opensips-cp 4

In the init script, the config check where it actually runs opensips (after running it with -c option) to check the config is very new and I just realized and fixed that it would sometimes not kill the test opensips after running it. I attached my fixed version in case you are interested. On Wed,

Re: [OpenSIPS-Users] Monit error under OpenSIPS 1.6.3 and Opensips-cp 4

You may not have monit running as a service. Check the current status of monit: monit status It will tell you if it is running. If it is not start it with: /etc/init.d/monit start I primarily use centos and fedora but I think that will work with debian systems too. I think that the default conf

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