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37518 discussions

[Users] Any trick to log ACK with siptrace?
by Rodrigo P. Telles 05 Nov '06

05 Nov '06

Hi Folks, Is there any trick to log ACK method using siptrace module? I'm trying with: ... modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) route { if (is_method("INVITE|ACK|BYE")) { setflag(22); }; ... but only INVITE and BYE method has been logged! I'm using OpenSER 1.1.0-notls. Thanks for any help Rodrigo Telles

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[Users] Difference between method and is_method
by Rodrigo P. Telles 05 Nov '06

05 Nov '06

Hi Folks, Today I realised something strange in openser-1.1.0-notls. I'm trying to mach any ACKs to do something (e.g. logging) but its doesn't work at all. I'm using "method" like this: If (method=="ACK") { xlog("L_ALERT","ACK received\n"); }; and I it doesn't work, but... if (is_method("ACK")) { xlog("L_ALERT","ACK received\n"); }; works perfectly! I could not find anything about it in archive mail list neither in doc. Rodrigo P. Telles

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[Users] Cisco-like Radius accounting
by Dimo 05 Nov '06

05 Nov '06

Hi, I have a radius accounting configuration I would like to share, if someone needs. Please can I get your feedback guys if this can be done in a better way. Just share your thoughts if you have done something similar. I need to account to my RADIUS the source IP (of the calling UA), the destination IP (called UA/gw) and the type of the leg - originating or terminating depending on whether the call is from the UA or to the UA. To do this I decided to use the cisco vendor-specific radius AVPs which have similar fields but for h323 calls. In my dictionary file of radiusclient-ng i added this according to the cisco specification: VENDOR Cisco 9 ATTRIBUTE Cisco-AVPair 1 string Cisco ATTRIBUTE H323-remote-address 23 string Cisco ATTRIBUTE H323-call-origin 26 string Cisco and in my openser.cfg i have: modparam("acc", "radius_extra", "Cisco-AVPair=$avp(s:c_src_ip); H323-remote-address=$avp(s:c_dst_ip); H323-call-origin=$avp(s:c_origin); Cisco-AVPair=$avp(s:c_call_id)") Now in my config i fill in the values of the c_src_ip, c_dst_ip, c_origin, and c_call_id avps according to the direction of the call. Here is how i do it for the originating leg, terminating is similar: route[14] { #ACCOUNT ORIGINATING if (is_direction("downstream")) { avp_printf("$avp(s:c_src_ip)", "call-origin-endpt=$si"); avp_printf("$avp(s:c_dst_ip)", "$ru"); avp_subst("$avp(s:c_dst_ip)", "/sip:.*@(.*)/h323-remote-address=\1/g"); avp_subst("$avp(s:c_dst_ip)", "/(.*):.*/\1/g"); avp_printf("$avp(s:c_origin)", "h323-call-origin=originate"); avp_printf("$avp(s:c_call_id)", "call-id=$ci"); } if (is_direction("upstream")) { avp_printf("$avp(s:c_src_ip)", "$ru"); avp_subst("$avp(s:c_src_ip)", "/sip:.*@(.*)/call-origin-endpt=\1/g"); avp_subst("$avp(s:c_src_ip)", "/(.*):.*/\1/g"); avp_printf("$avp(s:c_dst_ip)", "h323-remote-address=$si"); avp_printf("$avp(s:c_origin)", "h323-call-origin=originate"); avp_printf("$avp(s:c_call_id)", "call-id=$ci"); } setflag(2); } Now as you can see i take one of the IP addresses from the request-uri (and strip it with avp_subst), which I am not sure if always contains IP. Is there a better way - i can call this route after rewritehostport or lookup so the destination IP will hopefully be in the r-uri of the message, but maybe there is a better way of getting it? Any input/ideas will be greatly appreciated. Best, Dimo

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[Users] VoN Berlin & OpenSER Summit
by Daniel-Constantin Mierla 05 Nov '06

05 Nov '06

Hello everybody, I will see many of you at Berlin, next week. The summit has a BoF session on Tuesday, 15:45-17:00 and continues next days with morning and afternoon sessions: see http://www.openser.org for more. Anyone who wants to get in contact before or to have a private meeting, the can find us at booth 321. All participants registered before November 1 got full VON conference pass (valid for the three days, November 6-8, for exhibition and conference, including special events, like party, etc..). Cheers, Daniel

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[Serusers] LCS+SER+ASTERISK
by Felipe Louback 04 Nov '06

04 Nov '06

Howdy all! I am trying to setup the LCS communicating with SER and ASTERISK in the company I work for. I read the following tutorial: http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Int… But there the author points the necessity of 2 LCSs, but I just couldnt grasp why the need of a second LCS. Could anyone who has already setup that give me a hint of why it is necessary a second LCS? I thought I could route LCS to communicated directly with SER. I tried to use only one LCS(one that uses TLS) but it didnt communicate with SER. Thanks in advance, Felipe Louback -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "Come to me, all you who are weary and burdened, and I will give you rest." Matthews 11:28

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RE: [Serusers] how to search in usrloc?
by SER LIST 04 Nov '06

04 Nov '06

You do not need to check usrloc t to implement what you want. Define a timer and when you forward the INVITE to the PBX, you will automatically jump to the failure_route[] if you do not receive a response from the PBX (PBX down!) and you can put your logic there. >From: "sandrinho1976(a)libero.it" <sandrinho1976(a)libero.it> >To: "serusers" <serusers(a)lists.iptel.org> >Subject: [Serusers] how to search in usrloc? >Date: Fri, 3 Nov 2006 01:13:24 +0100 > >Hi everyone. > >I'm facing this problem when configuring a Ser as a proxy and a registrar. > >I have the Ser and a IP pbx: the pbx registers itself on the proxy as >pbx@<pbx_ip> (Contact header). >I had to manage INVITEs messages for incoming call that are going >towards the Ser, sent by UACs in the network. > >INVITEs are coming in the form <telephone_number>@<ser_ip> (Req-URI): >Ser had to decide if <telephone_number> is managed by the IP pbx (it has >this information) and eventually, do a rewrite uri to ><telephone_number>@<pbx_ip> and then forward the message to pbx. > >So far, it's all ok. > >Unfortunately, i have to handle also this situation: the ip pbx is down. >At the present, the INVITE is forwarded towards the pbx, the ip pbx does >not respond and after 10 seconds the requesting UACs sends a CANCEL. >This latency is not acceptable. >I'm looking for a way to implement this: >after the rewrite uri to <telephone_number>@<pbx_ip>, look if I have a >contact in usrloc with <pbx_ip> (i.e. the pbx is still registered): if >not, send immediately a response 4xx to UAC. >The problem is that i'm not able to access the usrloc db from the >routing script... any hint? > >Thanks a lot... > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Not only does Windows Live OneCare provide all-in-one PC care to keep your computer protected and well-maintained, but it also makes creating backup files a breeze. Try it today! http://ideas.live.com/programpage.aspx?versionid=b2456790-90e6-4d28-9219-5d…

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Re: [serusers]: trusted table(permissions module)
by Andrey Kuprianov 04 Nov '06

04 Nov '06

The one you attached, didnt have a semicolon. Anyway, try using a fresh copy of a ready made ser.cfg. Bests, Andrey. On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > Yup I always do the same, Might be copy paste error. Its having semicolon & working fine here. > > Regards > Kamal Mann > > -----Original Message----- > From: Andrey Kuprianov [mailto:andrey.kouprianov@gmail.com] > Sent: Thursday, November 02, 2006 10:34 AM > To: serusers(a)iptel.org > Subject: Re: [serusers]: trusted table(permissions module) > > Hi Kamal, > > Whenever you modified your ser.cfg did you restart SER after? Did you > check your ser.cfg for errors using "ser -c" command? I found some > errors in your ser.cfg code. For instance, inside the main route > block, your "if", which processes INVITEs, is missing a semicolon at > the end of the block. > > if (method=="INVITE") > { > # sl_send_reply("404", "INVITE "); > route(3); > break; > } <----- missing semicolon here > > > If you are comfortable with your current ser.cfg, that's ok, but I'd > suggest you start with a fresh one, make your trusted table work, and > then add a "REFER" processing, like in your currect ser.cfg. Just > download one fresh ser.cfg from iptel.org site and add your > allow_trusted() "if" clause there. > > Regards, > > Andrey. > > > On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > Hi > > Please find ser.cfg enclosed. > > > > Trusted table entry: > > +---------------+-------+--------------+ > > | src_ip | proto | from_pattern | > > +---------------+-------+--------------+ > > | 10.25.119.156 | any | ^sip:.*$ | > > +---------------+-------+--------------+ > > 1 row in set (0.00 sec) > > > > Thanks in anticipation > > Kamal Mann > > -----Original Message----- > > From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kuprianov > > Sent: Wednesday, November 01, 2006 7:03 PM > > To: serusers(a)iptel.org > > Subject: Re: [serusers]: trusted table(permissions module) > > > > Hi Kamal, > > > > I understand your setup. Nevertheless, it will be difficult to figure > > out your problem, w/o your ser.cfg file at hand. Also, send a trusted > > table query result (i.e. that row which contains entry for your > > SIP-AS). > > > > Regards, > > > > Andrey. > > > > On 11/1/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > > Hi All > > > In my scenario I need ser to communicate with my SIP-AS. This SIP-AS is having an application xyz running on it. This XYZ sends an invite to URI "A(a)xcv.de" which is a subscriber in SER and this is in TO header of Invite msg & FROM header contains another URI test(a)xcv.de but this URI (test(a)xcv.de) isn't registered / subscribed to SER. I need SER trust all packets from SIP-AS ip. BUT SER is sending 407 reply of INVITE to XYZ application! IP in trusted table is of SIP-AS on top of which XYZ application is running. > > > > > > Thanks in anticipation > > > Kamal Mann > > > > > > -----Original Message----- > > > From: Maciej Żwirski [mailto:mzwirek@poczta.fm] > > > Sent: Wednesday, November 01, 2006 3:20 PM > > > To: serusers(a)lists.iptel.org > > > Subject: Re: [serusers]: trusted table(permissions module) > > > > > > Kamal.Mann(a)t-systems.com wrote: > > > > > > > Hi All > > > > > > > > I made an entry /"10.25.119.156, any, ^sip:.*$"/ into trusted table so > > > > that any packet sent from 10.25.119.156 would be trusted by SER and > > > > don't ask for its credentials. But SER is replying *407- proxy* auth > > > > needed!! After this I tried with permissions.allow file: > > > > > > > Hi, > > > I had the same issue while interconnecting SER and Asterisk. It turned > > > out to be an issue on Asterisk side (I had the same user added for SER > > > and Asterisk, so the poor thing couldn't authorize either :)) So you > > > could check if the 407 you're getting is from SER or from the remote host. > > > > > > Regards, > > > Maciej Zwirski > > > > > > ---------------------------------------------------------------------- > > > Jestes kierowca? To poczytaj! >>> http://link.interia.pl/f199e > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > >

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[Users] State of T.38 and MediaProxy/RTPProxy
by Bastian Schern 04 Nov '06

04 Nov '06

Hello, does anybody knows the state of T.38 with MediaProxy or RTPProxy/NATHelper? Is it supported in MediaProxy v1.7.2 or the current RTPProxy(NATHelper release? Regards Bastian

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[Users] Syntax Error
by Brandon Armstead 03 Nov '06

03 Nov '06

Hello guys, I have been trying to get openser implemented with SEMS the past day or so although I keep running into problems, any advice on setting up openser with SEMS? (Sip Express Media Server). Although here is a syntax error straight from the documentation: ... modparam("tm","tw_append","append1:Email=avp[i:12];UA=hdr[User-Agent]") modparam("tm","tw_append","append2:body=msg[body]") ... t_write_fifo("voicemail/append1","/tmp/appx_fifo"); ... t_write_unix("logger/append2","/var/run/logger.sock"); Is the syntax for "t_write_unix". Although in my code I have the following: modparam("tm", "tw_append", "vm_email:P-Email-Address=avp[i:34]") Just like the example..... although I get error: 0(57475) ERROR:tm:parse_tw_append: parse error in <vm_email:P-Email-Address=avp[i:34]> around position 25(a) Any idea on what is going on? Thanks, all help is appreciated! -- Brandon Armstead

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[Users] Parallel calls with multiple contacts for 1 user
by Jean-François SMIGIELSKI 03 Nov '06

03 Nov '06

Hello, list! I currently use the debian package of OpenSER (v1.1.0-5), and I see a behaviour I do not explain with RFC's and OpenSER Documentation. I have registered several User-Agent under the same username/domain couple in an OpenSER instance with the registrar module. When I try to call "username@domain", I expected OpenSER to call the contact with the highest q-value, and keep the other contacts into transaction branches for later use (as said in the usrloc module's documentation, in the "lookup" function description). In spite of this, I observed that OpenSER called each contact in a parallel way, and the first UA that responds a 200 is the one that receive the final ACK, other branches are closed. Is it wanted / intentional / normal ? Is the documentation up to date? Whatever will your answer be, thank's a lot! Jean-François Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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