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37523 discussions

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

4 6

[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

1 0

[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/

2 1

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

03 Nov '06

Hello, I get syntax error on the following two commands, does anyone have any idea, thanks: avp_db_load("$ru", "i:34/$email_scheme") modparam("tm", "tw_append", "vm_email: P-Email-Address=avp[i:34]") -- Brandon Armstead

2 1

[Serusers] Multiple user registrations using same port number
by Alan 03 Nov '06

03 Nov '06

I am currently running load testing on SER and have a question regarding multiple contacts registered with the same IP and port number but different contact name. Will this work? Does each contact ID have to be defined with a different port number to establish a session? For example: Contact : sip:loadtest0100@10.180.0.71:10140 Contact : sip:loadtest0101@10.180.0.71:10140 Contact : sip:loadtest0102@10.180.0.71:10140 Contact : sip:loadtest0103@10.180.0.71:10140

2 1

[Users] About forking and unregister
by Bill Zhang 03 Nov '06

03 Nov '06

Hi, Anyone know how openser works with following: 1. Forking. Often times, I may have one account register with multiple SIP client, does it support that? I don't seem to make it working. 2. Unregister and register. Nowadays, there are a lot of SIP client will send an Unregister message to the SIP server and after received response, it will send the new registration again, this is because when a client behind the same router reboot multiple times, it might have registration in the SIP server and only the latest one reflect the current registration and all previous registration may not be valid as the router may already close the mapping port, so in RFC3261, it defines client can send a registration with contact head contains "*" and expeiration timer is 0 to unregister the previous registration. It does not seem that OPENSER can handle it. I know SER handles it. 3. Can I specify OPENSER to only send the INVITE (or other SIP request) to the latest AOR for a particular account? Thanks in advance. Bill

1 0

[Users] mediaproxy pulse / jitter and dropout
by Mark Price 03 Nov '06

03 Nov '06

I installed the openser.cfg at http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy (modified to authenticate against my domain, and to use exit instead of break). Calls between two phones on the domain work, but there is a lot of jitter. Well, not exacty jitter, but kind of a pulsing sound in the background of the call. Also, will occasionally drop out, and then come back after a little bit.

1 0

[Serusers] how to search in usrloc?
by sandrinho1976＠libero.it 03 Nov '06

03 Nov '06

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...

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