- sr-users - kamailio.org

[Serusers] Forward calls to external proxy that need authentication
by St�phane Muller 25 Apr '05

25 Apr '05

Hi everybody, I would like to know if it is possible to forward a call to an external proxy/pstn-gateway that need authentication ? I have read in the archives that SER itself can't do that, but it is possible by interacting with sipsak... Is it correct ? And if yes, exactly how ? Thank's a bunch for any answer. Steph

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[Serusers] missed caller call back
by Iqbal 25 Apr '05

25 Apr '05

Hi Has anyone ever managed to play around with dialing a missed caller. EG in the UK 1472 will replay the last caller So if in SER we did something like this if {prefix 1471} {route 7;} route[7] { lookup in missed_calls, then make a call to it, if [xx] dialed } Iqbal

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[Serusers] e164 exchanges
by Juha Heinanen 25 Apr '05

25 Apr '05

http://www.e164.org is another root registry aimed at CC '1' while the ENUM LLC launches its 1.e164.arpa trials. As for a list of companies that are participating in any registry, it's something I've yet to see anywhere. i tried last week to call a u.s. 800 number using www.e164.org and freenum.org enum, which both point to tf.voipmich.com. in summary: it was impossible to get connected. looks like their gws are full or broken. a service like that has to work reliably or it is good for nothing. -- juha

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Re: [Serusers] Sorry, Newbie
by Nucleo Wireless LSI 25 Apr '05

25 Apr '05

Hi, tanks for your attention, I've read the SIP introduction and it was very useful. But I have two problems: I've installed SER from the sources available in OnSIP to follow the getting started document, but I keep receiving error messages wen I try to run: [root@parati sbin]# ./ser Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.0.161.52 [10.0.161.52]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.0.161.52 [10.0.161.52]:5060 Aliases: tcp: parati.intranet:5060 tcp: localhost:5060 tcp: parati:5060 tcp: localhost.localdomain:5060 udp: parati.intranet:5060 udp: localhost:5060 udp: parati:5060 udp: localhost.localdomain:5060 stateless - initializing [root@parati sbin]# 0(0) Maxfwd module- initializing textops - initializing 0(0) ERROR: fix_actions: force_send_socket: could not resolve smaug ERROR: error -478 while trying to fix configuration Do you know how to solve this? And were is the ser.cfg? Thank You Again. Fabio Luiz --- the Mafia <thegalbd(a)gmail.com> wrote: > onsip.org <-- just follow this . if u need more help > mail me . > > Mafia > > On 4/17/05, the Mafia <thegalbd(a)gmail.com> wrote: > > yap onsip.org is really good one > > > > Salaque > > > > On 4/16/05, Greger V. Teigre <greger(a)teigre.com> > wrote: > > > http://onsip.org/ has a Getting Started > document. > > > g-) > > > > > > ----- Original Message ----- > > > From: "Nucleo Wireless LSI" > <ntsf_lsi(a)yahoo.com.br> > > > To: <serusers(a)lists.iptel.org> > > > Sent: Friday, April 15, 2005 11:28 PM > > > Subject: [Serusers] Sorry, Newbie > > > > > > > Hi every one, > > > > > > > > My name is Fabio and I work in a lab in my > university, > > > > and I want to use ser as an VoIP server in the > lab so > > > > every computer in our network is a SIP Client > with > > > > softfones making calls from one to another. > > > > > > > > This is an test scenario, I know that the use > of a SIP > > > > server is not necessary to implement such > > > > functionality, but we want to test SIP working > under > > > > this circumstances. > > > > > > > > If anyone knows were to find documentation, > HOW-TO's > > > > and step by step guides, I'd be very pleased. > > > > > > > > Tank You for the attention > > > > Fabio Luiz > > > > Newbie > > > > > > > > > > > > > > > > > > > > > > > > Yahoo! Acesso Grátis - Internet rápida e > grátis. > > > > Instale o discador agora! > http://br.acesso.yahoo.com/ > > > > > > > > > _______________________________________________ > > > > 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 > > > > > > __________________________________________________ Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/

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Re: [Serusers] Sorry, Newbie
by Nucleo Wireless LSI 25 Apr '05

25 Apr '05

Hi, tank you for the information, now the error messages are gone. There's another thing now, wen I run SER with the command /usr/local/sbin/ser the listening information is listed, but wen I look in the processes table SER doesn't shows. I tried help, but found no answer. Sorry for the inconvenience, Tank you again for the info. Fábio Luiz --- "Greger V. Teigre" <greger(a)teigre.com> escreveu: > Replace the ser.cfg file with the hello world > ser.cfg found at > http://onsip.org/ (there was an error in the > default ser.cfg file) > g-) > > Nucleo Wireless LSI wrote: > > Hi, > > > > tanks for your attention, I've read the SIP > > introduction and it was very useful. > > But I have two problems: > > I've installed SER from the sources available in > OnSIP > > to follow the getting started document, but I keep > > receiving error messages wen I try to run: > > [root@parati sbin]# ./ser > > Listening on > > udp: 127.0.0.1 [127.0.0.1]:5060 > > udp: 10.0.161.52 [10.0.161.52]:5060 > > tcp: 127.0.0.1 [127.0.0.1]:5060 > > tcp: 10.0.161.52 [10.0.161.52]:5060 > > Aliases: > > tcp: parati.intranet:5060 > > tcp: localhost:5060 > > tcp: parati:5060 > > tcp: localhost.localdomain:5060 > > udp: parati.intranet:5060 > > udp: localhost:5060 > > udp: parati:5060 > > udp: localhost.localdomain:5060 > > > > stateless - initializing > > [root@parati sbin]# 0(0) Maxfwd module- > initializing > > textops - initializing > > 0(0) ERROR: fix_actions: force_send_socket: could > not > > resolve smaug > > ERROR: error -478 while trying to fix > configuration > > Do you know how to solve this? > > And were is the ser.cfg? > > > > Thank You Again. > > > > Fabio Luiz __________________________________________________ Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/

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[Serusers] Can you Rewrite From in 180/183 Messages
by mark wehberg 25 Apr '05

25 Apr '05

I thought we had this working at one time or another. I can't seem to get it working now. I am trying to change the From in the invite, on the 180/183 messages back from the other side, I need to change the From tag back to its original value which is known. Replace doesn't seem to work, is there another tool? -Mark

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[Serusers] enum and lcr
by Iqbal 25 Apr '05

25 Apr '05

Hi I am having a little difficulty in setting up enum lookups. When a call comes in for a number prefixed with 00, for enum i would really have to replace this with a + prefix since enum requires that, however if this is done then I have problems with the lcr table and my prefixes, since they all are without + sign hence if no enum found I need to strip the plus off again before I pass it on to my pstn gateway. I was attempting to do something like this..would it be correct if (uri=~"^sip:00[0-9]*@") { strip(2); prefix("+"); }; # So now we have the + sign in front if (uri=~"^sip:\+[0-9]*@") { log(1, "+ International number detected\n"); if (!enum_query("e164.arpa") || !enum_query("e164.info")) { strip(1); route (8); # normal pstn }; else { route(1); # e164 result }; Problem is, whatif results were present in both trees, would that be a problem. Also if the e164 message works, I guess then you dont really need to do anything since its just goes out on SIP. Iqbal

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[Serusers] e164 lookups..confused
by Iqbal 25 Apr '05

25 Apr '05

Hi managed to get a few lookups happening, and this was the return I got from the lookup get_record: lookup(1.6.1.9.1.6.2.2.5.3.e164.arpa, 35) failed 4(27898) enum_query(): No NAPTR record found for 1.6.1.9.1.6.2.2.5.3.e164.arpa. 5(27899) udp_rcv_loop: probing packet received from 80.46.109.145 5118 4(27898) enum_query(): order 100, pref 10, flen 1, flags 'u', slen 7, services 'E2U+SIP', rlen 46, regexp '!^\+3522619161(.*)$!sip:\1@siprelay.e164.info!' 4(27898) reg_replace(): pattern: '^\+3522619161(.*)$', replacement: 'sip:\1@siprelay.e164.info', string: '+3522619161' 4(27898) enum_query(): resulted in replacement: 'sip:@siprelay.e164.info' 4(27898) TOI: rewrite_uri: Rewriting Request-URI with 'sip:@siprelay.e164.info' Now is that because there is a error in the enum module in ser , although I have managed to get other lookups working, the reg_replace seems to be okay anyone see if theres an error, I am querying the dns manually also, to see if there is error in record Iqbal

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[Serusers] transfer call to asterisk voicemail when UA is not accepting call
by Kamran Ahmad 25 Apr '05

25 Apr '05

hello Is this posible to CHECK if user is not accepting call. then transfer this call to asterisk voicemail. Kamran __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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RE: [Serusers] HELP!! Converting 7 digits to e164
by Matt Schulte 25 Apr '05

25 Apr '05

Does anyone have any suggestions? Please help, thanks. At this point we're learning C to write it into a module but aren't having much luck.. :/ -----Original Message----- From: Matt Schulte Sent: Saturday, April 23, 2005 10:47 AM To: Matt Schulte; serusers(a)lists.iptel.org Subject: RE: [Serusers] HELP!! Converting 7 digits to e164 Ok, duh there is a strip_tail command. I'm confusing myself now, that still doesn't help me accomplish this :) What I really need is a way to subst(replace) with an avp alias I guess? I don't see any way to do this though. -----Original Message----- From: Matt Schulte Sent: Saturday, April 23, 2005 10:32 AM To: serusers(a)lists.iptel.org Subject: [Serusers] HELP!! Converting 7 digits to e164 Importance: High All, we're still in a grind here we're trying to eliminate all external scripts, the last one we have converts 7 digits to e164 (eg 3212222 to +13143212222). The problem I'm having is I can't figure out a way to convert this inside SER. What needs to happen is Ser needs to take the area code from the user,eg 314 and convert the RURI to that persons areacode+phone number. If there was some kind of strip_tail command I would be in business. All of our users are based on e164, no names or anything. An example of what I mean: From: +13145281234 to/uri: 3212222 Script would take the "314" from the "from" and add it to the uri 3212222 ending up with 3143212222, from there I would simply prefix "+1" ending with +13143212222. The problem is getting ONLY the 314 out of the "from". Any thoughts? SER is bombing on us, the only lead is that we are using external cmds and we "should" get rid of them. Someone please help!! Matt _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] using "s inside replace
by mark wehberg 25 Apr '05

25 Apr '05

Hello, I want to replace the Caller-ID in quotes in the From Tag. Is there a switch that I can use in front of the " so that the SER treats it as another character and not the end of the first " ? replace("From: "xxxxxxxxxx"", "From: "yyyyyyyyyy""); -Mark

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[Serusers] Re: 2 Redundant Mysql DBs
by Helge Waastad 25 Apr '05

25 Apr '05

Hi, you can use a two-server(or more) mysql cluster (and a management server) and add a loadbalancere (LVS or linux-ha) layer to have a redundant system. If you have more servers to play around with, you can install the mysql API's and the ndbd's on different servers (actually quite fun to build ;-)) I also presume you have two or more SER's to handle your traffic, then you have to make sure only one SER writes location information to the cluster at a time. Replication can be stored in memory If you have a storage crash, you can rebuild the database from the other storage nodes. br hw Hi Guys, Is there a way to make 2 redundant mysql banks, I mean, making SER writing all the data to 2 different DB Servers ? I want to make my data redudant, just in case one of the Mysql Crashs. Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins <at> mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

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[Serusers] e164 exchanges
by Iqbal 25 Apr '05

25 Apr '05

Hi Does anyone know what VoIP companies are part of which exchange, is there a list out there somewhere, except e164.info I cant seem to find. Iqbal

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[Serusers] 2 Redundant Mysql DBs
by Felipe Martins 25 Apr '05

25 Apr '05

Thanks, That is just what I needed, I'm gonna read all the documentation and try to implement it, thansk a lot the entire List for the information and fast responding. Best Regards []'s -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

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[Serusers] 2 Redundant Mysql DBs
by Felipe Martins 25 Apr '05

25 Apr '05

Hi Guys, Is there a way to make 2 redundant mysql banks, I mean, making SER writing all the data to 2 different DB Servers ? I want to make my data redudant, just in case one of the Mysql Crashs. Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

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[Serusers] TLS: status, CVS, updates ...
by Cesc Santasusana 25 Apr '05

25 Apr '05

Hi, I have tested the free-tls code. It compiles and it works. I've tried with the minisip user agent connecting over TLS, and no major problems. Also, the interconnection of SER proxies (t_relay_to_tls between proxies, to force tls) also works. It seems to me as a rather functional piece of code, much wanted by the community, though it has not made its way through into the CVS. Why? I think it is time. This said, it also has some things that need to be fixed. I recently sent an email to peter and ser-dev ... but just got silence back. Anyway, here's the email, for those who came up with the same problems: sessioin resuming not supported (but unconfigured) by ser, and no peer certificate verification. Cesc >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >I found what i think it is a bug. I was testing with minisip, which supports TLS completely on the client side (even client certs, >incoming connections, etc). It would create the initial connection ok to SER. After 2 minutes, SER shuts down the socket. So far >so good. When minisip tries to register, it tries to create a new SSL connection, and as it supports session resuming, it would try >to resume the previous session. But SER does not support it ... and here is the bug. > >To fix it ... as simple as calling > SSL_CTX_set_session_cache_mode( ssl_ctx, SSL_SESS_CACHE_OFF ); > >This turns the cache off ... and when ssl receives a session resume request, it sends back a message indicating it is not possible, >the client then starts the handshake from scratch. >Another solution is to implement session catching ... but this may be too resource consuming in big servers or in embedded >systems ... so maybe better just to not support it by default ... maybe implement an option to turn it on at will. > >Another thing ... the verification of the certificates ... it is turned off. It should be turned on i think. > /* Set verification procedure > * The verification can be made null with SSL_VERIFY_NONE, or > * at least easier with SSL_VERIFY_CLIENT_ONCE instead of SSL_VERIFY_FAIL_IF_NO_PEER_CERT. > * For extra control, instead of 0, we can specify a callback function: > * int (*verify_callback)(int, X509_STORE_CTX *) > * Also, depth 2 may be not enough in some scenarios ... though no need > * to increase it much further */ > SSL_CTX_set_verify( _ctx, SSL_VERIFY_PEER | SSL_VERIFY_FAIL_IF_NO_PEER_CERT, 0); > SSL_CTX_set_verify_depth( _ctx, 2); > > Unclassified

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[Serusers] Call routing
by Edgardo O. Gonzales II 25 Apr '05

25 Apr '05

Any feedback from ser gurus.. ======================= Hi! I was able to install and make a call successfully. However, Im using number, for example 2222 as my login name and that will serve as my termination / extension number also. How can I define a specific or corresponding extension number to a specific user. for example, I use somebody as login name and I set 1111 as his extension number. With this information, I should able to login / register to my gatekeeper and using somebody as login name and my friends can call me using my 1111 voip number. Please help me if there something I need to modify on my ser.cfg Also I want to enable accounting. Please help.. thanks, ed

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[Serusers] Free TLS Implementation
by Nguyen Hong Tien 25 Apr '05

25 Apr '05

Dear Peter, I want trying TLS, Please sharing with me your TLS's source code. Thank you Nguyen Hong Tien

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[Serusers] Can SER location module work with Berkeley DB ?
by PHAN, Quang-Minh (Docteur Ordinateur) 25 Apr '05

25 Apr '05

Hi List, Is there a way to use SER with Berkeley DB for location module? I heard that Berkeley DB is ideal for location service and is better than MySQL when the location table is updated very frequently. Regards, Minh

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[Serusers] LCR-module extension for GW prefixes
by Andreas Granig 25 Apr '05

25 Apr '05

Hi all, I've extended Juha's LCR module to support prefixes for gateways. We use it for prefix-based setting of the bearer capability on our gateways. Note that the attached patch is against Juha's backport for rel_0_9_0 at http://tutpro.com/tmp/lcr-0.9.0.tgz because we've no CVS-HEAD running here. It works as follows: In the new table gw_cap you can define capabilities and their corresponding prefix, like: mysql root@ser> select * from gw_cap; +--------+-----------+------------+----------------------+ | cap_id | cap_value | cap_prefix | cap_description | +--------+-----------+------------+----------------------+ | 1 | 1 | | default phone trunk | | 2 | 1 | 01 | prefixed phone trunk | | 3 | 2 | 02 | prefixed fax trunk | +--------+-----------+------------+----------------------+ 3 rows in set (0.00 sec) In this case the cap_value of "1" means phone-capabilites, "2" means fax-capabilities. The table gw_grp is extended with an additional column cap_id. mysql root@ser> select * from gw_grp; +--------+-------------------------+--------+ | grp_id | grp_name | cap_id | +--------+-------------------------+--------+ | 1 | priority_1_phone | 1 | | 2 | priority_1_phone_prefix | 2 | | 3 | priority_1_fax_prefix | 3 | | 4 | priority_2_phone | 1 | | 5 | priority_2_phone_prefix | 2 | | 6 | priority_3_phone_prefix | 2 | +--------+-------------------------+--------+ 6 rows in set (0.00 sec) In ser.cfg you can then load the gateways for example like this if you use different gateways/trunks for phone and fax: if( /* phone call */) { if(!load_gws("1")) { sl_send_reply("500", "Internal server error, unable to load phone gateways"); break; } } else /* fax call */ { if(!load_gws("2")) { sl_send_reply("500", "Internal server error, unable to load fax gateways"); break; } } The usage of next_gw() is as usual. Also note that the gateway selection algorithm changes when you use load_gws("x") instead of load_gws(): The latter always selects the gateways with highest priority first. The new one uses the priority value as weight, so also gateways with lower priority are chosen. Hit ratio for 5000 rounds is as follows: priority-1-gw-1: 35% priority-2-gw-1: 14% priority-2-gw-3: 14% priority-2-gw-2: 14% priority-2-gw-4: 14% priority-3-gw-1: 9% Looking forward for comments/fixes/improvements, Andy

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[Serusers] How to setup Net Messenger with Ser
by tcchan 25 Apr '05

25 Apr '05

I downloaded the net messeger 5.0 linked in the ser web page. After installation, I wonder how to configure the messeger to get it work with SER. What do I used in the Option/connection, do I use SOCKS version4 or HTTP. How do I get the messeger to register with the proxy... Also which version of Messeger is tested woth SER. Thank for any help available.

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[Serusers] Call routing
by Edgardo O. Gonzales II 25 Apr '05

25 Apr '05

Hi! I was able to install and make a call successfully. However, Im using number, for example 2222 as my login name and that will serve as my termination / extension number also. How can I define a specific or corresponding extension number to a specific user. for example, I use somebody as login name and I set 1111 as his extension number. With this information, I should able to login / register to my gatekeeper and using somebody as login name and my friends can call me using my 1111 voip number. Please help me if there something I need to modify on my ser.cfg Also I want to enable accounting. Please help.. thanks, ed

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[Serusers] X-Lite Login Failed!
by IpTel List 24 Apr '05

24 Apr '05

Hi! I've set up SER 0.8.14 with mysql auth support. But now I can't login from X-Lite ... What's wrong with it? Any solutions? X-Lite settings: Display Name: 811223344 Username: 811223344 Authorization User: 811223344 Password: <pass for user> Domain/Realm: 192.168.1.1 SIP Proxy: 192.168.1.1 X-Lite log file is the following ... SEND TIME: 56703576 SEND >> 192.168.1.1:5060 REGISTER sip:192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.15:5060;rport;branch=z9hG4bK65689512B2CE11D995890010A72A9DE7 From: Alex <sip:811223344@192.168.1.1>;tag=3691525702 To: Alex <sip:811223344@192.168.1.1> Contact: "Alex" <sip:811223344@192.168.1.15:5060> Call-ID: 656894F4B2CE11D995890010A72A9DE7(a)192.168.1.1 CSeq: 30200 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 RECEIVE TIME: 56703582 RECEIVE << 192.168.1.1:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.15:5060;rport=5060;branch=z9hG4bK65689512B2CE11D995890010A72A9DE7 From: Alex <sip:811223344@192.168.1.1>;tag=3691525702 To: Alex <sip:811223344@192.168.1.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6c39 Call-ID: 656894F4B2CE11D995890010A72A9DE7(a)192.168.1.1 CSeq: 30200 REGISTER WWW-Authenticate: Digest realm="192.168.1.1", nonce="4269290d2de9736ea10d9bf16de4f02f93a01d18" Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.1:5060 "Noisy feedback tells: pid=456 req_src_ip=192.168.1.15 req_src_port=5060 in_uri=sip:192.168.1.1 out_uri=sip:192.168.1.1 via_cnt==1" SEND TIME: 56703593 SEND >> 192.168.1.1:5060 REGISTER sip:192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.15:5060;rport;branch=z9hG4bK65689514B2CE11D995890010A72A9DE7 From: Alex <sip:811223344@192.168.1.1>;tag=3691525702 To: Alex <sip:811223344@192.168.1.1> Contact: "Alex" <sip:811223344@192.168.1.15:5060> Call-ID: 656894F4B2CE11D995890010A72A9DE7(a)192.168.1.1 CSeq: 30201 REGISTER Expires: 1800 Authorization: Digest username="811223344",realm="192.168.1.1",nonce="4269290d2de9736ea10d9bf16de4f02f93a01d18",response="9e2b5978443135f37b2054ada6328fbe",uri="sip:192.168.1.1" Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 RECEIVE TIME: 56703602 RECEIVE << 192.168.1.1:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.15:5060;rport=5060;branch=z9hG4bK65689514B2CE11D995890010A72A9DE7 From: Alex <sip:811223344@192.168.1.1>;tag=3691525702 To: Alex <sip:811223344@192.168.1.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.cda8 Call-ID: 656894F4B2CE11D995890010A72A9DE7(a)192.168.1.1 CSeq: 30201 REGISTER WWW-Authenticate: Digest realm="192.168.1.1", nonce="4269290d2de9736ea10d9bf16de4f02f93a01d18" Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.1:5060 "Noisy feedback tells: pid=462 req_src_ip=192.168.1.15 req_src_port=5060 in_uri=sip:192.168.1.1 out_uri=sip:192.168.1.1 via_cnt==1"

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[Serusers] serweb click 2 dial authorization
by Sumeet Gupta 24 Apr '05

24 Apr '05

Hello, I was wondering how to get serweb to authorize its click 2 dial invite requests via proxy authorize. Or if it's possible to tell SER to NOT authorize serweb's requests. I tried this by telling SER to allow anything from the IP of the server where serweb is running. serweb talks to SER through the fifo so it makes sense that allowing the IP wouldn't work. I also tried to allow anything from the "controller" URI. Has anyone got click 2 dial working with authorization turned on in SER? I am using both stable releases of SER and SERweb. Thanks, SG _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Serusers] Serweb Can't login
by Clay Bryan 24 Apr '05

24 Apr '05

Hi, I am having a problem logging into the web pages. I do have a user in the database. I am rechecking all of my configurations right now. Just wondering if anyone can point me to the right place.. Thanks in advance.. Wercs Communications Clay Bryan Network Administrator Wercs Communications 400 East First Casper, WY 82601 <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe r%2C+WY++82601&country=us> CBryan(a)wercs.com <mailto:CBryan@wercs.com> tel: fax: mobile: 307-233-8359 307-233-8701 307-258-7371 Add me to your address book... <https://www.plaxo.com/add_me?u=21475170292&v0=646557&k0=282754785> Want a signature like this? <http://www.plaxo.com/signature>

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[Serusers] BYE submission by SER in case of mediaproxy timeout or credit limit reached. ?
by Paul van Schagen 24 Apr '05

24 Apr '05

Hello, I wonder if anyone has a solution for having a call being terminated by SER in case a certain credit limit has been reached or in case the connection was lost to either end of a call ? I am testing with mediaproxy and can have it timeout, it will tear down the RTP relaying nevertheless it wont teminate the actual call by means of a BYE request that was setup to the PSTN gateway. So this connection stays open. Is there a way to interface with SER and to have it generate a BYE to the PSTN gateway ? Can I use RADIUS to tear down the call in case of a prepaid credit limit ? Best regards Paul van Schagen

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[Serusers] Ser stateless
by Gin 24 Apr '05

24 Apr '05

Hi, I want to do a ser stateless sip server(because I dont have a big link (64k )), but I have same questions : - Dos it work good with NAT ? - I can to do a billing of the calls ? Gin

2 1

[Serusers] prepaid
by Kamran Ahmad 24 Apr '05

24 Apr '05

hi all i want to make a prepaid solution for ser+radius. 1. authorize request to radius. 2. if true then get time from the response of radius. 3. Pass this time to asterisk Dial application. or is there any other way to do this prepaid billing. waiting for your comments. Kamran __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

Rv: Re: Rv: Re: [Serusers] blocking certain countries
by Nhadie 23 Apr '05

23 Apr '05

Hi Sir, Thanks for your reply. I tried that also, what I did looks like this: if ((uri=~"^(sip:)?00246[0-9]*@([a-z]+\.)?mydomain\.com") || (uri=~"^(sip:)?00246[0-9]*@111\.111\.111\.111")) { if (isflagset(8)) {xlog("L_INFO", "Call to those locations are not allowed\n");}; sl_send_reply("404", "Not Found"); break; } else if ((uri=~"^(sip:)?00[0-9]*@([a-z]+\.)?mydomain\.com") || (uri=~"^(sip:)?00[0-9]*@111\.111\.111\.111")) { if (isflagset(8)) {xlog("L_INFO", "Call to anywhere in the World\n");}; # Strip first to digits (00) strip(2); # Prefix with 12345# prefix("12345#"); rewritehostport("222.222.222.222:5060"); # Remove credentials consume_credentials(); # Route call t_relay_to_udp("222.222.222.222", "5060"); break; But problem there is that I can't call anywhere. Any other problem you can think of sir? regards nhadie --------- Mensagem Original -------- From: Atle Samuelsen <clona(a)camaro.no> To: Nhadie <nhadie(a)cbcpworld.com> Cc: serusers(a)lists.iptel.org Subject: Re: Rv: Re: [Serusers] blocking certain countries Date: 23/04/05 22:54 > > if (uri =~"^003246"){ > sl_send_reply(403,"Go away"); > break; > }; > > * Nhadie <nhadie(a)cbcpworld.com> [050423 16:12]: > > Hi Sir, > > > > I need to block country codes 246, 871-874 and 881. For my customer to dial > > pstn, they simply dial 00 first then country code, etc. > > > > Any idea why my reg exp doesn't work? thank you > > > > regards, > > nhadie > > > > > > --------- Mensagem Original -------- > > From: Marian Dumitru <marian.dumitru(a)voice-sistem.ro> > > To: Nhadie <nhadie(a)cbcpworld.com> > > Cc: serusers(a)lists.iptel.org > > Subject: Re: [Serusers] blocking certain countries > > Date: 23/04/05 18:08 > > > > > > > > Hi Nhadie, > > > > > > Most probably there is something wrong with the regular express and the > > > dialled RURI doesn't match. > > > What RURI suppose to match and doesn't? > > > > > > Best regards, > > > Marian > > > > > > Nhadie wrote: > > > &gt; Hi All, > > > &gt; > > > &gt; I defined something like this on my config, to block pstn calls to > > > &gt; certain countries. Unfortunately it doesn't work. > > > &gt; > > > &gt; if > > ((uri=~&quot;^(sip:)?00(871-874|881|246)[0-9]*@([a-z]+.)?mydomain.com&quot;) > > > &gt; || (uri=~&quot;^(sip:)?00(871-874|881|246)[0-9]*@10.0.2.200&quot;)) { > > > &gt; if (isflagset(8)) {xlog(&quot;L_INFO&quot;, &quot;Call to > > those locations > > > &gt; are not allowedn&quot;);}; > > > &gt; sl_send_reply(&quot;404&quot;, &quot;Not Found&quot;); > > > &gt; break; > > > &gt; > > > &gt; } else if > > ((uri=~&quot;^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com&quot;) || > > > &gt; (uri=~&quot;^(sip:)?00[0-9]*@10.0.2.200&quot;)) { > > > &gt; if (isflagset(8)) {xlog(&quot;L_INFO&quot;, &quot;Call to > > anywhere in the Worldn&quot;);}; > > > &gt; # Strip first to digits (00) > > > &gt; strip(2); > > > &gt; # Prefix with 12345# > > > &gt; prefix(&quot;12345#&quot;); > > > &gt; > > rewritehostport(&quot;123.123.123.123:5060&quot;); > > > &gt; # Remove credentials > > > &gt; consume_credentials(); > > > &gt; # Route call > > > &gt; t_relay_to_udp(&quot;123.123.123.123&quot;, > > &quot;5060&quot;); > > > &gt; break; > > > &gt; } > > > &gt; } else { > > > &gt; ..... > > > &gt; ..... > > > &gt; ..... > > > &gt; ..... > > > &gt; > > > &gt; can anyone tell my error here? thank you in advance > > > &gt; > > > &gt; regards, > > > &gt; nhadie > > > > > > -- > > > Voice System > > > http://www.voice-system.ro > > > > > > > > > > > > > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > >

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RE: [Serusers] HELP!! Converting 7 digits to e164
by Matt Schulte 23 Apr '05

23 Apr '05

Ok, duh there is a strip_tail command. I'm confusing myself now, that still doesn't help me accomplish this :) What I really need is a way to subst(replace) with an avp alias I guess? I don't see any way to do this though. -----Original Message----- From: Matt Schulte Sent: Saturday, April 23, 2005 10:32 AM To: serusers(a)lists.iptel.org Subject: [Serusers] HELP!! Converting 7 digits to e164 Importance: High All, we're still in a grind here we're trying to eliminate all external scripts, the last one we have converts 7 digits to e164 (eg 3212222 to +13143212222). The problem I'm having is I can't figure out a way to convert this inside SER. What needs to happen is Ser needs to take the area code from the user,eg 314 and convert the RURI to that persons areacode+phone number. If there was some kind of strip_tail command I would be in business. All of our users are based on e164, no names or anything. An example of what I mean: From: +13145281234 to/uri: 3212222 Script would take the "314" from the "from" and add it to the uri 3212222 ending up with 3143212222, from there I would simply prefix "+1" ending with +13143212222. The problem is getting ONLY the 314 out of the "from". Any thoughts? SER is bombing on us, the only lead is that we are using external cmds and we "should" get rid of them. Someone please help!! Matt _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] HELP!! Converting 7 digits to e164
by Matt Schulte 23 Apr '05

23 Apr '05

All, we're still in a grind here we're trying to eliminate all external scripts, the last one we have converts 7 digits to e164 (eg 3212222 to +13143212222). The problem I'm having is I can't figure out a way to convert this inside SER. What needs to happen is Ser needs to take the area code from the user,eg 314 and convert the RURI to that persons areacode+phone number. If there was some kind of strip_tail command I would be in business. All of our users are based on e164, no names or anything. An example of what I mean: From: +13145281234 to/uri: 3212222 Script would take the "314" from the "from" and add it to the uri 3212222 ending up with 3143212222, from there I would simply prefix "+1" ending with +13143212222. The problem is getting ONLY the 314 out of the "from". Any thoughts? SER is bombing on us, the only lead is that we are using external cmds and we "should" get rid of them. Someone please help!! Matt

1 0

Rv: Re: [Serusers] blocking certain countries
by Nhadie 23 Apr '05

23 Apr '05

Hi Sir, I need to block country codes 246, 871-874 and 881. For my customer to dial pstn, they simply dial 00 first then country code, etc. Any idea why my reg exp doesn't work? thank you regards, nhadie --------- Mensagem Original -------- From: Marian Dumitru <marian.dumitru(a)voice-sistem.ro> To: Nhadie <nhadie(a)cbcpworld.com> Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] blocking certain countries Date: 23/04/05 18:08 > > Hi Nhadie, > > Most probably there is something wrong with the regular express and the > dialled RURI doesn't match. > What RURI suppose to match and doesn't? > > Best regards, > Marian > > Nhadie wrote: > > Hi All, > > > > I defined something like this on my config, to block pstn calls to > > certain countries. Unfortunately it doesn't work. > > > > if ((uri=~"^(sip:)?00(871-874|881|246)[0-9]*@([a-z]+.)?mydomain.com") > > || (uri=~"^(sip:)?00(871-874|881|246)[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to those locations > > are not allowedn");}; > > sl_send_reply("404", "Not Found"); > > break; > > > > } else if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || > > (uri=~"^(sip:)?00[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to anywhere in the Worldn");}; > > # Strip first to digits (00) > > strip(2); > > # Prefix with 12345# > > prefix("12345#"); > > rewritehostport("123.123.123.123:5060"); > > # Remove credentials > > consume_credentials(); > > # Route call > > t_relay_to_udp("123.123.123.123", "5060"); > > break; > > } > > } else { > > ..... > > ..... > > ..... > > ..... > > > > can anyone tell my error here? thank you in advance > > > > regards, > > nhadie > > -- > Voice System > http://www.voice-system.ro > > > > >

2 1

Rv: Re: [Serusers] blocking certain countries
by Nhadie 23 Apr '05

23 Apr '05

Hi Sir, I need to block country codes 246, 871-874 and 881. For my customer to dial pstn, they simply dial 00 first then country code, etc. Any idea why my reg exp doesn't work? thank you regards, nhadie --------- Mensagem Original -------- From: Marian Dumitru <marian.dumitru(a)voice-sistem.ro> To: Nhadie <nhadie(a)cbcpworld.com> Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] blocking certain countries Date: 23/04/05 18:08 > > Hi Nhadie, > > Most probably there is something wrong with the regular express and the > dialled RURI doesn't match. > What RURI suppose to match and doesn't? > > Best regards, > Marian > > Nhadie wrote: > > Hi All, > > > > I defined something like this on my config, to block pstn calls to > > certain countries. Unfortunately it doesn't work. > > > > if ((uri=~"^(sip:)?00(871-874|881|246)[0-9]*@([a-z]+.)?mydomain.com") > > || (uri=~"^(sip:)?00(871-874|881|246)[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to those locations > > are not allowedn");}; > > sl_send_reply("404", "Not Found"); > > break; > > > > } else if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || > > (uri=~"^(sip:)?00[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to anywhere in the Worldn");}; > > # Strip first to digits (00) > > strip(2); > > # Prefix with 12345# > > prefix("12345#"); > > rewritehostport("123.123.123.123:5060"); > > # Remove credentials > > consume_credentials(); > > # Route call > > t_relay_to_udp("123.123.123.123", "5060"); > > break; > > } > > } else { > > ..... > > ..... > > ..... > > ..... > > > > can anyone tell my error here? thank you in advance > > > > regards, > > nhadie > > -- > Voice System > http://www.voice-system.ro > > > > >

1 0

Rv: Re: [Serusers] blocking certain countries
by Nhadie 23 Apr '05

23 Apr '05

Hi Sir, I need to block country codes 246, 871-874 and 881. For my customer to dial pstn, they simply dial 00 first then country code, etc. Any idea why my reg exp doesn't work? thank you regards, nhadie --------- Mensagem Original -------- From: Marian Dumitru <marian.dumitru(a)voice-sistem.ro> To: Nhadie <nhadie(a)cbcpworld.com> Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] blocking certain countries Date: 23/04/05 18:08 > > Hi Nhadie, > > Most probably there is something wrong with the regular express and the > dialled RURI doesn't match. > What RURI suppose to match and doesn't? > > Best regards, > Marian > > Nhadie wrote: > > Hi All, > > > > I defined something like this on my config, to block pstn calls to > > certain countries. Unfortunately it doesn't work. > > > > if ((uri=~"^(sip:)?00(871-874|881|246)[0-9]*@([a-z]+.)?mydomain.com") > > || (uri=~"^(sip:)?00(871-874|881|246)[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to those locations > > are not allowedn");}; > > sl_send_reply("404", "Not Found"); > > break; > > > > } else if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || > > (uri=~"^(sip:)?00[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to anywhere in the Worldn");}; > > # Strip first to digits (00) > > strip(2); > > # Prefix with 12345# > > prefix("12345#"); > > rewritehostport("123.123.123.123:5060"); > > # Remove credentials > > consume_credentials(); > > # Route call > > t_relay_to_udp("123.123.123.123", "5060"); > > break; > > } > > } else { > > ..... > > ..... > > ..... > > ..... > > > > can anyone tell my error here? thank you in advance > > > > regards, > > nhadie > > -- > Voice System > http://www.voice-system.ro > > > > >

1 0

Rv: Re: [Serusers] blocking certain countries
by Nhadie 23 Apr '05

23 Apr '05

Hi Sir, I need to block country codes 246, 871-874 and 881. For my customer to dial pstn, they simply dial 00 first then country code, etc. Any idea why my reg exp doesn't work? thank you regards, nhadie --------- Mensagem Original -------- From: Marian Dumitru <marian.dumitru(a)voice-sistem.ro> To: Nhadie <nhadie(a)cbcpworld.com> Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] blocking certain countries Date: 23/04/05 18:08 > > Hi Nhadie, > > Most probably there is something wrong with the regular express and the > dialled RURI doesn't match. > What RURI suppose to match and doesn't? > > Best regards, > Marian > > Nhadie wrote: > > Hi All, > > > > I defined something like this on my config, to block pstn calls to > > certain countries. Unfortunately it doesn't work. > > > > if ((uri=~"^(sip:)?00(871-874|881|246)[0-9]*@([a-z]+.)?mydomain.com") > > || (uri=~"^(sip:)?00(871-874|881|246)[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to those locations > > are not allowedn");}; > > sl_send_reply("404", "Not Found"); > > break; > > > > } else if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || > > (uri=~"^(sip:)?00[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to anywhere in the Worldn");}; > > # Strip first to digits (00) > > strip(2); > > # Prefix with 12345# > > prefix("12345#"); > > rewritehostport("123.123.123.123:5060"); > > # Remove credentials > > consume_credentials(); > > # Route call > > t_relay_to_udp("123.123.123.123", "5060"); > > break; > > } > > } else { > > ..... > > ..... > > ..... > > ..... > > > > can anyone tell my error here? thank you in advance > > > > regards, > > nhadie > > -- > Voice System > http://www.voice-system.ro > > > > >

1 0

Rv: Re: [Serusers] blocking certain countries
by Nhadie 23 Apr '05

23 Apr '05

Hi Sir, I need to block country codes 246, 871-874 and 881. For my customer to dial pstn, they simply dial 00 first then country code, etc. Any idea why my reg exp doesn't work? thank you regards, nhadie --------- Mensagem Original -------- From: Marian Dumitru <marian.dumitru(a)voice-sistem.ro> To: Nhadie <nhadie(a)cbcpworld.com> Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] blocking certain countries Date: 23/04/05 18:08 > > Hi Nhadie, > > Most probably there is something wrong with the regular express and the > dialled RURI doesn't match. > What RURI suppose to match and doesn't? > > Best regards, > Marian > > Nhadie wrote: > > Hi All, > > > > I defined something like this on my config, to block pstn calls to > > certain countries. Unfortunately it doesn't work. > > > > if ((uri=~"^(sip:)?00(871-874|881|246)[0-9]*@([a-z]+.)?mydomain.com") > > || (uri=~"^(sip:)?00(871-874|881|246)[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to those locations > > are not allowedn");}; > > sl_send_reply("404", "Not Found"); > > break; > > > > } else if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || > > (uri=~"^(sip:)?00[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to anywhere in the Worldn");}; > > # Strip first to digits (00) > > strip(2); > > # Prefix with 12345# > > prefix("12345#"); > > rewritehostport("123.123.123.123:5060"); > > # Remove credentials > > consume_credentials(); > > # Route call > > t_relay_to_udp("123.123.123.123", "5060"); > > break; > > } > > } else { > > ..... > > ..... > > ..... > > ..... > > > > can anyone tell my error here? thank you in advance > > > > regards, > > nhadie > > -- > Voice System > http://www.voice-system.ro > > > > >

1 0

Rv: Re: [Serusers] blocking certain countries
by Nhadie 23 Apr '05

23 Apr '05

Hi Sir, I need to block country codes 246, 871-874 and 881. For my customer to dial pstn, they simply dial 00 first then country code, etc. Any idea why my reg exp doesn't work? thank you regards, nhadie --------- Mensagem Original -------- From: Marian Dumitru <marian.dumitru(a)voice-sistem.ro> To: Nhadie <nhadie(a)cbcpworld.com> Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] blocking certain countries Date: 23/04/05 18:08 > > Hi Nhadie, > > Most probably there is something wrong with the regular express and the > dialled RURI doesn't match. > What RURI suppose to match and doesn't? > > Best regards, > Marian > > Nhadie wrote: > > Hi All, > > > > I defined something like this on my config, to block pstn calls to > > certain countries. Unfortunately it doesn't work. > > > > if ((uri=~"^(sip:)?00(871-874|881|246)[0-9]*@([a-z]+.)?mydomain.com") > > || (uri=~"^(sip:)?00(871-874|881|246)[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to those locations > > are not allowedn");}; > > sl_send_reply("404", "Not Found"); > > break; > > > > } else if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || > > (uri=~"^(sip:)?00[0-9]*@10.0.2.200")) { > > if (isflagset(8)) {xlog("L_INFO", "Call to anywhere in the Worldn");}; > > # Strip first to digits (00) > > strip(2); > > # Prefix with 12345# > > prefix("12345#"); > > rewritehostport("123.123.123.123:5060"); > > # Remove credentials > > consume_credentials(); > > # Route call > > t_relay_to_udp("123.123.123.123", "5060"); > > break; > > } > > } else { > > ..... > > ..... > > ..... > > ..... > > > > can anyone tell my error here? thank you in advance > > > > regards, > > nhadie > > -- > Voice System > http://www.voice-system.ro > > > > >

1 0

Re: [Serusers] Question regarding *98, *67 etc.
by Dave 23 Apr '05

23 Apr '05

Elena, But from what I can see, 4xx, 5xx and 6xx refer to "errors" as per the RFC. It would be nice if there was another sequence (such as 9xx) for informational purposes only. Thanks --- Elena Ramona Modroiu <ramona(a)voice-system.ro> wrote: > You could you a new reply code from classes 4xx or > 5xx (these two look > more appropriate, although you can use a 6xx reply > too -- see chapter 21 > in RFC3261 about reply codes) with a special reason > phrase (Call > forwarding enabled). Many phones display the reason > phrase and that > should help the caller to understand what happened. > > A nicer way, imo, is to redirect the call to an > announcement server that > plays a confirmation message. > > Ramona > > Dave wrote: > > >I am using avpops to allow services such as > >call-forwarding, dnd, etc. I want to allow the > users > >to be able to enter sequences such as *982223333, > >implying that he wants call forwarding enabled and > the > >number to forward to is 2223333. > > > >Question: Once SER gets the INVITE and stores the > >number to forward in an AVP, what response should > SER > >send back to the UA? It cannot be 200 OK. I would > like > >to send a response that basically displays "Call > >forwarding enabled" and ends the transaction. > > > >Thanks in advance > > > >__________________________________________________ > >Do You Yahoo!? > >Tired of spam? Yahoo! Mail has the best spam > protection around > >http://mail.yahoo.com > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Auth* changes
by Bogdan-Andrei Iancu 23 Apr '05

23 Apr '05

I just committed some changes into 0.9.0 auth* modules - basically it's just a backport from the development branch. Mainly to solve some issues regarding the RPID handling (there were quite some reports about RPID problems in 0.9.0), which was one of the last drawbacks before 0.9.0 release. Interface changes: auth_db: param "load_credentials": list of '|' separated columns to be load from subscriber table at auth time. The values for each column is loaded in a corresponding AVP (string type) with same name as the column; Default value is "rpid". Empty value "" disable this feature. auth: param "rpid_avp": avp name (string) which carry the RPID value - must be synchronized with auth_db if RPID is fetched from DB. Default value is "rpid". Please all 0.9.0 users, give it a try a report any potential problems. bogdan

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[Serusers] Some questions about SER that I can't find a wiki answer to
by Chris St Denis 23 Apr '05

23 Apr '05

I have some questions about SER that I can't find answers to in the wiki. Does SER have T.38 support? Does SER have a module to send MWI notifications? Does the SER nat support replace a STUN server? Should it be used with one? Is stun better to use than it? Can SER handel more than one database type loaded? (Eg mysql and pgsql?) If so, how can you choose which one is used for what? Any (Unix) ODBC database module planned or under development? When routing a call through ser (from PSTN or another sip phone) how can I define the number of rings before the call gets sent to voicemail?

2 1

[Serusers] start voicemail if User is not accepting call
by Kamran Ahmad 23 Apr '05

23 Apr '05

hello i have SER+asterisk for voicemail. there are two cases 1. if UA is not registered at SER it will forward call to asterisk for voicemail. (working) 2. if UA is present at SER but not accepting call in 30 seconds. Ser should rout call to asterisk for voicemail.(this is not working Ser is only sending 486 busy to asterisk) failure_route[1] { if (t_check_status("408|486")){ revert_uri (); rewritehostport ("asterisk:5070"); append_branch(); #forward(asterisk,5070); } } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] blocking certain countries
by Nhadie 23 Apr '05

23 Apr '05

2 1

RE: [Serusers] if routeXX fails..
by Matt Schulte 23 Apr '05

23 Apr '05

Ok, I realized I already assumed your theory :-) This is always returning true, even though the last statement is returning false, the route ultimately is still returning true.. This was just an effort to consolidate code for those of our custs who send by IP (rather than auth). Am I going about this the right way or am I barking up the wrong tree? I just cant seem to get it to work :-( Matt code: route[6] { if lookup("location") { xlog("L_INFO", "LOG: Caught usrloc %mi %rm From <%fu> To <%tu> RURI <%ru>\n"); route(2); break; } if enum_query("blahblah") { xlog("L_INFO", "LOG: Caught ENUM %mi %rm From <%fu> To <%tu> RURI <%ru>\n"); route(2); break; } if (search("(f|From): .*(a)x.x.x.x")) { if (exec_dset("/usr/local/bin/aliases.pl")) { xlog("L_INFO", "LOG: Caught alias %mi %rm From <%fu> To <%tu> RURI <%ru>\n"); route(2); break; } } } -----Original Message----- From: Matt Schulte Sent: Thursday, April 21, 2005 6:03 PM To: marian.dumitru(a)voice-sistem.ro Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] if routeXX fails.. I see, that makes perfect sense :-).. So I just might need to flip around my "if" to get it to work. -----Original Message----- From: Marian Dumitru [mailto:marian.dumitru@voice-sistem.ro] Sent: Thursday, April 21, 2005 5:10 PM To: Matt Schulte Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] if routeXX fails.. Hi Matt, Basically you can do that, but be very careful since the return code of a route is the return code of the *last* command executed inside the route - and break always returns true! Best regards, Marian Matt Schulte wrote: > Is it possible to do this? or is it bad practice, I'm assuming the > "extra" route would have to return an error code to show a > nonpositive? I coulda swore I got this to work awhile back. > > if (!route(6)) { > xlog("L_INFO", "NOTICE: blah <%ru> sending 404\n"); > sl_send_reply("404", "Not Found"); > break; > } else { > # Else route6 ran another route and break > break; > } > -- Voice System http://www.voice-system.ro _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] BYE records for mysql
by Marian Dumitru 23 Apr '05

23 Apr '05

Hi Barry, try something like this: record_route(); # loose-route processing if (loose_route()) { if (method=="BYE") { setflag(1); setflag(2); } t_relay(); break; }; # account completed transactions via syslog setflag(1); setflag(2); Best regards, Marian Barry Murphy wrote: > This is what I have in my route {} > > record_route(); > # loose-route processing > if (loose_route()) { > t_relay(); > break; > }; > > > # account completed transactions via syslog > setflag(1); > setflag(2); > > Barry > > ----- Original Message ----- > From: "Marian Dumitru" <marian.dumitru(a)voice-sistem.ro> > To: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> > Cc: "Barry Murphy" <barry(a)unix.co.nz>; <serusers(a)lists.iptel.org> > Sent: Friday, April 22, 2005 10:07 AM > Subject: Re: [Serusers] BYE records for mysql > > > >>Hi Barry, >> >>Also make sure you do proper record_route() and sequential requests as >>BYEs hits your proxy. >> >>Best regards, >>Marian >> >>Klaus Darilion wrote: >> >>>Make sure that the BYE request hits the setflag(accountingflag) command. >>>Often BYEs are processed by the loose_route block. >>> >>>regards, >>>klaus >>> >>>Barry Murphy wrote: >>> >>> >>>>Hey guys, >>>> >>>>I don't seem to get BYE records from SER being sent to mysql, I get >>>>INVITE >>>>and ACK. Without BYE I can't see who terminated the call, caller or >>>>callee, >>>>and I can't determin the duration of the call. >>>> >>>>Any ideas? >>>> >>>>Thanks >>>>Barry >> >> >>-- >>Voice System >>http://www.voice-system.ro >> > > -- Voice System http://www.voice-system.ro

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RE: [Serusers] Question regarding *98, *67 etc.
by Matt Schulte 23 Apr '05

23 Apr '05

Better yet, does anyone have a Howto for feature codes? That would be neato keen.. I haven't touched avp yet and would like a crash course. I saw the "howto"'s on someones website, it's just the man pages :P Matt -----Original Message----- From: Elena Ramona Modroiu [mailto:ramona@voice-system.ro] Sent: Friday, April 22, 2005 5:44 PM To: Dave Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Question regarding *98, *67 etc. You could you a new reply code from classes 4xx or 5xx (these two look more appropriate, although you can use a 6xx reply too -- see chapter 21 in RFC3261 about reply codes) with a special reason phrase (Call forwarding enabled). Many phones display the reason phrase and that should help the caller to understand what happened. A nicer way, imo, is to redirect the call to an announcement server that plays a confirmation message. Ramona Dave wrote: >I am using avpops to allow services such as >call-forwarding, dnd, etc. I want to allow the users >to be able to enter sequences such as *982223333, >implying that he wants call forwarding enabled and the >number to forward to is 2223333. > >Question: Once SER gets the INVITE and stores the >number to forward in an AVP, what response should SER >send back to the UA? It cannot be 200 OK. I would like >to send a response that basically displays "Call >forwarding enabled" and ends the transaction. > >Thanks in advance > >__________________________________________________ >Do You Yahoo!? >Tired of spam? Yahoo! Mail has the best spam protection around >http://mail.yahoo.com > >_______________________________________________ >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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[Serusers] Confusion about terminology vis-a-vis AVP
by Dave 22 Apr '05

22 Apr '05

Reading AVP documentation is leading to some confusion - AVP is an attribute-value-pair. When we use the terminology "name of an AVP", does it refer to a name (e.g. i:34) that refers to an Attribute-Value-pair OR is the "AVP name" the attribute itself in the Attribute-Value-Pair. Thanks in advance __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Question regarding *98, *67 etc.
by Dave 22 Apr '05

22 Apr '05

I am using avpops to allow services such as call-forwarding, dnd, etc. I want to allow the users to be able to enter sequences such as *982223333, implying that he wants call forwarding enabled and the number to forward to is 2223333. Question: Once SER gets the INVITE and stores the number to forward in an AVP, what response should SER send back to the UA? It cannot be 200 OK. I would like to send a response that basically displays "Call forwarding enabled" and ends the transaction. Thanks in advance __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] AVPops: ERROR:avops:fixup_pushto_avp: bad param 1; expected : $[ruri|hdr_name|..]
by Martin Koenig 22 Apr '05

22 Apr '05

Hello, SER complains about the following error on startup: ERROR:avops:fixup_pushto_avp: bad param 1; expected : $[ruri|hdr_name|..] The corresponding line in ser.cfg is avp_pushto("P-Asserted-Identity/request","s:p-asserted-identity"); Any ideas what is going wrong here? Version is rel_0_9_0. With best regards, Martin

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RE: [Serusers] Authentication & MediaProxy
by info＠beeplove.com 22 Apr '05

22 Apr '05

Hello List: Are you guys taking any authentication measure to allow access on MediaProxy? Thanks, Mohammad Sorry, forgot to add a subject line, :p -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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[Serusers] (no subject)
by info＠beeplove.com 22 Apr '05

22 Apr '05

Hello List: Are you guys taking any authentication measure to allow access on MediaProxy? Thanks, Mohammad -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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FW: [Serusers] calls between UA´s behind same NAT using nathelper/rtpproxy
by jimmy 22 Apr '05

22 Apr '05

-----Original Message----- From: jimmy [mailto:jimmy_huang@uni.com.tw] Sent: Tuesday, March 15, 2005 10:01 AM To: 'rrisco(a)millicom.net.pe' Subject: RE: [Serusers] calls between UA´s behind same NAT using nathelper/rtpproxy Hi Rafael: It seems has two way to meet your requirement (1)use external script http://lists.iptel.org/pipermail/serusers/2005-February/015477.html (2)use avpops module http://www.voice-system.ro/docs/avpops/ Hope it's useful ... Jimmy -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Rafael J. Risco G.V. Sent: Tuesday, March 15, 2005 2:22 AM To: serusers(a)lists.iptel.org Subject: [Serusers] calls between UA´s behind same NAT using nathelper/rtpproxy Hi I would like to know if its possible to comunicate SIP Clients betwen them when they are behind the same nat, I am using Nathelper/RTPProxy... thanks Rafael -- rrgv _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] Bug # 2925
by Jiri Kuthan 22 Apr '05

22 Apr '05

I received questions for more detailed explanation, so here it is: branch value serves as transaction id, which (among others) differentiates retransmission of the same request from new requests. From RFC3261's standpoint, ACK for a 200 (as opposed to a [3-6]xx ACK) is a separate transaction, i.e., a different branch value is adequate. In case people do keep asking: it is really so, and receivers MUST be able to deal with ACK branch different from branch in INVITE. Let's think of a proxy that does not record-route and introduced branch 1234 to "its" Via. ACK coming later directly to UAS includes UAC's branch which is different from proxy's branch. It may or may not be the same as UAC's INVITE branch but it is different from what UAS sees. -jiri At 12:44 AM 4/20/2005, Darryl H. Thomas wrote: >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Tuesday, April 19, 2005 2:23 PM >To: Darryl H. Thomas; serusers(a)lists.iptel.org >Subject: Re: [Serusers] Bug # 2925 > >The bug report is invalid, ignore it. branch=0 in ACK to a 200 >is perfectly valid according to RFC3261. Your termination provider >must be using a buggy devices. > >-jiri > >At 07:04 PM 4/19/2005, Darryl H. Thomas wrote: >>Hello everyone, >> >>I was wondering whether anyone knew what was going on with the >following >>bug: >>http://developer.berlios.de/bugs/?func=detailbug&bug_id=2925&group_id=4 >80 >> >>For those not wanting to follow the links, this is a bug submitted by >>someone back in December wherein ser puts 'branch=0' in the Via header >it >>creates for an ACK to a 200 OK Invite response. >> >>I haven't seen any activity related to this, and it's affecting >>interoperability with one of our termination providers. >> >>0.9.0 users, have you seen the same problem?? >> >>Cheers, >>Darryl >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > >-- >Jiri Kuthan http://iptel.org/~jiri/ -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] ser 0.9.0+rpid
by raid＠gmx.net 22 Apr '05

22 Apr '05

hi list please can someone explain to use ser with rpid where is the best place to insert "append_rpid_hf("<", ">;party=calling;id-type=subscriber;screen=yes;privacy=off");" thanks regards raid -- +++ NEU: GMX DSL_Flatrate! Schon ab 14,99 EUR/Monat! +++ GMX Garantie: Surfen ohne Tempo-Limit! http://www.gmx.net/de/go/dsl

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[Serusers] PSTN => SER => PSTN
by Mik Cheez 22 Apr '05

22 Apr '05

I'm having an issue where my OUT PSTN is not getting a signal that a call has ended. My setup is as follows: IN OUT PSTN ==> SER ==> PSTN If I place a call and the person hangs up before the callee answers, the callee's phone just keeps ringing. I see a CANCEL event on the SER box, but nothing on the OUT PSTN. Here's a sample of my ser.cfg script: ------------------------------------ debug=3 fork=yes log_stderror=yes sip_warning=no check_via=no dns=no rev_dns=no port=5060 #children=4 fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method!="REGISTER") { record_route(); }; xlog("L_INFO", "#INFO# %rm\n"); if (loose_route()) { t_relay(); break; }; setflag(1); if (method=="CANCEL"){ sl_send_reply("487", "Terminated"); break; }; if (method=="INVITE") { if (uri=~"sip:[0-9]+@.*") { xlog("L_INFO", "Original URI: %ru \n"); exec_dset("insert script here"); xlog("L_INFO", "Relay URI = %ru\n"); if (!t_relay()) { sl_reply_error(); break; }; }; }; }

1 1

[Serusers] gw based on username
by Iqbal 22 Apr '05

22 Apr '05

Hi Currently I am using lcr to choose my gateway based upon prefix, now I wanted to choose a gateway based on username. I was thinking that if a user made a call to say 001xyz I could detect its username, see if in usr_prefences (using avp) it had a variable set eg:add_prefix, if so add a prefix of 2 to the number. Then in lcr add a false prefix of 2001 to the table with the new gateway, and at the gateway remove the prefix, before making the call, this way I dont have to ask users to dial any special prefixes. Any better ideas Iqbal

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[Serusers] gw based of username
by Iqbal Gandham 22 Apr '05

22 Apr '05

Hi Currently I am using lcr to choose my gateway based upon prefix, now I wanted to choose a gateway based on username. I was thinking that if a user made a call to say 001xyz I could detect its username, see if in usr_prefences (using avp) it had a variable set eg:add_prefix, if so add a prefix of 2 to the number. Then in lcr add a false prefix of 2001 to the table with the new gateway, and at the gateway remove the prefix, before making the call, this way I dont have to ask users to dial any special prefixes. Any better ideas -- Regards Iqbal Gandham, CEO IPClouds Ltd (http://www.ipclouds.co.uk) Baylis House ,Stoke Poges Lane Slough,Berkshire SL1 3PB V: iqbal(a)sip.ipclouds.co.uk ( call me free !!) M: +44 7866318220 F: +44 (0) 870 7474784 W: http://www.ipclouds.co.uk ------------------------- b5 18 b5 07 1c cd 15 41 ba 23 ad 94 38 30 b8 df -------------------------

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[Serusers] AS 5350 + Ser Problem
by Razvan Nemesiu 22 Apr '05

22 Apr '05

Hi all, I have a Cisco AS 5350 and two SER.On my CISCO I have an incoming pots dial-peer from PSTN (let's say that it matches 333XXX numbers).Then I have two outgoing voip dial-peers (both dial-peers match 333XXX numbers and are the outgoing routes for these numbers).One of these two outgoing dial-peers is set with a higher preference than the other in order to be the first choice for sending the calls.One of the dial-peers has session target one of the SER and the other has the other SER as session target.So, the main idea is that I want to do fallback between these two dial-peers: if the connection with the first SER (that is with higher preference) is down or there are network problems I want my CISCO to choose the other route to my second SER (and I want it to do that after a specified time).The question is how cand I do that ? And how can I set this timer ?(let's say 10 seconds -> after 10 seconds to choose the other route).What commands do I have to set on the dial-peers and what commands need to be set in the global configuration (timers,etc...)? Thank you very much and I appreciate any help.

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[Serusers] Help with authorization
by Steve Blair 22 Apr '05

22 Apr '05

I have a working 0.9.1 config to which I would like add server side features such as call forward all (cfwdall). I have a pretty good idea how to handle cfwdall using avp_ops however I'm stuck on the authentication. If a subscriber has local calling permissions (acl=local) and cfwdall their phone to a long distance number I need to respond with an informative response. In the INVITE processing in my config I have statements such as: if (is_user_in("credentials", "ld")) { setflag(11); }; These checks fail with the following errors: Apr 21 18:31:53 ser[498]: [SER]: AVP: Checking From gateway caller Apr 21 18:31:53 ser[498]: check_username(): No authorized credentials found (error in scripts) Apr 21 18:31:53 ser[498]: check_username(): Call {www,proxy}_authorize before calling check_* function ! Apr 21 18:31:53 ser[498]: [SER]: Flag for UMVM redirect successful. Apr 21 18:31:53 ser[498]: [SER]: AVP: Checking credentials Apr 21 18:31:53 ser[498]: is_user_in(): No authorized credentials found (error in scripts) I thought adding proxy_authorize("", "subscriber")), check_to and check_from calls prior to the is_user_in check would address these errors but that hasn't worked either. If I want to set a flag if the caller is an authorized subscriber, the callee is an authorized subscriber and then use "is_user_in" to determine if the called party has a particular credential what am I missing? Thanks,Steve

3 3

[Serusers] serweb path
by Llanos Serna García-Conde 22 Apr '05

22 Apr '05

5 4

[Serusers] SER 0.8.12 malformed outgoing packets with some UA
by Arek Bekiersz 22 Apr '05

22 Apr '05

Dear List, Some new UA device we are testing is causing strange behaviour of SER. When INVITE is forwarded by our SIP proxy (using SER 0.8.12 cvs + Radius backend), it contains some garbage characters and IP packet lenght is disrupted. Packet is cut on exact lenght of 1478 bytes (some MTU limit of 1500 bytes here???) See little debug below (I can send full debug on priv address). Notice the additional "."'s before INVITE and garbage in address line (U xx -> zz:5060 414@0:1480): I double checked SER log in debug mode, there are no differences between this abnormal situation and normal INVITE proxying, with other UA devices. Initial buffers are set to high values and nothing can go wrong.. Is this some bug in SER? Note: adresses were changed to xxx,yyy and so on: U xx -> yy:5060 SIP/2.0 100 trying -- your call is important to us.. Via: SIP/2.0/UDP yy:5060;branch=z9hG4bKDB20B3494FE0C715E1DA2AB3390D.. From: <sip:vvv@xx>;tag=6817B284750F6B75621571A17923.. To: <sip:www@xx>.. Call-ID: E5A4F1BBFB9D021B58C6B6440CB3@yy.. CSeq: 18 INVITE.. Server: Sip EXpress router (0.8.12-1rc6 (i386/freebsd)).. Content-Length: 0.. Warning: [...omited...] # U xx -> zz:5060 414@0:1480 <-- NOTICE garbage here ........INVITE sip:www@xxx SIP/2.0.. <-- NOTICE "dots" here Record-Route: <sip:www@xxx;ftag=6817B284750F6B75621571A17923;lr=on>.. Via: SIP/2.0/UDP xxx;branch=z9hG4bK8bef.e7a84e21.0.. Via: SIP/2.0/UDP yyy:5060;branch=z9hG4bKDB20B3494FE0C715E1DA2AB3390D.. From: <sip:vvv@xxx>;tag=6817B284750F6B75621571A17923.. To: <sip:www@xxx>.. Call-ID: E5A4F1BBFB9D021B58C6B6440CB3@yyy.. CSeq: 18 INVITE.. Contact: <sip:vvv@yyy;uniq=BE4F86E6E5C314AE11701C942248>.. Authorization: [...omited...] Max-Forwards: 69.. User-Agent: [...omited...] Supported: 100rel, replaces.. Allow-Events: telephone-event.. Allow-Events: refer.. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER.. Content-Type: application/sdp.. Accept: application/sdp.. Accept-Encoding: identity.. Content-Length: 418.. .. v=0.. o=user 20040204 20040204 IN IP4 yyy.. s=call.. c=IN IP4 yyy.. t=1114074899 1114078499.. m=audio 7078 RTP/AVP 8 0 2 102 100 99 18 101 13.. a=rtpmap:8 PCMA/8000.. a=rtpmap:0 PCMU/8000.. a=rtpmap:2 G726-32/8000.. a=rtpmap:102 G726-32/8000.. a=rtpmap:100 G726-40/8000.. a=rtpmap:99 G726-24/8000.. a=rtpmap:18 G729/8000.. a=rtpmap:101 telephone-event/8000.. a=fmtp:101 0-11.. a=rtpmap:13 CN/8000.. a=rt <--- NOTICE packet is cut here at 1478 bytes ######################################## U zzz:5060 -> xxx:5060 SIP/2.0 100 Trying.. Via: SIP/2.0/UDP xxx;branch=z9hG4bK8bef.e7a84e21.0.. Via: SIP/2.0/UDP yy:5060;branch=z9hG4bKDB20B3494FE0C715E1DA2AB3390D.. From: <sip:vvv@xxx>;tag=6817B284750F6B75621571A17923.. To: <sip:www@xxx>;tag=as7a954f71.. Call-ID: E5A4F1BBFB9D021B58C6B6440CB3@yyy.. CSeq: 18 INVITE.. User-Agent: [...omited...].. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:www@zzz>.. Content-Length: 0.. .. Here is tcpdump of this situation (note the packet lenght error): --------------------------------------------------------------------------- Packet 5 Timestamp: 01:00:07.969419 (0.068520) Source Ethernet Address: 00:00:5E:00:01:01 (<unknown>) Destination Ethernet Address: 00:E0:1E:DF:AE:82 (<unknown>) Encapsulated Protocol: IP IP Header Version: 4 Header Length: 20 bytes Service Type: 0x10 Datagram Length: 614 bytes Identification: 0x45FE Flags: MF=off, DF=off Fragment Offset: 0 TTL: 64 Encapsulated Protocol: UDP Header Checksum: 0xDF42 Source IP Address: xxx Destination IP Address: yyy UDP Header Source Port: 5060 (<unknown>) Destination Port: 5060 (<unknown>) Datagram Length: 594 bytes (Header=8, Data=586) Checksum: 0xB306 UDP Data SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP yyy:5060;branch=z9hG4bK003971715 B46D0300099FFA02356. From: <sip:vvv@xxx>;tag=3085BFD5D9E8D8717CD 04F54A2A5. To: <sip:www@xxx>. Call-ID: 37C1AE7089D5A7FA8302B724D791@yyy. CSeq: 13 INVITE. Server: Sip EXpress router (0.8.12-1rc6 (i386/freebsd)). Content-Length: 0. Warning: [...omited...]. . --------------------------------------------------------------------------- Packet 6 Timestamp: 01:00:07.969683 (0.000264) Source Ethernet Address: 00:00:5E:00:01:01 (<unknown>) Destination Ethernet Address: 00:0C:6E:A5:38:55 (<unknown>) Encapsulated Protocol: IP IP Header Version: 4 Header Length: 20 bytes Service Type: 0x10 Datagram Length: 1500 bytes Identification: 0xCBCD Flags: MF=on, DF=off Fragment Offset: 0 TTL: 64 Encapsulated Protocol: UDP Header Checksum: 0x3A55 Source IP Address: xxx Destination IP Address: zzz <*** Packet length corrupt ***> --------------------------------------------------------------------------- --------------------------------------------------------------------------- Packet 7 Timestamp: 01:00:07.971393 (0.001710) Source Ethernet Address: 00:0C:6E:A5:38:55 (<unknown>) Destination Ethernet Address: 00:00:5E:00:01:01 (<unknown>) Encapsulated Protocol: IP IP Header Version: 4 Header Length: 20 bytes Service Type: 0x10 Datagram Length: 538 bytes Identification: 0x00DE Flags: MF=off, DF=on Fragment Offset: 0 TTL: 64 Encapsulated Protocol: UDP Header Checksum: 0xE906 Source IP Address: zzz Destination IP Address: xxx UDP Header Source Port: 5060 (<unknown>) Destination Port: 5060 (<unknown>) Datagram Length: 518 bytes (Header=8, Data=510) Checksum: 0xF96E UDP Data SIP/2.0 100 Trying. Via: SIP/2.0/UDP xxx;branch=z9hG4bK487e.541ff4c2 .0. Via: SIP/2.0/UDP yyy:5060;branch=z9hG4bK003971715 B46D0300099FFA02356. From: <sip:vvv@xxx>;tag=3085BFD5D9E8D8717CD 04F54A2A5. To: <sip:www@xxx>;tag=as2f2ca5dd. Call-ID: 37C1AE7089D5A7FA8302B724D791@yyy CSeq: 13 INVITE. User-Agent: Perceval Voicemail Server Idun. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:www@zzz>. Content-Length: 0. . Thanks for help... Regards, Arek Bekiersz arek(a)perceval.net ------------------------------------------------------------------------ Perceval Technologies sa/nv Rue Tenbosch, 9 B-1000 Brussels Tel: +32-2-6409194 (ext.213) Fax: +32-2-6403154 URL: http://www.perceval.net/ E-mail for general informations: info(a)perceval.net E-mail for technical informations: helpdesk(a)perceval.net ------------------------------------------------------------------------ This e-mail message contains legally PRIVILEGED and CONFIDENTIAL information intended for the use of the addressee only. If you are not the intended recipient of this message, please notify the undersigned by telephone or e-mail reply and destroy this message and any attachments. Any views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of Perceval. The integrity and security of this message cannot be guaranteed and it may be subject to data corruption, interception, unauthorised amendment, viruses and unforesee

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[Serusers] Help Using SER FIFO "ser_fifo"
by Leo 22 Apr '05

22 Apr '05

Dear serusers, I have spent the past several nights trying to figure out how to use the ser_fifo to send commands and get info back from SER using PHP. My understanding of using the fifo in SER is to send it :fifo_command:result_filename. In the example below I am sending the "ul_dump" command to the fifo and telling to return the result to a file called "delthis.txt". The file "delthis.txt" never gets created and it doesn't look like this fifo thing is working at all. The examples in the admin guide are not helping me. Can anyone one help? The test PHP code is below: <html> <head> <title>Browser SERCTL </title> </head> <body> <?php function wrfifo() { $fifo_handle = fopen("/tmp/ser_fifo" , "w"); if (!$fifo_handle) { echo "cannot open ser_fifo<br>"; return 0; } if (fwrite($fifo_handle, ":ul_dump:/tmp/delthis.txt")==-1) { echo "cannot write to fifo"; fclose($fifo_handle); return 0; } fclose($fifo_handle); system("cat /tmp/delthis.txt"); return 1; } echo "begin<br>"; $success = wrfifo(); if ($success) { echo "success<br>"; } else { echo "failure<br>"; } ?> Leo Papadopoulos leo(a)ltcjp.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.18 - Release Date: 4/19/2005

2 3

[Serusers] Service Creation Options
by Aisling O'Driscoll 22 Apr '05

22 Apr '05

Hello, I am looking to develop services based on the SER platform which can be configured via a secure web interface e.g. sequential ringing, subscriber based call routing etc. It seems that the service languages available are: 1. CPL 2. SIP CGI 3. SIP Servlets Basically Im a little bit confused as to which one I should target. I believe CPL is for use on end user terminals so Im ruling that out. I was thinking of sip servlets...Any opinions on this?...Will this be able to manipulate SER easily and also is SerWeb in any way linked to this?....Would it be possible to use a combination of servlets and CPL?? Any information would be appreciated, Many Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 1

[Serusers] compilng Mysql.so
by Damian Bundred 22 Apr '05

22 Apr '05

hi, i've managed to get the individual components of mysql to compile, but i still get this error make[1]: Entering directory `/root/ser-0.9.0/modules/mysql' gcc -shared -Wl,-O2 -Wl,-E -pg db_con.o db_mod.o dbase.o my_con.o my_id.o my_pool.o res.o row.o utils.o val.o -L/usr/lib/mysql -L/usr/local/lib -L/usr/local/lib/mysql -L/usr/local/mysql/lib/mysql/ -L/usr/pkg/lib/mysql -L/usr/local/mysql/lib -L/usr/lib64/mysql -lmysqlclient -lz -o mysql.so /usr/lib64/gcc-lib/x86_64-suse-linux/3.3.4/../../../../x86_64-suse-linux/bin/ld: /usr/lib64/mysql/libmysqlclient.a(libmysql.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC /usr/lib64/mysql/libmysqlclient.a: could not read symbols: Bad value collect2: ld returned 1 exit status make[1]: *** [mysql.so] Error 1 make[1]: Leaving directory `/root/ser-0.9.0/modules/mysql' I havent used makefiles that much before, and could really use some help. Many thanks Damian.

1 0

[Serusers] record_route()
by Llanos Serna García-Conde 22 Apr '05

22 Apr '05

3 2

[Serusers]
by Grigory Fishilevich 22 Apr '05

22 Apr '05

Hi everybody, how can i (userfrendly) migrate from ser 0.8.14 to 0.9.0 i think the database strukture is different between these versions. can i dump, or whatever, the users table with serctl und import it into another ser? many thanks, Grigory Fishilevic

3 2

[Serusers] Feature Request: Log Contact hf of 3xx Response in Accounting
by Martin Koenig 22 Apr '05

22 Apr '05

Hello, I think it would make sense to log the Contact header field of a 3xx Response in Ser accounting in order to be able to bill a Call-Forwarding user. What do you think? With best regards, Martin Koenig

2 1

[Serusers] problem cancelling a call with CANCEL request
by Esteban Monturus 22 Apr '05

22 Apr '05

I am installing a SIP proxy in a Linux machine in the local network of my office. I have implemented a SIP proxy with SER 0.8.10. The function of the SIP proxy is to let users talk each other without knowing the IP where each one is reachable. My problem appears when the caller wants to cancel the call because the callee does not answer. Since I want the calle's phone to stop ringing, and I haven't received a final response, the caller's phone has to send a CANCEL request. When the proxy server receives the CANCEL, it replies with a 487 "Request cancelled". SER does this before forwarding the CANCEL to the callee's phone. These 487 response generated by SER makes the INVITE client transaction of the caller finish. When the INVITE server transaction finishes in the callee's phone, it sends another 487 response, which is forwarded by SER, but the caller's transaction is already finished, so it doesn't send the corresponding ACK. I would like to know how to solve this problem, since SER should wait for the callee's phone to send the 487 and then forward this response to the caller's phone. Best regards, Esteban.

2 1

Re: [Serusers] Free TLS implementation: UA setup?
by Alex Mack 22 Apr '05

22 Apr '05

Hi! I'm starting to monologise but I want to share with you what I found out. I managed to connect MS Messenger over TLS to ser. Works very well. But the question remains: Is there a tool (like sipsak) to generate SIP queries over TLS? Just in case when there's no messenger available. Alex Mack Alex Mack schrieb: > Hi! > > Or, to put it another way: How can I test SIP functionality "by foot" > (e.g. by "telnet proxy.mydomain.org 5060")? Or can sipsak be used to > work with openssl for an TLS-connection, e.g. piped to an openssl prompt? > > Alex Mack > > Alex Mack schrieb: > >> Hi! >> >> I've managed to implement and compile Peter Griffiths' free TLS >> implementation. It worked well, a test-connection with "openssl >> s_client" did also work. >> >> Now, how do I tell my UAs (snom 190, sipura SPA-1001) to use SIP over >> TLS? Both vendors of these UAs claim their UA is capable of SIPS, so >> how do I enable it? >> >> Alex Mack >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > > >

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[Serusers] compilation problems with SER
by Damian Bundred 22 Apr '05

22 Apr '05

Hi, i've downloaded ser 9.0 src, and ran make prefix=/usr/local all make prefix=/usr/local install, and it seemed to work fine until i realised that it hadn't compiled mysql module. when i ran make prefix=/usr/local exclude_modules="" modules and it gave errors in a few modules including this in the mysql section make[1]: Entering directory `/root/ser-0.9.0/modules/mysql' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -DNAME='"ser"' -DVERSION='"0.9.0"' -DARCH='"x86_64"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_x86_64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/include/mysql -I/usr/local/mysql/include -I/usr/pkg/include -I/usr/include/mysql -c dbase.c -o dbase.o dbase.c:34:25: mysql/mysql.h: No such file or directory dbase.c:35:26: mysql/errmsg.h: No such file or directory In file included from dbase.c:39: val.h:31:25: mysql/mysql.h: No such file or directory In file included from dbase.c:39: val.h:44: error: parse error before '*' token In file included from dbase.c:40: my_con.h:33:25: mysql/mysql.h: No such file or directory In file included from dbase.c:40: my_con.h:39: error: parse error before "MYSQL_RES" my_con.h:39: warning: no semicolon at end of struct or union my_con.h:40: warning: type defaults to `int' in declaration of `con' my_con.h:40: warning: data definition has no type or storage class my_con.h:41: error: parse error before "row" my_con.h:41: warning: type defaults to `int' in declaration of `row' my_con.h:41: warning: data definition has no type or storage class my_con.h:44: error: parse error before '}' token dbase.c: In function `submit_query': dbase.c:67: error: dereferencing pointer to incomplete type dbase.c:68: warning: implicit declaration of function `mysql_ping' dbase.c:68: error: dereferencing pointer to incomplete type dbase.c:72: error: dereferencing pointer to incomplete type dbase.c:90: warning: implicit declaration of function `mysql_query' dbase.c:90: error: dereferencing pointer to incomplete type dbase.c:93: warning: implicit declaration of function `mysql_errno' dbase.c:93: error: dereferencing pointer to incomplete type dbase.c:94: error: `CR_SERVER_GONE_ERROR' undeclared (first use in this function) dbase.c:94: error: (Each undeclared identifier is reported only once dbase.c:94: error: for each function it appears in.) dbase.c:94: error: `CR_SERVER_LOST' undeclared (first use in this function) dbase.c:98: warning: implicit declaration of function `mysql_error' dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c:98: error: dereferencing pointer to incomplete type dbase.c:98: warning: format argument is not a pointer (arg 3) dbase.c: At top level: dbase.c:138: error: parse error before '*' token dbase.c: In function `print_values': dbase.c:142: error: `_c' undeclared (first use in this function) dbase.c:142: error: `_b' undeclared (first use in this function) dbase.c:142: error: `_l' undeclared (first use in this function) dbase.c:142: error: `_v' undeclared (first use in this function) dbase.c:142: error: `_n' undeclared (first use in this function) dbase.c: At top level: dbase.c:166: error: parse error before '*' token dbase.c: In function `print_where': dbase.c:172: error: `_c' undeclared (first use in this function) dbase.c:172: error: `_b' undeclared (first use in this function) dbase.c:172: error: `_l' undeclared (first use in this function) dbase.c:172: error: `_k' undeclared (first use in this function) dbase.c:172: error: `_v' undeclared (first use in this function) dbase.c:172: error: `_n' undeclared (first use in this function) dbase.c:178: error: `_o' undeclared (first use in this function) dbase.c: At top level: dbase.c:207: error: parse error before '*' token dbase.c: In function `print_set': dbase.c:213: error: `_c' undeclared (first use in this function) dbase.c:213: error: `_b' undeclared (first use in this function) dbase.c:213: error: `_l' undeclared (first use in this function) dbase.c:213: error: `_k' undeclared (first use in this function) dbase.c:213: error: `_v' undeclared (first use in this function) dbase.c:213: error: `_n' undeclared (first use in this function) dbase.c: In function `store_result': dbase.c:304: error: dereferencing pointer to incomplete type dbase.c:304: warning: implicit declaration of function `mysql_store_result' dbase.c:304: error: dereferencing pointer to incomplete type dbase.c:305: error: dereferencing pointer to incomplete type dbase.c:306: warning: implicit declaration of function `mysql_field_count' dbase.c:306: error: dereferencing pointer to incomplete type dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c:311: error: dereferencing pointer to incomplete type dbase.c:311: warning: format argument is not a pointer (arg 3) dbase.c: In function `db_free_result': dbase.c:348: warning: implicit declaration of function `mysql_free_result' dbase.c:348: error: dereferencing pointer to incomplete type dbase.c:349: error: dereferencing pointer to incomplete type dbase.c: In function `db_query': dbase.c:398: error: dereferencing pointer to incomplete type dbase.c: In function `db_insert': dbase.c:471: error: dereferencing pointer to incomplete type dbase.c: In function `db_delete': dbase.c:516: error: dereferencing pointer to incomplete type dbase.c: In function `db_update': dbase.c:559: error: dereferencing pointer to incomplete type dbase.c:568: error: dereferencing pointer to incomplete type make[1]: *** [dbase.o] Error 1 make[1]: Leaving directory `/root/ser-0.9.0/modules/mysql' Am i trying to compile it correctly, and if so how do i get the mysql module sorted? Many thanks, Damian.

2 1

[Serusers] presence
by Giovanni Balasso 22 Apr '05

22 Apr '05

Had anyone luck with presence and eyebeam? What I do in my ser.cfg is routing SUBSCRIBEs and NOTIFYs just like INVITEs without nathelper functions (actually NOTIFYs hit loose route) and that works great with kphone. Still eyebeam refuses to tell me who's online (I changed to peer-to-peer mode). Does anyone know something about? What about pa module and patches from x-ten for eyebeam, will they be merged into cvs (with some docs)? thanks ciao -- Giovanni Balasso giaso(a)yahoo.it

2 1

[Serusers] Re: [Sems] Compile problems
by Torbjørn Lium 22 Apr '05

22 Apr '05

Well, I finally got SEMS to compile. The problem was with gcc version 2.95.4 on debian stable. When I tried gcc 3.0 instead, everything worked ok. Of course just installing it didn't help, I also had to change the symlinks /usr/bin/g++ and /usr/bin/gcc to get it to work. Is this compile problem specific to debian? If not wouldn't it be a good idea to put a little information about required gcc version on the webpage or in the README? Regards Torbjørn Lium

1 1

[Serusers] 0.9.0 + mysql.so
by Leon Sun 22 Apr '05

22 Apr '05

I installed 0.9.0 and 'ser' started well if I used default ser.cfg. It shows following when I enabled 'loadmodule "/usr/local/lib/ser/modules/mysql.so"'. Would please suggest? [root@sip ~]# ser -P /var/run/ser.pid ERROR: bad config file (1 errors)

2 1

[Serusers] Serweb CVS Head version: install notes and issues
by Cameron Beattie 22 Apr '05

22 Apr '05

For those who are interested, I am running SER 0.9.0 (got from onsip) with the onsip ser.cfg for mediaproxy. I also downloaded and installed serweb Head (see earlier post for how to get this). I followed the install instructions and then had to do a few additional things I had to do to get it working: - change permissions of compile_dir (/var/www/html/iptel/templates/ on my setup) to allow web server user to write to that directory - copy the directory img/butons to img/buttons. The directory storing button images is called butons. Some of the php scripts point to buttons. I didn't want to find all the errors which is why I cheated. I have noticed a few issues which may be of interest to the developer: - On accounting tab: DB Error: no such field - On my account tab: sorry -- cannot open write fifo - On user prefs: DB Error: no such table. Maybe it's looking for the table providers which doesn't exist? - Typo when logout. Should read " You have logged out. To login again, type your username and password below" - Type on Server monitoring tab. Should read "differential values". Also, no statistics are shown The features are impressive and I plan to do some more testing in the future. Regards Cameron

1 1

[Serusers] call transfer
by Iqbal 22 Apr '05

22 Apr '05

Hi I have my ser.cfg all setup handling calls, pstn and all that...even NAT seems to be behaving itself :-), BUT call transfer doesnt want to work. Is there any special routing that needs to be done which should handle call transfer, or is another message sent, I mean do I need to set a route for NOTIFY or other MESSAGE type. A calls B (call is fine) B then transfers to C (again B and C can hear each other) then when B hangs up A and C arent talking, and it seems as if B hasnt really hung up (maybe this is where the problem is) In order to debug this when using ngrep and SER end, I am wondering which IP address a,b,c to monitor or should it be all 3 Iqbal

1 0

[Serusers] what are/is AVPops ?
by Lucas Aimaretto 22 Apr '05

22 Apr '05

hi all, what are/is AVPops ? Why do I need them ? Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.1 - Release Date: 20/04/2005

2 2

RE: [Serusers] if routeXX fails..
by Matt Schulte 21 Apr '05

21 Apr '05

I see, that makes perfect sense :-).. So I just might need to flip around my "if" to get it to work. -----Original Message----- From: Marian Dumitru [mailto:marian.dumitru@voice-sistem.ro] Sent: Thursday, April 21, 2005 5:10 PM To: Matt Schulte Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] if routeXX fails.. Hi Matt, Basically you can do that, but be very careful since the return code of a route is the return code of the *last* command executed inside the route - and break always returns true! Best regards, Marian Matt Schulte wrote: > Is it possible to do this? or is it bad practice, I'm assuming the > "extra" route would have to return an error code to show a > nonpositive? I coulda swore I got this to work awhile back. > > if (!route(6)) { > xlog("L_INFO", "NOTICE: blah <%ru> sending 404\n"); > sl_send_reply("404", "Not Found"); > break; > } else { > # Else route6 ran another route and break > break; > } > -- Voice System http://www.voice-system.ro

1 0

[Serusers] Feature request: Textops function to remove ALL header fields of name x
by Martin Koenig 21 Apr '05

21 Apr '05

Hello, I think it would make sense to implement a function in textops similar to remove_hf(x) but remove_all_hf(x) in order to be sure a certain header field is not present any more. The specific scenario is setting up a RFC3323/3325-compilant platform, where it is imperative that untrusted nodes are not able to inject any "P-Asserted-Identity" header field into trusted nodes. I'm not sure remove_hf() does the job in this case. With best regards, Martin

2 1

[Serusers] ser_fifo permissions
by Cameron Beattie 21 Apr '05

21 Apr '05

This seems such a basic question but I haven't found the answer anywhere. How do I set the permissions on the ser_fifo file without running chmod? I tried setting user = and group = in ser.cfg but that didn't work. I don't want to have to chmod the file each time I start ser. I'm sure this is very simple and would appreciate any help. Regards Cameron

2 1

[Serusers] fix_actions error
by Dave 21 Apr '05

21 Apr '05

I have downloaded the version 0.9.0 from cvs and now each time I start SER, I get the following error: Apr 21 14:11:43 localhost /opt/ser/sbin/ser[20456]: hf_fixup(): Unsupported Header Field identifier Apr 21 14:11:43 localhost /opt/ser/sbin/ser[20456]: ERROR: fix_expr : fix_actions error I'll grateful for any suggestions -I have used the latest ser_mysql.sh script to create a new database. I have made sure to compile all modules. Thanks __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] ser and busy lines
by Ahmad Cheikh-Moussa 21 Apr '05

21 Apr '05

Hi! Following Situation: 1 x SER Server 1 x Gateway with two PRIs What happens if all lines are busy and a user wants to make another call to the PSTN? Will the user get a busy tone ? Is there a way that the ser server knows how many lines are busy ? Regards, Ahmad -- Ahmad Cheikh-Moussa NetUSE AG Dr.-Hell-Straße, 24107 Kiel, Germany Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499 Service: Service(a)NetUSE.DE -- http://NetUSE.DE/

2 1

[Serusers] SER on IPV6 ?
by Yin, Ming 21 Apr '05

21 Apr '05

Hallo everyone, Can anyone tell me, whether the SER can also be deployed in IPv6 network or not ? We have setup a ipv6 testbed and want to demonstrate SIP-based VoIP on that. Thanks in advance ! Kind Regards Ming

2 1

[Serusers] if routeXX fails..
by Matt Schulte 21 Apr '05

21 Apr '05

Is it possible to do this? or is it bad practice, I'm assuming the "extra" route would have to return an error code to show a nonpositive? I coulda swore I got this to work awhile back. if (!route(6)) { xlog("L_INFO", "NOTICE: blah <%ru> sending 404\n"); sl_send_reply("404", "Not Found"); break; } else { # Else route6 ran another route and break break; }

2 1

[Serusers] BYE records for mysql
by Barry Murphy 21 Apr '05

21 Apr '05

Hey guys, I don't seem to get BYE records from SER being sent to mysql, I get INVITE and ACK. Without BYE I can't see who terminated the call, caller or callee, and I can't determin the duration of the call. Any ideas? Thanks Barry

3 2

[Serusers] codecs with ser
by Barry Murphy 21 Apr '05

21 Apr '05

Hi, How do I tell ser to allow gsm codec? I can only make calls using ulaw for some reason :/ Barry

2 1

[Serusers] stop that
by WktE7＠aol.com 21 Apr '05

21 Apr '05

oy u stop sending for me any stupid e-mails i don't wont them, if u will send me another e-mail i will call to police for sending to me unwonted e-mails, thanks

3 2

[Serusers] SER high availability setup
by Atif Rasheed 21 Apr '05

21 Apr '05

Hi, I am trying to deploy a solution as shown below: SER<------>Asterisk1 | --->Asterisk2 | --->Asterisk3 Calls come in to the SER machine. I want to distribute calls among the asterisk machines(all with the same specifications). I initially tried using DNS round robin. It works well but doesn not have builtin failover mechanisms. So if a machine goes down calls will not be rerouted and sent to other available machines. I then tried using DNS SRV but, it seems, unfortunately SER does not support failover support yet. Now I am not sure how to distribute load amongst these machines. Any ideas suggestions will greatly be appreciated. Regards, Danish

2 1

[Serusers] Ser + Asterisk
by Elton Machado 21 Apr '05

21 Apr '05

Hi guys, I'm working of configurations examples, how to get Ser and Asterisk working together in peace ;) What I wan to do is to have ser doing all the hard work, registering, billing, proxying and help at nat stuff, and use asterisk to register myself at my sip providers, as I couldn't found a way to do it with ser. An idea how the things are or will be connected is... [ UAC's ] -> SER -> * / (Ser?) <-> [firewall full port redirect] -> router -> sip provider 192.168.20.x 192.168.20.1 192.168.20.10|192.168.30.10 192.168.30.1 192.168.30.x | 172.16.32.1 | 172.16.32.254 |external ip semi fixed or fixed I have all the stuff running fine if I use an IAX provider, problem is... all my providers use sip :( Does anyone has any clue to point me out ? Sip session goes fine but it just get stuck in rtp nat and I can't understand very well why. Another is the UAC's try to reach by itself the sip provider jupping over all 192.168.20.x/30.x thins goes fine if I use stun of course. My question is... Is there any way to do the same in asterisk? Like use a tun server or something, or using ser to help it to change any contact or sip headers field? HOw is the best way to acomplished that? Best Regards,

4 4

[Serusers] Ser + Asterisk for VoIP service provider
by John Best 21 Apr '05

21 Apr '05

Hello, I have recently been put in charge of starting up a VoIP project connecting voip trunk lines to business' existing PBX's. I have decided to go all open source and have figured out asterisk pretty well. The problem is with SER. There does not seem to be a lot of clear documentation on asterisk interoperability. My idea is to use SER for the bulk of the call routing and asterisk for voicemail, billing, and other apps. All calls will be sip to me and then passed onto a VoIP provider who will handle all Long distance and PSTN service. Does anyone have any suggestions on setup, or any config files they could email me please? Thanks

2 1

[Serusers] Serweb Problem
by Edgardo O. Gonzales II 21 Apr '05

21 Apr '05

Please give your help ... thanks ... Hi SerUser! Can somebody help me with this problem.. After I install and configure serweb as stated on configuration guide.. I got a blank page accessing admin and user_interface page. On logs of my apache, this is the feedback .. 210.23.237.236 - - [21/Apr/2005:23:51:12 +0800] "GET /serweb/user_interface/index.php?phplib_Session=4026e0eaf3aa336098b7e1d2926f49ff&phplib_Session=dc200ec3587d1985cf9aaf706d3b248c&phplib_Session=8d66aaa5f4418cb75126beda37c2acde&phplib_Session=14d85207dd2ec6439325db9f78bea914&phplib_Session=3a23bcfc1ecae59dcefead744c64aaf3&phplib_Session=6ff4d142925c0574d1d3614bcb679578&phplib_Session=5d83ca96f291e0310ab4777442a547f9&phplib_Session=8ac2999cee413bde7ef4769b8f73b860&phplib_Session=96c5b130e01b2db96d73bc050248183e&phplib_Session=9ae3f9631ab331d0c0299cf394e6f691&phplib_Session=ddb01b22fdc4d988a9093a37cf4ee00b&phplib_Session=d48993a3af2167a562d151f2cb285ce4&phplib_Session=2492b77a05e07417c38d229d30f58d9a&phplib_Session=a4edc5415a40e4a922c4dbf721f25dc0&phplib_Session=5d41839c67fbc842ac27f59a558e982a&phplib_Session=7b5b95a63d36ffdf9b412264bf941691&phplib_Session=507571c99dc312b6e36ea364de2644f4&phplib_Session=0992981f5a11f248824edbb95f5db9f3&phplib_Session=1f25b8f300e72b404a54ab4c2c4060e8&phplib_Session=870611806d28cda0d0021f8422454! 12a&ph p lib_Session=f5c422596455228e1749a081e1d9be1e&phplib_Session=2ecc3410c0ef13c4a66f12d51b02e7a3&phplib_Session=ff433b28f8596fe03518398fef080887&phplib_Session=aec18ca4299ec9009c236cde5d0bb733&phplib_Session=5937f583e214b20d67999da9947a9b7c&phplib_Session=854621f5353598bf6b6c9fd0f882766a&phplib_Session=0809e0fb3fddff21a14190e4240b814a&phplib_Session=22b80c053272e22c30c33a883182e014&phplib_Session=7c69db32533cd64b763462353c20d927&phplib_Session=c4eeba51a9991c5161269dafa9647d38&phplib_Session=472ca546c8ff6970ea9cae514940b86f&phplib_Session=c3c6871818a255d4eafe2d8a373228ef&phplib_Session=fded6f2f186c6d2dedd467a2b93f5c2a&phplib_Session=53033ca88aa7ec64b25d70f8ac424086&phplib_Session=62744ffaf3c7866e2f3324b1d6ca9f69&phplib_Session=d43a04d83057e27859e9a9db63782a4c&phplib_Session=bb309e7809c1341764212cab865e33ce&phplib_Session=002136a613c2bdee8d927d8a43be20ad&phplib_Session=82a678edfc54753212929c25f5fb5221&phplib_Session=2de93e7370b1de73041c6f4dc21a11df&phplib_Session=dfd10093b5f05cccd! 3178ad f cc3f8c62 HTTP/1.1" 302 - Is there something I miss during the configuration .. thanks, ed

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[Serusers] SER behind DNAT
by Oscar Patricio 21 Apr '05

21 Apr '05

Hi! I have the following scenario: UA1 ----> SER1+NAT ----> Internet <----DNAT ROUTER <---- SER2 <----UA2 Both SER1+NAT and SER2 have 2 network interfaces and work with mhomed=1. SER1 is on the same machine as the NAT, so SER1 listens on a public address. SER2 is behind a router doing full DNAT to it, so it doesn't directly listen in a public address. Communication works in both ways, regardless of who starts the call! This is the good thing. The bad thing is, that when UA1 hangs up, the BYE request is sent to the IP address of the outmost interface of SER2, which is behind the DNAT router. How do I make SER1 make the bye request go to the public interface of the router? I've tried changing the advertised_address and record_route_preset parameters, and haven't achieved success yet! Has anyone faced a similar problem, or can give me hand with this problem? Thanx a lot!!! Óscar

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[Serusers] SER authorization based on IP address
by Brian McCrary 21 Apr '05

21 Apr '05

Hello, I have some MC3810s that I want to use with SER over their PRI ports. The problem with these is that IOS 12.3 doesn't seem to support SIP registrar functions, so there is no way to get the DIDs to register with SER. Since these routers have static IP addresses, is it possible to have SER authenticate based on IP address? I know I could just add a ton of lines to my config file authorizing and routing them manually, but thoguht someone might know a better way to save their location like the registered users, and check the source IP vs the password field (something like put the IP address in the password field). I'm sure I'm far from the first that's ran into this, but couldn't find much so thought I'd ask for ideas. Thanks, Brian

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RE: [Serusers] SER authorization based on IP address
by Ricardo Martinez 21 Apr '05

21 Apr '05

Hello, I'm not sure if this could help, but i think the AVPOPS module can help you to solve this problem. You can add or remove the "trusted" ip's from your DB and let avpops do the work. So you don't need to modify your configuration file. Check this link : http://www.voice-system.ro/docs/avpops/ And check this example : http://www.voice-system.ro/docs/avpops/ar01s08.html#ex_trusting I think that could help you. I'm in the way to try this too. Good Luck Regards, Ricardo Martinez.- > -----Mensaje original----- > De: Brian McCrary [mailto:bmccrary@lwav.net] > Enviado el: Jueves, 21 de Abril de 2005 11:54 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] SER authorization based on IP address > > > Hello, > > I have some MC3810s that I want to use with SER over their PRI ports. > The problem with these is that IOS 12.3 doesn't seem to support SIP > registrar functions, so there is no way to get the DIDs to > register with > SER. > > Since these routers have static IP addresses, is it possible > to have SER > authenticate based on IP address? I know I could just add a > ton of lines > to my config file authorizing and routing them manually, but thoguht > someone might know a better way to save their location like the > registered users, and check the source IP vs the password field > (something like put the IP address in the password field). > > I'm sure I'm far from the first that's ran into this, but > couldn't find > much so thought I'd ask for ideas. > > Thanks, > > Brian > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] SEMS and SER VM??
by Erdem HAKI 21 Apr '05

21 Apr '05

Deal all, I am a newbie in SIP I am trying to add Voicemail service on my SER . As according to "SIP Express Router v0.11 --Admin Guide" SEMS is required to provide voicemail capability in SIP server. i have downloaded "sems-0.1.0.tar.gz" and installed in my machine where I have "ser-0.8.11" as a sip server. I have changed my ser.cfg scripts to provide voicemail capabilities and to load vm module available in SER. Now I get stuck How I have to use SEMS module ( I mean to say that "ans_machine" available in sems ) what is the difference & similarity between SEMS Voicemail and SER vm module. On what basis SEMS Voicemail and SER communicate to each other. Is their any configuration file is required inside the SEMS like SER (ex. ser.cfg). what is the proper step to perform a testing of voice mail using these modules.. (the same problem witk koyama) Thanks for your help.... Erdem H. HAKI

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RE: [Serusers] Serious Problems with SER - Not responding anythin g.
by Ricardo Martinez 21 Apr '05

21 Apr '05

Thanks again. I have heard about sipsak, but nothing about nagios. Is a tool that come with sipsak? How it works?. Is a monitoring tool?. I found the sipsak homepage but i don't see the relation with logging information from SER.... Can someone give more information about this tool? Thanks! Ricardo Martinez.- > -----Mensaje original----- > De: Jiri Kuthan [mailto:jiri@iptel.org] > Enviado el: Miércoles, 20 de Abril de 2005 19:57 > Para: Ricardo Martinez; 'serusers(a)lists.iptel.org' > Asunto: RE: [Serusers] Serious Problems with SER - Not responding > anythin g. > > > One more which came into my mind and has been reported to > block is syslog. > > > At 07:39 PM 4/20/2005, Ricardo Martinez wrote: > >Thanks Jiri. > >I remember have been checked the typical services when the > problem was > >happening. > >My Radius (i'm not use mysql) for example was working > without problems... > > > >How can i log information of SER failures? > > nagios/sipsak is an option. > > -jiri > > > >Regards, > >Ricardo.- > > > > > >> -----Mensaje original----- > >> De: Jiri Kuthan [mailto:jiri@iptel.org] > >> Enviado el: Miércoles, 20 de Abril de 2005 13:12 > >> Para: Ricardo Martinez; 'serusers(a)lists.iptel.org' > >> Asunto: Re: [Serusers] Serious Problems with SER - Not responding > >> anything. > >> > >> > >> a typical reason is SER is blocked waiting for some > external service. > >> It may be possibly unavailable mysql, DNS, or RADIUS... > >> > >> -jiri > >> > >> At 07:09 PM 4/20/2005, Ricardo Martinez wrote: > >> >Hello list. > >> > I'm experiencing problems with my SER. As i > >> mentioned in a past > >> >post > >> (http://lists.iptel.org/pipermail/serusers/2005-April/018324.html), > >> >from time to time (it is getting pretty often) my SER not > >> answering any > >> >request, when a REGISTER reach the server it is not > >> challenged, this happens > >> >too with the INVITE's messages. About two hours ago this > >> happened again. > >> >I asked in my last post how to LOG the internal errors or > >> messages from SER > >> >to see if i can get some idea about what's going on, but i > >> have no answer > >> >about it. > >> >This is the infornmation i was able to capture when this was > >> happening : > >> > > >> >the serctl moni command shows a lot of "failures". > >> > > >> >[cycle #: 11; if constant make sure server lives and fifo is on] > >> >Server: Sip EXpress router (0.8.14-3 (i386/linux)) > >> >Now: Wed Apr 20 12:56:29 2005 > >> >Up Since: Thu Apr 14 21:32:18 2005 > >> >Up time: 487451 [sec] > >> > > >> >Transaction Statistics > >> >Current: 6 (2612 waiting) Total: 38194 (0 local) > >> >Replied localy: 80752 > >> >Completion status 6xx: 4643, 5xx: 768, 4xx: 10664, 3xx: > >> 0,2xx: 22552 > >> > > >> >Stateless Server Statistics > >> >200: 5975004 202: 0 2xx: 0 > >> >300: 0 301: 0 302: 0 3xx: 0 > >> >400: 0 401: 205980 403: 0 404: 0 407: 22918 408: 0 483: 0 > >> 4xx: 19 > >> >500: 0 5xx: 0 > >> >6xx: 0 > >> >xxx: 0 > >> >failures: 24180 > >> > > >> >UsrLoc Stats > >> >Domain Registered Expired > >> >'location' 219 4552 > >> >-------------------------------------------------------------- > >> >[cycle #: 5; if constant make sure server lives and fifo is on] > >> >Server: Sip EXpress router (0.8.14-3 (i386/linux)) > >> >Now: Wed Apr 20 12:56:47 2005 > >> >Up Since: Thu Apr 14 21:32:18 2005 > >> >Up time: 487469 [sec] > >> > > >> >Transaction Statistics > >> >Current: 4 (2619 waiting) Total: 38219 (0 local) > >> >Replied localy: 80762 > >> >Completion status 6xx: 4643, 5xx: 770, 4xx: 10667, 3xx: > >> 0,2xx: 22552 > >> > > >> >Stateless Server Statistics > >> >200: 5975030 202: 0 2xx: 0 > >> >300: 0 301: 0 302: 0 3xx: 0 > >> >400: 0 401: 205995 403: 0 404: 0 407: 22922 408: 0 483: 0 > >> 4xx: 19 > >> >500: 0 5xx: 0 > >> >6xx: 0 > >> >xxx: 0 > >> >failures: 24380 > >> > > >> >UsrLoc Stats > >> >Domain Registered Expired > >> >'location' 220 4552 > >> >----------------------------------------------------------------- > >> >[cycle #: 2; if constant make sure server lives and fifo is on] > >> >Server: Sip EXpress router (0.8.14-3 (i386/linux)) > >> >Now: Wed Apr 20 12:56:52 2005 > >> >Up Since: Thu Apr 14 21:32:18 2005 > >> >Up time: 487474 [sec] > >> > > >> >Transaction Statistics > >> >Current: 4 (2620 waiting) Total: 38224 (0 local) > >> >Replied localy: 80762 > >> >Completion status 6xx: 4643, 5xx: 770, 4xx: 10667, 3xx: > >> 0,2xx: 22552 > >> > > >> >Stateless Server Statistics > >> >200: 5975034 202: 0 2xx: 0 > >> >300: 0 301: 0 302: 0 3xx: 0 > >> >400: 0 401: 206000 403: 0 404: 0 407: 22923 408: 0 483: 0 > >> 4xx: 19 > >> >500: 0 5xx: 0 > >> >6xx: 0 > >> >xxx: 0 > >> >failures: 24445 > >> > > >> >UsrLoc Stats > >> >Domain Registered Expired > >> >'location' 215 4557 > >> > > >> >What is happening to cause the "failures" counter grows > that much?. > >> > > >> >The free command shows this : > >> >[root@root]# free > >> > total used free shared > >> buffers cached > >> >Mem: 1030888 626464 404424 0 > >> 86692 369848 > >> >-/+ buffers/cache: 169924 860964 > >> >Swap: 2040244 4864 2035380 > >> > > >> >The top command shows not much use of CPU. The proccess > >> were running (the > >> >SER and mediaproxy) > >> >Despite of all this, the SER was not answering anything. > >> >Could this be a bug? or maybe a problem with my ser.cfg > file?. I'm > >> >attaching my ser.cfg file too to see if someone can tell me > >> if there is > >> >something wrong about it. > >> >I really hope that someone could help me, this is getting > >> really complicated > >> >every time it happens. > >> > > >> >Thanks > >> >Ricardo Martinez.- > >> > > >> >************************************************** > >> >SER.CFG > >> >************************************************** > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++++++++++++++++++++++++ > >> > > >> > > >> > > >> > > >> ># ----------- global configuration parameters > >> ------------------------ > >> > > >> >debug=0 # debug level (cmd line: -dddddddddd) > >> >fork=yes > >> >log_stderror=yes # (cmd line: -E) > >> > > >> >#Uncomment these lines to enter debugging mode > >> >#fork=no > >> >#log_stderror=yes > >> > > >> >listen=ip.of.my.ser > >> > > >> >alias=sip.mydomain.com > >> >alias=sip2.mydomain.net > >> >alias=ip.of.my.ser > >> > > >> > > >> >check_via=no # (cmd. line: -v) > >> >dns=no # (cmd. line: -r) > >> >rev_dns=no # (cmd. line: -R) > >> >port=5060 > >> >#children=4 > >> >fifo="/tmp/ser_fifo" > >> >fifo_mode=0666 > >> > > >> > > >> ># ------------------ module loading > >> ---------------------------------- > >> > > >> ># Uncomment this if you want to use SQL database > >> >#loadmodule "/usr/local//lib/ser/modules/mysql.so" > >> > > >> >loadmodule "/usr/local//lib/ser/modules/sl.so" > >> >loadmodule "/usr/local//lib/ser/modules/tm.so" > >> >loadmodule "/usr/local//lib/ser/modules/rr.so" > >> >loadmodule "/usr/local//lib/ser/modules/maxfwd.so" > >> >loadmodule "/usr/local//lib/ser/modules/usrloc.so" > >> >loadmodule "/usr/local//lib/ser/modules/registrar.so" > >> >loadmodule "/usr/local//lib/ser/modules/textops.so" > >> >loadmodule "/usr/local//lib/ser/modules/exec.so" > >> > > >> ># ++++++++++ Para trabajar con NAT > >> > > >> >loadmodule "/usr/local//lib/ser/modules/mediaproxy.so" > >> >loadmodule "/usr/local//lib/ser/modules/domain.so" > >> >loadmodule "/usr/local//lib/ser/modules/dbtext.so" > >> >loadmodule "/usr/local//lib/ser/modules/nathelper.so" > >> > > >> ># Uncomment this if you want digest authentication > >> ># mysql.so must be loaded ! > >> >loadmodule "/usr/local/lib/ser/modules/auth.so" > >> > > >> ># ++++++++++ Modulos de Accounting y Autorizacion via Radius > >> > > >> >loadmodule "/usr/local/lib/ser/modules/auth_radius.so" > >> >loadmodule "/usr/local/lib/ser/modules/group_radius.so" > >> >loadmodule "/usr/local/lib/ser/modules/acc.so" > >> > > >> ># ----------------- setting module-specific parameters > >> --------------- > >> ># -- rr params -- > >> ># add value to ;lr param to make some broken UAs happy > >> >modparam("rr", "enable_full_lr", 1) > >> > > >> ># +++++ module authorization > >> >modparam("auth_radius","radius_config","/usr/local/etc/radius > >> client/radiuscl > >> >ient.conf") > >> >modparam("auth_radius","service_type",15) > >> > > >> > > >> ># +++++ module accounting > >> >modparam("acc","radius_config","/usr/local/etc/radiusclient/r > >> adiusclient.con > >> >f") > >> >modparam("acc","log_level",1) > >> >modparam("acc", "service_type", 15) > >> >modparam("acc", "radius_flag", 1) > >> >modparam("acc", "radius_missed_flag", 3) > >> >modparam("acc", "failed_transactions", 1) > >> >modparam("acc", "report_ack", 0) > >> > > >> > > >> ># +++++ group radius > >> >modparam("group_radius","radius_config","/usr/local/etc/radiu > >> sclient/radiusc > >> >lient.conf") > >> >modparam("group_radius", "use_domain", 1) > >> > > >> > > >> ># +++++ module mediaproxy > >> >modparam("mediaproxy", "natping_interval", 60) > >> >modparam("mediaproxy", "mediaproxy_socket", > >> "/var/run/mediaproxy.sock") > >> >#modparam("mediaproxy", "sip_asymmetrics", > >> >"/usr/local/etc/ser/sip-asymmetrics-clients") > >> >#modparam("mediaproxy", "rtp_asymmetrics", > >> >"/usr/local/etc/ser/rtp-asymmetrics-clients") > >> >modparam("registrar", "nat_flag", 5) > >> > > >> ># +++++ module domain > >> >modparam("domain", "db_url", "/usr/local/etc/ser/domaintables") > >> >modparam("domain", "domain_table", "domain") > >> >modparam("domain", "domain_col", "domain") > >> > > >> ># +++++ module tm > >> >modparam("tm", "fr_timer", 15) > >> >modparam("tm", "fr_inv_timer", 22) > >> >modparam("tm", "wt_timer", 5) > >> > > >> ># +++++ module registrar (Con estos parametro haces que el > >> location lookup > >> >priorize solo los mas recientes contacts) > >> >modparam("registrar", "append_branches", 0) > >> >modparam("registrar", "desc_time_order", 1) > >> > > >> > > >> ># ------------------------- request routing logic > >> ------------------- > >> > > >> ># main routing logic > >> > > >> >route { > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> ++++++++++++++ > >> > # Llamada a Rutina Externa "translate" para > cambio de numero > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> ++++++++++++++ > >> > > >> > if (method=="INVITE" || method=="CANCEL") { > >> > exec_dset("/usr/local/etc/ser/translate"); > >> > }; > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> ++++++++++++++ > >> > # Initial sanity checks -- messages with > >> > # max_forwards==0, or excessively long requests > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> ++++++++++++++ > >> > > >> > if (!mf_process_maxfwd_header("10")) { > >> > sl_send_reply("483","Too Many Hops"); > >> > break; > >> > }; > >> > > >> > if ( msg:len > max_len ) { > >> > sl_send_reply("513", "Message too big"); > >> > break; > >> > }; > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> > # Marcamos todos los mensajes para que pasen por > >> nuestro servidor > >> > # a traves del record route. > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> > > >> > record_route(); > >> > > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> > # Permitimos el uso del campo Route en el ruteo > de la llamada > >> > # > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> > > >> > if (method=="BYE" || method=="CANCEL") { > >> > log(1, "NAT: BYE o CANCEL recibido --> > >> terminando la sesion > >> >de media\n"); > >> > end_media_session(); > >> > setflag(1); > >> > }; > >> > > >> > if (loose_route()) { > >> > t_relay(); > >> > break; > >> > }; > >> > > >> > > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+ > >> > # Manejo de los mensajes NOTIFY para Keep Alive de NAT > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+ > >> > > >> > if ((method=="NOTIFY") && search("^Event: keep-alive")) { > >> > sl_send_reply("200", "OK"); > >> > break; > >> > }; > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+ > >> > # Marcamos las llamadas para accounting > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+ > >> > > >> > setflag(1); > >> > > >> > > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+ > >> > # Test para clientes NATeados > >> > # > >> > # 1 : Chequea si el cliente tiene IP privada en el > >> campo Contact. > >> > # 2 : Chequea si el cliente se contacto con una IP > >> distinta a la del > >> >campo VIA. > >> > # 3 : Chequea si el cliente tiene IP privada en el > >> ultimo campo VIA. > >> > # > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+ > >> > > >> > > >> > if (client_nat_test("3")) { > >> > log(1, "NAT: Requerimiento de IP privada --> > >> fixed contact > >> >(en rutina principal)\n"); > >> > setflag(5); > >> > force_rport(); > >> > > >> > if (method=="REGISTER") { > >> > fix_nated_contact(); > >> > } else { > >> > fix_contact(); > >> > }; > >> > append_hf("P-hint: fixed NAT contact for > >> request\r\n"); > >> > }; > >> > > >> > > >> > if (uri==myself) { > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++ > >> > # Chequeo de REGISTRO para equipos del dominio 1 > >> > # Dominio Voiss : sip.mydomain.com > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++ > >> > > >> > if (method=="REGISTER" && > >> uri=~"^sip:.*sip.mydomain.com") { > >> > # Challenge/Response > >> > if ( > >> !radius_www_authorize("sip.mydomain.com")) { > >> > > >> www_challenge("sip.mydomain.com", "1"); > >> > break; > >> > }; > >> > > >> > # Mandamos un mensaje de Log si es > >> que un cliente > >> >NATEADO se registro. > >> > if (isflagset(5)) { > >> > log(1, "NAT: Cliente > >> NAT'eado registrado > >> >(dentro del chequeo de REGISTER)\n"); > >> > }; > >> > > >> > save("location"); > >> > break; > >> > }; # Fin chequeo de Registro para Dominio : > >> sip.mydomain.com > >> > > >> > > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++ > >> > # Chequeo de REGISTRO para equipos del dominio 2 > >> > # Dominio Mayoristas : sip2.mydomain.net > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++ > >> > > >> > if (method=="REGISTER" && > >> uri=~"^sip:.*sip2.mydomain.net") { > >> > # Challenge/Response > >> > if ( > >> !radius_www_authorize("sip2.mydomain.net")) { > >> > > >> www_challenge("sip2.mydomain.net", "1"); > >> > break; > >> > }; > >> > > >> > # Mandamos un mensaje de Log si es > >> que un cliente > >> >NATEADO se registro. > >> > if (isflagset(5)) { > >> > log(1, "NAT: Cliente > >> NAT'eado registrado > >> >(dentro del chequeo de REGISTER)\n"); > >> > }; > >> > > >> > save("location"); > >> > break; > >> > }; # Fin chequeo de Registro para Dominio : > >> >sip2.mydomain.net > >> > > >> > > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++ > >> > # CHEQUEO DEL INVITE PARA TODAS LAS LLAMADAS > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++ > >> > > >> > if (method=="INVITE" && src_ip!=xx.xx.xx.xx) { > >> > if ( !radius_proxy_authorize("")) { > >> > proxy_challenge("", "1"); > >> > log (1, "LOG: Llamada no > >> AUTORIZADA\n"); > >> > break; > >> > }; > >> > }; > >> > > >> > > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++++ > >> > # Llamadas hacia dominio H.323 > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++++ > >> > > >> > if ( (uri=~"^sip:717.*@.*") | > >> (uri=~"^sip:707.*@.*") | > >> >(uri=~"^sip:777.*@.*") | (uri=~"^sip:333.*@.*") | (uri=~"^sip > >> >:0.*@.*") ){ > >> > log(1,"FORWARDING : Llamada hacia > plataforma > >> >H.323\n"); > >> > route(1); > >> > break; > >> > }; > >> > > >> > > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++ > >> > # Llamadas SIP locales > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++ > >> > > >> > if (!lookup("location")) { #Usuarios que > no estan en > >> >"location" database. > >> > route(4); > >> > break; > >> > }; > >> > > >> > > >> > setflag(3); > >> > > >> > > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++ > >> > # Manejo para llamadas NAT'das > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++ > >> > > >> > if (method=="INVITE") { > >> > log(1, "ROUTING: Reply processing > >> (online user) > >> >enabled to handle NAT'd resonses\n"); > >> > t_on_reply("1"); > >> > if ( (isflagset(5)) && > >> (method=="INVITE" || > >> >method=="ACK") ) { > >> > log(1, "NAT: Invite > >> received --> > >> >enabling media proxy (else del lookup 2do if)\n"); > >> > use_media_proxy(); > >> > append_hf("P-hint: > >> request forced to > >> >media proxy\r\n"); > >> > }; > >> > }; > >> > > >> > # Do it ? (Ricardo Martinez) > >> > }; > >> > > >> > append_hf("P-hint: USRLOC\r\n"); > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+++++ > >> > # Efectuamos la llamada > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+++++ > >> > > >> > if (!t_relay()) { > >> > sl_reply_error(); > >> > break; > >> > }; > >> > > >> >} /* end of initial routing logic */ > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++++++++++++ > >> ># COMIENZO DE LA SECCION DE RUTAS > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++++++++++++ > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >++++++++++++++++++ > >> ># Route 1 : Llamadas con destino plataforma > >> >H.323#+++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+++++++++++++++++++++++ > >> >route[1] > >> >{ > >> > # send it out now; use stateful forwarding as it > >> works reliably > >> > # even for UDP2TCP > >> > > >> > # Esta NATEADO? > >> > if (isflagset(5)) { > >> > log(1, "NAT: At least one participant NAT'd > >> --> enable reply > >> >processing (hacia GW)\n"); > >> > t_on_reply("1"); > >> > if (method=="INVITE") { > >> > log(1, "NAT: Invite received --> > >> enabling proxied > >> >media session (hacia GW)\n"); > >> > setflag(1); > >> > use_media_proxy(); > >> > }; > >> > }; > >> > > >> > rewritehostport("mygw.mydomain.com:5060"); > >> > append_hf("P-hint: GATEWAY\r\n"); > >> > > >> > if (!t_relay()) { > >> > sl_reply_error(); > >> > break; > >> > }; > >> > > >> >} > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+++++++++++++++++ > >> ># Route 4 : Llamadas a usuarios Off-line > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+++++++++++++++++ > >> > > >> >route[4] { > >> > > >> > if (!t_newtran()) { > >> > sl_reply_error(); > >> > }; > >> > > >> > if (!t_reply("404", "Not Found IT!")) { > >> > sl_reply_error(); > >> > }; > >> > break; > >> >} > >> > > >> > > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+++++++++++++++++ > >> ># On Reply Route 1 : Equipos NATeados!! > >> >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > >> +++++++++++++++ > >> >+++++++++++++++++ > >> > > >> >onreply_route[1] { > >> > if ((isflagset(5) || client_nat_test("3")) && > >> >(status=~"(180)|(183)|2[0-9][0-9]")) { > >> > if (!search("^Content-Length:\ 0")) { > >> > use_media_proxy(); > >> > }; > >> > }; > >> > > >> > if (client_nat_test("1")) { > >> > fix_nated_contact(); > >> > }; > >> > > >> > > >> >} > >> > > >> > > >> > > >> > > >> >Ricardo Martinez.- > >> > > >> >_______________________________________________ > >> >Serusers mailing list > >> >serusers(a)lists.iptel.org > >> >http://lists.iptel.org/mailman/listinfo/serusers > >> > >> -- > >> Jiri Kuthan http://iptel.org/~jiri/ > >> > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/ >

2 1

[Serusers] Windows Messenger and SER
by Jffdsaklj dsjk 21 Apr '05

21 Apr '05

I'm testing SER with Windows Messenger 4.7 and I have problems with the presence service. Those contacts connected when the user connects, appear as connected (correct). But when another contact connects, it keeps appearing as disconnected, and after a long time (more or less one minute) shows the contact as connected. On the other hand, all the contacts appear as disconnected with Windows Messenger 5.1, independently that the contact is previously connected or connects later. Has someone problems with windows messenger?. I've checked that these contacts are connected in SER with "serctl ul show". So I imagine that it's some type of incompatibility with windows messenger.

6 8

[Serusers] ser sizing
by Ahmad Cheikh Moussa 21 Apr '05

21 Apr '05

Hi! Can anyone tell me which Hardware requirements I need for a SER server ? Is there anywhere a documentation for this ? Regards, Ahmad -- Ahmad Cheikh-Moussa NetUSE AG Dr.-Hell-Straße, 24107 Kiel, Germany Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499 Service: Service(a)NetUSE.DE -- http://NetUSE.DE/

2 3

[Serusers] RE: [Sipforum-discussion] regarding Authentication...
by Paul van Schagen 21 Apr '05

21 Apr '05

-----Oorspronkelijk bericht----- Van: Paul van Schagen [mailto:paul.van.schagen@adaline.net] Verzonden: donderdag 21 april 2005 9:02 Aan: 'Josekutty K.K' Onderwerp: RE: [Sipforum-discussion] regarding Authentication... Methods you really want to authenticate are INVITE ( outgoing calls ) and REGISTER ( incoming calls ). You try the request first and the switch answers requesting authentication. Then the authentication is sent by the UA to the switch. Remember the authentication also verifies the IP address of the sender is correct. By doing a double request you make sure that the original request came from where they say it came from. Lets say somebody fakes an IP address and sents the packet to the switch. The switch then sends back an authentication requests which arrives at another destination. The call never gets authenticated correctly and is rejected. Hopes this helps Can anybody help me with my mediaproxy timeout ? Best regards Paul van Schagen -----Oorspronkelijk bericht----- Van: sipforum-discussion-bounces(a)lists.su.se [mailto:sipforum-discussion-bounces@lists.su.se] Namens Josekutty K.K Verzonden: donderdag 21 april 2005 8:47 Aan: sipforum-discussion(a)lists.su.se Onderwerp: [Sipforum-discussion] regarding Authentication... Hi all, I am a beginner in SIP. Let me explain my doubt. Once a SIP request authenticates itself, (usually a REGISTER request), all further requests MUST include the authorization details .. I mean I MUST send the authorization details to subsequent requests say OPTIONS, INVITE etc . Or Should I try each request without authentication first? Regards Josekutty K K ------ To unsubscribe from the Sipforum-discussion mailing list visit https://mbox.su.se/mailman/listinfo/sipforum-discussion

1 0

[Serusers] Commercial SER Support
by Darren Nay 21 Apr '05

21 Apr '05

Hey All, Do any of you know of a reliable / knowledgeable source for commercial technical support for SER. If so, we may be interested. We have worked in the past directly with IPTel, and have discussed support options recently with Jiri. He suggested that we ask the fine folks of the SER Users mailing list for their suggestions. Thanks so much, Darren Nay Ionosphere, Inc dnay(a)ionosphere.net (864) 678-3158

3 2

[Serusers] how to get call duration in radius accounting stop
by Kamran Ahmad 21 Apr '05

21 Apr '05

hi how to get call duration in radius. there should be callduration in accounting stop but here in my case there is no callduration Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 8 User-Name = "4000@ser" Calling-Station-Id = "sip:4000@ser" Called-Station-Id = "sip:3000@ser" Sip-Translated-Request-URI = "sip:3000@ser" Acct-Session-Id = "CB13EC9C-9DB4-420F-BEF4-786E6398D714@UA" Sip-To-Tag = "as23df87de" Sip-From-Tag = "290015013442" Sip-Cseq = "2" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "ca12eca5fce40009" Timestamp = 1112912023 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Serusers] Ser 0.9.0 and Cisco Voice Gateway
by Ahmad Cheikh-Moussa 21 Apr '05

21 Apr '05

Hi Steve! Now it works properly. Like you wrote the port command was missed in the dial peer. dial-peer voice 1 pots description Default-Dial-peer fuer ausgehende Anrufe preference 3 service session max-conn 25 destination-pattern 0T progress_ind alert enable 8 direct-inward-dial port 0/0/1:15 Now I have activated the www_authorize section. # Uncomment this if you want to use digest authentication if (!www_authorize("netuse.de", "subscriber")) { www_challenge("netuse.de", "1"); break; }; On my cisco I see following when I enable the ccsip debugging: Apr 21 09:24:52.933 MEST: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 192.168.254.30:5060 Apr 21 09:24:52.933 MEST: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000 Apr 21 09:24:52.933 MEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.203:5060;branch=z9hG4bK2924EA From: "0*" <sip:0*@192.168.254.30>;tag=E7AF75C-13AE To: "0*" <sip:0*@192.168.254.30>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2283 Call-ID: E1A23104-B0DA11D9-81B582BA-75F29624 CSeq: 42 REGISTER WWW-Authenticate: Digest realm="netuse.de", nonce="426756705993d4a9785e2e828158f76faa5234d4", qop="auth" Server: Sip EXpress router (0.9.0 (sparc64/solaris)) Content-Length: 0 Warning: 392 192.168.254.30:5060 "Noisy feedback tells: pid=16087 req_src_ip=192.168.254.203 req_src_port=52773 in_uri=sip:192.168.254.30:5060 out_uri=sip:192.168.254.30:5060 via_cnt==1" Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table. Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x45D7009C key=E1A23104-B0DA11D9-81B582BA-75F29624 Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 45D7009C It looks like the cisco tries to registering to the server, but the server decline this. How can I enable the registering of my gateway in ser, without to disable the www_authorize ? Regards, Ahmad On Apr 20, 05, Steve Blair wrote: > > > Ahmad Cheikh Moussa wrote: > > >Hi! > > > >> > >>>Can I ask you some questions? > >>>How do I know, that my ser server only act as registrar and > >>>redirect Server ? With redirect I mean, any other calls, which > >>>starts with 0, redirect to the gateway. > >> > >> > >> > >>I'm not sure I understand. Are you concerned about people outside > >>of your user community routing calls through your proxy and out to the > >>PSTN? If so then you need to use check_from and check_to to > >>determine if the caller is authenticated to your proxy. > > > >I have following line in ser.cfg > >t_relay_to_udp("192.168.254.203", "5060"); > >when I understand you right, then I have to change this paramater to > >sl_relay_to_udp("192.168.254.203", "5060"); > > > No, the t_relay is fine. > > >This would mean stateless forwarding to this gateway ? > >correct ? > > > >Regars, > > Ahmad > > > > -- > > ISC Network Engineering > The University of Pennsylvania > 3401 Walnut Street, Suite 221A > Philadelphia, PA 19104 > > > voice: 215-573-8396 > > 215-746-8001 > > fax: 215-898-9348 > > sip:blairs@upenn.edu > -- Ahmad Cheikh-Moussa NetUSE AG Dr.-Hell-Straße, 24107 Kiel, Germany Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499 Service: Service(a)NetUSE.DE -- http://NetUSE.DE/

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