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37526 discussions

[Serusers] Radius accounting - User-Name parameter in the Start Acct-Status-Type
by tzieleniewski 09 Feb '07

09 Feb '07

Hi! Is it possible to configure acc_radius in such a way that it will send a User-Name parameter in the Accounting Packet because now it doesn't: rad_recv: Accounting-Request packet from host 127.0.0.1 port 32844, id=246, length=474 SER-Attr = "" Acct-Session-Id = "6012E956-D8D3-2367-979B-753579FE59FF(a)192.168.1.2" Sip-To-Tag = "572241288" SER-From = "tomix <sip:tomix@tezet.no-ip.org:7060>;tag=2020612254" SER-Flags = 8 SER-Original-Request-ID = "sip:hellboy@tezet.no-ip.org" Sip-Method = "INVITE" Sip-Cseq = 20886 Sip-Translated-Request-ID = "sip:hellboy@192.168.1.10:5060" Sip-Source-IP-Address = 192.168.1.2 Sip-From-Tag = "2020612254" SER-To = "<sip:hellboy@tezet.no-ip.org>;tag=572241288" SER-Digest-Username = "tomix" SER-Request-Timestamp = 1171023334 Calling-Station-Id = "sip:tomix@tezet.no-ip.org:7060" Sip-Source-Port = 7060 SER-Digest-Realm = "tezet.no-ip.org" Sip-Response-Code = 200 Called-Station-Id = "sip:hellboy@tezet.no-ip.org" SER-Response-Timestamp = 1171023341 Acct-Status-Type = Start Service-Type = IAPP-Register NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Cheers Tomasz

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[Serusers] SER 0.10.99 (Carpo) - NAT call - long ringing after hang up
by Theodor Meier 09 Feb '07

09 Feb '07

hello, using my fritzbox fon with an account from Carpo, ser 0.10.99, to call a cisco 7960 behind the same fritzbox fon, that uses as well an Carpo account, an hang up, the called cisco 7960 continues ringing for a minute. (fritzbox - ser - (same) fritzbox - 7960) Vice versa the same. (7960 - fritzbox - ser - fritzbox) Not so when I use the 7960 with a sipgate account (7960 - sipgate - telefonica gateway - Carpo ser - fritzbox) to call the fritzbox. Is this a ser problem? Perhaps a problem, when the call-phone and the called-phone use the same IP? Or is this a configuration problem of ser? What can I debug to find where the problem is? ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

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[Users] RTP fixes for mediaproxy
by Jeff Williams 09 Feb '07

09 Feb '07

Hi, I was having a few issues with mediaproxy when it came to re-invites and call setup where the SDP does not get sent with the first INVITE. I noticed this was due a couple couple of assumptions in mediaproxy which don't always hold: 1) That SDP is always set up first by the caller (i.e. in the INVITE) then updated by the called party (i.e. in the 200). This is not the case. The first SDP can be sent in the 200 response by the called party and the caller send their SDP in the ACK. 2) That RTP streams don't change once set up. This is wrong for reINVITEs. This would cause problems when a user agent sent a reINVITE to redirect RTP to on hold music for example - the RTP change would be ignored. I think this had an impact on some fax setups as well, but I haven't tested it. I found a patch to fix this by storing the streams in a hash by media type, but that is a limiting assumption also (maybe I want 2 audi streams?). The attached patch fixes this by changing the following: 1) This pathc allows a mediaproxy RTPSession to be set up by either the caller or called party. A 'caller' argument is passed to the RTPSession functions telling it if the caller or called party is doing the setup or update. 2) The official mediaproxy does not even look at SDP details for streams which are already set up. This patch lets every SDP packet update all of the RTP streams for that party (either caller or called). If you have issues with one-way-audio with media proxy after reINVITEs, have calls the first SDP from the called party or are just plain adventurous, please give it a go. It has been used in a production environment for a bit over a month now. The patch was made against mediaproxy 1.7.2, but will work against 1.8.0 as well since none of the rtphander.py code changed. Regards, Jeff

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[Serusers] failed to load mysql.so module
by Josef.Bajada＠go.com.mt 08 Feb '07

08 Feb '07

Hi, I am having this problem with the mysql module on ser. 0(31212) ERROR: load_module: could not open module </opt/ser/modules/mysql/mysql.so>: /opt/ser/modules/mysql/mysql.so: undefined symbol: log 0(31212) parse error (30,13-14): failed to load module ERROR: bad config file (1 errors) I can confirm that mysql.so exists in that directory. I have mysql 5.0.27 installed. I have already tried to clean and make all from scratch but the problem remained. Any ideas how to solve this? Thanks a lot, Josef

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[Users] set RURI with dispatcher module
by Klaus Darilion 08 Feb '07

08 Feb '07

Hi! I want to use the dispatcher module to use load-balancing/failover to gateways. Thus, I also want to change the domainpart of the RURI to be the IP address of the gateway - but dispatcher only sets the DURI. Is there an easy way to set the RURI/domain too or do I have to use: avp_write("$rU","$avp(i:666)"); ds_select_dst(...); avp_pushto("$ru","$du"); avp_pushto("$ru/username","$avp(i:666)"); Is there a more easy way? btw - I think above workaround would not work in failure_route as ds_next_dst adds a new branch but avp_pushto works against branches[0] - correct? thanks klaus

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[Users] SIP Proxy - CAC
by Michel Bensoussan 08 Feb '07

08 Feb '07

Hello, I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization, for example according to codec, and allow or disallow new sessions, depending on resources. I'm thinking using OpenSER for this purpose. On a first stage I want to install a SIP Proxy on the AP (embedded with OS Linux) and on a second stage adding the CAC feature. There is many modules in OpenSER and I'm not sure which one I should use for a SIP Proxy. I'm learning about SIP but my knowledge is limited (till last week I never heard about it). Thank you for advising. Regards, Michel.

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[Serusers] 0.9.x mysql.so: undefined symbol: log error
by Steve Blair 08 Feb '07

08 Feb '07

Hi: I keep getting the following error on a 0.9.x system. The latest 0.9.x version of SER was just download and built on this server but I continue to get this same error. I've check the mysql Makefile and it seems to be correct. What else could cause this error? ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Thanks,Steve

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[Serusers] sequential hunting
by Steve Blair 08 Feb '07

08 Feb '07

Hello: I would like to implement sequential hunting. Hunting is when a call arrives for a given subscriber, no one answers the original number so the proxy populates the username portion of the r-uri with another number (from the database) and tries again. This process continues until all numbers in the database are exhausted. I have the "failover" working but the r-uri field is not updated on any interation though the failure_route. Does anyone have any ideas how to make the following work? Thanks,Steve # hunting (serial forking) if (avp_db_load("$ruri/username", "s:allow_hunt")) { if (avp_check("s:allow_hunt", "eq/y/i")) { avp_delete("s:allow_hunt"); xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Enabled for <%ru>\n"); avp_db_load("$ruri/username", "s:huntlist/sam"); xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Started to <%ru>\n"); t_on_failure("6"); t_relay(); break; }; }; # Iterate through extension list for subscribers with call hunting enabled failure_route[6] { xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Failure Block #6: HUNTING for <%ru> from <%fu> at <%is>\n"); if (avp_pushto("$ruri/username", "s:huntlist/g")) { append_branch(); avp_delete("s:huntlist"); t_on_failure("6"); t_relay(); }; }

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[Users] RADIUS acc ok, RADIUS auth not
by Luca Corti 08 Feb '07

08 Feb '07

Hello, I'm having a hard time getting RADIUS auth to work at all. I've reached the point where I am debugging the network traffic with tshark. I can see RADIUS packets generated from the openser box for accounting, but no RADIUS authentication packets are sent. Also, it seems no response comes back from the RADIUS server even for accounting requests. radclient against my radius server returns me an Access-Reject at least and the tshark dump contains auth request and response. OpenSER 1.1.1 tells me: Feb 6 17:27:04 openser openser[22596]: rc_send_server: no reply from RADIUS server I suspect the issues lie in the radiusclient-ng layer. I'm using radiusclient-ng from Debian Etch (0.5.3-2), but I see many branches in CVS, like RELENG_0_5_5 which seems newer. thanks

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[Users] uac_replace_from() not work for Linksys ATA
by Howard Tang 08 Feb '07

08 Feb '07

HI All, I have a problem with uac_replace_from() function not working for Linksys Unit. This is what i have in the route[1], if (search("From:.*<sip:900*")) { uac_replace_from("anonymous","sip:anonymous@x.x.x.x"); } I have 0900xx as internal sip account, then I have to remove the 0900xx callerid before i send the call to PSTN gateway. Because 0900xx is not a valid number, Mobile phone providers block the call because of that, as a result, I need to replace the callerid to anonymous. I have tested with many other ATA and X-Lite without problem. The problem only ocurrs when i use Linksys ata. I have monitored the SIP message, the first request will change to anonymous then back to the original username. Anyone have an idea on how i can fix this ? -- Howard Tang ICQ : 259083 MSN : howard615(a)hotmail.com

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