sr-users October 2004

sr-users@lists.kamailio.org

189 participants
408 discussions

by Michael Shuler

I have the statement: If(src_ip == 192.168.1.80/27) { # Do something cool } Else { # Don't do something cool }; When the src_ip (which I verified by xlog) is 192.168.1.85 it doesn't match and goes to the else code. Am I doing something wrong? This is on 0.8.14. ---------------------------------------- Michael Shuler, C.E.O. BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP) 682 High Point Lane East Peoria, IL 61611 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: mike(a)bwsys.net Customer Service: (877) 976-0711

19 years, 7 months

[Serusers] Sipura SPA-2000

by Rodrigo P. Telles

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Folks, I've been experimenting Sipura SPA-2000 and I realised that it doesn't support G.729 in both lines simultaneously.. or be.. when I call to the first line it answers using the preferred codec (G.729) and at the same time I call to the second line and it answer using G711a. I tried to disable other codecs in the second line configuration (just G.729 enable) and tried again and then SPA-2000 answered as busy. I'm using the last firmware version (2.0.10(e)). Anyone have any idea/explanation to solve/about that? Thanks in advance. - -- ============================================ Rodrigo P. Telles <telles(a)devel-it.com.br> Project Manager Devel-IT - http://www.devel-it.com.br TDKOM Group ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBXXztiLK8unYgEMQRAtcKAJ4qR//3K4Ak4ocfto8EDuAbNWUU7wCeMUah Mx+uhzc6yi3Ko+ahGOBVJoE= =vMn7 -----END PGP SIGNATURE-----

19 years, 7 months

[Serusers] Associate an USERNAME with one or more number of phone-number

by Nicolas Ruiz

Hello, Is it possible to associate an Username with only one or more Phone Number ? My problem is when a UA send an invite with a BAD phone Number, he have to be stoped by SER. Is the "From" field, that i want to secure. U 62.39.69.11:20187 -> 80.118.128.5:5060 INVITE sip:0156383971@sip.vivaction.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 62.39.69.11:20187;branch=z9hG4bKacvxAsGIo..From: < sip:156384290@sip.vivaction.net>;tag=1c10095..To: <sip:0156383971@sip.vivaction.net>..Call-ID: 114402722127221Jakp-156384290--0 156383971@62.39.69.11..CSeq: 46418 INVITE..Contact: <sip:156384290@62.39.69.11:20187;user=phone>..Proxy-Authorization:Digest us ername="*********",realm="sip.vivaction.net",nc=00000001,cnonce="",nonce="41 5d5a9abfaa92e7223274a4cccbcdf474bededf",uri="sip:01 56383971(a)sip.vivaction.net",qop=auth,Algorithm="MD5",response="9b571eeddfd5c 5b56e8b48450b22b826"..Supported: em,timer..Accept-L anguage: en..Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE..Us er-Agent: Audiocodes-Sip-Gatewa y/MP-104 FXS/v.4.20.299.4192..Content-Type: application/sdp..Content-Length: 209....v=0..o=AudiocodesGW 13428 10067 IN IP4 62.3 9.69.11..s=Phone-Call..c=IN IP4 62.39.69.11..t=0 0..m=audio 20202 RTP/AVP 18 96..a=rtpmap:18 g729/8000..a=rtpmap:96 telephone-e vent/8000..a=fmtp:96 0-15..a=ptime:20.. Thanks a lot Best Regards Nicolas RUIZ

19 years, 7 months

[Serusers] Session-Timer at proxy level?

by kaiser

Hi, I am looking for a proxy with timer too. What proxy you test get this feature? Thanks kaiser

19 years, 7 months

[Serusers] What parameters with can be forwarded when using the exec command ?

by Wilhelm Farrugia

Hello, Would like to execute a script upon receipt of a bye request, and would like to forward the acct_session_id of the received packet to the executable. Is there a way with which this can be achieved ? Thank you, Regards, Wilhelm

19 years, 7 months

[Serusers] Billing

by Evgeny Ivanov

Hi all, I was wondering if you can point me a SER prepaid billing solution. Thank you in advance Evgeny Ivanov

19 years, 7 months

[Serusers] only one side voice with using ser

by ser die

hello firends, iam using sip-communicator with sip express router its registering and ringing but i could able to listen the voice with only one side i.e the destination could able to hear what source is saying and source number could not here any thing the dailer iam running in the public ips even when i test with the fxs the source i.e dialer couldnot able to listen any thing when session starts fxs endpoint could able to listen what dialer party is saying iam attaching the sip-communicator.xml here so if possible please tell me where am i going wrong? <?xml version="1.0" encoding="UTF-8"?> <configuration> <log4j> <rootLogger value="net.java.sip.communicator.common.Console.TraceLevel, RFLogger"/> <appender> <RFLogger value="org.apache.log4j.RollingFileAppender"> <layout value="org.apache.log4j.PatternLayout"> <ConversionPattern value="%r [%t] %p %c{2} %x - %m%n"/> </layout> <MaxBackupIndex value="1"/> <File value="log/sip-communicator.app.log"/> <MaxFileSize value="256KB"/> </RFLogger> </appender> </log4j> <net> <java> <sip> <communicator> <FIRST_LAUNCH value="true"/> <ENABLE_SIMPLE value="false"/> <media>  <PREFERRED_AUDIO_ENCODING value="0"/> <PREFERRED_VIDEO_ENCODING value="26"/> <MEDIA_SOURCE value=""/> <MEDIA_BUFFER_LENGTH value="100"/> <IP_ADDRESS value=""/> <AUDIO_PORT value=""/> <VIDEO_PORT value=""/> </media> <sip> <PUBLIC_ADDRESS value=""/> <TRANSPORT value=""/> <REGISTRAR_ADDRESS value="*.*.*.19"/> <USER_NAME value=""/> <STACK_PATH value="gov.nist"/> <PREFERRED_LOCAL_PORT value=""/> <DISPLAY_NAME value=""/> <REGISTRAR_TRANSPORT value="UDP"/> <REGISTRATIONS_EXPIRATION value="3600"/> <REGISTRAR_PORT value="5060"/> <FAIL_CALLS_ON_DEST_USER_MISMATCH value="false"/> <DEFAULT_DOMAIN_NAME value="*.*.*.19"/> <DEFAULT_AUTHENTICATION_REALM value="*.*.*.19"/> <WAIT_UNREGISTGRATION_FOR value="1100"/> <SAME_USER_EVERYWHERE value="true"/> <simple> <CONTACT_LIST_FILE value="contact-list.xml"/> <SUBSCRIPTION_EXP_TIME value="600"/> <MIN_EXP_TIME value="120"/> <LAST_SELECTED_OPEN_STATUS value="online"/> </simple> </sip>  <sipphone> <IS_RUNNING_SIPPHONE value="false"/> <MY_SIPPHONE_URL value="http://my.sipphone.com"/> </sipphone>  <gui> <AUTH_WIN_TITLE value="SIP Authentication!"/> <AUTHENTICATION_PROMPT value="Please enter login name and password for the specified realm:"/> <USER_NAME_LABEL value="User Name:"/> <USER_NAME_EXAMPLE value="Example: 1-747-555-1212"/> <PASSWORD_LABEL value="Password:"/> <GUI_MODE value="PhoneUiMode"/>  <imp> <CONTACT_LIST_X value=""/> <CONTACT_LIST_Y value=""/> <CONTACT_LIST_WIDTH value=""/> <CONTACT_LIST_HEIGHT value=""/> </imp> </gui> <common> <PREFERRED_NETWORK_INTERFACE value=""/> <PREFERRED_NETWORK_ADDRESS value=""/> </common>  <STUN_SERVER_ADDRESS value="stun01.sipphone.com"/> <STUN_SERVER_PORT value="3478"/> <VOICE_MAIL_ADDRESS value="17475551212"/> </communicator> </sip> </java> </net> <gov> <nist> <javax> <sip> <SERVER_LOG value="log/sip-communicator.stack.log"/> <TRACE_LEVEL value="16"/> </sip> </javax> </nist> </gov> <javax> <sip> <IP_ADDRESS value=""/> <STACK_NAME value="sip-communicator"/> <ROUTER_PATH value="net.java.sip.communicator.sip.SipCommRouter"/> <OUTBOUND_PROXY value="*.*.*.19:5060/udp"/> <RETRANSMISSON_FILTER value=""/> <EXTENSION_METHODS value=""/> <RETRANSMISSION_FILTER value="true"/> </sip> </javax> <java> <net> <preferIPv4Stack system="true" value="true"/> <preferIPv6Addresses system="true" value="false"/> </net> </java> </configuration> with regards serdiehard __________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail

19 years, 7 months

[Serusers] serweb accounting for cisco ata 186

by Jaime G. Ghirelli

I am running ser 0.8.12 and serweb_2004-01-04/ Serweb accounting works fine for xten softphones, but ciscos atas records are not shown. Any ideas on how to modify the serweb select query to show them ? The difference seems to be the aditional ";user=phone" on the sip_from, sip_to, and to_uri fields for atas example acc table record for cisco ata 186 <sip:8840@voip.brujula.net;user=phone>;tag=2306858... <sip:00541148029550@voip.brujula.net;user=phone> 200 ACK sip:00541148029550@200.68.120.81 sip:00541148029550@217.199.177.150 sip:8840@voip.brujula.net;user=phone sip:00541148029550@voip.brujula.net;user=phone 146901752(a)200.126.199.228 8840 voip.brujula.net 2306858569 n/a 2004-08-21 19:14:48 20040821161448 serweb accounting.php $q="select t1.to_uri, t1.sip_to, t1.sip_callid, t1.time, ". "t1.fromtag as invft, t2.fromtag as byeft, t2.totag as byett, ". "sec_to_time(unix_timestamp(t2.time)-unix_timestamp(t1.time)) ". "as length, ". "unix_timestamp(t2.time)-unix_timestamp(t1.time) ". "as seconds ". "from ".$config->table_accounting." t1, ". $config->table_accounting." t2 ". "where t1.username='".$auth->auth["uname"]."' and ". "t1.domain='".$config->realm."' and ". "t1.sip_callid=t2.sip_callid and ". "t1.sip_method='INVITE' and t2.sip_method='BYE' ". "order by t1.time desc"; $mc_res=mySQL_query($q); Thanks for the info Jaime Garcia http://voip.brujula.net

19 years, 7 months

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sr-users October 2004