sr-users February 2009

sr-users@lists.kamailio.org

76 participants
97 discussions

[Kamailio-Users] help me in ldap kamailio
by Husna Bte Mad Baguri 24 Feb '09

24 Feb '09

Can someone help me,i counter this error when try to install ldap in kamailio Compiling ldap_api_fn.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"ldap"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_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 -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c ldap_api_fn.c -o ldap_api_fn.o ldap_api_fn.c:38:18: error: ldap.h: No such file or directory In file included from ldap_api_fn.c:40: Plzzzzz help me regards Husna This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

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[Serusers] Cleaning bad npdi=yes and rn=VALUE tags from INVITEs
by Frank Durda IV 24 Feb '09

24 Feb '09

I have a client who occasionally sends calls where there is a npdi=yes tag (and there may or may not be a rn=VALUE tag present with it), BUT the answer these two tags are providing is wrong! That is, the number in question has or hasn't been ported, but the information provided in the INVITE message doesn't match reality. Either a rn=VALUE should have had value X and it instead has value Y, OR there is no rn=VALUE at all, and there should have been one, OR the number hasn't actually been ported, and no rn=VALUE should have been sent at all but we got a junk one, that probably isn't the right LRN for the true destination of the call. Because of these wrong npdi=yes tags, our PSTN gateway switch ends up failing these incorrectly-marked call, because it takes the provided rn=VALUE or lack of one to be correct and so it doesn't do its own LNP query, and the one provided in the SIP INVITE is sometimes just flat out wrong. If the PSTN gateway switch had ignored these suspect npdi=yes and rn=VALUE tages and simply done the LNP query itself, in virtually all cases these otherwise doomed calls would have completed successfully. For this particular customer, the problem of bogus values appears to be a permanent artifact. They either don't know how to fix it, or don't think it is important, of they are spoofing it to avoid being billed for LNP queries. Whatever the reason, it causes trouble tickets. So, we would like to have SER strip the npdi=yes and any rn=VALUE tags from the INVITE when it is this customer (known by src ip), so that the PSTN switch will do the LNP query itself on the SS7 side and get the call routing right. (We have already determined that getting the outfit that is sending the junk to do the right thing isn't going to happen any time soon.) Does anybody have any suggestions as how to write the ser.cfg code to edit-out a npdi=yes and rn=VALUE if they are present in INVITEs? We just want to always throw npdi=yes and any rn=VALUE that shows up if the call is coming from this given customer, basically treating all npdi=yes and rn= tags coming from this customer as being untrustworthy. There has to be an easier way than stripping the thing down and reassembling a new one with repeated strip(1) type operations, which looks to be extremely inefficient for this task and probably really meant only for stripping country codes and such off the phone numbers, not for cleaning up bad tags. Thanks, in advance!

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Re: [Kamailio-Users] LCR in Kamailio 1.4
by Ovidiu Sas 23 Feb '09

23 Feb '09

Make sure that from_uri is NULL and not a white space. Please copy the mailing list in your reply. Regards, Ovidiu Sas On Thu, Feb 19, 2009 at 11:25 AM, Asim Riaz <ariaz78(a)gmail.com> wrote: > > > On Thu, Feb 19, 2009 at 4:16 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >> Is your R-URI starting with a '0'? >> You have defined a single route for all '0' prefix destinations. > > yes R-URI is starting with 0. >> >> >> Regards, >> Ovidiu Sas >> >> On Thu, Feb 19, 2009 at 11:06 AM, Asim Riaz <ariaz78(a)gmail.com> wrote: >> > >> > >> > On Thu, Feb 19, 2009 at 3:53 PM, Daniel-Constantin Mierla >> > <miconda(a)gmail.com> wrote: >> >> >> >> Hello, >> >> >> >> On 02/19/2009 05:34 PM, Asim Riaz wrote: >> >>> >> >>> Hello, >> >>> I am using LCR, for some reason load_gws() function is not finding the >> >>> gateway in DB, it was working in the start then added some more routes >> >>> and >> >>> after that everything stopped working, I am not sure where to start >> >>> troubleshooting, any idea ? >> >> >> >> run kamailio with higher debug mode (e.g., debug=5) in configuration >> >> file. >> >> >> >> You should get more debug messages in the syslog, that may give you >> >> some >> >> hints. look for lines containing 'lcr'. >> >> >> >> Cheers, >> >> Daniel >> > >> > Hi Daniel, >> > I tried higher debug level, not getting anything related to LCR all i >> > am >> > getting XLOG messages "unable to find Gateway". >> > >> > Thanks, >> > >> > Asim >> > >> >> >> >>> >> >>> kamctl lcr show >> >>> >> >>> lcr routes >> >>> +----+--------+----------+--------+----------+ >> >>> | id | prefix | from_uri | grp_id | priority | >> >>> +----+--------+----------+--------+----------+ >> >>> | 9 | 0 | | 1 | 1 | >> >>> +----+--------+----------+--------+----------+ >> >>> lcr gateways >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> | gw_name | ip_addr | port | uri_scheme | transport | grp_id >> >>> | >> >>> strip | tag | flags | >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> | XXXX | XXXX | 5060 | 1 | 1 | 1 | >> >>> 0 >> >>> | | 0 | >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> >> >>> >> >>> >> >>> my routing script >> >>> >> >>> if(!load_gws()) { >> >>> xlog("L_INFO", "[ROUTE-X] Unable To Find Gateway "); >> >>> sl_send_reply("503", "unable to find gateway"); >> >>> exit; >> >>> } >> >>> Thanks in advance. >> >>> >> >>> Asim >> >>> >> >>> ------------------------------------------------------------------------ >> >>> >> >>> _______________________________________________ >> >>> Kamailio (OpenSER) - Users mailing list >> >>> Users(a)lists.kamailio.org >> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> >> Daniel-Constantin Mierla >> >> http://www.asipto.com >> >> >> > >> > >> > _______________________________________________ >> > Kamailio (OpenSER) - Users mailing list >> > Users(a)lists.kamailio.org >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > > >

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[Kamailio-Users] problem with 200ok
by BERGANZ François 23 Feb '09

23 Feb '09

Hello, I have Asterisk1---SER---Asterisk2. When I do INVITE from the left, --the asterisk2 send 200ok to the SER --the SER forward to the Asterisk1 --but the asterisk1 directly send the ACK to Asterisk2 Asterisk2 retransmit the 200ok… and error. I think that it need that the ACK come from the SER and not directly from the Asterisk1. So, how can I detect a 200ok and reply a ACK with my SER? Or, anyone have ever seen that problem? Thank you Next: my capture problem SIP/2.0 183 Session Progress Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653753 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 6§¢Iœÿ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653753 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv SIP/2.0 200 OK v: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.0 v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060 f: "francois berganz"<sip:170725014@IP_ASTERISK>;tag=as1551f6d3 t: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef i: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 INVITE Require: timer x: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I NFO m: <sip:URI_TEST@192.168.1.82:5060;user=phone> Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold c: application/sdp l: 206 v=0 o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82 s=- c=IN IP4 192.168.1.82 t=0 0 m=audio 41000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 200 OK v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060 f: "francois berganz"<sip:170725014@IP_ASTERISK>;tag=as1551f6d3 t: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef i: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 INVITE Require: timer x: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I NFO m: <sip:URI_TEST@IP_PHONE:5060;user=phone> Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold c: application/sdp l: 206 v=0 o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82 s=- c=IN IP4 192.168.1.82 t=0 0 m=audio 41000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ACK sip:URI_TEST@IP_PHONE:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Max-Forwards: 70 From: "francois berganz" <sip:170725014@IP_ASTERISK>;tag=as1551f6d3 To: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef Contact: <sip:170725014@IP_ASTERISK> Call-ID: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653754 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv SIP/2.0 200 OK Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653754 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ACK sip:URI_TEST@IP_PHONE:5060;user=phone SIP/2.0 Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Via: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.2 Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport=5060 Max-Forwards: 70 From: "francois berganz" <sip:170725014@IP_ASTERISK>;tag=as1551f6d3 To: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef Contact: <sip:170725014@IP_ASTERISK> Call-ID: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 ACK sip:URI_TEST@IP_ASTERISK SIP/2.0 Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK7a6d7cf4;rport From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Contact: <sip:170725014@IP_PROVIDER_ASTERISK> Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 ACK User-Agent: MARS Max-Forwards: 70 Content-Length: 0 Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] NAT traversal
by BERGANZ François 23 Feb '09

23 Feb '09

Hello, Is it possible to do NAT traversal without RTP-PROXY ? If yes, have you an example? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] how to prepend area code into a local call phone number
by Jinsong Hu 23 Feb '09

23 Feb '09

Hi, Sir: In the US, assume a person from 1-408-1234567 is dialing phone number 1234568. how do I figure out the caller's country code and area code and prepend to the local number ? I tried prefix($(from.user{s.substr, 0, 4})); and openser reported as parser error in config file, bad argument, string expected. Jimmy

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[Kamailio-Users] Using set_advertised_address with variable
by catalina oancea 23 Feb '09

23 Feb '09

Hi Does anybody know if there is a way to use the set_advertised_address function with an avp or variable parameter instead of plain string? I need to change the IP in the Via header with the value of a variable. Thanks Catalina

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[Kamailio-Users] missing file ldap.h
by Husna Bte Mad Baguri 23 Feb '09

23 Feb '09

Dear Whom May Concern, When I try to compile ldap module in kamailio I get error,ldap.h :No such file or directory. Where can I get that file.please someone help me. Regards Husna ________________________________ This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

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Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Daniel-Constantin Mierla 20 Feb '09

20 Feb '09

Hello, On 02/13/2009 12:08 PM, Abdul Hakeem wrote: > Hi, > Are you able to put me in touch with the guys using the SDP parser ? > direct your emails to mailing list all the time. Those people are there, the developer of SDP parser and others using it. You will get much faster response and maybe better than writing private emails. Thanks, Daniel > Cheers, > Abdul Hakeem > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: 11 February 2009 15:20 > To: alhakeem(a)ipextelecom.net; kamailio > Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc > > Hello, > > On 02/11/2009 03:20 PM, Abdul Hakeem wrote: > >> Hi, >> >> Just wanted to touch base on the parser to find out if you are still >> pursuing this. >> >> > it is still on my todo list, however didn't make it in the upcoming 1.5.0. > Ovidiu Sas did some commits to the SDP parser and I know couple of others > using it with success. > > Please open a feature request at: > http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=browse > > It is easier to track and not forgot about it for the next release. > > Thanks, > Daniel > > > >> I am in the mind of purchasing some Cisco equipment to do the Sip-MAP >> gateway from a GSM network. >> I have someone else from the OpenBTS with me on this. >> The object is to provide a full fledged sip based routing of voice, >> sms and multimedia calls from gsm handsets to BTS with a Sip server >> > backend. > >> Cheers, >> Abdul Hakeem >> >> -----Original Message----- >> From: users-bounces(a)lists.kamailio.org >> [mailto:users-bounces@lists.kamailio.org] On Behalf Of >> Daniel-Constantin Mierla >> Sent: 03 September 2008 17:45 >> To: Kristian Kielhofner >> Cc: users(a)lists.kamailio.org >> Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc >> >> Hello, >> >> On 09/03/08 19:10, Kristian Kielhofner wrote: >> >> >>> Hello everyone, >>> >>> Feel free to set $OSSPROXY to whatever you like = >>> Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). >>> >>> $OSSPROXY is my favorite SIP toolkit. In addition to being an >>> excellent proxy and registrar it serves as a platform to do SIP >>> mangling and wrangling. With that in mind let me propose a crazy >>> idea... >>> >>> How hard would it be to get $OSSPROXY to be able to parse what is >>> commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has >>> no business getting too involved in the SDP but I feel it makes sense >>> in some instances (like this one). What if I want to make a routing >>> decision based on an encapsulated ISUP parameter of some sort? Who >>> knows what it could be!?!? Maybe there are some other uses but that >>> makes the most sense and seems to be the most practical. It should >>> be possible to develop an $OSSPROXY module to: >>> >>> - Use the SDP parser to find the ISUP SDP part: >>> >>> http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing >>> >>> - Pass that part into a "real" ISUP parser (if necessary). I >>> couldn't seem to find the perfect library but perhaps something like >>> this could >>> work: >>> >>> http://www.openss7.org/isup.html >>> >>> - Make the ISUP parameters available via AVPs, pseudo variables, etc >>> for use in the $OSSPROXY script. >>> >>> I don't really have a need for this but it seems like it could be >>> cool. Am I nuts? >>> >>> >>> >>> >> I would expose the body as a dynamic structure, so components will be >> accessible in the script, e.g. (naming schema just for example), >> >> if($body(0=>content-type)=="application/sdp") >> { >> if($body(0=>m=>type) == "audio") >> { >> } >> } >> >> Based on the available content type parsers (sdp, isup), the structure >> has different attributes. >> >> Cheers, >> Daniel >> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] lookup("location") problem
by BERGANZ François 19 Feb '09

19 Feb '09

Hello, I do as the example in the module registrar doc: lookup("location"); switch ($?) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; And SER report me ERROR:core:pv_parse_spec: wrong char [$/36] in [$?] at [0 (0)] Know you why? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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sr-users February 2009