sr-users July 2006

sr-users@lists.kamailio.org

206 participants
415 discussions

[Users] How to handle "302 Moved temporarily" with TCP transport and changing port
by Ingmar Kliche 12 Jul '06

12 Jul '06

Hi all, I'm very new to OpenSER and I would like to ask for your advice to find a solution for my problem: I have a SIP gateway (A) which uses UDP transport and a SIP endpoint (B) which only supports TCP. I'd like to use OpenSER to convert UDP to TCP. The second issue is that the TCP endpoint (B) will redirect the incoming INVITE to itself to another TCP port using a "302 Moved temporarily" containing a CONTACT header. Here is my approach: My current understanding is that I would have to add something to the routing block to select the traffic for my TCP endpoint. Then I would have to add route where I would have to handle the TCP forwarding, something like: if (uri=~"...") { t_relay_to_tcp("<ip>", "5060"); } This looks very straight forward and should solve the UDP to TCP issue as I understand it. Is this approach correct? For the second issue I don't have a solution right now. As described above the TCP endpoint (B) will answer something like SIP/2.0 302 Moved Temporarily FROM: <sip:tel@HOST_A>;tag=1c552078080 TO: <sip:tel@HOST_B>;tag=7189e7deae CSEQ: 1 INVITE CALL-ID: 552077216117200615500@HOST_A MAX-FORWARDS: 70 VIA: SIP/2.0/TCP HOST_A;branch=z9hG4bKac552089789;alias CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6011.0 CONTACT: <sip:tel@HOST_B:1223;transport=tcp;maddr=HOST_B;x-mss-call-id=552077216117200615500%40HOST_B> I want OpenSER now to send out another INVITE via TCP to HOST_B Port 1223 (but the port changes for each call). In principle I would need a possiblity to extract the host and the port out of the CONTACT header of the message to send another t_relay_to_tcp("<host>", "<port>"). But how to do that? Or is there another possibility to tell OpenSER to take the target info out of the CONTACT header? Thanks in advance! Ingmar

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[Serusers] Textops with AVPs?
by sip 12 Jul '06

12 Jul '06

Is there a version of textops that can do substs with AVPs that will work on SER 0.9.6 or is that an openSER-only modification of the code? I'm curious because we're ALSO running into the issues of charging the call-forwarding user for forwarding a call to the PSTN instead of charging the calling party. Ideally, I'd like to rewrite the from address solely for the purpose of authenticating the user who's doing the forwarding and charging him for the call, but that would likely break things as there'd be no way to get back to the original user if I just rewrote the from username. SO, I thought, why not let the b2bua handle the details and just forward a uri with a prefix string that includes the user who's forwarding the call (the original RURI instead of just the rewritten one). And there's the trick. How do I craft a RURI out of bits and pieces of things into one long RURI? If it were all the same number, I could use prefix, but it's dynamic (as is the nature of most things), so prefix won't work. How do I take RURI=1105 And add to it: The rewritten RURI from the call forwarding info: 18005551212 AND the prefix for the b2bua auth: 9999 To make: new ruri: 9999110518005551212 N.

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[Serusers] European Ph.D VoIP programs
by Voipers Portugal 11 Jul '06

11 Jul '06

Hi everyone, I know these list aren't the best place to ask for assistance. However, I would like to know if anyone here knows what are the best European University to do a Ph.D degree in something related to VoIP. Anyone here already working on a particular project like this? All answers will be helpful. Thanks, Jose Simoes

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Re: [Serusers] Radius Accounting
by Greger V. Teigre 11 Jul '06

11 Jul '06

By making sure that the accounting flag is set also for those messages. g-) Caxton wrote: > Dear Teigre: > How can I slove this problem? > Thanks. > > Sincerely > Caxton > > > 2006/7/11, Greger V. Teigre <greger(a)teigre.com > <mailto:greger@teigre.com>>: > > The BYE is loose routed, thus it will go to route(1). > g-) > > Caxton wrote: > Hi all: > > I have installed radius server, and I also configure the > "ser.cfg" for the radius accounting. However, there is a problem > confues me. > > While the sip session have set up, the radiusclient send a > Request-Account packet to radius server immediately, than radius > server > can save the start-up time of the session into database. But > after the session terminated, radiusclient do nothing(it should > send a > Accounting-Stop packet to radius server)!! Therefore, I can > not have a complete time record. > > I try to look up the solution in seruser mailing list, but I > can not find what I want. Please help me, thanks. > > Sincerely > > Caxton > > -----ser.cfg----- > > [cut] > > # -- acc params -- > modparam("acc", "radius_config", > "/usr/local/etc/radiusclient-ng/radiusclient.conf") > modparam("acc", "log_level", 2) > modparam("acc", "radius_flag", 1) > modparam("acc", "report_ack", 0) > # ------------------------- request routing logic ------------------- > > # main routing logic > route{ > > # 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 >= 2048 ) { > sl_send_reply("513", "Message too big"); > break; > }; > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > if (!method=="REGISTER") record_route(); > > # subsequent messages withing a dialog should take the > # path determined by record-routing > if (loose_route()) { > # mark routing logic in request > append_hf("P-hint: rr-enforced\r\n"); > route(1); > break; > }; > > if (!uri==myself) { > # mark routing logic in request > append_hf("P-hint: outbound\r\n"); > route(1); > break; > }; > > # if the request is for other domain use UsrLoc > # (in case, it does not work, use the following command > # with proper names and addresses in it) > if (uri==myself) { > > if (method=="REGISTER") { > log(1, "REGISTER: Authenticating user\n"); > # Uncomment this if you want to use digest authentication > if > (!www_authorize("pc80.s206.tj.mcu.edu.tw > <http://pc80.s206.tj.mcu.edu.tw/>", "subscriber")) { > log(1, "REGISTER: Chanllenging > user\n"); > www_challenge(" > pc80.s206.tj.mcu.edu.tw <http://pc80.s206.tj.mcu.edu.tw/>", "0"); > break; > }; > save("location"); > break; > }; > if(method=="INVITE") { > log(1, "INVITE\n"); > setflag(1); > } > if(method=="MESSAGE") { > log(1, "MESSAGE\n"); > setflag(1); > } > if(method=="BYE"||method== "CANCEL") { > log(1, "BYE or CANCEL\n"); > setflag(1); > } > > [cut] > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Users] Declined INVITE invokes multiple Decline's at Proxy
by Augustin, Andrew (external) 11 Jul '06

11 Jul '06

Hello, I am using openser v1.0. I have a problem with the following SIP scenario with openser: - A invites B to a call session, B declines the invitation The following message trace occurs between A and B and various proxies. Openser v1.0 is running on Proxy-1. The messages occur in sequence and are numbered: F27, F28, ... through F107. Referring to the following trace for the above scenario I am expecting the call flow to terminate at message: F55 once the 'Decline' has routed its way back back from the B-Party to the A-Party and a final 'ACK' is sent by the A-Party to the previous proxy: Proxy-1. But what actually occurs is that openser running on Proxy-1 automatically resends a 'Decline' at message F56 identical to the 'Decline' it first sent at message F46. This in turn prompts further ACKs and 'Decline's until it finally times out at message F107. It appears that openser saves the original message and attempts to retry/resend the 'Decline' message when prompted by an ACK. The above behaviour also occurs for the following scenario: - A invites B to a call session, B responds with a Busy Can you help me to: - understand what is going on - terminate what appears to be an open 'transaction' for the 'Decline' and prevent further 'Decline's and ACKs being sent around the system. Best regards, Andrew A-Party Proxy-1 Proxy-2 Proxy-3 B-Party 10.1.195.134:5062 10.1.195.134:5060 127.0.0.1:5070 127.0.0.1:5080 10.1.195.208:5060 | | | | | | F26 INVITE (sdp | | | | |>--------------->| | | | | t to us 100 F27 | | | | |<---------------<| | | | | | F28 INVITE (sdp)| | | | |>--------------------------------->| | | | r call is important to us 100 F29 | | | |<---------------------------------<| | | | | sdp) INVITE F30 | | | | |<---------------<| | | | | F31 100 trying | | | | |>--------------->| | | | | F32 INVITE (sdp | | | | |>--------------->| | | | | t to us 100 F33 | | | | |<---------------<| | | | |(sdp) INVITE F34 | | | |<---------------------------------<| | | | F35 100 trying -- your call is im | | | |>--------------------------------->| | | | F36 INVITE (sdp)| | | | |>--------------------------------------------------->| | | | | Trying 100 F37 | | |<---------------------------------------------------<| | | | | Ringing 180 F38 | | |<---------------------------------------------------<| | | F39 180 Ringing | | | | |>--------------------------------->| | | | | Ringing 180 F40 | | | | |<---------------<| | | | | F41 180 Ringing | | | | |>--------------->| | | | | Ringing 180 F42 | | | |<---------------------------------<| | | Ringing 180 F43 | | | | |<---------------<| | | | | | | | Decline 603 F44 | | |<---------------------------------------------------<| | | F45 ACK | | | | |>--------------------------------------------------->| | | F46 603 Decline | | | | |>--------------------------------->| | | | | ACK F47 | | | |<---------------------------------<| | | | | Decline 603 F48 | | | | |<---------------<| | | | | F49 ACK | | | | |>--------------->| | | | | F50 603 Decline | | | | |>--------------->| | | | | ACK F51 | | | | |<---------------<| | | | | Decline 603 F52 | | | |<---------------------------------<| | | | F53 ACK | | | | |>--------------------------------->| | | Decline 603 F54 | | | | |<---------------<| | | | | F55 ACK | | | | |>--------------->| | | | | | F56 603 Decline | | | | |>--------------------------------->| | | | | ACK F57 | | | |<---------------------------------<| | | Decline 603 F58 | | | | |<---------------<| | | | | F59 ACK | | | | |>--------------->| | | | | | F60 603 Decline | | | | |>--------------------------------->| | | | | ACK F61 | | | |<---------------------------------<| | | Decline 603 F62 | | | | |<---------------<| | | | | F63 ACK | | | | |>--------------->| | | | | | F64 603 Decline | | | | |>--------------------------------->| | | Decline 603 F65 | | | | |<---------------<| | | | | | | Decline 603 F66 | | | | |<---------------<| | | | | F67 603 Decline | | | | |>--------------->| | | | | Decline 603 F68 | | | |<---------------------------------<| | | | F69 ACK | | | | |>--------------------------------->| | | F70 ACK | | | | |>--------------->| | | | | | | ACK F71 | | | |<---------------------------------<| | | | F72 603 Decline | | | | |>--------------------------------->| | | Decline 603 F73 | | | | |<---------------<| | | | | | | Decline 603 F74 | | | | |<---------------<| | | | | F75 603 Decline | | | | |>--------------->| | | | | Decline 603 F76 | | | |<---------------------------------<| | | F77 ACK | | | | |>--------------->| | | | | | F78 ACK | | | | |>--------------------------------->| | | | | ACK F79 | | | |<---------------------------------<| | | | F80 603 Decline | | | | |>--------------------------------->| | | Decline 603 F81 | | | | |<---------------<| | | | | | | Decline 603 F82 | | | | |<---------------<| | | | | F83 603 Decline | | | | |>--------------->| | | | | Decline 603 F84 | | | |<---------------------------------<| | | | F85 ACK | | | | |>--------------------------------->| | | | | ACK F86 | | | |<---------------------------------<| | | | F87 603 Decline | | | | |>--------------------------------->| | | Decline 603 F88 | | | | |<---------------<| | | | | | | Decline 603 F89 | | | | |<---------------<| | | | | F90 603 Decline | | | | |>--------------->| | | | | Decline 603 F91 | | | |<---------------------------------<| | | | F92 ACK | | | | |>--------------------------------->| | | | | ACK F93 | | | |<---------------------------------<| | | | F94 603 Decline | | | | |>--------------------------------->| | | Decline 603 F95 | | | | |<---------------<| | | | | | | Decline 603 F96 | | | | |<---------------<| | | | | F97 603 Decline | | | | |>--------------->| | | | | Decline 603 F98 | | | |<---------------------------------<| | | | F99 ACK | | | | |>--------------------------------->| | | | | ACK F100 | | | |<---------------------------------<| | | | F101 603 Decline | | | |>--------------------------------->| | | ecline 603 F102 | | | | |<---------------<| | | | | | | ecline 603 F103 | | | | |<---------------<| | | | | F104 603 Declin | | | | |>--------------->| | | | Decline 603 F105 | | | |<---------------------------------<| | | | F106 ACK | | | | |>--------------------------------->| | | | | ACK F107 | | | |<---------------------------------<| |

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RE: Re: [Serusers] Call Forward Billing
by Rosa De Santis 11 Jul '06

11 Jul '06

Thanks Jiri and Greger for your answers. I just wanted to make it simple, because I'm using a third party billing system. I'm using SER + Asterisk b2bua. My original idea was to add a Diversion header to the sip message in SER, and to modify the Radius message in Asterisk b2bua when i find that diversion header, to give the Radius message ready and modified to the billing system, so they don't have to make any additional process for the call forward case. Please, is this possible? any ideas? Thanks again. Rosa. ----------------------------------------------- > Date: Tue, 11 Jul 2006 08:18:59 +0200 > From: greger(a)teigre.com > To: jiri(a)iptel.org > CC: rosadesantis(a)hotmail.com; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Call Forward Billing > > Just to expand on "special-casing": Getting user actions into $ can be very tricky and is very dependent on your billing system. I.e. the capabilities of the type of invoice records that can be created (and manipulated; rating, VAT, no VAT, customer discounts, volume discounts, time of day etc etc). You normally have to start there and trace back: What should the normalized CDRs look like for the billing system to import and rate them correctly. Then you step backwards again: Your accounting records (start/stop), what should they look like. The tool taking start/stop records must be capable of interpreting the records and create normalized CDRs. If you gather accounting records from multiple sources (PSTN GWs, routers etc), it may make sense to consolidate the record collection and normalization (i.e. adapt your accounting records to be similar across sources so you simplify the normalization)- > Puh! This leads us to Jiri's comment: Once you know what you need, you can use additional AVPs/flags in acc table. > g-) > PS! In the PSTN world, the forwarding party/C-number is often added in a third column in the records and one set of start/stop will exist for each call leg. If present, the C-number will be used for charging. > Jiri Kuthan wrote: > It is a configuration thing. It appears a misconception to me to build special-casing in SER. acc module can store additional AVPs or flags in acc table. These can indicate reverse charging at your convenience, to be processed by your favorite CDR postprocesssing tool. Moreover, if done well, all the authentication-authorization-accounting process can be, if configured properly, brought to the same least common denominator, which is AVP -- you authenticate someone, check if his ACLs expressed as an AVP permit a call (e.g., 'international'), account it (possibly with additional AVPs/flags providing additional hints for how to interpret the CDR). -jiri At 19:49 10/07/2006, Rosa De Santis wrote: > Thanks for your answer. And this functionality is not present in SER ? Rosa. ----------------------------------------------- > Date: Sat, 8 Jul 2006 00:20:44 -0700 From: t_maron_lists(a)yahoo.com Subject: Re: [Serusers] Call Forward Billing To: rosadesantis(a)hotmail.com If you are locked by this, you can import a patch from openser, they have it for some time, see: http://openser.org/docs/modules/1.1.x/acc.html#MULTI-CALL-LEGS T. On 07/07/06 18:46, Rosa De Santis wrote: Hello all. Please, can somebody help about changing the billing when making call forward ? I'm trying to change the billed user when A calls B, B forwards to C (C is PSTN). I want to charge the call to B. I'm using Ser+Asterisk B2BUA. Any idea, please? Thanks Rosa. _________________________________________________________________ Be one of the first to try Windows Live Mail. http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-49… Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ----------------------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. > _________________________________________________________________ Be one of the first to try Windows Live Mail. http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-49… 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 _________________________________________________________________ Be one of the first to try Windows Live Mail. http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-49…

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[Serusers] Open source GSM/CDMA network
by CM0002＠aol.com 11 Jul '06

11 Jul '06

GSM suppliers working on that already and implementing softswitches in order to increase capacity and lower cost. However GSM is circuit switched - not packet switched (except GPRS) and here is the point cost can't really dropped. The way it will be more probably is that GSM CDMA might be used in the furture in the suburbans and in metropolitan areas we will see a WIMAX/VOIP scenario. This will result in a hybrid network for at least a transitional period. We already experienced it with AMPS/TDMA to GSM transition. CDMA (spread spectrum) is almost dead because high cost and leck of roaming capacity. WCDMA/UMTS (spread spectrum 3G) is very expencive and hard to manage (in Europe the launch was delayed by 2 years because hard and software problems). I recently started with a project (small scale) to setup such WIMAX based networks. The aim is to develop a low cost internet/phone network for rural communities in third world countries. Anybody with routing experience who wants to join me on a professional base is welcome. Regards Christian

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[Serusers] Re: Serusers Digest, Vol 39, Issue 12
by Caxton 11 Jul '06

11 Jul '06

Dear Teigre: How can I slove this problem? Thanks. Sincerely Caxton Message: 6 > Date: Tue, 11 Jul 2006 08:47:40 +0200 > From: "Greger V. Teigre" <greger(a)teigre.com> > Subject: Re: [Serusers] Radius Accounting > To: Caxton <kaifu.chan(a)gmail.com> > Cc: serusers(a)lists.iptel.org, serusers-request(a)lists.iptel.org > Message-ID: <44B3498C.7020003(a)teigre.com> > Content-Type: text/plain; charset="iso-8859-1" > > The BYE is loose routed, thus it will go to route(1). > g-) > > Caxton wrote: > > Hi all: > > > > I have installed radius server, and I also configure the "ser.cfg" > > for the radius accounting. However, there is a problem confues me. > > > > While the sip session have set up, the radiusclient send a > > Request-Account packet to radius server immediately, than radius server > > can save the start-up time of the session into database. But > > after the session terminated, radiusclient do nothing(it should send a > > Accounting-Stop packet to radius server)!! Therefore, I can > > not have a complete time record. > > > > I try to look up the solution in seruser mailing list, but I can > > not find what I want. Please help me, thanks. > > > > Sincerely > > > > Caxton > > > > -----ser.cfg----- > > > > [cut] > > > > # -- acc params -- > > modparam("acc", "radius_config", > > "/usr/local/etc/radiusclient-ng/radiusclient.conf") > > modparam("acc", "log_level", 2) > > modparam("acc", "radius_flag", 1) > > modparam("acc", "report_ack", 0) > > # ------------------------- request routing logic ------------------- > > > > # main routing logic > > route{ > > > > # 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 >= 2048 ) { > > sl_send_reply("513", "Message too big"); > > break; > > }; > > # we record-route all messages -- to make sure that > > # subsequent messages will go through our proxy; that's > > # particularly good if upstream and downstream entities > > # use different transport protocol > > if (!method=="REGISTER") record_route(); > > > > # subsequent messages withing a dialog should take the > > # path determined by record-routing > > if (loose_route()) { > > # mark routing logic in request > > append_hf("P-hint: rr-enforced\r\n"); > > route(1); > > break; > > }; > > > > if (!uri==myself) { > > # mark routing logic in request > > append_hf("P-hint: outbound\r\n"); > > route(1); > > break; > > }; > > > > # if the request is for other domain use UsrLoc > > # (in case, it does not work, use the following command > > # with proper names and addresses in it) > > if (uri==myself) { > > > > if (method=="REGISTER") { > > log(1, "REGISTER: Authenticating user\n"); > > # Uncomment this if you want to use digest authentication > > if (!www_authorize("pc80.s206.tj.mcu.edu.tw > > <http://pc80.s206.tj.mcu.edu.tw/>", "subscriber")) { > > log(1, "REGISTER: Chanllenging user\n"); > > www_challenge(" > > pc80.s206.tj.mcu.edu.tw <http://pc80.s206.tj.mcu.edu.tw/>", "0"); > > break; > > }; > > save("location"); > > break; > > }; > > if(method=="INVITE") { > > log(1, "INVITE\n"); > > setflag(1); > > } > > if(method=="MESSAGE") { > > log(1, "MESSAGE\n"); > > setflag(1); > > } > > if(method=="BYE"||method== "CANCEL") { > > log(1, "BYE or CANCEL\n"); > > setflag(1); > > } > > > > [cut] > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers

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[Users] Memory resolved?
by Helge Waastad 11 Jul '06

11 Jul '06

Hi, I've been away for 3 weeks and have'nt updated my CVS version since then. I just wonder if some memory problems have been solved? In my logs I've seen: Jul 10 22:06:16 proxy-02 /usr/sbin/openser[31698]: ERROR: build_auth_hf: no memory Jul 10 22:06:16 proxy-02 /usr/sbin/openser[31698]: ERROR: challenge: no mem w/cred Jul 10 22:06:16 proxy-02 /usr/sbin/openser[31700]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 542 Jul 10 22:06:17 proxy-02 /usr/sbin/openser[31698]: ERROR:usrloc:db_load_urecord: db_query failed Jul 10 22:06:17 proxy-02 /usr/sbin/openser[31698]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 424 Jul 10 22:06:17 proxy-02 /usr/sbin/openser[31698]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 470 Jul 10 22:06:17 proxy-02 /usr/sbin/openser[31700]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 424 Jul 10 22:06:17 proxy-02 /usr/sbin/openser[31700]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 470 Jul 10 22:06:18 proxy-02 /usr/sbin/openser[31698]: ERROR: build_auth_hf: no memory Jul 10 22:06:18 proxy-02 /usr/sbin/openser[31698]: ERROR: challenge: no mem w/cred Jul 10 22:06:18 proxy-02 /usr/sbin/openser[31700]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 533 Jul 10 22:06:18 proxy-02 /usr/sbin/openser[31700]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 579 Jul 10 22:06:19 proxy-01 /usr/sbin/openser[31492]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 478 Jul 10 22:06:19 proxy-02 /usr/sbin/openser[31696]: ERROR:usrloc:db_load_urecord: db_query failed Jul 10 22:06:19 proxy-02 /usr/sbin/openser[31696]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 424 Jul 10 22:06:19 proxy-02 /usr/sbin/openser[31696]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 470 Jul 10 22:06:19 proxy-01 /usr/sbin/openser[31496]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 478 Jul 10 22:06:21 proxy-01 /usr/sbin/openser[31492]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 478 Jul 10 22:06:21 proxy-02 /usr/sbin/openser[31700]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 478 Jul 10 22:06:22 proxy-01 /usr/sbin/openser[31496]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 424 Jul 10 22:06:22 proxy-01 /usr/sbin/openser[31496]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 470 Jul 10 22:06:24 proxy-02 /usr/sbin/openser[31696]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 478 Jul 10 22:06:25 proxy-02 /usr/sbin/openser[31700]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 478 If yes, I'll just start updating my packages. br hw

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RE: Re: [Serusers] Call Forward Billing
by Rosa De Santis 11 Jul '06

11 Jul '06

Thanks for your answer. And this functionality is not present in SER ? Rosa. ----------------------------------------------- > Date: Sat, 8 Jul 2006 00:20:44 -0700 > From: t_maron_lists(a)yahoo.com > Subject: Re: [Serusers] Call Forward Billing > To: rosadesantis(a)hotmail.com > > If you are locked by this, you can import a patch from openser, they have it for some time, see: > http://openser.org/docs/modules/1.1.x/acc.html#MULTI-CALL-LEGS > T. > On 07/07/06 18:46, Rosa De Santis wrote: > Hello all. > Please, can somebody help about changing the billing when making call forward ? > I'm trying to change the billed user when A calls B, B forwards to C (C is PSTN). I want to charge the call to B. I'm using Ser+Asterisk B2BUA. > Any idea, please? > Thanks > Rosa. > _________________________________________________________________ > Be one of the first to try Windows Live Mail. > http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-49… > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > ----------------------------------------------- > Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. _________________________________________________________________ Be one of the first to try Windows Live Mail. http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-49…

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sr-users July 2006