sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

[Serusers] posting from nonmembeers
by Jan Janak 29 Jan '04

29 Jan '04

Hello, I prohibited posting from non-members to the list again (such mails are held for review) due to large number of bogus messages received during last couple of hours. It will be enabled again later. I would like to encourage people who are using MS Outlook or similar bullshit from microsoft to try a different mail client. Outlook is evil and microsoft is doing nothing about that. During the last couple of hours I received 400 reports that my emails or emails from the mailing lists … [View More]

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RE: [Serusers] SER + Proxy + NAT
by dhiraj.2.bhuyan＠bt.com 29 Jan '04

29 Jan '04

Never tried it myself - but I think it should work the following way. If the Router (NAT) has DMZ support, you can put SER on a machine in the Router's DMZ interface. This will make the SER appear with a globally routable address (that of the gateway or something else if you have multiple IP address support). So when UA3 registers itself with SER (configured with NAT support), the nathelper module will detect that UA3 is behind a NAT gateway and will enable rtp-relaying. So the trick out here … [View More]is to make the SER appear to be on the Internet - rather than on the private network. Note that the router should allow communication between NATted devices and machines on the DMZ. Also note that by putting the SER on the internet, you are making it open to attacks. Add proper firewall rules. Let me know if it works :) Dhiraj Bhuyan Network Security Specialist, BT Exact Business Assurance Solutions -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Edson Gellert Schubert Sent: 29 January 2004 15:19 To: Adrian Georgescu Cc: Lista SER - IPTEL Subject: Re: [Serusers] SER + Proxy + NAT -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I thing that I didn't put myself very clear... Let me try an ASCII-Diagram... ;) UA1 ---- public IP ---- Internet ------ ADSL/GW ------ UA2 (Win Mes) (dialup) | | (IPTables) (Win Mes) | | /-----/ \-----------\ | | Router(NAT) ADSL/Win (Windows) | | UA3 ------------+ | (Win Mes) | UA4 (Win Mes) SER What I'm looking for is a Proxy to put in the ROUTER-Machine (could be a Linux/IPTables, FreeBSD, etc): I undestand what Jan explain, about the complications, and that's why I was asking about an "inteligent" Proxy to handle SIP traffic. Suppose that UA3 wants to talk with UA2 (were the ADSL/GW should have SIProxd installed). The communications flow would be UA3-SER-Router-Internet-ADSL/GW-UA2. Ok, in the Router appears the first challenge (how to transverse the NAT, keeping track from the flow?). Here comes the SIP-Proxy in action. It recieves the packet from SER, make desired changes and forward it through "Internet" to ADSL/GW. There, the SIProxyd recieves the packet, apply the related changes and forward it to UA2. Great. Is what we want. The reverse, that is, when UA2 (or UA1, or UA4) wants to talk with UA3 becomes the great challenge. How should the Proxy, in Router, knows where to send the packets that arrive from Internet? To SER? Directly to UA3? It's hard to make the decision. The Proxy had to have many from a SIP-Server functionalities. It has to maintain flows tables with users-ID, ports and servers IP used in each communication flow (other infos could help in other tasks, but I thing that these one are the minimum), so that it could decide to whom send each packet from each flow. So, do I make my doubts/points clear to You? If my understand is wrong, sorry and please correct me where necessary. Edson. P.S.: In my scenario there is no SER-2-SER communications, but another problem would be having two (or more) sites like the "Router" one. How to make than communicate each other through NAT GW/FW? - ----- Original Message ----- From: "Adrian Georgescu" < ag(a)ag-projects.com> To: < serusers(a)lists.iptel.org> Sent: Thursday, January 29, 2004 8:00 AM Subject: [Serusers] SER + Proxy + NAT > Edson, > > Putting a NAT traversal solution behing NAT is a chicken and eg > problem, isn't it? > > -- > Adrian > > > > Hi all... > > I look through the list's archives, but an not finding info to help me. > > The goal is use SER but not instaled in the GW/FW (it's not an > acceptable > option, well it's acceptable, but not for now). So I'm trying to put > the SER > in the Internal LAN (it could be installed in a DMZ also). So the > question > is if there is any proxy that could be putted on the GW/FW to handle > incomming calls (INVITEs) and forward it correctly to the SER machine > taken > over the NAT issues? > > I already look at SIProxd and RTProxy, but the first didn't forward > incomming calls, and the second demands that it be instaled, with SER > on the > GW/FW. I also am looking at SERMediaProxy (RTProxy alternative) but the > documentations aren't sufficient detailed to answer my question. Any > help > would be appreciated. > > Edson. > > - -------------------------------------------------------------------------------- > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > -----BEGIN PGP SIGNATURE----- Version: PGP 8.0 iQA/AwUBQBkkYDdMQB7Du1dpEQLGiQCfcjklZxwiAtG+rj+rKqCpKIORLA0AoOZF oBf1QhqGvX67oZ14W127mCxl =oB8Y -----END PGP SIGNATURE----- [View Less]

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[Serusers] SER + Proxy + NAT
by Adrian Georgescu 29 Jan '04

29 Jan '04

Edson, Putting a NAT traversal solution behing NAT is a chicken and eg problem, isn't it? -- Adrian Hi all... I look through the list's archives, but an not finding info to help me. The goal is use SER but not instaled in the GW/FW (it's not an acceptable option, well it's acceptable, but not for now). So I'm trying to put the SER in the Internal LAN (it could be installed in a DMZ also). So the question is if there is any proxy that could be putted on the GW/FW to handle incomming … [View More]

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[Serusers] Re: sems problem
by Raphael Coeffic 29 Jan '04

29 Jan '04

Hello, it seems like that Cisco 827-V router doesn't include any from-tag within its INVITEs...which is NOT RFC CONFORM!. It may be due to some kind of proprietary function mode. There is nothing we can do on the SEMS side to solve the problem. Does anybody else have such a gateway and got it work with voicemail? -Raphael. ----- Original Message ----- From: "Goh Sek Chye" <sekchye(a)lgatelecom.net> To: <raphael.coeffic(a)gmx.de> Sent: Monday, January 26, 2004 11:08 AM Subject: … [View More]sems problem > Hi Raphael, > > I am using your voice mail program sems and it is very > useful to us. Thank you very much! > > However, I have a problem here. > > I have a Cisco router 827-4V which is connected to a PBX using > FXS port. The Cisco router is configured to point to a SIP server > running SER and SEMS. > > When a PBX user calls a SIP user who is not online, > he hears a long death silence followed by a fast busy tone. > > I am clueless what is the problem and would like to seek your > help. > > The following is the debug message from the SER server and > the debug message from the Cisco router. Thanks in advance for your help. > > ================= > > SER debug messages > ==================== > > Jan 26 17:13:41 sip1 ser[25648]: SIP Request: > Jan 26 17:13:41 sip1 ser[25648]: method: <INVITE> > Jan 26 17:13:41 sip1 ser[25648]: uri: <sip:10650004@192.168.0.222;user=phone> > Jan 26 17:13:41 sip1 ser[25648]: version: <SIP/2.0> > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=1 > Jan 26 17:13:41 sip1 ser[25648]: end of header reached, state=5 > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: Via found, flags=1 > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: this is the first via > Jan 26 17:13:41 sip1 ser[25648]: After parse_msg... > Jan 26 17:13:41 sip1 ser[25648]: preparing to run routing scripts... > Jan 26 17:13:41 sip1 ser[25648]: DEBUG : is_maxfwd_present: searching for max_forwards header > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=128 > Jan 26 17:13:41 sip1 ser[25648]: end of header reached, state=9 > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: get_hdr_field: <To> [41]; uri=[sip:10650004@192.168.0.222;user=phone] > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: to body [<sip:10650004@192.168.0.222;user=phone>^M ] > Jan 26 17:13:41 sip1 ser[25648]: get_hdr_field: cseq <CSeq>: <101> <INVITE> > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: is_maxfwd_present: value = 6 > Jan 26 17:13:41 sip1 ser[25648]: end of header reached, state=9 > Jan 26 17:13:41 sip1 ser[25648]: check_self - checking if host==us: 13==13 && [192.168.0.222] == [192.168.0.222] > > Jan 26 17:13:41 sip1 ser[25648]: lookup(): '10650004' Not found in usrloc > Jan 26 17:13:41 sip1 ser[25648]: lookup(): '10650004' Not found in usrloc > Jan 26 17:13:41 sip1 ser[25648]: ************************ route[3]:vm:1 ************************************* > Jan 26 17:13:41 sip1 ser[25648]: route[3]:vm:2 > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_addifnew: msg id=789 , global msg id=787 , T on entrance=0xffffffff > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=-1 > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: get_hdr_body : content_length=134 > Jan 26 17:13:41 sip1 ser[25648]: found end of header > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=60 > Jan 26 17:13:41 sip1 ser[25648]: t_lookup_request: start searching: hash=43402, isACK=0 > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: proceeding to pre-RFC3261 transaction matching > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_lookup_request: no transaction found > Jan 26 17:13:41 sip1 ser[25648]: DBG: callback type 2, id 3 entered > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=44 > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: noisy_timer set for accounting > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_check: msg id=789 global id=789 T start=0x402e1070 > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_check: T alredy found! > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=-1 > Jan 26 17:13:41 sip1 ser[25648]: check_via_address(203.92.75.94, 203.92.75.94, 0) > Jan 26 17:13:41 sip1 ser[25648]: WARNING:vqm_resize: resize(0) called > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: reply sent out. buf=0x80d0cd0: SIP/2.0 1..., shmem=0x402d8cd8: SIP/2.0 1 > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_reply: finished > Jan 26 17:13:41 sip1 ser[25648]: route[3]:method==INVITE || REFER > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=-1 > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_check: msg id=789 global id=789 T start=0x402e1070 > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_check: T alredy found! > Jan 26 17:13:41 sip1 ser[25648]: vm: calculated route: > Jan 26 17:13:41 sip1 ser[25648]: vm: next r-uri: sip:41703@203.92.75.94:5060;user=phone > Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=-1 > Jan 26 17:13:41 sip1 ser[25648]: submit_query(): select email_address from subscriber where username='10650004' > Jan 26 17:13:41 sip1 ser[25648]: vm: write_to_vm_fifo: <0.1 voicemail INVITE 10650004 sekchye(a)yahoo.com 203.92.64 > .216 192.168.0.222 5060 sip:10650004@192.168.0.222;user=phone sip:41703@203.92.75.94:5060;user=phone "41703" <sip > :41703@203.92.75.94>^M <sip:10650004@192.168.0.222;user=phone> C475045A-4F1611D8-827089FC-A1038096(a)203.92.75.94 . > . 101 43402:300376901 . . . v=0^M o=CiscoSystemsSIP-GW-UserAgent 5931 2351 IN IP4 203.92.75.94^M s=SIP Call^M c= > IN IP4 203.92.75.94^M t=0 0^M m=audio 20416 RTP/AVP 0^M > > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: write_to_vm_fifo: write completed > Jan 26 17:13:41 sip1 ser[25648]: DEBUG: add_to_tail_of_timer[0]: 0x402e11bc > Jan 26 17:13:41 sip1 ser[25648]: receive_msg: cleaning up > Jan 26 17:13:41 sip1 Sems[17680]: Debug: version= <0.1> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.cmd= <voicemail> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.method= <INVITE> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.user= <10650004> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.email= <sekchye(a)yahoo.com> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.domain= <192.168.0.222> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.dstip= <192.168.0.222> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.port= <5060> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.r_uri= <sip:10650004@192.168.0.222;user=phone> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.from_uri= <sip:41703@203.92.75.94:5060;user=phone> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.from= <"41703" <sip:41703@203.92.75.94>> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.to= <<sip:10650004@192.168.0.222;user=phone>> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.callid= <C475045A-4F1611D8-827089FC-A1038096(a)203.92.75.94> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.from_tag= <> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.to_tag= <> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cseq_str= <101> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cseq= <101>(101) > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.key= <43402:300376901> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.route= <> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.next_hop= <> > Jan 26 17:13:41 sip1 Sems[17680]: Debug: hdrs: `' > Jan 26 17:13:41 sip1 Sems[17680]: Debug: body: `v=0 o=CiscoSystemsSIP-GW-UserAgent 5931 2351 IN IP4 203.92.75.94 > s=SIP Call c=IN IP4 203.92.75.94 t=0 0 m=audio 20416 RTP/AVP 0 ' > Jan 26 17:13:41 sip1 Sems[17680]: Error: invalid FIFO command: cmd.from_tag is empty !!! > Jan 26 17:13:48 sip1 ser[25685]: MSILO:clean_silo: cleaning stored messages - 874200 > Jan 26 17:13:48 sip1 ser[25685]: MSILO:clean_silo: cleaning expired messages > Jan 26 17:13:48 sip1 ser[25685]: submit_query(): delete from silo where exp_time<=1075108428 > Jan 26 17:13:55 sip1 ser[25685]: DEBUG: timer routine:0,tl=0x402e11bc next=(nil) > Jan 26 17:13:55 sip1 ser[25685]: DEBUG: FR_handler:stop retr. and send CANCEL (0x402e1070) > Jan 26 17:13:55 sip1 ser[25685]: DEBUG: FR_handler:stop retr. and send CANCEL (0x402e1070) > Jan 26 17:13:55 sip1 ser[25685]: ->>>>>>>>> T_code=100, new_code=408 > Jan 26 17:13:55 sip1 ser[25685]: DBG: callback type 6, id 4 entered > Jan 26 17:13:55 sip1 ser[25685]: submit_query(): insert into missed_calls (from_uri,to_uri,sip_method,i_uri,o_uri > ,sip_from,sip_callid,sip_to,sip_status,username,totag,fromtag,time ) values ('sip:41703@203.92.75.94','sip:106500 > 04(a)192.168.0.222;user=phone','INVITE','sip:10650004@192.168.0.222;user=phone ','sip:10650004@192.168.0.222;user=ph > one','"41703" <sip:41703@203.92.75.94>','C475045A-4F1611D8-827089FC-A1038096(a)203.92.75.94' ,'<sip:10650004@203.92. > 64.216;user=phone>','408 Request Timeout','10650004','n/a','n/a','2004-01-26 09:13:55') > Jan 26 17:13:55 sip1 ser[25685]: rc_get_seqnbr: couldn't open sequence file /var/run/radius.seq: Permission denie > d > Jan 26 17:13:55 sip1 ser[25685]: DEBUG: relay_reply: branch=0, save=0, relay=0 > Jan 26 17:13:55 sip1 ser[25685]: parse_headers: flags=-1 > Jan 26 17:13:55 sip1 ser[25685]: check_via_address(203.92.75.94, 203.92.75.94, 0) > Jan 26 17:13:55 sip1 ser[25685]: DEBUG: reply relayed. buf=0x80cef08: SIP/2.0 4..., shmem=0x402d47a8: SIP/2.0 4 > Jan 26 17:13:55 sip1 ser[25685]: DBG: callback type 7, id 1 entered > Jan 26 17:13:55 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[4]: 0x402e1128 > Jan 26 17:13:55 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[0]: 0x402e1140 > Jan 26 17:13:55 sip1 ser[25685]: DEBUG: final_response_handler : done > Jan 26 17:13:55 sip1 ser[25647]: SIP Request: > Jan 26 17:13:55 sip1 ser[25647]: method: <ACK> > Jan 26 17:13:55 sip1 ser[25647]: uri: <sip:10650004@192.168.0.222;user=phone> > Jan 26 17:13:55 sip1 ser[25647]: version: <SIP/2.0> > Jan 26 17:13:55 sip1 ser[25647]: parse_headers: flags=1 > Jan 26 17:13:55 sip1 ser[25647]: end of header reached, state=5 > Jan 26 17:13:55 sip1 ser[25647]: parse_headers: Via found, flags=1 > Jan 26 17:13:55 sip1 ser[25647]: parse_headers: this is the first via > Jan 26 17:13:55 sip1 ser[25647]: After parse_msg... > Jan 26 17:13:55 sip1 ser[25647]: DEBUG : sl_filter_ACK: to late to be a local ACK! > Jan 26 17:13:55 sip1 ser[25647]: preparing to run routing scripts... > Jan 26 17:13:55 sip1 ser[25647]: DEBUG : is_maxfwd_present: searching for max_forwards header > Jan 26 17:13:55 sip1 ser[25647]: parse_headers: flags=128 > Jan 26 17:13:55 sip1 ser[25647]: DEBUG: add_param: tag=baea774f5e5e7e26323ab05f178242c9-2bd7 > Jan 26 17:13:55 sip1 ser[25647]: end of header reached, state=29 > Jan 26 17:13:55 sip1 ser[25647]: DEBUG: get_hdr_field: <To> [83]; uri=[sip:10650004@192.168.0.222;user=phone] > Jan 26 17:13:55 sip1 ser[25647]: DEBUG: to body [<sip:10650004@192.168.0.222;user=phone>] > Jan 26 17:13:55 sip1 ser[25647]: DEBUG: is_maxfwd_present: value = 6 > Jan 26 17:13:55 sip1 ser[25647]: end of header reached, state=9 > Jan 26 17:13:55 sip1 ser[25647]: check_self - checking if host==us: 13==13 && [192.168.0.222] == [192.168.0.222] > > Jan 26 17:13:55 sip1 ser[25647]: lookup(): '10650004' Not found in usrloc > Jan 26 17:13:55 sip1 ser[25647]: lookup(): '10650004' Not found in usrloc > Jan 26 17:13:55 sip1 ser[25647]: ************************ route[3]:vm:1 ************************************* > Jan 26 17:13:55 sip1 ser[25647]: receive_msg: cleaning up > Jan 26 17:13:56 sip1 ser[25685]: DEBUG: timer routine:4,tl=0x402e1128 next=(nil) > Jan 26 17:13:56 sip1 ser[25685]: DEBUG: retransmission_handler : reply resending (t=0x402e1070, SIP/2.0 4 ... ) > Jan 26 17:13:56 sip1 ser[25685]: DEBUG: reply retransmitted. buf=0x42237da0: SIP/2.0 4..., shmem=0x402d47a8: SIP/ > 2.0 4 > Jan 26 17:13:56 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[5]: 0x402e1128 > Jan 26 17:13:56 sip1 ser[25685]: DEBUG: retransmission_handler : done > Jan 26 17:13:58 sip1 ser[25685]: DEBUG: timer routine:5,tl=0x402e1128 next=(nil) > Jan 26 17:13:58 sip1 ser[25685]: DEBUG: retransmission_handler : reply resending (t=0x402e1070, SIP/2.0 4 ... ) > Jan 26 17:13:58 sip1 ser[25685]: DEBUG: reply retransmitted. buf=0x42237da0: SIP/2.0 4..., shmem=0x402d47a8: SIP/ > 2.0 4 > Jan 26 17:13:58 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[6]: 0x402e1128 > Jan 26 17:13:58 sip1 ser[25685]: DEBUG: retransmission_handler : done > Jan 26 17:14:02 sip1 ser[25685]: DEBUG: timer routine:6,tl=0x402e1128 next=(nil) > Jan 26 17:14:02 sip1 ser[25685]: DEBUG: retransmission_handler : reply resending (t=0x402e1070, SIP/2.0 4 ... ) > Jan 26 17:14:02 sip1 ser[25685]: DEBUG: reply retransmitted. buf=0x42237da0: SIP/2.0 4..., shmem=0x402d47a8: SIP/ > 2.0 4 > Jan 26 17:14:02 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[7]: 0x402e1128 > Jan 26 17:14:02 sip1 ser[25685]: DEBUG: retransmission_handler : done > Jan 26 17:14:06 sip1 ser[25685]: DEBUG: timer routine:7,tl=0x402e1128 next=(nil) > Jan 26 17:14:06 sip1 ser[25685]: DEBUG: retransmission_handler : reply resending (t=0x402e1070, SIP/2.0 4 ... ) > Jan 26 17:14:06 sip1 ser[25685]: DEBUG: reply retransmitted. buf=0x42237da0: SIP/2.0 4..., shmem=0x402d47a8: SIP/ > 2.0 4 > Jan 26 17:14:06 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[7]: 0x402e1128 > Jan 26 17:14:06 sip1 ser[25685]: DEBUG: retransmission_handler : done > Jan 26 17:14:10 sip1 ser[25685]: DEBUG: timer routine:0,tl=0x402e1140 next=(nil) > > > > > > > > > > Debug message from Cisco 827-V router > ===================================== > Jan 26 17:13:41.158 SGT: Sent: > INVITE sip:10650004@192.168.0.222;user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.202:5060 > From: "41703" <sip:41703@192.168.0.202> > To: <sip:10650004@192.168.0.222;user=phone> > Date: Mon, 26 Jan 2004 17:13:41 GMT > Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202 > Cisco-Guid: 3162002760-1326846424-2188282364-2701361302 > User-Agent: Cisco-SIPGateway/IOS-12.x > CSeq: 101 INVITE > Max-Forwards: 6 > Timestamp: 1075108421 > Contact: <sip:41703@192.168.0.202:5060;user=phone> > Expires: 180 > Content-Type: application/sdp > Content-Length: 136 > > v=0 > o=CiscoSystemsSIP-GW-UserAgent 5931 2351 IN IP4 192.168.0.202 > s=SIP Call > c=IN IP4 192.168.0.202 > t=0 0 > m=audio 20416 RTP/AVP 0 > > Jan 26 17:13:41.170 SGT: Received: > SIP/2.0 100 Trying -- just a second > Via: SIP/2.0/UDP 192.168.0.202:5060 > From: "41703" <sip:41703@192.168.0.202> > To: <sip:10650004@192.168.0.222;user=phone> > Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202 > CSeq: 101 INVITE > Server: Sip EXpress router (0.8.11 (i386/linux)) > Content-Length: 0 > > > > Jan 26 17:13:55.338 SGT: Received: > SIP/2.0 408 Request Timeout > Via: SIP/2.0/UDP 192.168.0.202:5060 > From: "41703" <sip:41703@192.168.0.202> > To: <sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9 -2bd7 > Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202 > CSeq: 101 INVITE > Server: Sip EXpress router (0.8.11 (i386/linux)) > Content-Length: 0 > > > > Jan 26 17:13:55.358 SGT: Sent: > ACK sip:10650004@192.168.0.222;user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.202:5060 > From: "41703" <sip:41703@192.168.0.202> > To: <sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9 -2bd7 > Date: Mon, 26 Jan 2004 17:13:41 GMT > Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202 > Max-Forwards: 6 > Content-Length: 0 > CSeq: 101 ACK > > > > Jan 26 17:13:56.346 SGT: Received: > SIP/2.0 408 Request Timeout > Via: SIP/2.0/UDP 192.168.0.202:5060 > From: "41703" <sip:41703@192.168.0.202> > To: <sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9 -2bd7 > Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202 > CSeq: 101 INVITE > Server: Sip EXpress router (0.8.11 (i386/linux)) > Content-Length: 0 > > > > > Jan 26 17:14:02.406 SGT: Received: > SIP/2.0 408 Request Timeout > Via: SIP/2.0/UDP 192.168.0.202:5060 > From: "41703" <sip:41703@192.168.0.202> > To: <sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9 -2bd7 > Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202 > CSeq: 101 INVITE > Server: Sip EXpress router (0.8.11 (i386/linux)) > Content-Length: 0 > > > > Jan 26 17:14:06.446 SGT: Received: > SIP/2.0 408 Request Timeout > Via: SIP/2.0/UDP 192.168.0.202:5060 > From: "41703" <sip:41703@192.168.0.202> > To: <sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9 -2bd7 > Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202 > CSeq: 101 INVITE > Server: Sip EXpress router (0.8.11 (i386/linux)) > Content-Length: 0 > > > > > > > > > > > > > > > > > -- > Best regards > Goh Sek Chye > LGA Telecom Pte Ltd > PGP KeyID: AE6D04A2 > > PS: To import my PGP key, > gpg --recv-keys --keyserver wwwkeys.pgp.net AE6D04A2 > > --------------------------------------------------------------- > Privileged, confidential and/or copyright information may be > contained in this e-mail. This e-mail is for the use only of > the intended addressee. If you are not the intended addressee, > or the person responsible for delivering it to the intended > addressee, you may not copy, forward, disclose or otherwise use > it or any part of it in any way whatsoever. To do so is > prohibited and may be unlawful. If you receive this e-mail by > mistake please advise the sender immediately by using the > reply facility in your e-mail software. This message is subject > to and does not create or vary any contractual relationship > between Little Green Apples Pte Ltd (the Company) or any of > its subsidiaries (comprising of LGA Technologies Pte Ltd; > LGA International Pte Ltd; & LGA Telecom Pte Ltd) and you. > By opening any attachment to this message, you also agree to > accept the risk that it may contain a virus or damaging code, > and you agree that the sender and/or the Company and its > subsidiaries will not be liable for any loss or damage > thereby caused. > --------------------------------------------------------------- > [View Less]

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[Serusers] Voicemail timers and functionality Question
by Glenn Dalgliesh 28 Jan '04

28 Jan '04

After reviewing the documentation and mailing list I think I can conclude that their is no way currently to have a calls end up at a Voicemal process or server based on timeout values, at this time provided by fr_ commands, without setting this value for all calls and having all call remaining stateful. If true has anyone found any work arounds to this problem? I sawl mention from an email from Jan that it was a feature on the list anyone seen any update related to it? Thanks

3 2

[Serusers] Symmetric audio video UA
by dhiraj.2.bhuyan＠bt.com 28 Jan '04

28 Jan '04

Hello List, Does anyone know of any user agent - commercial or opensource - that supports symmetric RTP for both audio and video? Thanks in advance, Dhiraj Bhuyan Network Security Specialist, BT Exact Business Assurance Solutions Tel: +44 1473 643932 Mob: +44 7962 012145 Email: dhiraj.2.bhuyan(a)bt.com

2 1

[Serusers] caller id
by M.I. Deen 28 Jan '04

28 Jan '04

Dear List, I have a Cisco 1700 FXO and set up a dial peer to talk to ser, on a sip session. I am trying to establish how I could identify a sip registered user in ser, based on the caller id value obtained from the sip session record passed from the Cisco to sip server. The identified user in ser will be recepiet (To:) in the INVITE message. Appreciate yours assistance. Deen

1 0

[Serusers] cpl-c module in 0.8.12
by Mario Kolberg 28 Jan '04

28 Jan '04

Hi, I'm trying to use the CPL-C module in ser version 0.8.12 (stable). However, I cannot get it to compile. I'm getting the following error message: gcc -fPIC -DPIC -g -O9 -funroll-loops -Wall -Wa,-xarch=v8plus -DNAME='"cpl-c.so"' -DVERSION='"0.8.12"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 2.95"' -D__CPU_sparc64 -D__OS_solaris -DCFG_DIR='"/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_TCP -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT … [View More]

2 1

Re: [Serusers] Re: Peering SER and Cisco Call Manager
by Jan Janak 28 Jan '04

28 Jan '04

Please always CC the list. Read the admin guide or search through the mailing lists archives. Look for t_relay_to_udp, forward or similar functions. If you gateway is running on IP 1.2.3.4 then you can use t_relay_to_udp("1.2.3.4"); http://iptel.org/ser/admin.html Jan. On 27-01 12:46, Raymond May wrote: > Jan, > Sorry to bother you again but I am trying to figure > out how to configure SER to forward digits to the > SIP/H.323 gateway ( Forward Digits from SER to > SIP/H.… [View More]

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RE: [Serusers] Voicemail
by Joao Sampaio 28 Jan '04

28 Jan '04

Thank you Raphel, What a mistake, I was blind! I have now the voicemail working, at least answering to the caller, I didn't yet test if it records the message and send it to the user email. I have to work now in other things and later I will return to SEMS. Thank very much. Best regards, João -----Original Message----- From: Raphael Coeffic [mailto:rco@iptel.org] Sent: terça-feira, 27 de Janeiro de 2004 16:04 To: Joao Sampaio; serusers(a)lists.iptel.org Subject: Re: [Serusers] … [View More]

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sr-users January 2004