sr-users June 2003

sr-users@lists.kamailio.org

55 participants
81 discussions

by Maxim Sobolev

Folks, This question isn't directly related to a SER, but does anybody know why Cisco IOS inserts several identical `c=<IP>' lines in SDP which it generates? No other UA I know about does that. I wonder because it creates a problems with SDP rewriting in nathelper module and I need to decide whether I have to modify nathelper to cope with that or open a Cisco TAC case instead. Any comments are appreciated! -Maxim SIP/2.0 200 OK Via: SIP/2.0/UDP 67.104.130.102:5061 From: <sip:0118000@67.104.130.102:5061;user=phone>;tag=bb4b39670c51e5c737c5de99b5450cc5 To: <sip:61412722777@64.180.102.72;user=phone>;tag=357A3908-2F6 Date: Tue, 17 Jun 2003 07:35:12 GMT Call-ID: 2920698556(a)192.168.1.100 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: <sip:61412722777@67.104.130.107:5060> Content-Type: application/sdp Content-Length: 285 v=0 o=CiscoSystemsSIP-GW-UserAgent 7069 1539 IN IP4 67.104.130.107 s=SIP Call c=IN IP4 67.104.130.107 ^^^^^^^^^^^^^^^^^^^^^^^ t=0 0 m=audio 19044 RTP/AVP 4 19 101 c=IN IP4 67.104.130.107 ^^^^^^^^^^^^^^^^^^^^^^^ a=rtpmap:4 G723/8000 a=rtpmap:19 CN/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15

20 years, 10 months

[Serusers] Processing of OPTIONS Request

by Chintan Thakker

Hi, Should the server produce a 100 - TRYING (Stateful)/200 - OK (Stateless) to the calling party when an OPTIONS message is sent to it. [11.2 processing of OPTIONS request @ rfc3261 - "The response code chosen MUST be the same that would have been chosen had the request been an INVITE"] Currently the server just forwards the OPTIONS message (to the called party) without sending a (200/100) back to the calling party. Thanks, Platform: GNU/Linux 2.4.18-14 i686 Server: Sip EXpress router (0.8.11pre29 (i386/linux)) -- Start OPTIONS message -- 2003/06/19 12:40:05.819450 192.1.2.88:5060 -> 192.1.2.17:5060 OPTIONS sip:9727610001@192.1.2.17 SIP/2.0.Via: SIP/2.0/UDP 192.1.2.88:5060; branch=z9hG4bKkdjuw.Max-Forwards: 70.From: "9727619271" <sip:9727619271@ 192.1.2.88>;tag=323.To: sip:9727610001@192.1.2.17.Call-ID: 517928895(a)192.1. 2.88.CSeq: 1 OPTIONS.. -- End OPTIONS message --

20 years, 10 months

[Serusers] ser 0.8.10 (i386/freebsd)

by Marcelo Schmidt

I installed ser yesterday and I was trying to get an UA to connect to it. I created just one user: <sip:marcelo@marcelo.homeunix.com> and used an iptel.org account to send a message, I got: 483 Too Many Hops. I attached the ser.cfg and a capture file from tcpdump. Thanks, Marcelo.

20 years, 10 months

[Serusers] Ser 0.8.11Pre29 Number of processes

by Damien Vicq

Hi, I have just upgraded our ser 0.8.10, which worked fine, to ser 0.8.011Pre29, with the Jabber and PA modules running. The problems I am encountering are: when in debug mode I often get error messages: XJAB:xjab_check_workers: worker has exited, status=0, err=-1, errno=10. I have not found the meaning of this error code. More serious: ser seems to create many processes (up to more than a hundred after some hours of use), which after a day or so crashes the servers it runs on (Linux Redhat 9, bi PIII-500). Do you know where this can come from ? Thanks in advance, Damien Vicq Bell Labs

20 years, 10 months

[Serusers] How to register MSN Messenger4.6 to non-default port of ser

by Zen Kato

Hi, I have been running ser-0.8.10/FreeBSD-4.7 for 6 months without problem. I changed ser default port,i.e.,5060 to 5065 on ser.cfg because some sip phone needs 5060 for their STUN/NAT traversal now. It works fine for sip hardphones using UDP:5065 of ser for REGISTER, but MSN Messenger4.6 on windows98 could not REGISTER eventhough the REGISTER packet was sent from the Messenger4.6 to the ser's 5065 as bellow; -------------------sample ------------------- U 2003/06/19 10:10:08.426264 192.168.0.21:1079 -> 192.168.0.26:5065 REGISTER sip:tel.no-ip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.21:14236..Fr om: <sip:050021@tel.no-ip.com>;tag=acbb1c1c-a23c-11d7-8627-0090997e7578..To : <sip:050021@tel.no-ip.com>..Call-ID: acbb1c1d-a23c-11d7-8627-0090997e7578 @192.168.0.21..CSeq: 2 REGISTER..Contact: <sip:192.168.0.21:14236>;methods= "INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"..User -Agent: Windows RTC/1.0..Expires: 0..Content-Length: 0.... --------------------sample end----------------- Messenger4.6 does not accept "aaa@foo.bar.com:5065" and never change UDP port: REGISTER sip:foo.bar.com From: <sip:aaa@foo.bar.com> To: <sip:aaa@foo.bar.com> I tried "iptel.org SIP Express Router v0.8.10 -- Admin's Guide, Troubleshooting,Q: Windows Messenger authentication fails. " but it does not work. Does someone know how to register Messenger4.6 to non-default port of sip proxy? Regards, Zen Kato

20 years, 10 months

[Serusers] Are Max-Forwards Mandatory ?

by Chintan Thakker

Hi, Since RFC3262, it seems that Max-Forwards header is mandatory in ACKs sent by the server. [ 28.1 Major Functional Changes @ rfc3261] ]Currently the server returns ACKs that do not have it. Thanks, Platform: GNU/Linux 2.4.18-14 i686 Server: Sip EXpress router (0.8.11pre29 (i386/linux)) - Start ACK Message - Raw_Data.1=0000: 41 43 4b 20 73 69 70 3a 39 37 32 37 36 31 30 30 ACK.sip:97276100 Raw_Data.2=0010: 30 31 40 31 39 32 2e 31 2e 32 2e 32 32 33 20 53 01(a)192.1.2.223.S Raw_Data.3=0020: 49 50 2f 32 2e 30 0d 0a 56 69 61 3a 20 53 49 50 IP/2.0..Via:.SIP Raw_Data.4=0030: 2f 32 2e 30 2f 55 44 50 20 31 39 32 2e 31 2e 32 /2.0/UDP.192.1.2 Raw_Data.5=0040: 2e 31 37 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 .17;branch=z9hG4 Raw_Data.6=0050: 62 4b 32 65 61 33 2e 61 32 61 65 33 34 32 36 2e bK2ea3.a2ae3426. Raw_Data.7=0060: 30 0d 0a 46 72 6f 6d 3a 20 39 37 32 37 36 31 39 0..From:.9727619 Raw_Data.8=0070: 32 37 31 20 3c 73 69 70 3a 39 37 32 37 36 31 39 271.<sip:9727619 Raw_Data.9=0080: 32 37 31 40 31 39 32 2e 31 2e 32 2e 38 38 3e 3b 271(a)192.1.2.88>; Raw_Data.10=0090: 74 61 67 3d 31 31 33 34 0a 43 61 6c 6c 2d 49 44 tag=1134.Call-ID Raw_Data.11=00a0: 3a 20 35 39 38 36 33 32 39 34 31 40 31 39 32 2e :.598632941@192. Raw_Data.12=00b0: 31 2e 32 2e 38 38 0a 54 6f 3a 20 39 37 32 37 36 1.2.88.To:.97276 Raw_Data.13=00c0: 31 30 30 30 31 20 3c 73 69 70 3a 39 37 32 37 36 10001.<sip:97276 Raw_Data.14=00d0: 31 30 30 30 31 40 31 39 32 2e 31 2e 32 2e 31 37 10001(a)192.1.2.17 Raw_Data.15=00e0: 3e 3b 74 61 67 3d 38 32 30 32 0a 43 53 65 71 3a >;tag=8202.CSeq: Raw_Data.16=00f0: 20 31 20 41 43 4b 0d 0a 55 73 65 72 2d 41 67 65 .1.ACK..User-Age Raw_Data.17=0100: 6e 74 3a 20 53 69 70 20 45 58 70 72 65 73 73 20 nt:.Sip.EXpress. Raw_Data.18=0110: 72 6f 75 74 65 72 28 30 2e 38 2e 31 31 70 72 65 router(0.8.11pre Raw_Data.19=0120: 32 39 20 28 69 33 38 36 2f 6c 69 6e 75 78 29 29 29.(i386/linux)) Raw_Data.20=0130: 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e 67 74 68 ..Content-Length Raw_Data.21=0140: 3a 20 30 0d 0a 0d 0a - End ACK Message -

20 years, 10 months

[Serusers] routing question

by Steve Blair

Hello: I have some Cisco phones registered to SER running on FreeBSD 4.7. Users of these phones want to dial a telephone number and have the call correctly routed out to the PSTN via a Cisco gateway on the network. The users of course only key-in a telephone number. No domain name or IP address. I thought the following code would correctly append the IP address and port of the gateway to their telephone number so the call could get routed correctly. This is not working. Instead either the IP address of the calling phone or the domain name for my domain is appended. Any thought on how to correct this would be appreciated. # check if it's about PSTN destinations through our gateway; if (uri=~"sip:9{10}") { route(4); break; }; route[4] ..... some account statements followed by ..... rewritehostport("ip-address-of-pstn-gateway:5060"); Thanks,Steve

20 years, 10 months