June 2004 - sr-users - kamailio.org

[Fwd: Re: [Serusers] cpled authentication problem]
by Bogdan-Andrei IANCU 12 Nov '04

12 Nov '04

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[Serusers] Voice Mail options for an SER environment
by Steve Blair 17 Jul '04

17 Jul '04

Hello: I'm interested in how this community is providing voicemail service to IP phone users registered to a SER proxy. If you can speak to this point I'd appreciate any feedback you can provide. Specifically I'm interested in: 1) What product is providing voice mail service? 2) With this system can users manipulate messages via the telephone set? 3) If the answer to #2 is yes then how is this functionality implemented. Do you provision a lead number for users to call into voicemail? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-7903 fax: 215-898-9348 sip:blairs@upenn.edu

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[Serusers] Parallel forking and reply error
by Zappasodi Daniele 12 Jul '04

12 Jul '04

Hi, all I'm implementing a black list, the control access to the gateway and other control policies and I have a problem with the call forking. In case of a call that generates a parallel forking I want the capability to select if accept or refuse the call for each single contact, but if the first reply that I send to the caller is an error message 4xx, the UA caller shuts down the call. What would I have to do to add my reject (caused by my access control policy) to the call forking administration and so reply an error only if all contacts can not be reachable? I'm trying to implement a solution to reply an error only if all the contacts can not be reachable. To do this thing I need to know the number of contacts that the lookup function has found for the request-uri that is arrived ( Is it possible to know if I have a single contact or a parallel forking?) and, if there are more than one contact associated to the sip uri, how can I go to serve the next contact in list without sending a reply for the destinations that I have to block? Any suggestion? Thanks Daniele

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[Serusers] Problems with aliases
by Bo Byrd 12 Jul '04

12 Jul '04

I cannot seem to get my system to work with aliases. I have the userloc and uri modules enabled as per the manual, as well as lookup("aliases"); command. I tried entering aliases with "serctl alias add 1001 sip:bo@domain.net" as well as "serctl alias add 1001(a)domain.net sip:bo@domain.net". Both commands successfully complete but I always get a busy signal on the other phone when I call 1001. 0(14359) SIP Request: 0(14359) method: <INVITE> 0(14359) uri: <sip:1001@domain.net;user=phone> 0(14359) version: <SIP/2.0> 0(14359) parse_headers: flags=1 0(14359) end of header reached, state=5 0(14359) parse_headers: Via found, flags=1 0(14359) parse_headers: this is the first via 0(14359) After parse_msg... 0(14359) preparing to run routing scripts... 0(14359) DEBUG : is_maxfwd_present: searching for max_forwards header 0(14359) parse_headers: flags=128 0(14359) end of header reached, state=9 0(14359) DEBUG: get_hdr_field: <To> [50]; uri=[sip:1001@domain.net;user=phone] 0(14359) DEBUG: to body [<sip:1001@domain.net;user=phone> ] 0(14359) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(14359) DEBUG: get_hdr_body : content_length=249 0(14359) found end of header 0(14359) DEBUG: is_maxfwd_present: max_forwards header not found! 0(14359) DEBUG: add_param: tag=2562321765 0(14359) end of header reached, state=29 0(14359) parse_headers: flags=256 0(14359) find_first_route(): No Route headers found 0(14359) loose_route(): There is no Route HF 0(14359) check_self - checking if host==us: 26==13 && [domain.net] == [x.x.x.x] 0(14359) check_self - checking if port 5060 matches port 5060 0(14359) lookup(): '1001' Not found in usrloc 0(14359) lookup(): '1001' Not found in usrloc 0(14359) parse_headers: flags=-1 0(14359) check_via_address(x.x.x.x, x.x.x.x, 0) 0(14359) receive_msg: cleaning up 0(14359) SIP Request: 0(14359) method: <ACK> 0(14359) uri: <sip:1001@domain.net;user=phone> 0(14359) version: <SIP/2.0> 0(14359) parse_headers: flags=1 0(14359) end of header reached, state=5 0(14359) parse_headers: Via found, flags=1 0(14359) parse_headers: this is the first via 0(14359) After parse_msg... 0(14359) parse_headers: flags=4 0(14359) DEBUG: add_param: tag=4ddff14623b40c25b5f0541000aec456.33da 0(14359) end of header reached, state=29 0(14359) DEBUG: get_hdr_field: <To> [92]; uri=[sip:1001@domain.net;user=phone] 0(14359) DEBUG: to body [<sip:1001@domain.net;user=phone>] 0(14359) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(14359) receive_msg: cleaning up # ser -V version: ser 0.8.12 (i386/freebsd) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.168 2003/10/12 15:09:08 andrei Exp $ main.c compiled on 14:15:39 Nov 21 2003 with gcc 2.95 Bo Byrd

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[Serusers] Asterisk/Digium as PSTN gateway and SER as backend soft switch?
by Tony Lum 12 Jul '04

12 Jul '04

Hi, Has any one done using Asterisk/Digium T1 card as PSTN gateway, and SER as the backend soft switch? I am looking to install Asterisk boxes in a few cities to act as PSTN gateways and using SER to accept these calls. SER will have to communicate with other long distance carriers to allow world wide calling. Will this combination work? I like to know your comments or suggestions on this combination. thanks Tony Lum

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RE: [Serusers] SER, ATA-186 & Cisco AS5300
by CM Rahman 12 Jul '04

12 Jul '04

Please email your ser.cfg file Thanks &*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. CTO CCNP, MCSE Security "Secure your self by securing your System" CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13704 Research Blvd. Building O-Suite 4 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Linus Surguy Sent: Saturday, June 19, 2004 12:08 PM To: serusers(a)lists.iptel.org Subject: [Serusers] SER, ATA-186 & Cisco AS5300 I hope this is not an FAQ, I've tried searching the available info. Using Ser 0.8.12, Cisco AS5300 with IOS 12.3.6 & a ATA-186. ATA-186 (behind NAT) -> SER -> AS5300 On the INVITE, we get: 10w6d: httpish_msg_process_network_msg: Content Length 261, Bytes Remaining 256 10w6d: HandleUdpSocketReads: SIP Message incomplete, trashed ser.cfg is just using t_relay_to_udp and not doing anything fancy. Any ideas!? Linus (Full debug from debug ccsip all) 10w6d: Received: INVITE sip:448000664901@gk.magrathea.net;user=phone SIP/2.0 Max-Forwards: 10 Record-Route: <sip:448000664901@213.166.5.135;ftag=651568306;lr=on> Via: SIP/2.0/UDP 213.166.5.135;branch=z9hG4bKbf06.7adc86c4.0 Via: SIP/2.0/UDP 213.162.109.118:5060 From: sip:linus@gk.magrathea.net;tag=651568306 To: <sip:448000664901@gk.magrathea.net;user=phone> Call-ID: 82733180(a)213.162.109.118 CSeq: 1 INVITE Contact: <sip:linus@213.162.109.118:5060;transport=udp> User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Expires: 300 Content-Length: 261 Content-Type: application/sdp v=0 o=linus 115389 115389 IN IP4 213.162.109.118 s=ATA186 Call c=IN IP4 213.162.109.118 t=0 0 m=audio 16384 RTP/AVP 4 8 0 101 a=rtpmap:4 G723/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 10w6d: httpish_msg_process_network_msg: Content Length 261, Bytes Remaining 256 10w6d: HandleUdpSocketReads: SIP Message incomplete, trashed _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] voicemail, again
by Milivoje Mirovic 12 Jul '04

12 Jul '04

Hello everybody, I'm trying to configure voicemail using sems. I've figured out that I have problem with 'vm' module in second ser instance. In cfg file I have problem with these two lines: loadmodule "/usr/lib/ser/modules/vm.so" modparam("voicemail", "db_url","sql://root:@127.0.0.1/ser") I recieve error: ser[4525]: DBT:dbt_cache_get_db: database [sql://root:@127.0.0.1/ser] does not exists! ser[4525]: DBT:dbt_init: cannot get the link to database ser[4525]: ERROR; vm_init_child: could not init db sql://root:@127.0.0.1/ser ser[4525]: init_mod_child(): Error while initializing module voicemail ser[4525]: init_child failed However I use database properly with same parameters for other modules. I tried with another database (created with 'ser_mysql.sh copy vm'), unfortunately I have same results. I recompiled ser, but I didn't found lines to uncomment to activate voicemail, so I concluded that voicemail is enabled by default in ser 0.8.12. Can you help me solve this problem? Milivoje __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail

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[Serusers] get_contact_uri: Error while parsing Contact body
by Carlo Pires 12 Jul '04

12 Jul '04

Hi, I'm getting this message when my grandstream ht286 trys to register. It seens ht286 isn't sending authorization header field to SER. Or, ht286 sending strange contact (*) disturbs SER to process REGISTER requests ? This message starts few days ago. As I changed my ser box to last cvs version these days I'm confuse if it's a grandstream's problem or a ser's problem. ngrep logs: attached -- []s Carlo Pires U 193.126.48.33:5060 -> 200.193.194.142:5060 REGISTER sip:sip.televoip.net SIP/2.0..Via: SIP/2.0/UDP 192.168.0.111;branch=z9hG4bK4977acbbc428883a..From: "Reginaldo Paula Souza" <sip:reginaldo_ptt@sip.televoip.net>;tag=307110492b802983..To: <sip:reginaldo_ptt@sip.televoip.net>..Contact: *..Call-ID: 6221319848b4afe5@192.168.0.111..CSeq: 106 REGISTER..Expires: 0..User-Agent: Grandstream HT286 1.0.4.68..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 200.193.194.142:5060 -> 193.126.48.33:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.0.111;branch=z9hG4bK4977acbbc428883a;rport=5060;received=193.126.48.33..From: "Reginaldo Paula Souza" <sip:reginaldo_ptt@sip.televoip.net>;tag=307110492b802983..To: <sip:reginaldo_ptt@sip.televoip.net>;tag=a791a9ab0b94eb4c05eab9b15c0fec4f.f9b6..Call-ID: 6221319848b4afe5@192.168.0.111..CSeq: 106 REGISTER..P-NATed-Caller: Yes..WWW-Authenticate: Digest realm="sip.televoip.net", nonce="40d46cd7c8d155389bef79bb516a0a2b2a5524f8"..Server: Sip EXpress router (0.8.13-dev-30-db_api (i386/linux))..Content-Length: 0..Warning: 392 200.193.194.142:5060 "Noisy feedback tells: pid=11293 req_src_ip=193.126.48.33 req_src_port=5060 in_uri=sip:sip.televoip.net out_uri=sip:sip.televoip.net via_cnt==1"....

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[Serusers] (no subject)
by CM Rahman 12 Jul '04

12 Jul '04

Can somebody explain the error ? U 216.236.168.188:4566 -> 216.236.160.11:5060 REGISTER sip:ccsi.com SIP/2.0..Via: SIP/2.0/UDP 216.236.168.188:11185..Max- Forwards: 70..From: <sip:pappusip@ccsi.com>;tag=efbf15e06e864dd396fb40e4423 2f5d0;epid=f8501f5093..To: <sip:pappusip@ccsi.com>..Call-ID: cae57e249e2e4e f6be07a82d964b2946@216.236.168.188..CSeq: 1 REGISTER..Contact: <sip:216.236 .168.188:11185>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CA NCEL, NOTIFY, ACK, REFER"..User-Agent: RTC/1.2.4949 (Messenger 5.0.0482)..E vent: registration..Allow-Events: presence..Content-Length: 0.... ######################################################################## ######################################################################## ######################################################################## ############################################ U 216.236.160.11:5060 -> 216.236.168.188:11185 SIP/2.0 477 Unfortunately error on sending to next hop occured (477/TM)..Vi a: SIP/2.0/UDP 216.236.168.188:11185..From: <sip:pappusip@ccsi.com>;tag=efb f15e06e864dd396fb40e44232f5d0;epid=f8501f5093..To: <sip:pappusip@ccsi.com>; tag=2f560d56b3e8a9ab008f885604d9e30a-f375..Call-ID: cae57e249e2e4ef6be07a82 d964b2946@216.236.168.188..CSeq: 1 REGISTER..Server: Sip EXpress router (0. 8.12 (i386/freebsd))..Content-Length: 0..Warning: 392 216.236.160.11:5060 " Noisy feedback tells: pid=13999 req_src_ip=216.236.160.11 req_src_port=506 0 in_uri=sip:ccsi.com out_uri=sip:ccsi.com via_cnt==62".... ######################################################################## ######################################################################## ###############################^Cexit &*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. CTO CCNP, MCSE Security "Secure your self by securing your System" CompTI Security Plus Certified CCS Internet http://www.ccsi.com <http://www.ccsi.com/> 13704 Research Blvd. Building O-Suite 4 Austin, TX 78750 Tel: 512-257-2274 Ex: 115

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[Serusers] SUBSCRIBE response (200) does not include Expires header
by Doug Raum 12 Jul '04

12 Jul '04

I'm starting to use the pa module, and noticed that when it gets a SUBSCRIBE, the 200 response doesn't have an expires header, as it should. (RFC 3265, 3.1.1) SUBSCRIBE sip:vmail@192.168.0.5 SIP/2.0 Via: SIP/2.0/UDP voipdev3.zoom.com:33050;rport From: <sip:8100100@192.168.0.5:5060>;tag=a67e989d-13c4-173-5ac26-24be To: <sip:vmail@192.168.0.5> CSeq: 1 SUBSCRIBE Call-ID: b5bb80-a67e989d-13c4-173-5ac26-5440 Expires: 3600 Event: presence Max-Forwards: 70 Supported: replaces Contact: <sip:8100100@192.168.0.5:5060> Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP voipdev3.zoom.com:33050;rport=33050;received=192.168.0.10 From: <sip:8100100@192.168.0.5:5060>;tag=a67e989d-13c4-173-5ac26-24be To: <sip:vmail@192.168.0.5>;tag=92462461b2a59d6c6be4b5af5841f077-b62f CSeq: 1 SUBSCRIBE Call-ID: b5bb80-a67e989d-13c4-173-5ac26-5440 Content-Length: 0 I haven't spent a lot of time looking at the code for pa, but should this be an easy thing to fix? Thanks, -- Doug

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