sr-users December 2003

sr-users@lists.kamailio.org

108 participants
257 discussions

[Serusers] Problems with Grandstream budgetone and handytone

by Maxim Sobolev

Folks, It is not really ser-specific problem, but I think that it is a good place to ask. We are having problems with getting BT/HT working with our system - phones with firmware versions 1.0.4.26 and 1.0.4.17 don't send ACK for final 200 OK if the in-band alerting (183 session progress) is enabled in the remote UA. Can somebody confirm this problem? And it seems that version 1.0.3.81 is OK, but unfortunately there is no way to perform firmware downgrade. Regards, Maxim

20 years, 7 months

[Serusers] pstn relay help

by Madan

Hi Can somebody please send me working cfg for ser and dial peer plan for cisco regards, Madan

20 years, 7 months

[Serusers] pstn relay test

by Madan

Hi, can someebody at list help me with pstn relay test, i have got cisco voice box, i need a step by step guide on how to do this i ll really appreciate, i have already tried thru mailing list regards, Madan

20 years, 7 months

[Serusers] Re: PSTN relay help

by Madan

MessageHi I have ser up and running and i can make calls from/to SIP UA but i want to test PSTN termination as well for that purpose i have one cisco voice box at the end we have dial peer plan as 441 and 442 now i want to make call to a pstn no say 1126102911 how do i do that..? i have tried to follow the mailing list but all vain..cna somebody help me to test this out. PSTN relevant ser.cfg if (uri=~"sip:[4-9]+@(mydomain|202\.71\.135\.219)") { t_relay_to_udp("212.74.110.243", "5060"); break; }; regards, Madan

20 years, 7 months

[Serusers] SIP UA ACK Timedout!

by Madan

Hi, I have been facing a strange problem of ACK Failed. I m using SER with SJ Phone and Xlite as softphone to make calls..but after say 10-15 mins I m getting ACK Timedout! message on SJPhone..is this SIP UA bug or something to do with SER or config? regards, Madan get_hdr_field: cseq <CSeq>: <28684> <ACK> 0(8193) DEBUG: is_maxfwd_present: value = 70 0(8193) DEBUG: add_param: tag=2124396779 0(8193) end of header reached, state=29 0(8193) parse_headers: flags=256 0(8193) check_self - checking if host==us: 14==14 && [202.71.135.212] == [202.71.135.219] 0(8193) check_self: host != me 0(8193) check_self - checking if host==us: 14==14 && [202.71.135.219] == [202.71.135.219] 0(8193) ral(): Topmost route URI: 'sip:1111@202.71.135.219;ftag=2124396779;lr=on' is me 0(8193) parse_headers: flags=256 0(8193) DEBUG: get_hdr_body : content_length=0 0(8193) found end of header 0(8193) fnr(): No next Route HF found 0(8193) ral(): No next URI found 0(8193) lookup(): '' Not found in usrloc 0(8193) Warning: sl_send_reply: I won't send a reply for ACK!!

20 years, 7 months

[Serusers] Ser and Voicemail

by joao vianna

I new here, comming from Vocal-dev, I'm trying run ser and voicemail in different process. Can I have a simplest configuration for ser.cfg and ser-vm.cfg tested in (v) 0.8.12 ? Thanks, Joao Vianna. _________________________________________________________________ Enjoy the holiday season with great tips from MSN. http://special.msn.com/network/happyholidays.armx

20 years, 7 months

RE: [Serusers] SER supported feature list

by Hernandez, Cesar

I agree that features are not the only thing to look at for decision-making. The purpose for this table is to have a compiled list of SIP proxy specific features that are already available in SER (maybe it can be added to the SER admin manual), and have an idea of what may need development if required. >>* Private number (permanent Caller-ID blocking): >A way I do maintain privacy is I use a stupid user id >(You certainly know how much identity is dislocsed if you >send from mr_goerge_w_bush(a)hotmail.com....) Sure, but this may not be possible if some sort of authentication is required. Isn't this better handle on the SIP proxy server? >>* Caller blocking: >Yes -- we block requests from source causing annoying traffic. Is caller blocking system wide or configurable per user? >>* Call Transfer: >phone feature Don't the SIP proxy needs to support the SIP REDIR too? >>* Support for call time limit: >call-limitation is part of separate software Is there software already available or are you just saying that this will need to be developed as separate software? >>* Redundant server configuration: >available as separate software Again is there some software developed for SER or are you thinking about clustering software and/or making usage of DNS SRV resource records which will provide some kind of redundancy? Thanks, your input was appreciated. -Cesar -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Wednesday, December 17, 2003 3:42 PM To: Hernandez, Cesar; 'serusers(a)lists.iptel.org' Subject: Re: [Serusers] SER supported feature list At 09:19 PM 12/17/2003, Hernandez, Cesar wrote: >Hello, > >I'm compiling a table of features supported in SER, such table will be helpful to compare SER capabilities with other SIP proxy servers. Keep in mind that's a hard job. Some of the features may be end-device feature and some may have different mimics than you would expect. Also, it is not only specific telephony features which matter: importantly, there are things such as operational characteristics (performance, scalability) and ability to introduce new features and integrate with web. >* Private number (permanent Caller-ID blocking): A way I do maintain privacy is I use a stupid user id (You certainly know how much identity is dislocsed if you send from mr_goerge_w_bush(a)hotmail.com....) >* Caller-ID blocking per call (like Bell *67): see above, a phone may maintain multiple identities, one of them less informational than the other one. >* Caller blocking: Yes -- we block requests from source causing annoying traffic. >* Call Waiting (support more than one call): phone feature (see for example www.iptel.org/tt/ for such a phone) >* International routing dial plan: >* Least-cost routing: all trivial config options >* URI/Phone manipulation: Yes >* Find-me parallel calling: No find-me is built-in (known under the SIP term "parallel forking") >* Find-me consecutive calling: No unavailable now >* Call Transfer: phone feature >* Conference Calls (3-way calling): Yes >* NAT STUN: Yes just for sake of clarity: we have a STUN impelmentation, but it is separate from SER >* Billing Flat-file CDR: Yes >* Authentication: Yes >* Authorization: Yes >* RADIUS authentication/Authorization: Yes >* RADIUS accounting: Yes >* Support for call time limit: call-limitation is part of separate software >* ENUM support : Yes >* IM&P (Instant Messaging and presence): Yes (2G/SMS, SIMPLE/XMPP Jabber) >* WEB Provisioning Interfaces: Yes >* Programming interface to remotely provision from an OSS system (add/remove/modify user/settings): provisioning available as command-line or web tools >* Programming interface to remotely modify system settings from an OSS system (routing rules, gateways, etc.): available as server configuration options >* Redundant server configuration: available as separate softwarea >* Call return (like Bell *69): >* Call screening: see missed calls and click-to-dial in SERweb >* MWI (SIP NOTIFY): not now -jiri

20 years, 8 months

RE: [Serusers] Problems with Grandstream budgetone and handytone

by Klaus Darilion

Hi Max! I have one BT100 with versions: Program--1.0.3.81 Bootloader--1.0.0.7 HTML--1.0.0.18 I tested it and it sends ACK after a 200 OK, even if it receveid a 183 prior to the 200 OK. Klaus > -----Original Message----- > From: Maxim Sobolev [mailto:sobomax@portaone.com] > Sent: Wednesday, December 17, 2003 9:36 PM > To: serusers(a)lists.iptel.org > Cc: Oleksandr Kapitanenko > Subject: [Serusers] Problems with Grandstream budgetone and handytone > > > Folks, > > It is not really ser-specific problem, but I think that it is a good > place to ask. We are having problems with getting BT/HT > working with our > system - phones with firmware versions 1.0.4.26 and 1.0.4.17 > don't send > ACK for final 200 OK if the in-band alerting (183 session > progress) is > enabled in the remote UA. Can somebody confirm this problem? And it > seems that version 1.0.3.81 is OK, but unfortunately there is > no way to > perform firmware downgrade. > > Regards, > > Maxim > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

20 years, 8 months

RE: [Serusers] RE: Using Sip Sak

by Klaus Darilion

hi! 1. always CC to the list 2. if (uri=="bbcom-ser.bbcominc.com") will test the complete request uri, not only the domain part! 2 solutions: use regular expression comparision like: # this condition is true if request URI matches # the regular expression "@bat\.iptel\.org" if (uri=~"@bat\.iptel\.org") { # ... or better: use (uri == myself) from http://www.iptel.org/ser/doc/seruser/seruser.html#OPERATORS the expression uri==myself is true if the host part in request URI equals a server name or a server alias (set using the alias option in configuration file) -----Original Message----- From: Peter David [mailto:PDavid@bbcominc.com] Sent: Wednesday, December 17, 2003 9:34 PM To: Klaus Darilion Subject: RE: [Serusers] RE: Using Sip Sak Klaus I did what you recommended but it did not work. So tried this on the ser.cfg file. # 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=="bbcom-ser.bbcominc.com") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("bbcom-ser.bbcominc.com", "subscriber")) { www_challenge("bbcom-ser.bbcominc.com", "0"); break; }; adn still did not work ** request ** OPTIONS sip:pdavid@bbcom-ser.bbcominc.com SIP/2.0 Via: SIP/2.0/UDP radius01.bbcominc.com:32806 From: <sip:sipsak@radius01.bbcominc.com:32806> To: <sip:pdavid@bbcom-ser.bbcominc.com> Call-ID: <mailto:1694144032@radius01.bbcominc.com> 1694144032(a)radius01.bbcominc.com CSeq: 1 OPTIONS Contact: <sip:sipsak@radius01.bbcominc.com:32806> Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.8.6 Accept: text/plain message received: SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP radius01.bbcominc.com:32806;received=66.234.143.178 From: <sip:sipsak@radius01.bbcominc.com:32806> To: <sip:pdavid@bbcom-ser.bbcominc.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58 .8e13 Call-ID: 1694144032(a)radius01.bbcominc.com CSeq: 1 OPTIONS Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 66.234.143.180:5060 "Noisy feedback tells: pid=17507 req_src_ip=66.234.143.180 req_src_port=5060 in_uri=sip:pdavid@bbcom-ser.bbcominc.com out_uri=sip:pdavid@bbcom-ser.bbcominc.com via_cnt==71" ** reply received 6368.226 ms after first send and 2811.766 ms after last send ** SIP/2.0 483 Too Many Hops final received -----Original Message----- From: Klaus Darilion [mailto:darilion@ict.tuwien.ac.at] Sent: Wednesday, December 17, 2003 10:52 AM To: Peter David; serusers(a)lists.iptel.org Subject: RE: [Serusers] RE: Using Sip Sak -----Original Message----- From: Peter David [mailto:PDavid@bbcominc.com] Sent: Wednesday, December 17, 2003 7:43 PM To: Peter David; serusers(a)lists.iptel.org Subject: [Serusers] RE: Using Sip Sak Hello everyone, I did the config in accordance with the DOCS that was on the site....Isthere anything on this cfg file that I did not do correctly [root@bbcom-ser ser]# more ser.cfg # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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 > max_len ) { 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); 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==bbcominc.com) { maybe =="bbcominc.com" works better. klaus if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("bbcominc.com", "subscriber")) { www_challenge("bbcominc.com", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } Hello Everyone, I think I have my SER setup I just need to runn some test. Do I have to run SIpsak on the ser or another Box like a linux box. I did the following: [root@radius01 root]# sipsak -s sip:pdavid@bbcom-ser.bbcominc.com -vv New message with Via-Line: OPTIONS sip:pdavid@bbcom-ser.bbcominc.com SIP/2.0 Via: SIP/2.0/UDP radius01.bbcominc.com:32799 From: <sip:sipsak@radius01.bbcominc.com:32799> To: <sip:pdavid@bbcom-ser.bbcominc.com> Call-ID: 203787683(a)radius01.bbcominc.com CSeq: 1 OPTIONS Contact: <sip:sipsak@radius01.bbcominc.com:32799> Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.8.6 Accept: text/plain ** request ** OPTIONS sip:pdavid@bbcom-ser.bbcominc.com SIP/2.0 Via: SIP/2.0/UDP radius01.bbcominc.com:32799 From: <sip:sipsak@radius01.bbcominc.com:32799> To: <sip:pdavid@bbcom-ser.bbcominc.com> Call-ID: <mailto:203787683@radius01.bbcominc.com> 203787683(a)radius01.bbcominc.com CSeq: 1 OPTIONS Contact: <sip:sipsak@radius01.bbcominc.com:32799> Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.8.6 Accept: text/plain ** timeout after 500 ms** ** request ** OPTIONS sip:pdavid@bbcom-ser.bbcominc.com SIP/2.0 Via: SIP/2.0/UDP radius01.bbcominc.com:32799 From: <sip:sipsak@radius01.bbcominc.com:32799> To: <sip:pdavid@bbcom-ser.bbcominc.com> Call-ID: <mailto:203787683@radius01.bbcominc.com> 203787683(a)radius01.bbcominc.com CSeq: 1 OPTIONS Contact: <sip:sipsak@radius01.bbcominc.com:32799> Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.8.6 Accept: text/plain message received: SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP radius01.bbcominc.com:32799;received=66.234.143.178 From: <sip:sipsak@radius01.bbcominc.com:32799> To: <sip:pdavid@bbcom-ser.bbcominc.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58 .743f Call-ID: <mailto:203787683@radius01.bbcominc.com> 203787683(a)radius01.bbcominc.com CSeq: 1 OPTIONS Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 66.234.143.180:5060 "Noisy feedback tells: pid=1621 req_src_ip=66.234.143.180 req_src_port=5060 in_uri=sip:pdavid@bbcom-ser.bbcominc.com out_uri=sip:pdavid@bbcom-ser.bbcominc.com via_cnt==71" ** reply received 1195.150 ms after first send and 678.739 ms after last send ** SIP/2.0 483 Too Many Hops final received ======================================================================== ============ [root@radius01 root]# sipsak -T -s sip:pdavid@bbcom-ser.bbcominc.com warning: IP extract from warning activated to be more informational 0: 66.234.143.180 (18.961 ms) SIP/2.0 483 Too Many Hops 1: 66.234.143.180 (15.510 ms) SIP/2.0 483 Too Many Hops 2: 66.234.143.180 (41.453 ms) SIP/2.0 483 Too Many Hops 3: 66.234.143.180 (39.659 ms) SIP/2.0 483 Too Many Hops 4: 66.234.143.180 (56.881 ms) SIP/2.0 483 Too Many Hops 5: 66.234.143.180 (66.248 ms) SIP/2.0 483 Too Many Hops 6: 66.234.143.180 (69.779 ms) SIP/2.0 483 Too Many Hops 7: 66.234.143.180 (85.664 ms) SIP/2.0 483 Too Many Hops 8: 66.234.143.180 (100.143 ms) SIP/2.0 483 Too Many Hops 9: 66.234.143.180 (111.643 ms) SIP/2.0 483 Too Many Hops 10: 66.234.143.180 (121.099 ms) SIP/2.0 483 Too Many Hops 11: 66.234.143.180 (140.994 ms) SIP/2.0 483 Too Many Hops 12: 66.234.143.180 (140.692 ms) SIP/2.0 483 Too Many Hops 13: 66.234.143.180 (177.798 ms) SIP/2.0 483 Too Many Hops 14: 66.234.143.180 (163.843 ms) SIP/2.0 483 Too Many Hops 15: 66.234.143.180 (193.053 ms) SIP/2.0 483 Too Many Hops 16: 66.234.143.180 (209.526 ms) SIP/2.0 483 Too Many Hops 17: 66.234.143.180 (229.255 ms) SIP/2.0 483 Too Many Hops 18: 66.234.143.180 (225.985 ms) SIP/2.0 483 Too Many Hops 19: 66.234.143.180 (236.770 ms) SIP/2.0 483 Too Many Hops 20: 66.234.143.180 (269.157 ms) SIP/2.0 483 Too Many Hops 21: 66.234.143.180 (295.315 ms) SIP/2.0 483 Too Many Hops 22: 66.234.143.180 (302.434 ms) SIP/2.0 483 Too Many Hops 23: 66.234.143.180 (352.886 ms) SIP/2.0 483 Too Many Hops 24: 66.234.143.180 (320.909 ms) SIP/2.0 483 Too Many Hops [root@radius01 root]# Seems to time out. Should I even doit from another box or should I do it from the SER server? Any suggestions to make it work....any help is goods help thank you....all. Peter David Network Engineer BBCOM, Inc. Office 213-489-2156 x228 pdavid(a)bbcominc.com

20 years, 8 months

[Serusers] SER supported feature list

by Hernandez, Cesar

Hello, I'm compiling a table of features supported in SER, such table will be helpful to compare SER capabilities with other SIP proxy servers. I went quickly through the documentation, and I'm still reading, but I though that I may ask some of the gurus to help complete the table since I found no mention yet in the documentation, which does not necessary mean it's not supported, and that is why you will see a blank instead of NO. You may also think of extra cool features not mentioned in the table. Thanks in advance for any input, Regards, Cesar. ---------------------------------------------------------------------------- ----------------------------- SER SIP proxy feature list: * User-Based Access Control (Web): Yes * Call forwarding to alternate phone number on busy/noanswer: Yes * Call forwarding to alternate SIP phone on busy/noanswer: Yes * Call forwarding to external SIP voicemail on busy/noanswer: Yes * Support for multiple VoIP/PSTN gateways: Yes * Caller-ID: Yes * Private number (permanent Caller-ID blocking): * Caller-ID blocking per call (like Bell *67): * Caller blocking: * Call Waiting (support more than one call): * International routing dial plan: * Least-cost routing: * URI/Phone manipulation: Yes * Find-me parallel calling: No * Find-me consecutive calling: No * Call Transfer: * Conference Calls (3-way calling): Yes * NAT STUN: Yes * Billing Flat-file CDR: Yes * Authentication: Yes * Authorization: Yes * RADIUS authentication/Authorization: Yes * RADIUS accounting: Yes * Support for call time limit: * ENUM support : Yes * IM&P (Instant Messaging and presence): Yes (2G/SMS, SIMPLE/XMPP Jabber) * WEB Provisioning Interfaces: Yes * Programming interface to remotely provision from an OSS system (add/remove/modify user/settings): * Programming interface to remotely modify system settings from an OSS system (routing rules, gateways, etc.): * Redundant server configuration: * Call return (like Bell *69): * Call screening: * MWI (SIP NOTIFY): ---------------------------------------------------------------------------- --------------------------------

20 years, 8 months

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sr-users December 2003