- sr-users - kamailio.org

[Serusers] FCP Server Howto
by Kofi Obiri-Yeboah 18 Feb '05

18 Feb '05

Paul, Here is a cut and paste of a posting you did back on Dec 08, 2004. You said "Here is a complete ser.cfg I posted a few days ago which uses rtpproxy." However you did not include the cfg info as you intended Could you please point to where you posted that info. Thanks KOFI Success is a state of Mind

1 0

[Serusers] SIP_DOMAIN and bootup question
by Dana Olson 18 Feb '05

18 Feb '05

I've been looking through the admin manual and haven't found what I'm looking for yet. I point my SIP client to, say, domainname.com, and on my DNS server, I point this to my SER system's IP address, say, 10.10.10.10. When I had SIP_DOMAIN=10.10.10.10, I experienced long delays between pressing hangup and the call actually being hung up. When I changed this to SIP_DOMAIN=domainname.com and ran /etc/init.d/ser restart, the behaviour is as expected. I put export SIP_DOMAIN=domainname.com into /etc/profile and reboot the system, I try dialing and then hanging up again, and it's back to the same behaviour I mentioned above. If I run /etc/init.d/ser restart, then it is fixed again. Am I doing something wrong here? I don't want to have to run a manual restart after every system boot. Thanks in advance.

1 0

[Serusers] ser 0.8.14 and what sems version?
by Rosario Pingaro 18 Feb '05

18 Feb '05

I'd like to run sems couppled with two ser sessions. I have to run 0.8.14 because i need serweb. What sems version I have to use and where to download it? In fact I tried the cvs and I got some very strange problem and using the 0.1.0 I get: wrong FIFO interface version! please help! Thanks Rosario

1 0

[Serusers] error on 0.9.0 ext module
by Nhadie 18 Feb '05

18 Feb '05

Hi All, I'm trying to install 0.9.0, I had some problems compiling: ext and extcmd is not getting compiled. Are these modules critical? This is the error on ext modules: ext.c: In function `ext_rewrite': ext.c:340: warning: passing arg 4 of `append_branch' makes pointer from integer without a cast ext.c:340: error: too few arguments to function `append_branch' make: *** [ext.o] Error 1 This is the error on the extcmd.c extcmd_funcs.c: In function `tuac_callback': extcmd_funcs.c:327: error: structure has no member named `cbp' extcmd_funcs.c:336: error: structure has no member named `cbp' extcmd_funcs.c: In function `send_sip_req': extcmd_funcs.c:435: warning: passing arg 7 of pointer to function from incompatible pointer type make: *** [extcmd_funcs.o] Error 1 Thank you in advanced nhadie

1 0

RE: [Serusers] SER stops responding
by Samuel Osorio Calvo 18 Feb '05

18 Feb '05

Hi, I'm not an expert so maybe my guessings are not wright, but I'll try to help: There are several possiblities for these "SER time-outs". The first is the DNS resolving process. When a SER child is making a DNS query it is blocked, and it does not answer to further incoming queries. Therefore if all your childs are making this DNS queries, no process will be available for responding the new requests. To solve this situation, use stateful behaviour because only the first request within a transaction requiring DNS resolution will *block*. In the other hand, if you use stateless SER, ALL requests will require a DNS query... (by the way, this information is taken from the SER's admin guide....have all of the mailing-list subscribers read it? I recomend it....) Another possibility is the flexible-but-unreliable exec calls from SER config file. It is also stated in the admin guide that the use of exec reduces a lot SER performance and you should be really careful using them.......having a SER child stoped for an uncertain amount of time until the exec command finishes can lead to "blocking" phases, for instance. One comment: why not changing the current DNS blocking queries for a non-blocking queries such as reSIProcate does with ARES library??? I have no idea if it would be a high effort in the SER core...it's just a possible enhancement.... I hope I was not *completely* wrong and I could help you, Samuel. Unclassified. >>> "Matt Schulte" <mschulte(a)netlogic.net> 02/18/05 03:55PM >>> I've had this problem too, especially running syslog debug or external processes. what ver of ser are you running? We had to ditch a lot of our external processes because I assumed they took too long to run, no one on the list was any help so I just gave up. Not being a coder is very frustrating.. -----Original Message----- From: AJ Grinnell [mailto:ajgrinnell@gmail.com] Sent: Friday, February 18, 2005 8:44 AM To: serusers(a)lists.iptel.org Subject: [Serusers] SER stops responding I am having a strange problem with 0.9.0. At random times, SER will just stop responding. I run ethereal, and can see INVITEs, REGISTERs and OPTIONs requests, but SER does not respond to any of them. The process does not die, and moni tells me that SER is still running, just no respones. Sometimes SER will recover after a few minutes, most of the time it doesnt. Any ideas? _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] SIP phone Ringing for ever.......
by Srinath Ramachandra 18 Feb '05

18 Feb '05

Dear All, I am testing a SIP based CPE which is a System On Chip (SOC). SER is the SIP server. When I make a call from A to B and if B is not picking up the call, A recieves ringing tone until it is hung up. My question is: 1. Is the CPE under test failing from the RFC 3261 point of view ? 2. Is there a provision in the SER config file, to terminate an INVITE transaction if there is no response within a specified amount of time, say 2 min ? thanks in advance for your kind replies.

2 1

[Serusers] aliases and table and ringing two phones using same ID
by Iqbal 18 Feb '05

18 Feb '05

Hi I am trying to get a call blasting kind of thing setup, where a call to alias 101 goes to abc(a)xyz.com and qwe(a)xyz.com In 0.8.x this seemed possible by adding alias 101 for both addresses, not sure if this is/was the correct way of doing it, however in 0.10 it seems as if the username in alias table is set as a primary key, which means i cant do this, is this a feature, if so, whats the best way of ringing multiple phones Iqbal

1 1

[Serusers] ser cvs head
by Matt Schulte 18 Feb '05

18 Feb '05

what's the correct location to get the latest cvs head? the website points to old stuff

2 1

RE: [Serusers] Re: [Serdev] least cost routing module
by Matt Schulte 18 Feb '05

18 Feb '05

The point of LCR is to lower your own costs, not benefit your customers. -----Original Message----- From: Juha Heinanen [mailto:jh@lohi.tutpro.com] Sent: Thursday, February 17, 2005 3:03 PM To: Martin Koenig Cc: serusers(a)lists.iptel.org Subject: [Serusers] Re: [Serdev] least cost routing module Martin Koenig writes: > Another question, wouldn't it make sense to allow the lcr engine to > somehow rewrite the RURI for a specific call destination, i.e. add a > prefix or suffix to the RURI? Or maybe even different header fields for > each destination? everything is possible, but as i said before, i add features that my customers are not asking only if i have spare time. -- juha _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

RE: [Serusers] CANCEL "forbidden"
by Matt Schulte 18 Feb '05

18 Feb '05

FYI... if (!method == "CANCEL") { if (!is_user_in("Credentials", "int")) { sl_send_reply("403", "No access"); break; } } -----Original Message----- From: Matt Schulte Sent: Thursday, February 17, 2005 1:24 PM To: serusers(a)lists.iptel.org Subject: [Serusers] CANCEL "forbidden" Someone posted awhile back how to make cancel's *not* challenge/auth? I can't find it anywhere, anyone have that post? _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

RE: [Serusers] SER stops responding
by Matt Schulte 18 Feb '05

18 Feb '05

I've had this problem too, especially running syslog debug or external processes. what ver of ser are you running? We had to ditch a lot of our external processes because I assumed they took too long to run, no one on the list was any help so I just gave up. Not being a coder is very frustrating.. -----Original Message----- From: AJ Grinnell [mailto:ajgrinnell@gmail.com] Sent: Friday, February 18, 2005 8:44 AM To: serusers(a)lists.iptel.org Subject: [Serusers] SER stops responding I am having a strange problem with 0.9.0. At random times, SER will just stop responding. I run ethereal, and can see INVITEs, REGISTERs and OPTIONs requests, but SER does not respond to any of them. The process does not die, and moni tells me that SER is still running, just no respones. Sometimes SER will recover after a few minutes, most of the time it doesnt. Any ideas? _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] SER stops responding
by AJ Grinnell 18 Feb '05

18 Feb '05

I am having a strange problem with 0.9.0. At random times, SER will just stop responding. I run ethereal, and can see INVITEs, REGISTERs and OPTIONs requests, but SER does not respond to any of them. The process does not die, and moni tells me that SER is still running, just no respones. Sometimes SER will recover after a few minutes, most of the time it doesnt. Any ideas?

1 0

Re: [Serusers] How to authorize sip proxies
by Pavel Siderov - Hostmates 18 Feb '05

18 Feb '05

I'm using proxy_authorize function but something goes wrong because I check also whether user is in defined group to make calls. In other words if sip proxy w/ ip 12.34.56.78 forwards traffic to sip proxy w/ ip 10.10.10.10 and user:1234@12.34.56.78 doesnt exsist in 10.10.10.10 group table it's call is rejected. Thanks, Pavel ----- Original Message ----- From: Pavel Siderov - Hostmates To: serusers(a)lists.iptel.org Sent: Friday, February 18, 2005 11:23 AM Subject: [Serusers] How to authorize sip proxies Hi there, Can you help me how to stop other sip proxies to forward calls to my ser and how to authorize some of them to forward calls to me. Thanks and regards, Pavel Siderov ------------------------------------------------------------------------------ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] ser-0.814 & rtpproxy stability
by support 18 Feb '05

18 Feb '05

Hi everyone, It seems that ser-0.8.14 will become "dead" after running for a few days. I am using rtpproxy and nathelper with ser. After I restarted ser and rtpproxy, everything becomes fine again. Would this problem be caused by ser-0.814 or rtpproxy? I don't know ser-0.814 or rtpproxy will be stable or not. Have you encountered such problem? Also, is ser-0.9 more stable than ser-0.8.14? Best Regards, Thomas

2 1

[Serusers] Authentication digest
by Raÿfffffffffffal Cumplido 18 Feb '05

18 Feb '05

Hi, I'm trying to register my sip application to the SER Register using MD5 authentication but I don't know what parameters I had to concatenate to make the response, and how to cancatenate it. If someone can help me. Thanks. I have readed something about the MD5 of (username+password+nonce+method+uri), but concatenated with a point? Is my first time with the authentication ... :P Raúl ______________________________________________ Renovamos el Correo Yahoo!: ¡250 MB GRATIS! Nuevos servicios, más seguridad http://correo.yahoo.es

2 1

[Serusers] SER NAT problems
by sad sack 18 Feb '05

18 Feb '05

I have SER and one Cisco 7960 phone in public network. One cisco 7960 in a private network behind NAT. I have configured SER with nathelper and rtpproxy (both are downloaded via CVS). Call from public to private network works fine. But when I call from private to public network, then phone rings fine, but there is no audio. I'm wondering how can I get both ways to work ? I have just one ser-server, do I have to two servers, which one is proxy and another is used as outbound proxy. __________________________________ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com

1 0

[Serusers] Call forwarding on SER
by Ritesh Jalan 18 Feb '05

18 Feb '05

How to configure call forwarding on SER?? Thanks & Regards Ritesh Jalan

2 2

[Serusers] How to authorize sip proxies
by Pavel Siderov - Hostmates 18 Feb '05

18 Feb '05

Hi there, Can you help me how to stop other sip proxies to forward calls to my ser and how to authorize some of them to forward calls to me. Thanks and regards, Pavel Siderov

1 0

[Serusers] Questions about avp_db_load() at avpops module
by Charles Wang 18 Feb '05

18 Feb '05

Dear ALL: I have defined these lines at my ser.cfg. modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") modparam("avpops", "avp_aliases", "voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702") Q1: But I don't know what it mean about 700 or 701. Is it a numeric ID of one UA? And I have refered to http://www.voice-system.ro/docs/avpops And find many examples in it. But I still can't understand what it mean about avp_db_load(). my avp table is named as "usr_preferences" and it is empty before I use avpops modules. Q2: Is it should have some initilized records in it? Otherwise, what will be loaded with avp_db_load()? For this web page said: avp_db_load("$ruri/domain","i:/domain_preferences"); - loads all AVPs with ID from 'domain_preferences' table for domain from RURI Q3: Is "$ruri" the caller user's URI? and the "domain" is the same? Is "domain_preferences" table name? Is it equal to my "usr_preferences"? Or it is just a virtual table in memory? Thank you for your answer my stupid questions. Charles

1 0

[Serusers] nathelper ERROR: extract_mediaip: no `c=' in SDP
by Charles Wang 18 Feb '05

18 Feb '05

Thanks Marian's help and answer. I can make a call from domainA to domainB via PDT module. But I got another error message from nathelper or rtpproxy. It is "ERROR: extract_mediaip: no `c=' in SDP" from /var/log/messages. I also find this message at source code of nathelper. So I think it should happen from this module. It seems voice channel can't pass throught it. I make a call from UA_A ( under NAT ) from domainA and using PDT to redirect to UA_B via domainB. UA_B has ring and it can answer. I can hear the UA_A's sound from UA_B. The sound of UA_B is failed to UA_A. UA_A(NAT) ==> domainA(PDT) ==> domainB(nathelper+rptproxy) ==> UA_B(NAT) SOUND_A ==========================================> UA_B UA_A <===== XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX === SOUND_B Do I put the prefix2domain() in a wrong place?? Or Is it a bug ofi nathelper & rtpproxy? And my ser.cfg as below: Best Regard. Charles # # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=8 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ listen=xxx.xxx.xxx.xxx alias=ser.xxx.net alias=ser alias=xxx.xxx.xxx.xxx check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ 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" loadmodule "/usr/local/lib/ser/modules/textops.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" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/speeddial.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/pdt.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) modparam("usrloc", "db_mode", 1) # -- 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) # -- nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # -- db_url params -- modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc|pdt", "db_url", "mysql://ser:heslo@localhost/ser") # -- use_domain params -- modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", "use_domain", 0) # -- permissions params -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- accounting params -- modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 0) # -- domain params -- modparam("domain", "db_mode", 1) # -- avp params -- modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") #modparam("avpops", "use_domain", "1") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") modparam("avpops", "avp_aliases", "voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702") # To use more than one tables example #modparam("avpops", "db_scheme", "scheme1:table=subscriber;uuid_column=uuid;value_column=first_name") # -- tm params -- modparam("tm", "fr_timer", 15) modparam("tm", "fr_inv_timer", 22) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # -- pft params -- modparam("pdt", "db_table", "prefix_domain") modparam("pdt", "prefix", "") #modparam("pdt", "start_range", 2000) #modparam("pdt", "terminator", 0) modparam("pdt", "hsize_2pow", 2) modparam("pdt", "sync_time", 300) modparam("pdt", "clean_time", 600) # ------------------------- 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; }; if (method=="INVITE" || method=="BYE") { setflag(1); }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("SER: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # 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; }; prefix2domain(); 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") { # Uncomment this if you want to use digest authentication if (!www_authorize("ser.xxx.net", "subscriber")) { www_challenge("ser.xxx.net", "0"); break; }; save("location"); break; }; - lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } On Thu, 17 Feb 2005 12:49:10 +0100, Marian Dumitru <marian.dumitru(a)voice-sistem.ro> wrote: > Hi Charles, > > You have to use interdomain prefixes. Set on domainA a prefix for > recognizing dialed numbers in domainB. > The logic is like: > PrefixToB+numberInB@domainA -> numberinB@domainB > > you can do this directly from script or by using PDT module. > > Best regards, > Marian

2 2

[Serusers] UA records are lost while UA is still online
by szj 18 Feb '05

18 Feb '05

Hi: I use the latest code from SER cvs. the mysql db and ser are located on two computer. I registered 3 sip UA on the registrar which alse is a proxy. all the ua will expire after 1 hour. But after a while, i don't know how long it is. I found the location table has lost the ua record. my three sip ua is still online. I wonder whether the registrar and usrloc have other condition I don't know. What happens to them?? Thanks for your advice and help! Best Regards Sun Zongjun

1 0

[Serusers] SIP user authentication
by Lakshmi Sharma 18 Feb '05

18 Feb '05

Hello, I've installed the latest version of SER on SUSE linux 9.0. Setup MYSQL as well to test sutehntication/challenge. When SER receives Register request, I'm seeing these msgs on console: "Received SIP_msg. No mem for the SIP_msg". It would be great if someone could help me with this. Thanks & Regards Lakshmi

1 0

[Serusers] SER/Asterisk 183 with SDP message
by scm-j＠nuntius.com 18 Feb '05

18 Feb '05

I am using Asterisk for IVR in conjunction with SER. My service provider prefers that I send a 183 progress message before sending a 200 OK, in response to their INVITE. I am using the "progressinband" option in sip.conf along with the Ringing() in extensions.conf for this purpose. When I set progressinband=no, Asterisk (i.e Ringing() command) sends a 100, followed by 180, followed by 200OK in response to an INVITE. However, when I set progressinband=yes, Asterisk sends a 100, followed by 180, followed by 183 (with SDP) followed by 200OK. I want asterisk to send only the 183 without the 180. i.e 100, followed by 183, followed by 200 OK, in response to an INVITE. How do I do that? regards, SCM

1 0

[Serusers] Fax problem with SER/Asterisk
by scm-j＠nuntius.com 17 Feb '05

17 Feb '05

I am facing some problems with sending fax calls. I am using G.711 codec. However, my service provider (Level-3) does not support T.38 fax relay. According to Level-3 they "think" the problem lies in the "Media Description" field of the SDP message I am sending them. The problem is related to the "Media Format" sub-field (value 101) that I am sending, which according to them should be removed. The SDP "Media Description" field I am sending, looks as follows: "m=audio 20744 RTP/AVP 0 101" which is decoded by Ethereal as follows: Media Type: audio Media Port: 20744 Media Proto: RTP/AVP Media Format: ITU-T G.711 PCMU Media Format: 101 My call is initiated by Asterisk which is sent to Level-3 via SER Asterisk ---> SER ---> Level-3 The SIP call flow looks fine for the call. The sending fax sends the page and then gives an error. The receiving side gives me an error and prints a blank page. My questions are as follows: 1) Could the "Media Format" 101 be the problem? If so, how do I remove the 101? This change is probably in Asterisk, as it initiates the call, and not SER, or probably both. 2) If not, should I be looking elsewhere in SER/Asterisk to solve this problem? regards, SCM

3 5

[Serusers] LCR Module Question
by Java Rockx 17 Feb '05

17 Feb '05

Hi All. I've got a question about the new LCR module that Juha commited to CVS. We're located in the US so this question may pertain only to ser installations that follow the North American Numbering Plan. For us to _really_ perform LCR decisions we need to look at the NPA and NXX (ie, the first 6 digits of a 10-digit phone number) of the origination and termination numbers. This this we would decide where to route the call. The reason is that it matters here we get DIDs from (sometimes they're provided by 3rd party PSTN gateway operators) and where our softswitch(s) are physically located. So at a high level the matrix of routing decisions would look something like this: Orig NPANXX Term NPANXX PSTN Gateway --------------------- ----------------------- ----------------------- 407566 321251 10.10.0.40 814332 202442 67.93.11.31 So can the LCR module accomodate something like this? Regards, Paul

2 3

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users