sr-users August 2006

sr-users@lists.kamailio.org

184 participants
360 discussions

by Ladislav Andel

some gateways might or might not support G.729.. You should see in SDP body of 200 OK message from gateway if you get one, what codecs the gateway supports. Ladislav ravi reddy wrote: > Hi > > I just tried to call a pstn number but call is not implemented even > thoough PSTN supports G729 > > So, here what is going wrong ? any suggestion please > > On 8/2/06, *Ladislav Andel* <ladia6(a)centrum.cz > <mailto:ladia6@centrum.cz>> wrote: > > Hi, > Just make sure. Do both phones support G.729? if the handytone > does the > other party also has to support G.729. Otherwise a phonecall can > not be > setup. > > Ladislav > > ravi reddy wrote: > > > Hi users, > > > > I have one doubt here please try to give some information > to me. > > > > I am using SER-0.9.6 with mediaproxy-1.7.2 > > > > and for testing purpose i am using grandstream adapter handytone-496 > > > > If i make changes in codecs in grandstream that it should only use > > g729 codec --the call is not implemented > > > > "here what i want to know is does grandstream support G729 codec or > > the adapter compress the voice in G729 format when we select g729"? > > > > if later is correct , so we must have to implement g729 codec in > SER > > or in mediaproxy so how we can do that? > > > > if i put G711 only,- the call is implementing or other wise the call > > cannot be made by the SER. > > > > so please give me some information regarding this :-( > > > > > > Thank You. > > > > Regards, > > Ravi. > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Serusers mailing list > >Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

18 years, 3 months

[Serusers] G729 codecs --help

by ravi reddy

Hi users, I have one doubt here please try to give some information to me. I am using SER-0.9.6 with mediaproxy-1.7.2 and for testing purpose i am using grandstream adapter handytone-496 If i make changes in codecs in grandstream that it should only use g729 codec --the call is not implemented "here what i want to know is does grandstream support G729 codec or the adapter compress the voice in G729 format when we select g729"? if later is correct , so we must have to implement g729 codec in SER or in mediaproxy so how we can do that? if i put G711 only,- the call is implementing or other wise the call cannot be made by the SER. so please give me some information regarding this :-( Thank You. Regards, Ravi.

18 years, 3 months

[Users] OpenSER + pa_mod

by Bernard Chan

Hello, I have OpenSER 1.0.1 (and upgraded to OpenSER 1.1.0 today). I am now experimenting with the use of pa_mod. The module is functioning pretty well, with one problem. If the SIP client is terminated and restarted right away, the SIP client's presence view is not updated. As I traced, I found that both times the SUBSCRIBE requests are received by OpenSER. From the packet sniffer I see some NOTIFYs are indeed sent to the client, but are rejected, I think probably because the Call-ID does not match the current one (it is from the earlier instance), and the client returned the "subscription does not exist" error status to the server. Then OpenSER seemed not to send further NOTIFYs, so there is no update to the presence information. If instead I restart the SIP client after subscription expires, the behaviour seems to be normal. Being pretty new to OpenSER, I would like to ask if this is normal? And anything I can do to solve the problem? Thank you. ------------------------------------------------------- [...] loadmodule "/usr/local/lib/openser/modules/pa.so" [...] #modparam("pa", "default_expires", 10) modparam("pa", "timer_interval", 3) modparam("pa", "watcherinfo_notify", 0) modparam("pa", "new_tuple_on_publish", 1) modparam("pa", "callback_update_db", 1) ------------------------------------------------------- Regards, Bernard Chan.

18 years, 3 months

[Users] Mysql & Radius

by Max Clark

Hi all, OpenSER supports (and can use) a mysql database to store various things including usernames/passwords, aliases, call forwarding, location, etc... I am curious why for accounting a Radius server has to be added into the mix. Given that most Radius implementations are just writing to mysql now anyways is there any way to bypass Radius and write the accounting data directly to mysql? Thanks in advance, Max -- Max Clark http://www.clarksys.com

18 years, 3 months

[Serusers] error: 500 Command 'ul_show_contact' not found

by trulyliu

hello: I download ser with presence support from http://ftp.iptel.org/pub/ser/presence/ser-0.10.99-dev35-pa-4.1_src.tar.gz use make make install and configure my mysql server follow tutorial serctl start Starting SER : started pid(3666) I use this command to add a user serctl add 222 222 222@localhost and I get following replay MySql password: error: 500 Command 'ul_show_contact' not found here is logs in /var/log/messages Aug 2 10:49:34 liuqc dhclient: bound to 172.19.1.81 -- renewal in 3564 seconds. Aug 2 11:01:27 liuqc ser: WARNING: fix_socket_list: could not rev. resolve 172.19.1.81 Aug 2 11:01:27 liuqc ser: WARNING: fix_socket_list: could not rev. resolve 172.19.1.81 Aug 2 11:01:27 liuqc ser: init_tcp: using epoll_lt as the io watch method (auto detected) Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3666]: AVPops - initializing Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3666]: Maxfwd module- initializing Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3666]: using 'none' rls-subscription authorization! Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3666]: using 'none' subscription authorization for watcher info! Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3666]: INFO: udp_init: SO_RCVBUF is initially 105472 Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3666]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3666]: INFO: udp_init: SO_RCVBUF is initially 105472 Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3666]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3690]: INFO: fifo_server.c:957: fifo process starting: 3690 Aug 2 11:01:27 liuqc /usr/local/sbin/ser[3690]: INFO: fifo_server.c:972: fifo server up at /tmp/ser_fifo... Aug 2 11:01:37 liuqc /usr/local/sbin/ser[3690]: ERROR: fifo_server.c:748: Command must begin with :: 222@localhost Aug 2 11:01:37 liuqc /usr/local/sbin/ser[3690]: ERROR: fifo_server.c:1061: No reply pipe (null) does someone can help me ? why does serctl add 222 222 222@localhost return error? -- trulyliu(a)gmail.com

18 years, 3 months

Re: [Users] How to add subscribers into OPENSER database(can not do it with openserctl)?

by Klaus Darilion

On Mon, July 31, 2006 22:14, Bill Zhang said: > Strange, I used to be able to use serctl to add subscribers, but with > openserctl, the choice are very limited, why? All those nice command line > options seem to be gone:-(. > openserctl was splitted into modules, and the functions to manipulate the DB are only installed if the DB packages are installed (mysql ...). Maybe there is a problem with this modularization. regards klaus > Best Regards, > Bill Zhang > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

18 years, 3 months

[Users] Accounting wrong BYEs

by Dimo

Hi all, I have a problem with accounting wrong BYEs: I have noticed that some UAs like the spa are sending BYE to cancel an INVITE instead of a CANCEL message. Now openser is interpreting this correctly, but the problem is with accounting which will account this BYE as call stop. Since there was never OK to the invite, there is no call start and just a call stop which makes my billing very unhappy. Is it possible to have something in the accounting module that will prevent such "fake" byes of being accounted? I am also thinking how to script it in my config, what i have thought of so far is when receiving a BYE to check if there is ongoing transaction and if there is, do not flag the BYE for accounting. This however will probably result in some real BYEs not being accounted too, because (correct me if wrong) the transaction will still exist in mem for some short time after the OK is received. Any help or suggestions will be greatly appreciated. Best, Dimo

18 years, 3 months

[Serusers] Force SER to send calls using TO header

by Mark Anthony C. Delfin

Hello Guys, Just like to request assistance in trying to figure out how can I route the call from SER as seen on TO header. Below is the snippet of the sip log: 0(20457) DEBUG: get_hdr_field: <To> [34]; uri=[sip:8810844@24.90.219.179 :8700] 0(20457) DEBUG: to body [<sip:8810844@24.90.219.179:8700> ] 0(20457) get_hdr_field: cseq <CSeq>: <1154241348> <INVITE> 0(20457) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(20457) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(20457) check_via_address(10.10.10.21, 10.10.10.21, 0) 0(20457) Sending: INVITE sip:8810844@10.10.10.86 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.86;branch=0 Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bK13666f91365343 From: <sip:2589@mandela>;tag=cba-0094-44cc5343 To: <sip:8810844@24.90.219.179:8700> Call-ID: 317e120dd2385173-0094-44cc5343-282c(a)10.10.10.21 CSeq: 1154241348 INVITE Contact: <sip:2589@10.10.10.21> Date: Sun, 30 Jul 2006 06:35:47 GMT User-Agent: BRSIP v2.0.0.11 Max-Forwards: 16 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Allow-Events: keep-alive, message-summary Supported: timer Session-Expires: 1800 Min-SE: 600 Expires: 300 Content-Type: application/sdp Content-Length: 220 v=0 o=BRSDP 177 177 IN IP4 10.10.10.21 s=BRSDP Session c=IN IP4 63.116.254.21 t=0 0 m=audio 15000 RTP/AVP 4 18 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 I need SER to send the call based on the TO HEADER URI seen on get_hdr_field. This value changes depending on what another sip proxy is sending to the SER. The t_relay is not working as i like it to behave. Any help is greatly appreciated. Thanks in advance.

18 years, 3 months

[Serusers] Call between two SER Server

by Asep Saifudin

Hi, I have two running SER Server but I stil got problem to make call from my first SER server to my secondary SER server. My first SER server info : 1. IP address = 192.168.100.100 2. sample subscriber number is 10999 My second SER Server info : 1. IP address = 192.168.200.200 2. sample subscriber number is 20099 My question is : How suberiber 10099 can call to subscriber 20099 ? What additional config should I put in each ser.cfg ? or should I use gatekeeper/softswitch ? Thanks

18 years, 3 months

[Serusers] Mediaproxy bug??

by Robert Zorop

Hi, i'm using mediaproxy 1.7.2 and ser. I found a problem when trying to use accounting features of mediaproxy. It simply does not account anything. If i just only pass one path throught media proxy, (INVITE SIDE), it does account media, and Kb data. Here is my config. Please people.. any clue about this!?. route[1] { t_on_reply("1"); if (method == "INVITE") { use_media_proxy(); } else if (is_method("BYE|CANCEL")) { end_media_session(); }; # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } onreply_route[1] { if (status=~"(180)|(183)|(2[0-9][0-9])") { use_media_proxy(); }; }

18 years, 3 months

Jump to page:

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 August 2006