sr-users May 2005

sr-users@lists.kamailio.org

230 participants
522 discussions

[Serusers] avpops - avp_pushto bug - maybe a do_action bug too.

by Marco Figus

Using avpops to change ruri/domani I found a strange behaviour: If you try to modify the ruri/domain of a previously received msg using avp_push to, you can't handle with the port. Suppose to receive a msg with ruri: name@domain:port and you wanna change it to name@newdomain:newport it seems to be impossible. if you do something like: avp_write("newdomain","s:_newdomain"); avp_pushto("$ruri/domain","s:_newdomain"); the result is: name@newdomain:port the do_action method just change the domain using the SET_HOST_T action type and leaves the final part of the string unchanged!! But I wanna change it too setting the newport value. A possible solution could be to recall the do_action wiht action SET_PORT_T with an empty value to reset the :port in the ruri field of the received domain, so I can set entire new domain using: avp_write("newdomain:newport","s:_newdomain"); avp_pushto("$ruri/domain","s:_newdomain"); having as a result: name@newdomain:newport and not name@newdomain:newport:port as happens now. Or (the best way I think) add a new value for the first parameters in method avp_pushto(destination, name) that allows to have the domainport as destination (invoking do_action whit SET_PORT_T action). I'm disposable to work on it, let me know. Marco.

19 years, 6 months

[Serusers], [Semsusers] sems pstn gw with ser, bad voice stream quality

by Grigory Fishilevich

hi guys, i have running ser with sems as pstngw (eicon diva pri card). both product are cvs snapshots, downloaded few days ago. calls from sip to pstn have bad quality, some packets be dropped. has anybody experiences with such environment? sipcalls between serusers have a very good voice quality. i think it caused by sems. bg, Grigory Fishilevich

19 years, 6 months

[Serusers] Password mysql changed

by Sandra Donaire

Hi!! I change the default password for the mysql database, an I change the SerWeb conf and runs ok, but wuehn I try restar the server, it's fail in db_connect. I can't find where its defined tha password for the module. Can somebody help me? Thanks. -- Sandra Donaire Arroyo Telefónica I+D División de "Tecnologías de acceso a redes IP"

19 years, 6 months

[Serusers] where is my /etc/profile?

by Bharat More

Hi all... I has installed ser 8.12.0, ser is running,and i exported my domain as novanet.co.in its gone successfully. but the next step is to export the domain in etc/profile file but where it is? Its supposed to be in the folder etc/profile but there is nthing in this folder?... Need urgent Help...Thanks for any hint in Adv. ________________________________________________________________________ Yahoo! India Matrimony: Find your life partner online Go to: http://yahoo.shaadi.com/india-matrimony

19 years, 6 months

[Serusers] UA sending 302 Moved Temp message

by Lucas Aimaretto

Hi all, I'm getting a rare message from an UA when receivin an INVITE. The UA replies with a SIP/2.0 302 Moved Temporarily. Any ideas of what does it mean ?? How to solve it ?? This UA can make calls anywhere, but can not receive a call at all. Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.12 - Release Date: 17/05/2005

19 years, 6 months

RE: [Serusers] SER b2bua agent (+more)?

by Wojciech Ziniewicz

so ok.. it,s my phones case since they are rather old and cheap :) > -----Original Message----- > From: Iqbal [mailto:iqbal@gigo.co.uk] > Sent: Thursday, May 19, 2005 8:32 PM > To: Wojciech Ziniewicz > Cc: 'serusers(a)lists.iptel.org' > Subject: Re: [Serusers] SER b2bua agent (+more)? > > > ur sipphones should be deciding the RTP port, u can usually > set them to > pick a random port from a range. > > iqbal > > Wojciech Ziniewicz wrote: > > >questions inline > > > > > > > >>-----Original Message----- > >>From: Iqbal [mailto:iqbal@gigo.co.uk] > >>Sent: Thursday, May 19, 2005 12:34 PM > >>To: Wojciech Ziniewicz > >>Cc: 'serusers(a)lists.iptel.org' > >>Subject: Re: [Serusers] SER b2bua agent (+more)? > >> > >> > >> > >>>2. I've got simple pstn model > >>> > >>>LAN ---> SER > >>> \ | > >>> \ | > >>> PSTN > >>> > >>>I get simple disconnetion and loose rtp stream when calling > >>> > >>> > >>out to PSTN when > >> > >> > >>>making more than 5 calls. > >>> > >>>I don't actually know whether it's SER or my E1 Gateway > >>> > >>> > >>problem. Logs from > >> > >> > >>>Linksys PAP2 tells me that it's the "RTP PORT DUPLICATE" . > He's right > >>>because the sniffer tells the same . I see that two phones > >>> > >>> > >>connect their RTP > >> > >> > >>>stream on port 3000. > >>> > >>>Any ideas ? > >>> > >>> > >>> > >>> > >>Um..nope, it should not be a problem with SER, because SER > >>really does > >>not care about the RTP stream, and 5 connections to ser is really > >>nothing, I think its in linksys, I dont use the device > >>myself, but does > >>it have a limit on how many natted clients can sit behind it > >>(assuming u > >>r using NAT) or how many DHCP connections (I had this problem > >>once), or > >>even how many RTP streams it can handle > >> > >> > > > >I mean - my Sipphones connected to syslog say : > >"RTP PORT DUP:30000" > >but my E1 Gateway (Multitech Multivoip3010) does not say > anything. It looks > >like allocation of RTP is not random - every phone wants to > get on port > >30000. Any ideas (again) ? > > > >I'm quite sure it's not SER's case, but who knows. > > > >. > > > > > > >

19 years, 6 months

Fwd: RES: [Serusers] Cisco 3640 X SER X PSTN

by Welesley Sibelson Dias

Many thks Jose Ricardo and "Ser Users" thks for your attetion more one time Jose Ricardo i have other problems now ahahahahahahah my call hang up and dont voice works now the caller rings e accept but cisco send me byes i send ethereal log to you into anex my IP is SER 200.184.153.54 Cisco 200.184.152.129 and Phone Voip 200.184.153.52 thks a lot this message below is in Portuguese: Muito Obrigado Jose Ricardo Falew pela forca desculpe meu pessimo ingles ---------- Mensagem reenviada ---------- Subject: RES: [Serusers] Cisco 3640 X SER X PSTN Date: Qui 19 Mai 2005 13:04 From: Jose Ricardo Maia Moraes <jrm(a)uol.com.br> To: serusers(a)lists.iptel.org Hi, I use R2 Digital on all of my AS5300 connected to PABX Nortel, Lucent and some carriers. Look the sample of my config: controller E1 0 framing NO-CRC4 clock source line primary ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani cas-custom 0 country brazil metering category 2 answer-signal group-b 1 voice-port 0:0 compand-type a-law cptone BR dial-peer voice 10 pots destination-pattern xxx direct-inward-dial port 0:0 Good luck. -----Mensagem original----- De: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] Em nome de Welesley Sibelson Dias Enviada em: quinta-feira, 19 de maio de 2005 06:33 Para: serusers(a)lists.iptel.org Assunto: [Serusers] Cisco 3640 X SER X PSTN Hi All I have try connect one E1 using SER and CISCO 3640 but i have some problems my PSTN Carrier Need Number for "A" but i don't send this , i need to complete calls in my carrier. Please have someone using cisco in R2 Digital send my one example of cfg . thks a lot

19 years, 6 months

RE: [Serusers] SER b2bua agent (+more)?

by Wojciech Ziniewicz

questions inline > -----Original Message----- > From: Iqbal [mailto:iqbal@gigo.co.uk] > Sent: Thursday, May 19, 2005 12:34 PM > To: Wojciech Ziniewicz > Cc: 'serusers(a)lists.iptel.org' > Subject: Re: [Serusers] SER b2bua agent (+more)? > > >2. I've got simple pstn model > > > >LAN ---> SER > > \ | > > \ | > > PSTN > > > >I get simple disconnetion and loose rtp stream when calling > out to PSTN when > >making more than 5 calls. > > > >I don't actually know whether it's SER or my E1 Gateway > problem. Logs from > >Linksys PAP2 tells me that it's the "RTP PORT DUPLICATE" . He's right > >because the sniffer tells the same . I see that two phones > connect their RTP > >stream on port 3000. > > > >Any ideas ? > > > > > > Um..nope, it should not be a problem with SER, because SER > really does > not care about the RTP stream, and 5 connections to ser is really > nothing, I think its in linksys, I dont use the device > myself, but does > it have a limit on how many natted clients can sit behind it > (assuming u > r using NAT) or how many DHCP connections (I had this problem > once), or > even how many RTP streams it can handle I mean - my Sipphones connected to syslog say : "RTP PORT DUP:30000" but my E1 Gateway (Multitech Multivoip3010) does not say anything. It looks like allocation of RTP is not random - every phone wants to get on port 30000. Any ideas (again) ? I'm quite sure it's not SER's case, but who knows.

19 years, 6 months

[Serusers] Re: [Serdev] [Fwd: Support for INFO method]

by Maxim Sobolev

Bayan Towfiq wrote: > Hi Maxim, > Do you know if anyone has RFC2833 support done for SEMS? Yes, we've done. It will be committed into the SEMS's cvs in few days from now. -Maxim > > Thanks, > Bayan > > Maxim Sobolev wrote: > >> Folks, >> >> Please approve attached patch, which adds support for INFO method in the >> parser and in the FIFO. This support is necessary for getting >> out-of-band DTMF support working in SEMS (yes, we do have this patch for >> SEMS already :). >> >> Thanks! >> >> -Maxim >> >> ------------------------------------------------------------------------ >> >> Index: stats.h >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/stats.h,v >> retrieving revision 1.10 >> diff -d -u -d -u -r1.10 stats.h >> --- stats.h 24 Aug 2004 08:45:10 -0000 1.10 >> +++ stats.h 16 May 2005 22:49:34 -0000 >> @@ -47,6 +47,7 @@ >> case METHOD_ACK: stats->dir##_requests_ack++; >> break; \ >> case METHOD_CANCEL: stats->dir##_requests_cnc++; >> break; \ >> case METHOD_BYE: stats->dir##_requests_bye++; >> break; \ >> + case METHOD_INFO: stats->dir##_requests_info++; break; \ >> case METHOD_OTHER: stats->dir##_requests_other++; >> break; \ >> default: LOG(L_ERR, "ERROR: unknown method in rq stats >> (%s)\n", \ >> #dir); \ >> Index: modules/tm/t_fifo.c >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/modules/tm/t_fifo.c,v >> retrieving revision 1.22 >> diff -d -u -d -u -r1.22 t_fifo.c >> --- modules/tm/t_fifo.c 25 Apr 2005 09:06:33 -0000 1.22 >> +++ modules/tm/t_fifo.c 16 May 2005 22:49:34 -0000 >> @@ -805,6 +805,7 @@ >> str_uri.len,str_uri.len ? str_uri.s : ""); >> >> if ( REQ_LINE(msg).method_value==METHOD_INVITE || + >> REQ_LINE(msg).method_value==METHOD_INFO || >> (twi->append && twi->append->add_body) ) { >> /* get body */ >> if( (body.s = get_body(msg)) == 0 ){ >> Index: parser/msg_parser.h >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/parser/msg_parser.h,v >> retrieving revision 1.56 >> diff -d -u -d -u -r1.56 msg_parser.h >> --- parser/msg_parser.h 28 Feb 2005 15:28:24 -0000 1.56 >> +++ parser/msg_parser.h 16 May 2005 22:49:34 -0000 >> @@ -67,7 +67,7 @@ >> >> /* number methods as power of two to allow bitmap matching */ >> enum request_method { METHOD_UNDEF=0, METHOD_INVITE=1, >> METHOD_CANCEL=2, METHOD_ACK=4, - METHOD_BYE=8, METHOD_OTHER=16 }; >> + METHOD_BYE=8, METHOD_INFO=16, METHOD_OTHER=32 }; >> >> #define FL_FORCE_RPORT 1 /* force rport */ >> #define FL_FORCE_ACTIVE 2 /* force active SDP */ >> Index: parser/parse_fline.c >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/parser/parse_fline.c,v >> retrieving revision 1.8 >> diff -d -u -d -u -r1.8 parse_fline.c >> --- parser/parse_fline.c 24 Aug 2004 09:01:25 -0000 1.8 >> +++ parser/parse_fline.c 16 May 2005 22:49:34 -0000 >> @@ -1192,6 +1192,7 @@ >> else IFISMETHOD( CANCEL, 'C') >> else IFISMETHOD( ACK, 'A' ) >> else IFISMETHOD( BYE, 'B' ) + else IFISMETHOD( INFO, 'I' ) >> /* if you want to add another method XXX, include METHOD_XXX in >> H-file (this is the value which you will take later in >> processing and define XXX_LEN as length of method name; >> Index: parser/parse_fline.h >> =================================================================== >> RCS file: /cvsroot/ser/sip_router/parser/parse_fline.h,v >> retrieving revision 1.6 >> diff -d -u -d -u -r1.6 parse_fline.h >> --- parser/parse_fline.h 24 Aug 2004 09:01:25 -0000 1.6 >> +++ parser/parse_fline.h 16 May 2005 22:49:34 -0000 >> @@ -51,6 +51,7 @@ >> #define CANCEL_LEN 6 >> #define ACK_LEN 3 >> #define BYE_LEN 3 >> +#define INFO_LEN 4 >> >> >> struct msg_start { >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serdev mailing list >> serdev(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serdev >> >> > > >

19 years, 6 months

[Serusers] ser childs vs radius's RPID

by Mihaly Zachar

Hi, I'm trying to play with SER and auth_radius, with "Sip-RPId" attribute, because I would like to change something in every registered UA's number. When SER runs alone with "fork=no" everything is correct... If run ser with "fork=yes" and "children=4", in the sip packet's Remote-Party-ID header field contains the last registered user's Sip-RPId. Why??? That is why, the showed caller's number seems to be a random number... What can I do??? Thanks, Misi

19 years, 6 months

← Newer
1
...
18
19
20
21
22
23
24
...
53
Older →

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 May 2005