sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

Re: [Serusers] Re: Does SER works on TWO network interfaces with public and private IP addresses?

by Charles Wang

Mike: Before I test these two interfaces(Private and Public), I have only a public interface on my SER proxy. My NATed clients are XLite or any SIP IP phones or SIP gateways. They work fine with SER and each other. When I try to make these clients register from private interface, the problem happens. So I don't think it is the problem of NAT functions at XLite or Gateways. On 9/18/05, Mike Williams <mwilliams(a)etc1.net> wrote: > I was having problems with NAT myself, and found this. I thought it sounded like your problem. > > http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.h… > > ---Mike > > 5.1. SIP with NAT or Firewalls ( Back to Tutorials Page ) > > 1.1. Problem Description: > > > Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on itself they only carry call signaling (call setup, teardown,… and use RTP to carry the audio samples. > > The signaling usually uses fixed and standardized ports, but the RTP uses random ports to exchange both call legs (incoming and outgoing audio). > > Most firewalls/NATs are unable to link the signalling protocol packets with the audio packets and are in some cases unable to tell where to send the audio to. > > When making a call, everything will seem to go normal, caller id will get passed, ringing will start, you can pick up and hangup the call, but no audio in one or both directions. > > > > -----Original Message----- > From: Charles Wang [mailto:lazy.charles@gmail.com] > Sent: Sat 9/17/2005 2:40 AM > To: Mike Williams; serusers(a)iptel.org > Subject: Re: [Serusers] Re: Does SER works on TWO network interfaces with public and private IP addresses? > > Hi, Mike: > the default gateway is 192.168.11.254 not 192.168.11.1. > So i dont think the flow is 192.168.11.2 to 192.168.11.1. > > On 9/17/05, Mike Williams <mwilliams(a)etc1.net> wrote: > > I believe your problem is simple. With the SIP protocol, you are sending > > the streams like this: > > > > 192.168.11.2 -> 192.162.11.1 -> 221.21.X.X > > > > After you answer, the clients negotiate for RTP traffic, and try to send > > data directly from 192.168.11.2 to 221.21.X.X, not using the SIP server. > > You are probably having problems actually routing the data (trying > > pinging the 221.21.X.X box from your 192.168.11.2 client) or you're > > having NAT issues. Is far as I know, you must have a direct route from > > the caller to the callee to pass RTP streams; you can't proxy them > > through the SIP server. > > > > Good luck, and let me know if you have any more questions. > > > > Mike Williams (mwilliams(a)etc1.net) > > > > Charles Wang wrote: > > > > >On 9/13/05, Charles Wang <lazy.charles(a)gmail.com> wrote: > > > > > > > > >>Hi, ALL: > > >> > > >>I use ser + mediaproxy + PSTN support, and my ser with two interfaces. > > >>One is public IP address such as 211.21.xxx.xxx. > > >>Another one is private IP address such as 192.168.11.1. > > >> > > >>And I use XLite (192.168.11.2) register to SER's private interface(via > > >>HUB only). > > >>It can register sucessfully. > > >> > > >>But when I make a call to PSTN with this XLite, the callee rings and I > > >>answer it. > > >>I can not hear any sounds from each side. > > >> > > >>I try to register another XLite(192.168.11.3), and make a call to > > >>another private XLite(192.168.11.2). I can hear rings but it is still > > >>no any sounds from each side. > > >> > > >>Can anyone tell me what it happens? > > >> > > >>Best Regard > > >>Charles > > >> > > >> > > >> > > > > > > > > > > > > > > > > > -- > > Best Regards > Charles > > > -- Best Regards Charles -- Best Regards Charles

19 years, 2 months

[Serusers] SER Load balancing

by Marcello Lupo

Hi to all, did you have suggestions how to accomplish to load balancing load of SER between 2 machine running both SER and DB to check routing entries?? I was thinking to use LVM but i don't know how to let the same call messages to go to the same machine. There should be something line an LVM module able to check the CallID field and keep track of it. May be some other ways?? Using DNS records or other? Another question is if there is a searchable archive of this Mailing list. Thanks, Bye, Marcello

19 years, 2 months

Re: [Serusers] Re: Does SER works on TWO network interfaces with public and private IP addresses?

by Charles Wang

Hi, Mike: the default gateway is 192.168.11.254 not 192.168.11.1. So i dont think the flow is 192.168.11.2 to 192.168.11.1. On 9/17/05, Mike Williams <mwilliams(a)etc1.net> wrote: > I believe your problem is simple. With the SIP protocol, you are sending > the streams like this: > > 192.168.11.2 -> 192.162.11.1 -> 221.21.X.X > > After you answer, the clients negotiate for RTP traffic, and try to send > data directly from 192.168.11.2 to 221.21.X.X, not using the SIP server. > You are probably having problems actually routing the data (trying > pinging the 221.21.X.X box from your 192.168.11.2 client) or you're > having NAT issues. Is far as I know, you must have a direct route from > the caller to the callee to pass RTP streams; you can't proxy them > through the SIP server. > > Good luck, and let me know if you have any more questions. > > Mike Williams (mwilliams(a)etc1.net) > > Charles Wang wrote: > > >On 9/13/05, Charles Wang <lazy.charles(a)gmail.com> wrote: > > > > > >>Hi, ALL: > >> > >>I use ser + mediaproxy + PSTN support, and my ser with two interfaces. > >>One is public IP address such as 211.21.xxx.xxx. > >>Another one is private IP address such as 192.168.11.1. > >> > >>And I use XLite (192.168.11.2) register to SER's private interface(via > >>HUB only). > >>It can register sucessfully. > >> > >>But when I make a call to PSTN with this XLite, the callee rings and I > >>answer it. > >>I can not hear any sounds from each side. > >> > >>I try to register another XLite(192.168.11.3), and make a call to > >>another private XLite(192.168.11.2). I can hear rings but it is still > >>no any sounds from each side. > >> > >>Can anyone tell me what it happens? > >> > >>Best Regard > >>Charles > >> > >> > >> > > > > > > > > > -- Best Regards Charles

19 years, 2 months

[Serusers] SER proxy and SER+SEMS media server

by Teemu Harju

Hi, I'm implementing following kind of system. I have two separate machines where I've installed SER on. The other machine has also SEMS installed. Like you probably already figured out other is used as a plain proxy/registrar and the other as a sort of UAS serving voicemail, conference etc with the help of SEMS. Now my question is, what is the best way to forward messages from proxy/registrar to the media server? I've done the forwarding according to the examples I've found and it uses rewritehostport() to forward messages to voicemail. Voicemail needs to get the email address from mysql database located in proxy. This is no problem connecting to the database, but isn't it so that when I've rewritten the Request-URI that the voicemail cannot find the email from database beacause it's rewritten and has the media servers host part in the URI. How have you guys implemented this? I would also need other SEMS plug-ins like conference, so I would also need to differentiate those requests also. Could I rewerite the Request-URI like voicemail@mediaserver? Can the voicemail still find out who was I actually trying to call to since the URI is rewritten? Is there difference in implementing the media server with asterisk? Wouldn't I need then two different databases? One in asterisk and one in SER. Thanks... -- Teemu Harju http://www.teemuharju.net

19 years, 2 months

[Serusers] Building SEMS on Mac OS..share your experience..

by Amos Nungu

Hi, I am Installing SER/SEMS on Mac. SER is running fine Building SEMS give some error but it does build to the end, and I can run "make Install" BUT can't start SEMS.. error below ser:~ root# sems bash: /usr/local/sbin/sems: cannot execute binary file and compilation error is.. #make all cd gsm-1.0-pl10; make ./lib/libgsm.a make[5]: `lib/libgsm.a' is up to date. cc -o gsm.so gsm.o gsm-1.0-pl10/lib/libgsm.a -shared powerpc-apple-darwin8-gcc-4.0.0: unrecognized option `-shared' /usr/bin/ld: gsm.o bad magic number (not a Mach-O file) /usr/bin/ld: warning empty table of contents: gsm-1.0-pl10/lib/libgsm.a (can't load from it) collect2: ld returned 1 exit status make[4]: *** [gsm.so] Error 1 make[3]: [all] Error 2 (ignored) make[4]: Nothing to be done for `deps'. make[5]: Nothing to be done for `all'. make[4]: Nothing to be done for `deps'. make[4]: `../../lib/apps/isdngw.so' is up to date. make[4]: Nothing to be done for `deps'. g++ -o ivr.so Ivr.o IvrDtmfDetector.o IvrEvents.o IvrMediaHandler.o IvrPython.o -lm -ldl -lsocket -lnsl -lpthread -shared -ldl -lpthread -lutil -lm -Xlinker --export-dynamic -L/usr/lib/python2.3/config -lpython2.3 /usr/lib/python2.3/lib-dynload/time.so powerpc-apple-darwin8-g++-4.0.0: unrecognized option `-shared' /usr/bin/ld: unknown flag: --export-dynamic collect2: ld returned 1 exit status make[4]: *** [ivr.so] Error 1 make[3]: [all] Error 2 (ignored) make[4]: Nothing to be done for `deps'. cc -fPIC -I ../../amci -Wall -I./lame-3.96/include -c mp3.c -o mp3.o mp3.c:37:18: error: lame.h: No such file or directory mp3.c: In function 'MP3_create': mp3.c:88: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:88: error: (Each undeclared identifier is reported only once mp3.c:88: error: for each function it appears in.) mp3.c:88: error: 'gfp' undeclared (first use in this function) mp3.c:93: warning: implicit declaration of function 'lame_init' mp3.c:98: warning: implicit declaration of function 'lame_set_errorf' mp3.c:99: warning: implicit declaration of function 'lame_set_debugf' mp3.c:100: warning: implicit declaration of function 'lame_set_msgf' mp3.c:102: warning: implicit declaration of function 'lame_set_num_channels' mp3.c:103: warning: implicit declaration of function 'lame_set_in_samplerate' mp3.c:104: warning: implicit declaration of function 'lame_set_brate' mp3.c:105: warning: implicit declaration of function 'lame_set_mode' mp3.c:106: warning: implicit declaration of function 'lame_set_quality' mp3.c:108: warning: implicit declaration of function 'id3tag_init' mp3.c:109: warning: implicit declaration of function 'id3tag_set_title' mp3.c:110: warning: implicit declaration of function 'lame_init_params' mp3.c:118: warning: control reaches end of non-void function mp3.c: In function 'MP3_destroy': mp3.c:122: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:122: error: parse error before ')' token mp3.c: In function 'Pcm16_2_MP3': mp3.c:138: warning: implicit declaration of function 'lame_encode_buffer' mp3.c:138: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:138: error: parse error before ')' token mp3.c:143: warning: control reaches end of non-void function mp3.c: In function 'MP3_close': mp3.c:224: warning: implicit declaration of function 'lame_encode_flush' mp3.c:224: error: 'lame_global_flags' undeclared (first use in this function) mp3.c:224: error: parse error before ')' token mp3.c:228: warning: implicit declaration of function 'lame_mp3_tags_fid' mp3.c:228: error: parse error before ')' token mp3.c:221: warning: unused variable 'mp3buffer' mp3.c:229: warning: control reaches end of non-void function make[4]: *** [mp3.o] Error 1 make[3]: [all] Error 2 (ignored) . . . make[4]: Nothing to be done for `deps'. make[4]: `../../lib/apps/voicemail.so' is up to date. make[4]: Nothing to be done for `deps'. make[4]: `../../lib/audio/wav.so' is up to date. Regards

19 years, 2 months

[Serusers] Authentication

by Sebastian Kühner

Hello, I want to authenticate the users that wants to make a PSTN call on every INVITE. I have one user that can't make PSTN calls, because the message "407 Proxy Authentication Required" goes back to the wrong port (the IP is correct). I checked it with ngrep. Has anyone of you an idea how could I fix that problem? Thanks! Sebastian

19 years, 2 months

[Serusers] Re: 400 Bad Request

by Alan Chiu

This INVITE came from a client device but not sure why SER 0.8.14 (i386/linux) would reject it with 400 Bad Request and there's no reason given. Any insights would help. ***************************************************** Session Initiation Protocol Request line: INVITE sip:4519@10.100.0.111:5060;transport=udp SIP/2.0 Message Header To: <sip:4519@10.100.0.222> From: "PHONE TEST"<sip:1111@172.31.1.1>;tag=5b4ae231 Via: SIP/2.0/UDP 172.31.1.20:5060;branch=z9hG4bK-d87543-ffe1a8337da8f24e89ef-1-cHA5ZGU5Mz E0ZWU3YTE3YjUzNjgxYg..-d87543- Call-ID: 4132a9661e815dd517bec43512d032db CSeq: 267297881 INVITE Contact: <sip:4AmIBQqlz52H8OKUlL_xUG_l--zuAAngOdmZbN4wa1rlMGPDB0hIkU7H3mat-JyuSGS Vy4.@172.31.1.20:5060> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Date: Thu, 15 Sep 2005 16:34:32 GMT User-Agent: Server Authorization: Digest username="testuser(a)domains.com",realm="companyX.com",algorithm="MD5",uri ="sip:4519@10.100.0.222",no nce="432993119128c2b1a32ca43b9436e2fe0bca62f0",response="e42bc4a17da0282 552265396c0eb461e",opaque="" Content-Length: 243 Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 27663 1963582334 IN IP4 172.31.1.20 Owner Username: root Session ID: 27663 Session Version: 1963582334 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 172.31.1.20 Session Name (s): session Connection Information (c): IN IP4 172.31.1.20 Connection Network Type: IN Connection Address Type: IP4 Connection Address: 172.31.1.20 Time Description, active time (t): 0 0 Session Start Time: 0 Session Start Time: 0 Media Description, name and address (m): audio 10492 RTP/AVP 0 8 101 Media Type: audio Media Port: 10492 Media Proto: RTP/AVP Media Format: 0 Media Format: 8 Media Format: 101 Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Attribute Value: 101 0-16 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 101 telephone-event/8000 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Session Initiation Protocol Status line: SIP/2.0 400 Bad Request Message Header To: <sip:4519@10.100.0.222>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6a43 From: "PHONE TEST"<sip:1111@172.31.1.1>;tag=5b4ae231 Via: SIP/2.0/UDP 172.31.1.20:5060;branch=z9hG4bK-d87543-ffe1a8337da8f24e89ef-1-cHA5ZGU5Mz E0ZWU3YTE3YjUzNjgxYg..-d87543- Call-ID: 4132a9661e815dd517bec43512d032db CSeq: 267297881 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 10.100.0.111:5060 "Noisy feedback tells: pid=6024 req_src_ip=172.31.1.20 req_src_port=5060 in_uri=sip:4519@10. 100.0.111:5060;transport=udp out_uri=sip:4519@10.100.0.111:5060;transport=udp via_cnt==1" *****************************************************

19 years, 2 months

[Serusers] Does SER works on TWO network interfaces with public and private IP addresses?

by Charles Wang

Hi, ALL: I use ser + mediaproxy + PSTN support, and my ser with two interfaces. One is public IP address such as 211.21.xxx.xxx. Another one is private IP address such as 192.168.11.1. And I use XLite (192.168.11.2) register to SER's private interface(via HUB only). It can register sucessfully. But when I make a call to PSTN with this XLite, the callee rings and I answer it. I can not hear any sounds from each side. I try to register another XLite(192.168.11.3), and make a call to another private XLite(192.168.11.2). I can hear rings but it is still no any sounds from each side. Can anyone tell me what it happens? Best Regard Charles

19 years, 2 months

[Serusers] SIP VXML

by Abdul Lateef

Hi all, Can any suggest me to developing IVR using vxml for SIP Gatekeeper, Is there any SIP Vxml open source available? Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 2 months

FW: [Serusers] Call forwarding Question (following issue 5)

by Aisling

Sorry I never attached the messages... -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 16 September 2005 11:31 To: 'Greger V. Teigre'; 'serusers(a)lists.iptel.org' Subject: RE: [Serusers] Call forwarding Question (following issue 5) I have attached a complete ngrep of the messages. There were no errors in the var/log/messages file. Originally 3500 and 5000 were on the same machine so I changed the scenario so that 3500 (Windows messenger) was calling 2092 (BT100 - which is busy) and should be forwarded to 2009 (KPhone, same pc as SER). However it still doesn't forward the call.....Its the last two messages confuse me...I don't understand why a 407 Proxy Authentication required would be sent back to original caller.... My rule in the usr_preferences table of the mysql database is: Username (2092) Attribute (fwdbusy) Value (sip:2009@serveraddress) Any ideas? Many thanks, Aisling. -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: 15 September 2005 18:43 To: Aisling; serusers(a)lists.iptel.org Subject: Re: [Serusers] Call forwarding Question (following issue 5) Yes, we have verified that the configs lack the append_branch you mention. The fix has been done (https://siprouter.onsip.org/trac/changeset/15) and new configs are available through the regular downloads at onsip.org. To your other problem: No clue, Aisling. Sounds very strange indeed. The config should only trigger proxy authentication on an INVITE (not REGISTER). Maybe what you are experiencing is Messenger sending a new INVITE. (It should not, as it should receive a 100 Trying from ser). Only a *complete* ngrep trace will help in understanding what's happening (and any error messages in /var/log/messages with the timestamp for matching) g-) Aisling wrote: > Hi, > > I am testing the call forwarding features demonstrated in Issue 5 of > the > onsip getting started document. I found that blind call transfer > worked > perfectly but fwdbusy & fwdnoanswer gave me errors: > > ERROR: t_forward_nonack: non branched for forwarding > ERROR: w_t_relay(failure mode): forwarding failed > ERROR: sl_reply_error used: I'm terribly sorry, server error occurred. > > On the onsip site I noted that someone else had this problem and it > was > solved by putting append_branch in the fwdbusy and fwdnoanswer > sections in > the failure route. > > Thankfully that fixed those errors. However when I went to test > fwdbusy > again, it doesn't give any errors but still doesn't work. The call > scenario > was as follows: > > Windows Messenger client 3500 ring BT100 2092. 2092 is off the hook > (thereby > sending a 486 busy message) and the call should be forwarded to xlite > client > 5000. > > i.e. 3500 -> 2092(busy) -> 5000 > > The message sequence showed that everything was correct up to 2092 > sending > the 486 busy to SER and then SER sending an ACK back to 2092. But > then SER > sends a 407 proxy authentication required to 3500 which replies with > an > ACK....and that's it... > > Can someone explain why SER would send a 407 Proxy authentication to > the > original caller?...I thought this should only be in response to a > register > message?.... > > Any help appreciated, > Thanks, > Aisling. > > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only for the person to whom it is addressed. Its contents may be > protected by legal and/or professional privilege. Should it be > received by you in error please contact the sender at the above > quoted email address. Any unauthorised form of reproduction of this > message is strictly prohibited. The Institute does not guarantee the > security of any information electronically transmitted and is not > liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the > sender nor for any delay in its receipt. > > > > >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

19 years, 2 months

← Newer
1
...
22
23
24
25
26
27
28
...
51
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 September 2005