sr-users January 2005

sr-users@lists.kamailio.org

199 participants
399 discussions

by Darren Westlake

Hi all, Could you please help me with specifying a PC for use as an SER server? We currently run a maximum of about 100-150 simultaneous calls but potentially this could go to up 500 within about a year so it would need to handle that many. What processor, RAM, hard disk etc. would be required to run such a service? thanks, Darren

19 years, 10 months

Re: RES: [Serusers] "Can't connect to RTP proxy"

by Samuel Osorio Calvo

Please, take a look at older threads regarding the comparison between them. I think there are some good posts about differences. Sorry but I haven't tried them so I can not help you, Samuel Unclassified. >>> "Humberto Aicardi" <humberto(a)aicardi.com.br> 01/09/05 01:49PM >>> Chris, I have just configured RTPProxy and currently is working, but after reading about Media-proxy I too would like to have a comparison between RTPProxy and Media-Proxy. If someone could be kind enough to explain the difference I will appreciate it. Thanks, Humberto -----Mensagem original----- De: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] Em nome de Chris Enviada em: Saturday, January 08, 2005 1:53 PM Para: serusers(a)lists.iptel.org Assunto: [Serusers] "Can't connect to RTP proxy" Hi all, I have 0.9.0 running with customized version of ser.cfg posted by Paul, I think. It works to the extent that the .cfg has so far been developed. However on "serctl start" and with traffic I get multiple logs which all seem to stem from: "send_rtpp_command: can't connect to RTP proxy". Can any one point me in the right direction to "resolve" this? On a connected note (I think), The cfg file is based, it seems, on nat-helper operation. I understand that nat-helper and media-proxy are mutually exclusive. but that media-proxy seems to be more powerful and well suitable for smaller systems that do not need to scale greatly. Is there any document which explains what are the fundamental differences in capabilities between the two, and ser.cfg deltas required to go from nat-helper-->media-proxy. I have found text which references one or the other but not in a comparative sense. thanks Chris --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.809 / Virus Database: 551 - Release Date: 09/12/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.809 / Virus Database: 551 - Release Date: 09/12/2004

19 years, 10 months

[Serusers] RE: Configuring SER FOR MULTICAST ADDRESS

by Samuel Osorio Calvo

Hi all, >From version 0.9 and some 0.8-xx-dev versions, multicast listening is working by just adding the address in the config file as if it was another address listen = 224.0.1.75 There are a couple of parameters to configure muilticast (see NEWS file): <- new config variables: < mcast_loopback = <yes/no> - loopback sent multicast datagram, default no. < mcast_ttl = number - set multicast ttl, default OS specific (usually 1). So, if you just want to work in your LAN without loopback, just add the multicast address in the config file and SER is going ot listen to it. Remember to have fork=yes in order to enable SER listening to more than one address, just in case you want SER to listen to the multicast and to the "public IP". Sending to multicast should be as sending to a "normal" UDP address, so from SER's config file I guess you can try to send to a multicast address. Cheers, Sam. Unclassified. >>> <innovation.interops(a)wipro.com> 01/10/05 07:44AM >>> Dear all, Any suggestions on configuring SER for multicast addressing. Thanks karthikeyan.k ________________________________ From: Innovation Interops Sent: Thu 1/6/2005 5:14 PM To: serusers(a)lists.iptel.org Subject: Configuring SER FOR MULTICAST ADDRESS Dear all, How to configure SER to listen on Multicast address? Regds karthikeyan.k Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

19 years, 10 months

[Serusers] RE: Configuring SER FOR MULTICAST ADDRESS

by innovation.interops＠wipro.com

Dear all, Any suggestions on configuring SER for multicast addressing. Thanks karthikeyan.k ________________________________ From: Innovation Interops Sent: Thu 1/6/2005 5:14 PM To: serusers(a)lists.iptel.org Subject: Configuring SER FOR MULTICAST ADDRESS Dear all, How to configure SER to listen on Multicast address? Regds karthikeyan.k Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

19 years, 10 months

[Serusers] problem regarding compilation of exluded modules of ser

by pranav_mnit

Dear sir, As the radius accounting modules are not include in the tar ball. so to include them i tried to install the ser from its source and edit the make file by ommitting radius related modules from the exclude section .but when make install it gives the error "radius.O not compiled" what should i do? pranav MNIT Indiatimes Email now powered by APIC Advantage. Help! HelpClick on the image to chat with me

19 years, 10 months

[Serusers] 408 problem - accounting

by Vitor Brasileiro

Hi list, During a call between X-Pro and Sipura, when Sipura disconnects first, X-Pro doesn't receive any message signaling that call ended and keeps the call on connected state. The bigger problem is about the accounting issue. On my accounting table, I have two enters for this call (two BYE messages, one w/ 200 OK and the other w/ 408). I'm thinking about remove all 408 records from accounting table, and put a crontab enter to do this everyday. Nbody have a tip? Thanks, Vitor Brasileiro.

19 years, 10 months

[Serusers] A-Z route

by mohammad

Hi ALL; A-Z route over H323 is available. 1) several backup routes per one destination 2) availability of test 3) technical support Plz contact me offlist for rate and further informations MSN:hezare3@hotmail.com Mohammad

19 years, 10 months

[Serusers] ser-0.9.0 and rpid

by Helge Waastad

Hello, is there a bug in append_rpid_hf in ser-0.9.0 ? I get ascii characters in my rpid even if MySQL version 4.1.8 I've added the INVITE from ser to the gateway (cisco) [removed some private information...] The Remote-Party address in subscriber table is 67512387 and comes out like 'edc allo'........... best regards, hw INVITE sip:<name>@<cisco>:5060 SIP/2.0 Record-Route: <sip:<ser>;ftag=1952781227;lr> Via: SIP/2.0/UDP <ser>;branch=z9hG4bKc863.7c873544.0 Via: SIP/2.0/UDP 10.0.0.4:5060;received=<client IP>;rport=64532;branch=z9hG4bK5497F64D2C5444D0BDF03D0F9B7E38E3 From: <name> <sip:<name>@<relam>;tag=1952781227 To: <sip:<destination>@<realm>> Contact: <sip:<name>@<client IP>:64532> Call-ID: 1E1BCEDE-DE4C-4B5D-A0F7-D6F3048427B1(a)10.0.0.4 CSeq: 59604 INVITE Max-Forwards: 16 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 298 Remote-Party-ID: <sip:edc allo@<realm>;party=calling;screen=no;privacy=off <mailto:allo@<realm>;party=calling;screen=no;privacy=off> P-RTP-Proxy: YES P-NATed-Calee: YES P-hint: GATEWAY v=0 o=67830064 202717921 202717968 IN IP4 10.0.0.4 s=X-Lite c=IN IP4 <ser> t=0 0 m=audio 35706 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15

19 years, 10 months

[Serusers] DTMF problems

by Humberto Aicardi

Hi, I'm a newbie on SER so please be patient. I have been using * for quite a while and now decided to use SER as a front-end for my SIP users. I have everything setup including *, SER, RPTProxy, STUN and everything works except for DTMF tones, for example, when I access * voicemail through SER the tones do not get correctly recognized by *. Asterisk is configured to use RFC2833 as a transport for DTMF tones. Can anyone please help me solve this issue? Thanks in advance, Humberto

19 years, 10 months

RES: [Serusers] DTMF problems

by Humberto Aicardi

Hello, My current configuration is: RedHat 9.0 Asterisk HEAD version SER latest CVS version www.vovida.org STUN server eyeBeam XTen softphone When I dial from the softphone, which is connected to the SER server, to the * box (voicemail) it will request to input the mailbox number and password. When I try to do this the * box get some but not all digits I have pressed, so basically, it looses some digits. When I'm connected directly it works ok. Here is the context in * for the SER server: [300] type=peer host=xxx.xxx.xxx.xxx nat=no disallow=all allow=g729 allow=gsm allow=ulaw dtmfmode=rfc2833 qualify=yes context =from-ser ; Default for incoming calls insecure=very If you need any further any information please let me know. Thanks, Humberto -----Mensagem original----- De: Yair Hakak [mailto:yhakak@gmail.com] Enviada em: Saturday, January 08, 2005 1:37 PM Para: Humberto Aicardi Assunto: Re: [Serusers] DTMF problems Hello, what endpoints are you using? -yair p.s. not that it should have any relevance to the question, but what stun server are you using, and are you running it on the same machine as asterisk and/or SER? I am also running an asterisk/SER combination and I want to run my own STUN server as well. On Sat, 8 Jan 2005 10:23:17 -0200, Humberto Aicardi <humberto(a)aicardi.com.br> wrote: > > > Hi, > > > > I'm a newbie on SER so please be patient. I have been using * > for quite a while and now decided to use SER as a front-end for my SIP > users. I have everything setup including *, SER, RPTProxy, STUN and > everything works except for DTMF tones, for example, when I access * > voicemail through SER the tones do not get correctly recognized by *. > Asterisk is configured to use RFC2833 as a transport for DTMF tones. > > > > Can anyone please help me solve this issue? > > > > Thanks in advance, > > Humberto > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

19 years, 10 months

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sr-users January 2005