sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

Re: [Serusers] Contacts in 200 OK
by CURRO_DOMINGUEZ 20 Jan '04

20 Jan '04

Hello, Bernie and list I think my point of view is wrong. The behaviour of SER is absolutely OK about contacts. My last explanation in previous message has been terribly bad. Our problem is about phonebook. We want to allow users to register from PC, mobile phones, smart phones, iPaq,... in a multimedia environment. They should be able to make voice calls and IM. But we have to find out the way to avoid users write its phonebook every time they change from PC to mobil phone, for example. (And all of this over IPv6, but it will be later) So, I think the question is: Can SER manage phonebooks via SUSBRIBE/NOTIFY or any other method? Should the client ask to SER for phonebook table or it's a SER initiative? And if SER is not designed for this, does anyone know a software, module, whatever which can handle this? Or do you suggest any other way to attack this problem? Thank you very much for your time Curro ----- Mensaje Original ----- De: Bernie Hoeneisen <bhoeneis(a)switch.ch> Fecha: Lunes, Enero 19, 2004 3:35 pm Asunto: Re: [Serusers] Contacts in 200 OK > Hi Curro! > > On Mon, 19 Jan 2004, CURRO_DOMINGUEZ wrote: > > > Hello, > > > > Thanks to the list for your help. I can say that SER is running > OK and > > we are doing a lot of very interesting tests. > > > > I have a question about contacts. As Jiri said to me, SER sends all > > contacts in replies to REGISTER. This reply is a 200 OK and has > so many > > Contact fields as the user has in database. > > This is the correct behavior as standardized in RFC 3261. > > > The problem is that SIP clients like Windows Messenger doesn't care > > about this Contact Fields. I know this isn't very important if > you're> thinking of SIP Phones, but we are very interested in > presence. We work > > with SIP applications which need to know about user's contacts from > > database. > > I don't see the link between "Windows Messenger" not caring about > otherContacts and a Presence feature in the database (I guess you > want SER to > support this?). > > > So, I would like to know if someone has tested a SIP software that > > works on this issue. We have looked at RFC 3261 about this, but > there> isn't any comment. > > What do you mean by "this issue"? Could you explain this a bit more. > > > T: Bernie > > >

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[Serusers] Forwarding calls from SER to H323 gw or another SIP
by Fábio Silvestri 20 Jan '04

20 Jan '04

Hi, I'm in trouble with my sip proxy, I have to forward some calls to a h323 gateway or to a vocal sip proxy? So does anybody knows how can I do this? -- |o |o |o Fabio Silvestri |o fabio(a)informatec.com.br |o ICQ: 1667351 |o

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[Serusers] Serweb Error : 500 fifo_uac: no mem for hf block
by CURRO_DOMINGUEZ 20 Jan '04

20 Jan '04

Hello I continue with my tests on SER. Things are going OK, so really thanks. I have this error in serweb. I connect as an user and go to phonebook. When I click on sip address of my contacts, I get this message on a newwindow: 500 fifo_uac: no mem for hf block. ¿Is this normal? And I have this problem too: I can't send IM from serweb. When I click on send, I get 478 fifo_uac: Unresolveable destination (478/FIFO/UAC) However, the SIP clients are running OK. Again, thank you very much Curro

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Re: [Serusers] NATHelper
by Kevin Chu 20 Jan '04

20 Jan '04

Hi Maxim, I am highly interested for the ser/rtpproxy on NAT box solution. May I know when that solution will be available? Thanks, Kevin ---------------------------------------------------------------------------- --- Date: Tue, 20 Jan 2004 12:58:02 +0200 From: Maxim Sobolev <sobomax(a)portaone.com> Subject: Re: [Serusers] NATHelper To: Craig Graham <craig(a)twolips-translations.co.uk> Cc: serusers(a)lists.iptel.org Message-ID: <400D09BA.70608(a)portaone.com> Content-Type: text/plain; charset=us-ascii; format=flowed Craig Graham wrote: > I have a very small setup; at the office there's a few machines behind an > Intertex IX66 NAT box, that does a fairly good job of SIP as far as we can > tell from the few connections we've had to other sites. > > At home, I have a few machines behind a Linux box, doing NAT. All user > machines are running MS Messenger. > > For a long time I've been trying to get a decent SIP service at home. I've > tried both Partysip and Siproxd, both of which give me voice connections but > no presence, IM, video or additional streams for whiteboards or Messenger's > "application sharing." > > I've been picking up snippets about NATHelper, which as I understand it > allows SER to run on a NAT box and allow people to call across the NAT box. > However, I've also seen a couple of messages that suggest that as of October > this module was only handling single audio streams, which is the same as I > have at the moment with Partysip and Siproxd. Is my understanding right and > I'll have to wait a bit longer before trying SER, or does it already do > multistream? Multistream is video + audio, right? If so, then yes, only single audio stream is supported at the moment, though it would be trivial to add support for multistream as well. Please also note that currently I am working on next version of nathelper/rtpproxy, which would allow "bridging", i.e. the configuration when ser/rtpproxy runs on NAT box, listens on both private and public addresses and relays SIP/RTP traffic between public and private interfaces. -Maxim

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[Serusers] Kphone + Video Calls
by dhiraj.2.bhuyan＠bt.com 20 Jan '04

20 Jan '04

Hello List, Did anyone have success with video calls using Kphone (3.14 and 4.0)? I am using Redhat 9 - with self configured kernel 2.4.24 and pwc video driver. I installed vic - but cound not get video calls working. vic detects the webcam when I try to run it from command prompt. Anyone know how to get this thing working? Any help will be greatly appreciated. Thanks, Dhiraj Bhuyan Network Security Specialist, BT Exact Business Assurance Solutions Tel: +44 1473 643932 Mob: +44 7962 012145 Email: dhiraj.2.bhuyan(a)bt.com

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[Serusers] NATHelper
by Craig Graham 20 Jan '04

20 Jan '04

I have a very small setup; at the office there's a few machines behind an Intertex IX66 NAT box, that does a fairly good job of SIP as far as we can tell from the few connections we've had to other sites. At home, I have a few machines behind a Linux box, doing NAT. All user machines are running MS Messenger. For a long time I've been trying to get a decent SIP service at home. I've tried both Partysip and Siproxd, both of which give me voice connections but no presence, IM, video or additional streams for whiteboards or Messenger's "application sharing." I've been picking up snippets about NATHelper, which as I understand it allows SER to run on a NAT box and allow people to call across the NAT box. However, I've also seen a couple of messages that suggest that as of October this module was only handling single audio streams, which is the same as I have at the moment with Partysip and Siproxd. Is my understanding right and I'll have to wait a bit longer before trying SER, or does it already do multistream? -- Dr. Craig Graham, Software Engineer Advanced Analysis and Integration Limited, UK. http://www.aail.co.uk/

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[Serusers] CC-Diversion and Cisco 3725
by Marcello Lupo 20 Jan '04

20 Jan '04

Hi to all, i have a Cisco 3725 as PSTN-VOIP gateway. I have to implement the call transfer service (from VOIP to PSTN) using the CC-Diversion header field. I understood, from the documentation, that the Cisco use the CC-Diversion field only on a 3xx answer. Is this true or can i use it in an originating call too?? Becouse when i receive a call from PSTN i have a script (used with exec_dset) that check if the call forward is active for that number, and if it is active, it rewrite the URI with the destination number changed with the number selected from the customer to transferr the call to. My problem is that i need to put the original destinatin number in the CC-Diversion field so the gateway allow the call to pass through otherwise it will negate the call becouse it is originatig from a not allowed number on that PRI. Using the CC-Diversion field the cisco should put in the ISDN Setup the Redirecting number as the original destination for that call and our provider will check that field to allow the call out. But if i use this process tha cisco will manage the call forward as a new call originating from the proxy and not a 302 answer Moved temporarly. This is becouse i asked if the cisco will manage the CC-Diversion field only on a 302 answer or in all cases. I tried to use the sl_send_reply with this sintax: if(uri=~"sip:123456@10.10.10.10.*"){ exec_dset("/usr/local/etc/ser/rewriteuri_cf_unconditioned $URI"); append_hf("CC-Diversion:<sip:123456@20.20.20.20:5060>;reason=unconditional\r\n"); rewritehost("30.30.30.30"); // Cisco IP sl_send_reply("302", "Moved Temporarily"); }; but when the cisco receive the packet there is not the CC-Diversion field in it. It seems to be as the SER is rewriting completely the URI to send the replay. This probably due to my lack of experience on SIP can someone help me? Thanks, BYe, MArcello

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[Serusers] voip-billing
by Murali Krishnan 20 Jan '04

20 Jan '04

Hello, I'm doing a project on voip billing coupled with SER SIP servers. I've completely installed SER SIP server,and I've installed trabas voip billing system on the same Linux 8.0 system. i did it according to the install manual, but when i run the browser with the url of the virtual host on my mozilla browser, it says, 'Connection was refused when attempting to contact abc.def.ghi.jkl'. I'll explain the detailed conf that i did. The first step of your detailed installation manual was easy and did the second one up to creating two databases, dw_voip, and dw_amrita. After that i couldn't grant a user and password to access those databases. could you please tell me how to? I completed the third step of configuring virtual host. i did it through redhat --> server settings --> http server tab. I gave the name of virtual host as voiphost. but am having doubt with the 'document root directory' field. should i give the path to the normal http server document root, or should i give it as /root/voip-billing/resources/ ? on host information, i gave it as IP based virtual host, with the system's ip and the server's name 'localhost'. after these, when i run the url 'http://abc.def.ghi.jkl/', it returs an error message saying 'Connection was refused when attempting to contact abc.def.ghi.jkl'... I WOULD REALLY APPRECIATE IF YOU COULD THOROW SOME LIGHT ON THESE ISSUES... __________________________________ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus

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RE: [Serusers] Recording SIP calls
by Klaus Darilion 20 Jan '04

20 Jan '04

sipps (www.ahead.de) has a built-in recording machine. regads, klaus > -----Original Message----- > From: Darshan Uka [mailto:ukadarshan_2001@hotmail.com] > Sent: Monday, January 19, 2004 5:00 AM > To: serusers(a)lists.iptel.org > Cc: nils(a)iptel.org > Subject: [Serusers] Recording SIP calls > > > > Hi, > Is anyone aware of any SIP user agent that can record a SIP > call and store > it to a .wav file? > Is it possible to configure the SIP Express Router to record > the SIP voice > call and save it to a file rather than sending it to the other end? > If yes, please let me know the details. > Thanks, > > > -Darshan. > > _________________________________________________________________ > Let the new MSN Premium Internet Software make the most of > your high-speed > experience. http://join.msn.com/?pgmarket=en-us&page=byoa/prem&ST=1 > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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[Fwd: Re: [Serusers] ser not stopping]
by mdehn＠yknet.ca 20 Jan '04

20 Jan '04

---------------------------- Original Message ---------------------------- Subject: Re: [Serusers] ser not stopping From: mdehn(a)yknet.ca Date: Mon, January 19, 2004 10:18 pm To: "Annie Sasidar" <asasidar(a)mail.unomaha.edu> -------------------------------------------------------------------------- The reason you see "Stopping SER: Failed" is that it was not running in the first place. Then when you see "Starting SER: OK", your system has started SER. If you were to do another restart you would see that is successfully stops and starts again. Mike > > > > > Hi, > When i reboot the pc ser does not stop while shutdown. > Also when i run the /etc/rc.d/init.d/ser restart, i get > Stopping ser: Failed > Starting ser: OK. > > Why is ser not stopping? > Is it a problem? > > Please help. > Thanks, > Annie > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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sr-users January 2004