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Hi, First, here is my config: Kernel version 2.4.25 on a Fedora distro, Asterisk and a Fritz! Isdn PCI Card (v2). I try to make the CAPI drivers deal with Asterisk but I can't try to figure out to get of this issue. As I see, Fritz modules are integrated with the kernel, so I directly loaded the 'hisax_fcpcipnp' module from it. I install also Capi modules by downloading archives of the

Hi there, Is it possible to manage UUI with asterisk and ISDN (T0 Fritz card). Basically, is it possible to send User to User Information using the D-channel, while making a call?

...u put this patch on line? (I don't think it's too big...) In my mind, the main objective is to create a special field and force its value in chan_capi.c and check wether it goes through asterisk or not... What do you think of that? Regards ---------------------- > >jean-marie.goupil@telintrans.fr wrote: >> OK, so I'll do that... Is there any infos I need to know about chan_sip.c >> (because I suppose it's it that I need to play with)? > >Some stuff is already there. This is a "capi debug" trace where i SEND >UUS1 from a normal ISDN Phone TO an aster...

...i.c) Now I would like to know where this field is instanciated for outgoing calls in order to control this step? I am looking for that but I don't really know in which file I need to search...(apparently, it is not in chan_capi.c) Can anyone help me? >Hi Again, > >jean-marie.goupil@telintrans.fr wrote: > >>Can you put this patch on line? (I don't think it's too big...) >> >> >Sorry, i simply don't have a patch, IIRC all i did was inserting >something like > >693a694 > CONNECT_REQ_USERUSERDATA(&CMSG) = "testing"; > &g...

I tried that but it still doesn't work... I think I don't have the correct approach. Have I to install any ISDN drivers (=modules ?) BEFORE dealing with CAPI ? If yes, why shouldn't I use the hisax drivers (which are kernel ones) instead of fcpci drivers (which doesn't seems to work, by the way...) And finally, how is it possible to link the two modules together? As you can see,

If I do that, the modprobe doesn't find any module called fcpci and is looking for any module called hisax_fcpcipnp as it's the one install for the isdn fritz card. # capiinit modprobe: Can't locate module fcpci ERROR: failed to load driver fcpci Maybe, I should reinstall the fritz card drivers itselves? (however, I don't think so...)

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><div>Finally, i will get back to a RedHat 9 distrib as I see that it works with that distribution...</div></FONT>

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><div>I am interested too so I think it would a good thing to post an URL (as you said)</div><DIV>&nbsp;</DIV><DIV>Thank you!</DIV></FONT>

I've configured asterisk to connect a MySQL database for CDR, Voicemail and SIP/IAX2 peers. - CDR are reccorded - Voicemail config is readen directly in the database but SIP/IAX2 peers still have to be declared in sip/iax2.conf to make calls... However, when I restart Asterisk: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found ==

OK, so I'll do that... Is there any infos I need to know about chan_sip.c (because I suppose it's it that I need to play with)? Does anyone know an interesting website where I can find infos about UUI in ISDN (with CAPI maybe?) ? Thanks for your help.

Here is what you should write in extensions.conf: exten => _5.,1,Dial(IAX2/iax-a2:secret@a1.mystrx.com /${EXTEN}@inbound-calls So when you will dial anything beginning with 5, the call will be dialed in the context inbound-calls of a1.mystrx.com -----asterisk-users-admin@lists.digium.com a ?crit : ----- Pour: asterisk-users@lists.digium.com De: willy@yponeinc.com Envoy? par:

Does anyone knows wether it's possible or not to do SIP seeding with MYSQL_FRIENDS config (sip.conf is coded via a MySQL table) ?

Hi there! Does anyone knows how to test Asterisk load with sipp? I am using uac.xml to call a 'playback extensions' via a SIP channel. When I increase the Call rate (about 20cps), I begin to have INVITE/200/BYE retransmissions meanwhile the RedHat box is not loaded at all (made a TOP). Where is the pb? [root@10.54.196.38 sipp]# sipp 10.54.196.32 -s 9001 -sf uac.xml -d 100 -i 10.54.196.38

You maybe have to create a SIP user called like it is declared in your UAC/UAS xml file. I think it should be 'sipp' or something like that... -----asterisk-users-admin@lists.digium.com a ?crit : ----- Pour: <asterisk-users@lists.digium.com> De: "C. Johnson" <javadude@cedrick.net> Envoy? par: asterisk-users-admin@lists.digium.com Date: 31-05-2004 08:03 Objet: RE:

Hi there, I just give you the script (in Python) I have just written in case of someone would like to implemant this. I think it is more simple than the one we can see over the net... It uses DISA (security issues ==> limit access with contexts and the password !!) and CAPI but it should work with type of channel. Basically, you ring your asterisk and the line goes down after 1 ring. Asterisk

Hi, sipp (http://sipp.sourceforge.net/) seems to be a good app. Take a look at http://www.voip-info.org/wiki-SIPP on the wiki to have more info about it... Basically, there is scenario which are describe there and I personnally generated about 3,000,000 calls before having to restart asterisk and i placed about 90 concurrent calls. Good luck! -----asterisk-users-admin@lists.digium.com a ?crit :