Re: [asterisk-users] Completing my Configuration
Dear Anselm,
I am sorry about the big traffic in the newsgroup.
I tried to send my post to the newsgroup for 3 days now - once a day,
but it did not appear. Today I tried putting it in cc also with, then it
worked out ...
I will carefully read your answer.
thank you veryy much
Original-Nachricht
Datum: Tue, 25 Sep 2007 09:34:14 +0200
Von: Anselm Martin Hoffmeister [EMAIL PROTECTED]
An: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Completing my Configuration
Am Dienstag, den 25.09.2007, 07:37 +0200 schrieb Guenther Sohler:
Hallo Group,
I have basically set up a small asterisk system,
which ahs 4 peers:
* registers at 2 Sipgates
* 2 hardware phones connected to it
Both Hardware phones can phone outwards(cheaper sipgate is selected with
dialplan)
Calls from both sipgates make my hardware phones ring
But here comes the challenges:
Is it possible to configure asterisk in such a way that in the phone:
* there are names instead of numbers in my hardware phone displayed
Depends on the hardware phones. In theory, with each SIP call connecting
to the phone, both a name and a number can be transferred. AFAIK sipgate
defaults to setting both to the usual callerID. That is exactly the
reason why you can set the variables ${CALLERID(num)} and
${CALLERID(name)}.
Some hardware phones (I assume, the better ones ;-) display both; my
Allnet for example seems to only display the name, but store the number
for the call back list. My Fritz!Boxen seem to forward both name and
number to ISDN devices on the internal S0-bus, just not many ISDN phones
can actually display text numbers.
Let your asterisk have an ast database, looking like
callerid/420123456789 = Doe, John Q.
callerid/492240224922 = Mustermann, Dr. Peter
Then you could expand your dialplan logic a little. If you have a line
exten = 12345,4,Dial(SIP/phone1,60)
or whatever that looks like in your SIP-incoming context, insert those
lines before it [and change the 4, 5, 6, 7s ;-) ]
exten = 12345,4,Set(CALLERID(name)=${DB(callerid/${CALLERID(num)})})
exten = 12345,5,GotoIf($[${CALLERID(name)} = ]?6:7)
exten = 12345,6,Set(CALLERID(name)=-- ${CALLERID(num)})
exten = 12345,7,Dial(SIP/phone1,60)
Line 6 treats the case that the number is not in your database and sets
the callerid-name to -- NUMBER_OF_CALLER
You can manually add data to the astdb from the asterisk CLI with
database set callerid 420456789 Silly, Roger M.
You should check that both your SIP providers provide incoming CLI in
the international formatting, without country prefix or +. In my
experience some SIP providers send numbers like
492240224922, others send +49... or 0049..., some send national format
02240... for all national calls, some even omit the leading 0 there,
and some just change the behaviour depending from which network (T-Com
landline, Arcor landline, T-Mobile cell phone, O2 cellphone, foreign
callers...) the call originates. If you have more than two providers,
this can be a PITA - you will need some dialplan logic to sanitize the
callerid in those cases, and sometimes you are just left for guessing,
for example when the provider signals calls from T-Mobile as 16177554224
and calls from Boston, MA, USA the very same. Germany does not have
fixed-length numbers, even in the mobile phone networks the length
differs, and the number given might be valid for both circumstances.
/rant
* The Ringtone is different for special call numbers
If your phone supports that, yes, you can do it. The common method for
this seems to be sending an additional header. There will be docs on
SIPAddHeader(blah) or similar on www.voip-info.org, and you might want
to also use a database here to find out wether special ringtones are to
be activated or not.
* it is displayed, in which sipgate the call came from
You could use the CALLERID(name) field for that, by adding the provider
short name in front of the caller's name, like
exten = 12345,4,Set(CALLERID(name)=at-${DB(callerid/${CALLERID(num)})})
for calls via the at provider - or whatever seems stylish enough.
I personally have a logic that makes use of the dial-around prefix in
use here in Germany: From a regular T-Com landline you can select the
provider that will carry the next call by dialling 010[1-9]X or 0100XX.
Those prefixes of course do not work on SIP provider lines, and my
asterisk does not have landlines connected. So I use those for my own
purposes, e.g. selecting the SIP account that the call may go out
through. Dialplan logic detects 010XX (100 possible accounts are
enough, I just ignore 0100XX as additional number field here) and
selects the outgoing provider accordingly.
If I wished to have the incoming line signalled to me, I would prefix
the incoming CALLERID(num) with the provider code. Callbacks would go
Re: [asterisk-users] Anyone else having problems with the list
me too :) Original-Nachricht Datum: Tue, 25 Sep 2007 12:57:25 +0200 Von: Michiel van Baak [EMAIL PROTECTED] An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] Anyone else having problems with the list On 22:43, Tue 25 Sep 07, Carlos Hernandez wrote: Yes for me. Carlos -- Julian Lyndon-Smith wrote: I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian I have similar problems. Some mails arrive, some dont. If I check the listarchive on the web I see more emails then in mutt. I already disabled greylisting etc and browsed thru the spam quarantine but nothing. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GMX FreeMail: 1 GB Postfach, 5 E-Mail-Adressen, 10 Free SMS. Alle Infos und kostenlose Anmeldung: http://www.gmx.net/de/go/freemail ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Completing my Configuration
Hallo Group, I have basically set up a small asterisk system, which ahs 4 peers: * registers at 2 Sipgates * 2 hardware phones connected to it Both Hardware phones can phone outwards(cheaper sipgate is selected with dialplan) Calls from both sipgates make my hardware phones ring But here comes the challenges: Is it possible to configure asterisk in such a way that in the phone: * there are names instead of numbers in my hardware phone displayed * The Ringtone is different for special call numbers * it is displayed, in which sipgate the call came from * using an extension in my call number redirects the call just to one sip phone ? And What about Asterisk web server: I was told you can sue it to configure asterisk via web. I turned it on an connected to it, but I can only read 404 Object not found Asterisk Webserver Whats wrong ? Thank you very much for your inspirations! -- Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP and Firewall
Hallo, I'd like to correctly set up my firewall in my system for udp and asterisk I have got a server, which has got one static ip adress to the internet. Asterisks is running on this server. It registers at sipgate.at and mujtelefon.com The Server also does nat to the my intranet, where my pc and my hardware sip phone sits. The Hardware sip phone registers to asterisk on my server from its intranet ip adress. Everything works fine. The question is just: How to code good stateful firewall rules with iptables and netfilter_sip ? What would be apropriate to my system ? rds -- GMX FreeMail: 1 GB Postfach, 5 E-Mail-Adressen, 10 Free SMS. Alle Infos und kostenlose Anmeldung: http://www.gmx.net/de/go/freemail ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP and Firewall
Dear Group! I want to improve the firewall rules for SIP and I already compiled the linux kernel with additional SIP netfilter settings Now I found this on the internet: modprobe ip_conntrack_sip ip_nat_sip Set IPtables filter rules iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT iptables -A INPUT -p udp --dport 5060 -j ACCEPT Set IPtables NAT rules iptables -A FORWARD -o eth0 -p udp --dport 5060 -j ACCEPT iptables -t nat -A POSTROUTING -o eth0 -j SNAT --to-source ip.add.dr.ess -- But I do not understand it. where is eth0 connected to in this example ? and what would be the source ip adress ? what will happen if i amn connected to 2 sip gateways ? -- Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newcomer Question
Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at both providers(numbers) at the same time. And If I call someone, calls to austria shall use sipgate, whereas calls to czech shall use mujtelefon. So far, i read through the tutorial at digium.com and got the impression, that asterisk might be able to do this and makes phoning very conveniant. YEsterday evening i managed to get a compiled version of asterisk running on my server. I just font yet have a complete idea, what is to be changed * get asterisk running on my server as phone central(registrar) * which firewall settings ? before/after nat * my hardware phone registers with asterisk at my server * which files to i have to change ? dialplan, sip.conf? How do I achieve this ? Thanx in advance! -- Pt! Schon vom neuen GMX MultiMessenger gehört? Der kanns mit allen: http://www.gmx.net/de/go/multimessenger ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newcomer Question
My Phone identifies as USer-Agent: ALL7950 02.09.23 I suppose its AllNet 7950 Hope this helps :) Original-Nachricht Datum: Thu, 20 Sep 2007 08:36:59 +0200 Von: Jan De Coster [EMAIL PROTECTED] An: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] Newcomer Question hi, and first off all ... welcome! now it would be handy if you provide us with the name of your phone for ex 'a linksys spa942' or somthing kr, Jan de Coster Guenther Sohler wrote: Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at both providers(numbers) at the same time. And If I call someone, calls to austria shall use sipgate, whereas calls to czech shall use mujtelefon. So far, i read through the tutorial at digium.com and got the impression, that asterisk might be able to do this and makes phoning very conveniant. YEsterday evening i managed to get a compiled version of asterisk running on my server. I just font yet have a complete idea, what is to be changed * get asterisk running on my server as phone central(registrar) * which firewall settings ? before/after nat * my hardware phone registers with asterisk at my server * which files to i have to change ? dialplan, sip.conf? How do I achieve this ? Thanx in advance! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newcomer Question
Hallo Martin and Group! Thank you very much for your perfect introduction into asterisk. I managed to * get asterisk server running * configuring the internal numbers * registering to 2 sip gateways * outbound phoning to sipgate works perfect * outbound phoning to mujtelefon not yet tested The problem i am having now is, that i cant be reached by inbound phone calls from neither sipgate nor mujtelefon i used my mobile to call this numbers. sipgate tells me on the phone:Das Endgeraet ist fuer diesen Service nicht konfiguriert. bei mujtelefon kommt nur die mobilbox In asterisk cli i dont see anything about that. in the sipgate login page there is neither mentioned. If this is working, I intend to have nice music played for incoming calls until the phone call is accepted. What is very confusing for sipgate is, that my number(734365) is different from my user name(1734365). Can anybopdy check, if all settings are ok according to that ? Please find below my sip.conf, only the passwords are scrambled. If you directly reply to me, also reply to [EMAIL PROTECTED] because i am afraid missing your answer in the much traffic in that mailing list. Thank you very much! [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm musicclass=default language=de dtmfmode=rfc2833 sipdebug=no register = 1734365:[EMAIL PROTECTED]:5060/00437201734365 register = 272048160:[EMAIL PROTECTED]:5060/00420272048160 [sipgateat] host=sipgate.at secret=NMMTNMKP username=1734365 fromuser=1734365 fromdomain=sipgate.at srvlookup=yes context=sipgateat-in canreinvite=no nat=no type=friend qualify=yes
[asterisk-users] Newcomer Question
Hallo Martin and Group! Thank you very much for your perfect introduction into asterisk. I managed to * get asterisk server running * configuring the internal numbers * registering to 2 sip gateways * outbound phoning to sipgate works perfect * outbound phoning to mujtelefon not yet tested The problem i am having now is, that i cant be reached by inbound phone calls from neither sipgate nor mujtelefon i used my mobile to call this numbers. sipgate tells me on the phone:Das Endgeraet ist fuer diesen Service nicht konfiguriert. bei mujtelefon kommt nur die mobilbox In asterisk cli i dont see anything about that. in the sipgate login page there is neither mentioned. If this is working, I intend to have nice music played for incoming calls until the phone call is accepted. What is very confusing for sipgate is, that my number(734365) is different from my user name(1734365). Can anybopdy check, if all settings are ok according to that ? Please find below my sip.conf, only the passwords are scrambled. If you directly reply to me, also reply to [EMAIL PROTECTED] because i am afraid missing your answer in the much traffic in that mailing list. Thank you very much! [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm musicclass=default language=de dtmfmode=rfc2833 sipdebug=no register = 1734365:[EMAIL PROTECTED]:5060/00437201734365 register = 272048160:[EMAIL PROTECTED]:5060/00420272048160 [sipgateat] host=sipgate.at secret=NMMTNMKP username=1734365 fromuser=1734365 fromdomain=sipgate.at srvlookup=yes context=sipgateat-in canreinvite=no nat=no type=friend qualify=yes
