Hi all, -------- beacause I am a newbie in the asterisk ralm and the existing documentation could not satisfy I'd like to ask you some Questions: 1. Does somewhere in the Internet exist additional documentations for asterisk configuration ? 2. Does Asterisk work as a standard SIP Proxy ? 3. I am just installing a Asterisk PBX in our institute and additionally I purchased some ot the Snom 100 SIP devices. For SIP account I have registered with iptel.org and fwd.pulver.com. Both work fine with the Snom 100 device. I tried the Snom 100 from home via a Linksys DSL router wich supports UPnp and port range forwarding to the private network. I faced some troubles with registering the Snom with IPTEL.ORG from my NAT'ed network. Because I'd like to setup a Asterisk PBX to my private Network as well my approach is to connect the SNOM to the asterisk at home network and establish a server-server connection via IAX to the institutes Asterisk server. The institute's Asterisk can login at iptel.org and fwd.pulver.com. If soembody wants to call me he should first try to reach me at my Snom phone at institutes premisses and after a defined time this call should be forwarded to my home asterisk server to reach the SNom 100 at home. BTW - for calling out I want to use the AVM Fritz ISDN card with ISDN4LINUX. I could not find any doc for Asterisk- ISDN4LINUX configuration. Later I'd like to upgrade to a Digium device to support conferencing. Wich of the Digium's devices you propose for a normal EuroISDN line (2 B-channels) ? I have attached a small figure for my SIP configuration aaproach. Thank you for your help regards Olaf -- Dipl. Ing. Olaf Menzel - System Engineer FOKUS - Fraunhofer Institute for Open Communication Systems: - Competence Center for Advanced Satellite Communication Schloss Birlinghoven, 53754 Sankt Augustin, Germany Phone: +49-2241-14-3494 Mobile: +49-175-2616161 Fax: +49-2241-14-43494 email: olaf.menzel@fokus.fhg.de Internet: http://www.fokus.fhg.de/satcom -------------- next part -------------- A non-text attachment was scrubbed... Name: asterisk-conf.gif Type: image/gif Size: 15802 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030607/17fdacd3/asterisk-conf.gif
>Hi all, >-------- >beacause I am a newbie in the asterisk ralm and the existing documentation >could not satisfy I'd like to ask you some Questions: > >1. Does somewhere in the Internet exist additional documentations for asterisk >configuration ?http://www.digium.com/handbook-draft.pdf In addition, there are a variety of home-built pages. http://www.automated.it/guidetoasterisk.htm http://asterisk.gnuinter.net/>2. Does Asterisk work as a standard SIP Proxy ?No. Asterisk can perform basic SIP redirection, but it is not a standard SIP proxy in many of the ways that you might expect. It can do _some_ of the features of a SIP proxy, but to call it a SIP proxy would be an overstatement.>3. I am just installing a Asterisk PBX in our institute and additionally I >purchased some ot the Snom 100 SIP devices. For SIP account I have >registered with iptel.org and fwd.pulver.com. Both work fine with the Snom >100 device. I tried the Snom 100 from home via a Linksys DSL router wich >supports UPnp and port range forwarding to the private network. I faced some >troubles with registering the Snom with IPTEL.ORG from my NAT'ed network. >Because I'd like to setup a Asterisk PBX to my private Network as well my >approach is to connect the SNOM to the asterisk at home network and establish >a server-server connection via IAX to the institutes Asterisk server. The >institute's Asterisk can login at iptel.org and fwd.pulver.com. If soembody >wants to call me he should first try to reach me at my Snom phone at >institutes premisses and after a defined time this call should be forwarded >to my home asterisk server to reach the SNom 100 at home. BTW - for calling >out I want to use the AVM Fritz ISDN card with ISDN4LINUX. I could not find >any doc for Asterisk- ISDN4LINUX configuration. Later I'd like to upgrade to >a Digium device to support conferencing. Wich of the Digium's devices you >propose for a normal EuroISDN line (2 B-channels) ? I have attached a small >figure for my SIP configuration aaproach. >My suggestion is that if your Asterisk server is behind a NAT, the only hosts it should talk with are devices that are also behind the NAT, or other Asterisk servers via IAX or IAX2 which may be on the outside of the NAT. Anything else causes more headaches than it's worth. I'm afraid I have no experience with the EuroISDN equipment, so I am not qualified to answer that part of your questions. JT> >Thank you for your help > >regards > >Olaf > > >-- >Dipl. Ing. Olaf Menzel - System Engineer >FOKUS - Fraunhofer Institute for Open Communication Systems: >- Competence Center for Advanced Satellite Communication >Schloss Birlinghoven, 53754 Sankt Augustin, Germany >Phone: +49-2241-14-3494 Mobile: +49-175-2616161 Fax: +49-2241-14-43494 >email: olaf.menzel@fokus.fhg.de Internet: http://www.fokus.fhg.de/satcom >Attachment converted: PrivateSpace:asterisk-conf.gif (GIFf/prvw) (000324C8)
shido
2003-Jun-07 15:32 UTC
[Asterisk-Users] sip channel driver causes asterisk to crash when talking to quintum A800
This is the sip debug when the call went through........
Sip read:
INVITE sip:2877433@64.42.218.157 SIP/2.0
Call-ID: call-80B9AC74-7A97-D711-0006@64.42.218.146
Contact: <sip:2044808000@64.42.218.146>
Content-Length: 157
Content-Type: application/sdp
CSeq: 1 INVITE
From: <sip:2044808000@64.42.218.146>;tag=402ada92-5
To: <sip:2877433@64.42.218.157>
User-Agent: Quintum/1.0.0
Via: SIP/2.0/UDP 64.42.218.146;branch=z9hG4bK-tenor-64.42.218.146-5
Quintum: 0c01030b0239380501
v=0
o=Quintum 4 4 IN IP4 64.42.218.146
s=VoipCall
c=IN IP4 64.42.218.146
t=0 0
m=audio 10240 RTP/AVP 0
c=IN IP4 64.42.218.146
a=rtpmap:0 pcmu/8000/1
11 headers, 8 lines
Using latest request as basis request
Sending to 64.42.218.146 : 5060 (non-NAT)
Capabilities: us - 4, them - 4, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
This is the backtrace from asterisk........
(gdb) bt
#0 0x42080e72 in strncpy () from /lib/i686/libc.so.6
#1 0x41fd8971 in get_calleridname (input=0x44e20fbc "",
output=0x44e20f7c
"<sip:2044808000@64.42.218.146>;tag=402ada92-5")
at chan_sip.c:3345
#2 0x41fd8a3d in check_user (p=0x0, req=0x0, cmd=0x0, uri=0x0, reliable=0)
at chan_sip.c:3365
(gdb)
The quintum is running firmware - OBCSM[98]: Release from peer
user agent
quintum/1.0.0
Ok in the quintum debug we have:
64.42.218.146IP/2.01-8(*SPROTO :270367494:From:
<sip:2044808000@64.42.218.146>;tag=402ada92-4:Call-ID: call-0081F509-6E9
SPROTO :270161169:c=IN IP4 64.42.218.146
SPROTO :270367494:To: <sip:2877433@64.42.218.157>0161169:t=0
09932583:Contact: <sip:2
SPROTO :2701
SPROTO :270367494:User-Agent: Quintum/1.0.0owed}
For each of t
S
SPROTO :27016116
SPROTO :270367494:Via: SIP/2.0/UDP
64.42.218.146;branch=z9hG4bK-tenor-64.42.218.9:a=rtpmap:0 pcmu/8000/1
lamreplace: num {lam
SPROTO :2701611
146-483:CS
SPROTO :270367494:Quintum:
0c01030b0239370501ansaction::SendRequest()startTransaction succ
SPROTO :270367494:92-1u want PPPoE c
SPROTO :270367494:v=0
SPROTO :270367494:o=Quintum 3 3 IN IP4
64.42.218.146ansaction::stateRetransmissionTimer()rerint current
SPROTO :270367494:s=VoipCall-Agent: Quintum/1.0.0 exi
SPROTO :270367494:c=IN IP4 64.42.218.146TO :269932583:Via: SIP
SPROTO :270367494:t=0 0
EXCP :270367495:Socket: pid=0x3b desc=6 send() returned errno=0x5020
lr1=0x
I hope this helps....
Greg Merriweather
Hi John, ------------ thank you for pointing me to to some of the additional Asterisk documentation stuff.> > http://www.digium.com/handbook-draft.pdf > > In addition, there are a variety of home-built pages. > http://www.automated.it/guidetoasterisk.htm > http://asterisk.gnuinter.net/ >I found a lot of helpful hints.> My suggestion is that if your Asterisk server is behind a NAT, the > only hosts it should talk with are devices that are also behind the > NAT, or other Asterisk servers via IAX or IAX2 which may be on the > outside of the NAT.All my SIP devices in my home network are behind the NAT. The asterisk server at our institute has a official public IP address. Maybe the existing firewall ca make some trouble ?? Thank you for your fast reply and help Olaf
On Sunday 08 June 2003 08:45, Olaf Menzel wrote:> Hi John, > ------------ > thank you for pointing me to to some of the additional Asterisk > documentation stuff. > > > http://www.digium.com/handbook-draft.pdf > > > > In addition, there are a variety of home-built pages. > > http://www.automated.it/guidetoasterisk.htm > > http://asterisk.gnuinter.net/ > > I found a lot of helpful hints. > > > My suggestion is that if your Asterisk server is behind a NAT, the > > only hosts it should talk with are devices that are also behind the > > NAT, or other Asterisk servers via IAX or IAX2 which may be on the > > outside of the NAT. > > All my SIP devices in my home network are behind the NAT. The > asterisk server at our institute has a official public IP address. > Maybe the existing firewall ca make some trouble ??Two solutions: your existing asterisk server could be given an interface on the internal network (without forwarding packets). Second, you could deploy an additional internal Asterisk server whose sole purpose would be to translate SIP calls to IAX, to relay to the outside Asterisk server. -Tilghman