Re: [asterisk-users] SEMI OFF-TOPIC - Fail2ban
On 01/08/2015 11:37 PM, ricky gutierrez wrote: Hi list , someone on the list has seen this type of connection attempts in asterisk, fail2ban does not stop 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: SecurityEvent=ChallengeSent,EventTV=1420750787-386840,Severity=Informational,Service=SIP,EventVersion=1,AccountID=sip:100@173.230.133.20,SessionID=0x169f528,LocalAddress=IPV4/UDP/173.230.133.20/5060,RemoteAddress=IPV4/UDP/63.141.229.58/5078,Challenge=770e84a3 [2015-01-08 15:20:20] SECURITY[21515] res_security_log.c: SecurityEvent=ChallengeSent,EventTV=1420752020-854997,Severity=Informational,Service=SIP,EventVersion=1,AccountID=sip:102@173.230.133.20,SessionID=0x169f528,LocalAddress=IPV4/UDP/173.230.133.20/5060,RemoteAddress=IPV4/UDP/198.204.241.58/5074,Challenge=23965594 I modified the fail2ban with the filter, but still not detected Do you really want to detect ChallengeSent? That should occur also on legitimate login processes... -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Commas is variables problem
Hi,
I'm running into a strange problem with commas is variables. I have the
following contexts:
[messages]
exten = _+.,1,Noop(External SMS)
same = n,Set(ACTUALTO=${CUT(CUT(MESSAGE(to),@,1),:,2)})
same = n,Macro(goip_sendsms,${ACTUALTO},${MESSAGE(body)})
same = n,Hangup()
[macro-goip_sendsms] ;Call Macro(goip_sendsms,number,message)
exten = s,1,Noop(SMS via GOIP)
same = n,Set(MESSAGE(body)=${URIDECODE(${ARG1}%0A${ARG2})})
same = n,MessageSend(sip:goip)
When I send
Test,test
through this, the receiver gets an SMS saying
Test
The console shows
-- Executing [+3584172x@messages:1]
NoOp(Message/ast_msg_queue, External SMS) in new stack
-- Executing [+3584172x@messages:2] Set(Message/ast_msg_queue,
ACTUALTO=+3584172x) in new stack
-- Executing [+3584172x@messages:3]
Macro(Message/ast_msg_queue, goip_sendsms,+3584172x,Test,test)
in new stack
-- Executing [s@macro-goip_sendsms:1] NoOp(Message/ast_msg_queue,
SMS via GOIP) in new stack
-- Executing [s@macro-goip_sendsms:2] Set(Message/ast_msg_queue,
MESSAGE(body)=+3584172x
-- Test) in new stack
-- Executing [s@macro-goip_sendsms:3]
MessageSend(Message/ast_msg_queue, sip:goip) in new stack
-- Executing [+3584172x@messages:4]
Hangup(Message/ast_msg_queue, ) in new stack
== Spawn extension (messages, +3584172x, 4) exited non-zero on
'Message/ast_msg_queue'
Is there a way to make * ignore the comma in the string here?
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
Description: S/MIME Cryptographic Signature
--
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[asterisk-users] Get last dialed number in a context?
Hi,
I would like to implement an auto-redial function in a context. The idea
is about like this:
Dial a number
Hear busy
Hangup
Pick up again
Dial a code like *123
= jumps into a context which redials until callresult is not busy
Maybe like this:
[autoredial]
exten = s,1,Set(number=${CHANNEL(lastdialed)})
exten = s,2,Dial(SIP/${number}@account,60,g)
exten = s,3,Wait(15)
exten = s,4,GotoIf( [ ${DIALSTATUS} = BUSY ]?2)
exten = s,5,Hangup
For that I'd need to somewhere get the last dialed number from the
channel/line I'm initiating the call from. Is something like this
already implemented?
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
Description: S/MIME Cryptographic Signature
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Get last dialed number in a context?
On 06/03/2014 01:53 PM, Patrick Laimbock wrote: Have you looked at Call Completion Supplementary Services (CCSS)? https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5243096 PSTN doesn't support that here. -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get last dialed number in a context?
On 06/03/2014 06:06 PM, Eric Wieling wrote: Have you tried RetryDial()? I want it to be a conscious decision and not just automatically in every call. For the vast majority of my call I can just try some time later but some people I need to get a hold of ASAP sometimes. -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get last dialed number in a context?
On 06/03/2014 12:44 PM, Israel Gottlieb wrote: you could save the info in astdb for the last call per extension and then pull it from there I guess I'll have to do this then :). -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get last dialed number in a context?
On 06/03/2014 08:10 PM, Carlos Chavez wrote: Why is the redial function on the phone not suitable for this? Why dial *123 when you can just hit redial on your phone? None of my phones (and none that I know) has any redial until you actually get somebody-function. It's only press the button and when it's busy again, you have to again hang up and press the button again... -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP fraud IP blacklist
Hi, in case, anyone is interested... I have started compiling a blacklist of hosts and networks from which SIP fraud attempts occur. My criteria currently are: To block an IP: - Minimum 3 attacks within one week from the same IP To block a network: - Attacks from minimum 3 IPs from that network within 2 weeks Common criteria: - Provider does not react to complaints OR - Provider sends autoreply but attacks don't stop within a week Definition of attack: - Minimum 5 attempts to make an unauthorized phone call to a non-PBX-internal number OR - Minimum 10 attempts to make an unauthorized phone call to a PBX-internal number OR - Minimum 10 failed authentication attempts If this happens, the IP gets auto-banned (iptables) for 24 hours and goes to my watch list. The watch list is the base for my further decisions. Currently, I don't remove IPs or networks from the list. If I have time and/or motivation I might create some kind of removal process later - also, depending on how big the list gets and how many people use it. The list is yet pretty short but for me, it has reduced the noise on my PBX from 20-30 attacks per day to about 2 or 3 per week, especially after most of the Palestinian networks ended up on the list. You're free to use the list - own your own responsibility and risk. It's in the ipdeny.com format, so a simple script can be used to CURL the list and create iptables rules from it. A sample script for something like that is also on my website (check the Linux section). That's the website for the list: http://stefan.gofferje.net/it-stuff/sipfraud/sip-attacker-blacklist And that's the download URL: http://stefan.gofferje.net/sipblocklist.zone Note that the list is updated every 6h so polling it more often doesn't help anything. Please limit polling to once a day or so. -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Numbers hackers call
On 03/26/2014 05:05 PM, Michelle Dupuis wrote: I see a lot of attempts by hackers to call 00972595301123 or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... Those lame hacking attempts aren't the big issue - unless you have an insecure SIP-PBX. Germany just got hit with a wave of hacks of Fritz!Box home routers with integrated SIP, causing hundreds of thousands in damage. The big issue is that the ISPs worldwide don't give a crap about complaints! And that's not only some backwater-ISPs in some 3rd world countries! It's mainly the big names, like Hetzner, L3, etc. who - oh well, yeah - send you an autoreply but in the end don't bother doing anything. Just recently was an article, again in a German IT-newsticker, about Hetzner's abuse handling. They just forward the complaint to their customer, including full contact data - which is pretty much illegal (privacy protection, etc.) - but they don't follow up. I got so fed up that I now put the top 20 of attacking IPs to my website... Current top 5: 1. iWeb (Canada) 2. Level 3 (USA) 3. Dacom (S-Korea) 4. Intergenia (Germany) 5. OVH (France) See http://stefan.gofferje.net/it-stuff/sipfraud Really, if everybody would run statistics on attacks and publish them, those ISPs would pretty quickly not only start reacting to fouled servers but probably start monitoring proactively because being in the top 20 of attacker-IPs ain't good for their reputation... -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Numbers hackers call
On 03/27/2014 08:36 PM, Eric Wieling wrote:
I have an iptables file which blocks all traffic except traffic from networks
allocated by ARIN or are Legacy networks. I pulled the information from
http://www.iana.org/assignments/ipv4-address-space/ipv4-address-space.xhtml
My iptables script can be found at the link below.
http://help.nyigc.net/tmp/iptables_geoblock
It might be helpful to someone.
Below's my solution. I specifically block China, Korea and Palestine.
That already massively reduced my amount of attacks. I can't block as
much as you because I do allow unregistered inbound SIP calls to
sip:ste...@home.mylastname.net. CN, KR and PS are currently the only
attack origins from where I wouldn't expect legit inbound traffic.
Here's my script (pulls data from ipdeny.com). The script is called in
my primary IPTABLES script after flushing and before my specific ruleset.
And it runs on my perimeter firewall.
WARNING: That's about 5000 networks to stuff into the tables! My fw is a
Phenom 8650 3-core machine and it takes about 8.5 minutes to stuff all
the rules into the kernel!
#!/bin/bash
IPTABLES=/sbin/iptables
ANY=0.0.0.0/0
BLOCKDIR=blocklist.d
if ! test -d ${BLOCKDIR}; then
mkdir ${BLOCKDIR}
fi
DATE=$(date)
echo Country blocking rules...
echo Downloading rules...
curl -s http://www.ipdeny.com/ipblocks/data/countries/cn.zone -o
${BLOCKDIR}/cn.zone || echo Warning: Couldn't download CN zone
curl -s http://www.ipdeny.com/ipblocks/data/countries/kr.zone -o
${BLOCKDIR}/kr.zone || echo Warning: Couldn't download KR zone
curl -s http://www.ipdeny.com/ipblocks/data/countries/ps.zone -o
${BLOCKDIR}/ps.zone || echo Warning: Couldn't download PS zone
echo Done downloading. Setting rules...
for FILE in ${BLOCKDIR}/*zone; do
for ADDRESS in $(cat ${FILE}); do
echo Blocking network: ${ADDRESS}...
$IPTABLES -A INPUT -s ${ADDRESS} -d $ANY -j DROP
$IPTABLES -A INPUT -s ${ADDRESS} -d $ANY -j LOG --log-prefix
Packet log: COUNTRY DROP
$IPTABLES -A FORWARD -s ${ADDRESS} -d $ANY -j DROP
$IPTABLES -A FORWARD -s ${ADDRESS} -d $ANY -j LOG --log-prefix
Packet log: COUNTRY DROP
done
done
echo Done. Started: ${DATE}, finished: $(date)
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler Koch - the original point and click interface
smime.p7s
Description: S/MIME Cryptographic Signature
--
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[asterisk-users] Is this list dead? Or the project?
Hi, I'm tinkering with Asterisk for * for about 12 years now and since about 10 years, it's my home PBX. I was off the list for something like 7 years - had other things to do. But... I remember, then, sometimes came over 1000 mails in 24h. Now it's hardly 50 new mails per week. Is the list dead? Or is the project dead? Or is nobody tinkering any more and everybody buying some turnkey-stuff? Just wondering... -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile and Nokie E51 = noise
Hi, I'm playing with * for about 12 years now and since about 10 years, it's my home PBX. I can do pretty much everything I want but one thing I haven't managed yet... Mobile connection via bluetooth... I'm still using a Nokia E51 and the setup and everything works fine. However, on the second or third call, the incoming audio is noise. I have tried alignmentdetection=yes and also forcemaster but it doesn't make a difference. That's my bt-dongle: home:~ # hwinfo --bluetooth 02: USB 00.0: 11500 Bluetooth Device [Created at usb.122] Unique ID: CiZ2.nQKjiuCfL84 Parent ID: uIhY.kllrQr_lFX9 SysFS ID: /devices/pci:00/:00:02.0/usb3/3-3/3-3:1.0 SysFS BusID: 3-3:1.0 Hardware Class: bluetooth Model: Cambridge Silicon Radio Bluetooth Dongle (HCI mode) Hotplug: USB Vendor: usb 0x0a12 Cambridge Silicon Radio, Ltd Device: usb 0x0001 Bluetooth Dongle (HCI mode) Revision: 31.64 Driver: btusb Driver Modules: btusb Speed: 12 Mbps Module Alias: usb:v0A12p0001d3164dcE0dsc01dp01icE0isc01ip01in00 Driver Info #0: Driver Status: btusb is active Driver Activation Cmd: modprobe btusb Config Status: cfg=new, avail=yes, need=no, active=unknown Attached to: #1 (Hub) I tried with a different bt-dongle with the same results. Additionally, I recently started getting this message in the syslog - and quite a lot of those: Jan 26 20:06:22 home kernel: [3023015.007826] Bluetooth: hci0 SCO packet for unknown connection handle 65328 Mobile integration is the last bit that my * is missing before being the perfect PBX, so I hope, somebody here could help me with that :). -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP/GSM-gateway recommendation?
Hi, can anybody recommend a priceworthy SIP/GSM-gateway that's known to work flawlessly with asterisk? Should especially support CLIP/CLIR in both directions and it would be perfect if it would send notifications e.g. if the incoming call is diverted or if the remote party puts me on hold. I don't favor GSM-PCI-cards because I'm just building a new asterisk based an an Atom board in a small casing. --Stefan -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk32H6YACgkQbQKZlCdPOMPq4QCgknv5BoRc2q18JjsO/2a9Sz8O gAsAoKER5vgiSu0ro+46OhBqbXsX6Qwx =HwKD -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP/IAX guest access?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I have a general question about SIP access for nonregistered users. I would like to make it possible for basically anybody to make a SIP call to my asterisk without having to have a user account, but in a specific context. So that e.g. somebody could make a SIP call to SIP/ste...@my.asterix.pbx and it would go like this: [incoming_guest] exten = stefan,1,dial(SIP/300SIP/301) exten = stefan,2,voicemail(300,u) For IAX I created a user [guest] with a specific context [incoming_guest] in which I handle the calls but I also haven't really figured out how to create the stefan@.. solution. To reach this context, people have to call IAX/gu...@my.asterisk.pbx How do I create a context in which all calls from nonregistered clients are handled? - -S - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk3xBXYACgkQbQKZlCdPOMN6qgCfR+TBVpVCSKDZyzUJk6r53VYS dvYAoJrbb76zqFY3c1K0YzA9j3dowPE4 =2SCj -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX guest access?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, On 06/09/2011 08:50 PM, Jamie A. Stapleton wrote: Guest calls go to the context specified in [general] of sip.conf. Thx. Is this valid for IAX2 also? - -S - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk3xDBEACgkQbQKZlCdPOMMogQCeOX1QWdLQJ9SQGnSHNoh9UGFO iWkAnjwp4oBhbNdGn+lz0fHb3hokH+/f =la5a -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jabber / facebook chat?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 05/18/2011 06:23 PM, Jason Parker wrote: To clarify, does that mean that you were able to successfully use facebook chat with sasl? This is correct. - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk3UGSwACgkQbQKZlCdPOMMj7QCdGZpt3CZEN6rP6sKBAxz2CcsM FnsAn1/Duexn+Seb3GaIcQ17L2Po7ELA =Ozxv -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jabber / facebook chat?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 04/17/2011 02:13 AM, Stefan Gofferje wrote: has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. I finally figured it out. For facebook chat to work you have to use usetls = no usesasl = yes Chat.facebook.com offers TLS but it seems, it's incompatible to res_jabber. - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk3SZ50ACgkQbQKZlCdPOMOWhACgso4Yse7GeGSKUI/4+8n523zu Ec0An2IecBY6Aelg1DRNoNFnxYGjO1aI =qn6C -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Repost: Jabber / GTalk / hints
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 04/17/2011 02:28 AM, Stefan Gofferje wrote: Hi! Are hints not yet implemented in res_jabber? I have this here: exten = 3000,hint,gtalk/gtalk_account/mari....@gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003@internal : SCCP/6003 State:Unavailable Watchers 0 6002@internal : SCCP/6002 State:IdleWatchers 0 6001@internal : SCCP/6001 State:IdleWatchers 0 6000@internal : SCCP/6000 State:IdleWatchers 0 6004@internal : SIP/sgofferj State:IdleWatchers 0 6200@internal : SCCP/6200 State:Unavailable Watchers 0 3000@internal : gtalk/gtalk_account/ State:Idle Watchers 1 Funnily, the gtalk hint is the only one with a watcher although all hints are hooked in various phones... Any ideas, comments, etc...? -S - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk3JE9QACgkQbQKZlCdPOMPFBQCfcGBlAppalPYIoCsPKbBUQ1UU 3hgAnRCp2HirVgI2aYmKoJsskG7dcVnC =9O5t -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Repost: Jabber / facebook chat?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 04/17/2011 02:13 AM, Stefan Gofferje wrote: Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. -S - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk3JE/oACgkQbQKZlCdPOMMAswCgoSK4Vlz6+VNVTNF5P9XcHeWY sLMAoKl0GuyRP/2GeL3PFgO5cP6KRK/G =4A13 -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?
On Thursday 28 April 2011, Bruce B wrote: How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? I am appreciate experienced inputs. Force the switch port which the asterisk is connected to 10MBit/s half-duplex and then fire a ping -f -s 65507 asterisk-host from a machine with a gigabit-link to the switch. That should get the line quality pretty much to the bottom. -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Repost: Jabber / facebook chat?
On Sunday 17 April 2011, Stefan Gofferje wrote: Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Repost: Jabber / GTalk / hints
On Sunday 17 April 2011, Stefan Gofferje wrote: Hi! Are hints not yet implemented in res_jabber? I have this here: exten = 3000,hint,gtalk/gtalk_account/mari....@gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003@internal : SCCP/6003 State:Unavailable Watchers 0 6002@internal : SCCP/6002 State:IdleWatchers 0 6001@internal : SCCP/6001 State:IdleWatchers 0 6000@internal : SCCP/6000 State:IdleWatchers 0 6004@internal : SIP/sgofferj State:IdleWatchers 0 6200@internal : SCCP/6200 State:Unavailable Watchers 0 3000@internal : gtalk/gtalk_account/ State:Idle Watchers 1 Funnily, the gtalk hint is the only one with a watcher although all hints are hooked in various phones... Any ideas, comments, etc...? -S -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile: Dropping incompatible voice frame
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I have no audio on chan_mobile but this message repeats continuously: Dropping incompatible voice frame on Mobile/DNA-54f4 of format slin since our native format has changed to 0x0 (nothing) Can somebody point me to the right direction? Asterisk SVN-branch-1.6.2-r313579 - -Stefan - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk2tr3YACgkQbQKZlCdPOMOLkwCfYm/jdPx3uOYdcvZ5XsZeKWAg sD8AoL4ygna6jWsKLY9sEwzU2VjRek/T =hbxJ -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Jabber / GTalk / hints
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten = 3000,hint,gtalk/gtalk_account/mari....@gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003@internal : SCCP/6003 State:Unavailable Watchers 0 6002@internal : SCCP/6002 State:IdleWatchers 0 6001@internal : SCCP/6001 State:IdleWatchers 0 6000@internal : SCCP/6000 State:IdleWatchers 0 6004@internal : SIP/sgofferj State:IdleWatchers 0 6200@internal : SCCP/6200 State:Unavailable Watchers 0 3000@internal : gtalk/gtalk_account/ State:Idle Watchers 1 Funnily, the gtalk hint is the only one with a watcher although all hints are hooked in various phones... Any ideas, comments, etc...? - -S - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk2qJgYACgkQbQKZlCdPOMN1xgCaA1Fk82FXF41AdImMU358VzDy kfUAoIkDK7qCx2Xjwn2bd/osg1rvuqBP =Z8lO -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Jabber / facebook chat?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk2qIpsACgkQbQKZlCdPOMOEcACfcVot6VqUDB/99PNXT2C+Bv5l QBwAnAr4yQjIg03IcwOHg4hnSCv5LrLT =Y112 -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- 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] Asterisk 1.4.22 and 1.6.0 Released
Asterisk Development Team schrieb: [Release info] Did anyone notice bug #0013531 (http://bugs.digium.com/view.php?id=13531)? It seems that the hold logic / MOH logic in chan_sip is somehow broken in 1.6.0... Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Dropping SIP support?
Michael Graves schrieb: Earlier today I glanced at Junction Networks blog and was surprised to find a post indicating that Cisco was dropping SIP support in their 79xx series phones. Here's t link: http://www.junctionnetworks.com/blog/charlotte/2008/09/19/junction-netwo rks-lab-cisco-7960-phones Is this true? What are they thinking? Only SCCP? AFAIK the other way around is true. Cisco is dropping SCCP. The new firmware is for SIP only but it's with some Cisco extensions as the latest CCMs are using SIP as preferred protocol. Could be that Cisco drops the standard SIP FW though. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
Hi, [EMAIL PROTECTED] schrieb: I have done what you told me to do, but nothing changed. Always the same problem. If I understand your dialplan right, your * is still calling itself via SIP, right? This is what is called a loop. You should review your dialplan and replace all dial(SIP/[EMAIL PROTECTED]) by goto(respective_context,exten,pri). Or are you trying to call SIP clients which are registered to the box? In that case you don't dial(SIP/[EMAIL PROTECTED]) but dial(SIP/accountname) while accountname is what stands in [] for that client in your sip.conf. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
Hi, [EMAIL PROTECTED] schrieb: In fact, after entering in Asterisk for the first time, my call is redirected to an other component of my system. This other equiment redirect the same call to Asterisk a second time. Hm, I suppose, your equipment is using reinvites for that redirection. The only idea to solve this I can think of would be having your equipment stay in the media path, i.e. making that redirection a brand new call. Then * shouldn't complain. But in my opinion that would be pretty ugly by means of scalability and ressources. Maybe a redesign of your callflow in general would be a better option. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.0-rc6 - SIP hold logic broken?
Hi, I have the following symptoms: Call X-lite / Nokia E51 X-lite press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH Call X-lite / SCCP phone MOH works as supposed Call SCCP phone / Nokia E51 SCCP press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH In addition, the BLF on the SCCP phones does NOT show the hinted SIP extension on hold. With 1.4 latest, everything worked as supposed. As this problem appears also between SIP clients, it is NOT a chan_sccp-related issue. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
Hi, [EMAIL PROTECTED] schrieb: I have a SIP request which comes from an Asterisk and which has to re-enter in the same Asterisk (during the same session), but during the second passage in Asterisk, it send me a 482 Loop Detected. So is it a bug or Asterisk control the session and considere it as a loop ? If it is not a bug, how could I resolve this problem ? Handle it in the dialplan logic. You don't want loops. Loops need- and senselessly burn ressources. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP request send me 482 error
[EMAIL PROTECTED] schrieb: Thanks for help, but I don't understand what you say. How is it possible to handle the error in the dialplan if my request return a 482 after entering Asterisk, but before accessing the dialplan ? Ok :). I meant, you should handle the whole thing in the dialplan without creating a loop. A loop is when a request is originating from the same PBX as it is directed to. Example: Your Asterisk is at IP 192.168.1.1. You have a phone context and an IVR context [phones] exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]) ;Loop, VERY BAD! exten = 4321,1,Goto(ivr_context,s,1) ;This is how it should be [ivr_context] exten = s,1,Background (welcome) ... Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SVN 1.6.0 / current does not compile
[CC] chan_agent.c - chan_agent.o chan_agent.c: In function ‘unload_module’: chan_agent.c:2496: error: void value not ignored as it ought to be make[1]: *** [chan_agent.o] Error 1 make: *** [channels] Error 2 Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Specific SIP answers on incoming calls?
Hi, when I still had ISDN, I was using Hangup(causecode) to send e.g. Wrong number to unwelcome callers. Meanwhile, I am only using SIP providers (no PSTN lines any more) and I would like to do similar, i.e. send specific SIP headers. Besides wrong number, I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to send a specific reply out of the dialplan? Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restrict SIP registration to one ip address only?
Remco Barendse schrieb: Suprising that this feature isn't used much, i would suspect that many asterisk installations (including mine) have very simple (short) extension numbers which makes brute forcing them rather easy. Extension numbers and SIP account basically have nothing to do with each other. If you name your SIP accounts after the respective extension number, you have a security issue in your design which you should solve first! A SIP peer definition can be like [Remcossoftclientathislaptop] type=friend secret=verysecretpassword ... And then in the diaplan you just do something like [internalcontext] exten = 10,1,Dial(SIP/Remcossoftclientathislaptop,30) exten = 10,2,Hangup() ... So, the username for you SIP client would be Remcossoftclientathislaptop while the dialled extension would be 10. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
Barton Fisher schrieb: It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP aware. Apparently, this current firmware/programming is not, one way audio problems. Is there a version that support VoIP directly for this router? Do you have firewall feature set? Then you could simply activate the SIP protocol inspection. Without firewall feature set, I guess, it's impossible. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Cisco 1841 - Can it be made SIP aware?
Kristian Kielhofner schrieb: IMNSHO, the less SIP aware the better... I have to disable SIP inspection on every IOS/PIX device I come across. Fix the one-way audio problems on your proxy, registrar, etc (in the case, Asterisk). Most SIP ALGs are broken. Interesting. I have my Asterisk with RFC-1918 IPs behid a NATting PIX and the FIXUP SIP of the PIX makes it very easy for me to use my * as server for external clients as well as as client for SIP providers. The PIX nicely replaces the RFC-1918 IP in the SIP-traffic with the current (dynamic) public IP of itself and keeps track of the RTP traffic. Actually, it also chages the ports in the RTP negotiation and then automatically forward the RTP traffic to the ports, the * was offering. Very very convenient. If the IOS firewall in the newer routers make problems, maybe I should not change to an ISR as I planned :). Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which internet phone protocol best to choose
No worries - I forgot a smiley. I didn't mean to appear annoyed or otherwise negative. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which internet phone protocol best to choose
http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which internet phone protocol best to choose
Alex Balashov schrieb: The short answer is SIP. Maybe not behind a firewall which you don't have control over. IAX is a single-port-protocol and as such much less problematic with firewalls and NAT. Read the second link in my previous mail. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which internet phone protocol best to choose
Julien Claassen schrieb: IAX I can basically understand, although I wasn't aware in the slightest, that other standard softphones supported it. They don't. Well - it depends, what you see as standard. There are very good multi-platform combined SIP/IAX clients like Zoiper. But Zoiper is not as popular as X-Lite because it wasn't adopted by lots of providers yet. In short: I am up for the longer answer. :-) My short answer contained links to pretty long explanations and a list of clients. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN or BRIstuff ...
Gordon Henderson schrieb: So comments, ponderings or anecdotes, etc. ... ? Bristuff worked perfectly fine for me for about 5 years. HOWEVER, you should keep in mind, that bristuff are very extensive patches against the zaptel dirvers and also against the core. So regarding updates you are totally depending on Junghanns. That means especially that there can be noticeable delay after an update of asterisk until the bristuff update is available. This *can* be bad if the update is a security update. Technically, I had very little problems. Even using a HFC-S card as internal ISDN bus worked like a charm. I just went to full IP because SIP trunks are cheaper in Finland than ISDN connections. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP TLS / Nokia E51
Hi, did anybody get SIP TLS working with E51? If I enable security in the phone's SIP config, the E51 attempts a REGISTER via 5060 UDP with method TLS, digest. My asterisk (latest SVN) just answers 401 UNAUTHORIZED. Is there some comprehensive howto for configuring SIP TLS? Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Automatic call to voicemail on login?
Hi, I would like to arrange that when an IAX client logs in / registers with my * AND there are unread voicemails, this IAX client will be automatically called and connected to the respective voicemail box. One possibility is to have a cronjob that creates a callfile - let's say - every five minutes which checks ChanIsAvail and connect to the voicebox if new messages are there. But with lots of IAX clients, this does not exactly scale very well. If there any other native way to execute an action on login or logout of a client? Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5 min limitation on phone calls! how to!
RoLaNd RoLaNd schrieb: Hello all! my last month's phone bill sky rocketed after i setup asterisk with softphones all over the house! could someone help me set up a limitation for my wife and kids not to be able to talk for more than 5 min at a time! or like 20 min per week! or whtever limitation i could set for this! Set(TIMEOUT(absolute)=seconds) http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 SVN / dahdi / meetme / - unable to open pseudo device
Hi, I was switching from zaptel to dahdi and got latest SVN from everything. Compiling works fine. kernel module dahdi_dummy is loaded. /dev/dahdi/pseudo exists Trying to go into a meetme does not work: [Aug 11 14:04:45] -- Executing [EMAIL PROTECTED]:1] MeetMe(SCCP/6000-0001, 444|dcIM) in new stack [Aug 11 14:04:45] WARNING[4184]: app_meetme.c:775 build_conf: Unable to open pseudo device [Aug 11 14:04:45] -- SCCP/6000-0001 Playing 'conf-invalid' (language 'en') [Aug 11 14:04:49] == Spawn extension (client_int_sgmobile, 8001, 1) exited non-zero on 'SCCP/6000-0001' Asterisk SVN-branch-1.4-r137138 Funnily, 1.6-trunk works with that dahdi version... Any ideas? Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 SVN / dahdi / meetme / - unable to open pseudo device
Kevin P. Fleming schrieb: Fixed in revision 137188; this module apparently did not get any DAHDI conversion work at all, but I don't know how it got missed. Thanks for the testing! Confirmed. Works fine now under all (extensively) tested conditions. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 encryption - LAN. no, INET: yes???
Russell Bryant schrieb: You'd have to provide a packet capture to see exactly what is happening. It sounds like on the call leg between your client and Asterisk, it isn't offering encryption as a capability, so it doesn't get used. However, when your friend calls you, and Asterisk makes a call out to your client, it offers encryption, and your client accepts it. Hm, not sure if I get your point. This is the infrastructure (exempt): Zoiper --LAN-- Asterisk --INET-- Zoiper (my) | (friend) | Cisco phone When I dial the Cisco phone from my Zoiper, wireshark shows unencrypted packets. When my friend calls the Cisco phone from her Zoiper, wireshark shows unknown = encrypted(?) packets. We are both using the same Zoiper release, just she on MAC and I on Windows PC. I also now tested to make a call from the Cisco phone to my Zoiper - also no encryption. Would it make sense to introduce a parameter forceencryption=yes per peer in iax.conf? In sensitive environments, people want to be certain that a call is encrypted. They probably rather want a call to fail than have a call that might be unencrypted without knowing it. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile: scrambled audio, no MOH, no call signalization
This is how it sounds: http://stefan.gofferje.net/chan_mobile_distorted.wav Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 encryption - LAN. no, INET: yes???
Russell Bryant schrieb: Interesting. Here are a couple more sanity checks you can do. First, double check to ensure that your entry in iax.conf has encryption=yes set. Also, when you make the call into Asterisk, set the verbose setting up a bit. You should see output from chan_iax2 which indicates what peer you are authenticating as. Make sure that the call is matching the entry that you think it is. I will do some more testing as you suggested. Also, is there any encryption option in Zoiper that you have to enable? Not to my knowledge. I will send an issue report to asteriskguru also. Would it make sense to introduce a parameter forceencryption=yes per peer in iax.conf? In sensitive environments, people want to be certain that a call is encrypted. They probably rather want a call to fail than have a call that might be unencrypted without knowing it. That is a good suggestion. Opened a bug for that (0013285) :). Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-addons-1.6.0-beta4 compile error
Hi, addons 1.6 don't compile here. Any ideas? Terve, Stefan [EMAIL PROTECTED]:/usr/src/asterisk-addons-1.6.0-beta4 make CC=gcc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent makeopts make[1]: Entering directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect' make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect' make[1]: Entering directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect' make[1]: `makeopts' is up to date. make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect' CC=gcc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent make[1]: Entering directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect' make[2]: Entering directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect' gcc -g -c -D_GNU_SOURCE -Wall -c -o menuselect.o menuselect.c gcc -g -c -D_GNU_SOURCE -Wall -c -o strcompat.o strcompat.c gcc -g -c -D_GNU_SOURCE -Wall-c -o menuselect_curses.o menuselect_curses.c make[3]: Entering directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect/mxml' gcc -O -Wall -c mxml-attr.c gcc -O -Wall -c mxml-entity.c gcc -O -Wall -c mxml-file.c gcc -O -Wall -c mxml-index.c gcc -O -Wall -c mxml-node.c gcc -O -Wall -c mxml-search.c gcc -O -Wall -c mxml-set.c gcc -O -Wall -c mxml-private.c gcc -O -Wall -c mxml-string.c /bin/rm -f libmxml.a /usr/bin/ar crvs libmxml.a mxml-attr.o mxml-entity.o mxml-file.o mxml-index.o mxml-node.o mxml-search.o mxml-set.o mxml-private.o mxml-string.o a - mxml-attr.o a - mxml-entity.o a - mxml-file.o a - mxml-index.o a - mxml-node.o a - mxml-search.o a - mxml-set.o a - mxml-private.o a - mxml-string.o ranlib libmxml.a make[3]: Leaving directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect/mxml' gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a -lncurses make[2]: Leaving directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect' make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.0-beta4/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Entering directory `/usr/src/asterisk-addons-1.6.0-beta4' make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.0-beta4' make[1]: Entering directory `/usr/src/asterisk-addons-1.6.0-beta4' make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.0-beta4' make[1]: Entering directory `/usr/src/asterisk-addons-1.6.0-beta4' make[1]: Leaving directory `/usr/src/asterisk-addons-1.6.0-beta4' make[1]: Entering directory `/usr/src/asterisk-addons-1.6.0-beta4/channels' [CC] chan_mobile.c - chan_mobile.o chan_mobile.c:177: warning: ‘struct ast_cli_args’ declared inside parameter list chan_mobile.c:177: warning: its scope is only this definition or declaration, which is probably not what you want chan_mobile.c:178: warning: ‘struct ast_cli_args’ declared inside parameter list chan_mobile.c:179: warning: ‘struct ast_cli_args’ declared inside parameter list chan_mobile.c:182: error: initializer element is not constant chan_mobile.c:182: error: (near initialization for ‘mbl_cli[0].cmda[0]’) chan_mobile.c:183: error: initializer element is not constant chan_mobile.c:183: error: (near initialization for ‘mbl_cli[0].cmda[1]’) chan_mobile.c:184: error: initializer element is not constant chan_mobile.c:184: error: (near initialization for ‘mbl_cli[0].cmda[2]’) chan_mobile.c:247: warning: ‘struct ast_cli_args’ declared inside parameter list chan_mobile.c:248: error: conflicting types for ‘handle_cli_mobile_show_devices’ chan_mobile.c:177: error: previous declaration of ‘handle_cli_mobile_show_devices’ was here chan_mobile.c: In function ‘handle_cli_mobile_show_devices’: chan_mobile.c:256: error: ‘CLI_INIT’ undeclared (first use in this function) chan_mobile.c:256: error: (Each undeclared identifier is reported only once chan_mobile.c:256: error: for each function it appears in.) chan_mobile.c:257: error: ‘struct ast_cli_entry’ has no member named ‘command’ chan_mobile.c:262: error: ‘CLI_GENERATE’ undeclared (first use in this function) chan_mobile.c:266: error: dereferencing pointer to incomplete type chan_mobile.c:266: error: request for member ‘argc’ in something not a structure or union chan_mobile.c:266: warning: comparison between pointer and integer chan_mobile.c:267: error: ‘CLI_SHOWUSAGE’ undeclared (first use in this function) chan_mobile.c:267: warning: return from incompatible pointer type chan_mobile.c:269: error: dereferencing pointer to incomplete type chan_mobile.c:269: error: request for member ‘fd’ in something not a structure or union chan_mobile.c:269: warning: passing argument 1 of ‘ast_cli’ makes integer from pointer without a cast chan_mobile.c:273: error: dereferencing pointer to incomplete type chan_mobile.c:273: error: request for member ‘fd’ in something not a structure or union chan_mobile.c:275: warning: passing argument 1 of ‘ast_cli’ makes integer from pointer without
Re: [asterisk-users] asterisk-addons-1.6.0-beta4 compile error
Hi, Russell Bryant schrieb: It looks like you're trying to compiled Asterisk-addons 1.6 against Asterisk 1.4. You will need to install Asterisk 1.6 before you can compile and install Asterisk-addons 1.6. So, 1.6 must be _installed_ before compiling addons? It's not enough to have it readily compiled in the neighbour dir? I'll try that, thx. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-addons-1.6.0-beta4 compile error
Stefan Gofferje schrieb: So, 1.6 must be _installed_ before compiling addons? It's not enough to have it readily compiled in the neighbour dir? Confirmed - works. Thank you! Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-addons-1.6.0-beta4 compile error
Russell Bryant schrieb: Stefan Gofferje wrote: So, 1.6 must be _installed_ before compiling addons? It's not enough to have it readily compiled in the neighbour dir? That is correct, at least for the easy case. Alternatively, you can specify the Asterisk location as an argument to the configure script. -addons-1.6.0$ ./configure --with-asterisk=/path/to/asterisk-1.6.0 However, as Tzafrir noted in another reply, it is worth mentioning that regardless of which method you use, Asterisk-addons 1.6.0 modules _must_ be used with Asterisk 1.6.0. Yes, of course! I am running 1.4.21.2 and wanted to compile 1.6 comletely before shutting down the * and installing the new stuff. Reduce downtime. And it was a good idea as current chan-sccp-b trunk does not compile with 1.6 as it turned out. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile: scrambled audio, no MOH, no call signalization
Hi, I started testing chan_mobile. Target is having some old phone with a duosim (second card with same number) put to silent somewhere in the rack with the *. That phone should mainly take incoming calls and after 45secs put them to the mailbox AND permit me to talk via my nice Cisco desktop phones. I'm using latest trunk of everything. First I tried with a 6310i. Outgoing calls had badly scrambled audio and incoming calls were not signalled. Now I have a Sony Ericsson Z600. The first outgoing call had scrambled audio. After an incoming call with clear audio, the following calls had also good audio. I remember I had a similar problem when I wanted to use my Win PC as a headset. Maybe it's in the dongle. I use a CSR chipset BT dongle. [EMAIL PROTECTED]:~ hwinfo --bluetooth 06: USB 00.0: 11500 Bluetooth Device [Created at usb.122] UDI: /org/freedesktop/Hal/devices/usb_device___noserial Unique ID: FKGF.nQKjiuCfL84 Parent ID: pBe4.T_tl6i7A1LE SysFS ID: /devices/pci:00/:00:10.1/usb2/2-1/2-1:1.0 SysFS BusID: 2-1:1.0 Hardware Class: bluetooth Model: Cambridge Silicon Radio Bluetooth Dongle (HCI mode) Hotplug: USB Vendor: usb 0x0a12 Cambridge Silicon Radio, Ltd Device: usb 0x0001 Bluetooth Dongle (HCI mode) Revision: 5.25 Driver: hci_usb Driver Modules: hci_usb Speed: 12 Mbps Module Alias: usb:v0A12p0001d0525dcE0dsc01dp01icE0isc01ip01 Driver Info #0: Driver Status: hci_usb is active Driver Activation Cmd: modprobe hci_usb Config Status: cfg=new, avail=yes, need=no, active=unknown Attached to: #2 (Hub) [EMAIL PROTECTED]:~ I also noticed that there is no MOH at all (also not started according to CLI) when a local phone holds the chan_mobile call. This is valid for SCCP, SIP and IAX clients. I would be interested in further testing and bugsearching as chan_mobile could save me from the need of a landline :). Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 encryption - LAN. no, INET: yes???
Hi, I have configured all IAX clients with encryption. I use Zoiper as a softphone. When I make a call in the LAN from desktop-PC to *, the call is - according to wireshark not encrypted. Wireshark identifies the packets as normal G.711 mu-law packets. However, * reports the client as encrypted: k-tanco*CLI iax2 show peers Name/UsernameHost Mask Port Status sgofferj RFC-1918 IP(D) 255.255.255.255 4570 (E) OK (2 ms) Funnily, if my friend calls me from internet - also with Zoiper - Wireshark cannot identify the packets so I conclude, the call is encrypted. Does this make any sense? Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP TLS error: ast_make_file_from_fd: FILE * open failed
That does not make too much sense to me... Configuration should be ok... [Aug 8 23:30:13] SSL certificate ok [Aug 8 23:30:13] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Aug 8 23:30:13] WARNING[23835]: tcptls.c:463 ast_make_file_from_fd: FILE * open failed! Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Action on login
Hi, is there meanwhile the possibility for some actions besides dialling in *? Namely, I would like that if a remote IAX or SIP user logs in AND there are new messages, they automatically get a call and be connected to the voicemail. The only method I know by now is make a context in the dialplan, checking if the user has logged in and then initiate the call. And of course firing a callfile to every x minutes to that context for each remote user. That does not scale very well. It would be much nicer to have some kind of login / logout action parameter in sip.conf or so. --Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Autoanswer in Nokia SIP clients?
Olivier schrieb: Well, if I'm in a GSM call, the Nokia SIP client anyway signals busy to a SIP call. Is it certain ? Yes, just tested it myself. Phone answers with Busy here if in a GSM call. From my understanding, Symbian applications MUST leave this decision type to an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. Well, there are Symbian application implementing a local answering machine on the phone. There even is a spy application which autoanswers a call from a specific number without any indication and rejects all other calls, so it must be possible for a Symbian app to autoanswer a call, even from GSM. I just don't see the point why it shouldn't be possible for a SIP client to autoanswer a call instead of waiting for the green button, given that the phone is not in a GSM call. -S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Autoanswer in Nokia SIP clients?
Olivier schrieb: As a dual GSM/WiFi mode phone might be already engaged in a GSM conversion while a SIP call occurs, I think Symbian application dev rules would impose any application to centralize microphone and speaker allocation to a Symbian provided resource manager. So I think automatic answer of any kind are currently not supported by this Symbian provided resource manager. Well, if I'm in a GSM call, the Nokia SIP client anyway signals busy to a SIP call. It could do the same to a SIP call with autoanswer request... It's just a question if you have to press the green button or not on an incoming SIP call... -S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Autoanswer in Nokia SIP clients?
Olivier schrieb: Which company publishes this Symbian application implementing a local answering machine on the phone, for instance ? There are several. For instance, rock your mobile comes to my mind. http://www.rock-your-mobile.com/ http://www.rock-your-mobile.com/answering-machine.php -S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Autoanswer in Nokia SIP clients?
Hi, anybody knows if it is possible to make the Nokia SIP client in the phones autoanswer a call in speakerphone mode? --Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push to talk over cellular with asterisk (was: Autoanswer in Nokia SIP clients?)
Gordon Henderson schrieb: On Mon, 4 Aug 2008, Patrick wrote: Hi Stefan, Stefan Gofferje wrote: Hi, anybody knows if it is possible to make the Nokia SIP client in the phones autoanswer a call in speakerphone mode? I looked into this for my N95 but if it's possible then it isn't documented. At least I could not find any public documentation how to do it. I did not dig into the Symbian developer docs. Maybe those contain the answer. Similarly for my E90. There's nothing obvious in it that'll make it auto-answer. I was hoping that Nokia was sloppy and tried sending SIP INVITES with PoC headers but the phone was ignoring them or answering with a unsupported media type or similar. Unfortunately, I also wasn't able to register the PoC application with my asterisk. Anybody knows if somebody works on an PoC / push to talk over cellular implementation for *? I studied the 3GPP / Nokia drafts about 2 or 3 years ago. It doesn't look too complicated but I'm not exactly a gifted programmer :). --Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push to talk over cellular with asterisk
Gordon Henderson schrieb: Seriously - Push to talk - Half duplex Communications. How ancient is that! It's really reminiscent of ancient US style truckers - Smokey and the Bandit and all that. That's just so last century. Lets put all that behind us and get with the 21st century! We all have hands-free, full duplex communications now, so lets just forget all that old rubbish and get on with the programe. Well, push to talk makes very much sense whereever you have to reach a group of recipients at once. Push to talk over cellular is quite popular among US law enforcement agencies e.g. When I was still living in Germany, I was heavily utilizing PTT to coordinate my service team. It's just practical. One talks, everybody listens. Radio simpleness but GSM reliabilty and quality. Besides that was the original reason for me looking for the Nokia autoanswer feature. After PTT didn't work with *, I thought of simply calling a list of phones with autoanswer and throwing them into a meetme. As I do with SCCP phones here now. If I really want PTT, then I'll go out buy a pair of Motorola handsets. Which would reach how far? 500m? Surely not from Helsinki to Oulu or even internationally... --Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No MOH on SIP hold nor on park
Hi, when I put a call on hold from my Nokia E51 (SIP client), the other side does NOT hear music on hold although sip debug / wireshark shows that the E51 tells the asterisk that it now holds the call. Canreinvite is set to no. Also, when parking a call (features.conf), the parked caller does not hear music on hold. In queues, when using # and when using the hold functions of my Cisco 7960 (SCCP), music on hold works without problems. I'm running Asterisk 1.4.21.1. IIRC, MOH on parked calls was working earlier but I didn't use the park functions extensively so I don't remember exactly when that was. Any ideas? --Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN
Hi folks, I'm currently trying to get some early audio, i.e. audio without a connection, to the caller to give some cost-free info while the alerting phase. The WIKI's info on the Progress() application says, that just Progress() before e.g. a Background(soundfile|n) should work but it doesn't. In fact, the call times out (The person you've called is temporarily not available) if there is no ringing indication for longer than 30 secs or so (long sound). As soon as the Dial application is called, the caller gets a normal ringing indication. Dial(...|m) causes the caller to hear nothing and the call to timeout. So, I guess, in addition to Progress(), * has to tell the PSTN that the call is proceeding AND that there is audio, i.e. inband information, available and it should be passed to the caller. I guess, the focus should be on the progress indication as the call timeout shows, that the PSTN isn't satisfied with just what Progress() sends. Does anyone got this Early-B3 working with BriStuff and HFC-S? Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r | chan_sccp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skinny.conf and sccp.conf
René Enskat [Teamware GmbH] schrieb: Hi, I want to try the skinny/sccp protocol. Somebody can give me a working config for a cisco 7960 or 7970 ip phone? Isit possible to forward a SIP extension to the skinny phones? Coz i use normally a sip phone and i only want to forward this calls to the skinny phone. http://chan-sccp.org/ -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 trunks encrypted?
Hi folks, I understand that IAX2 supports public key authentication. Is the transmission also encrypted or is it possible to encrypt an IAX2 trunk between 2 *s? Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SCCP support is making good progress
Hi folks, whoever owns a Cisco phone and is unhappy about slow firmware, incomplete XML support etc... should really have a look at Sergio Chersovani's rewrite of chan-sccp! It is in fact far away from any sccp-channel drivers we used to know just half a year ago. Sergio did a complete rewrite and meanwhile it's not only running stable with asterisk stable and CVS - it's also very feature reach and supports almost any feature, the real Callmanager has and a few more. But the greatest advantage is - one can use the native Skinny firmware for the Cisco phones. And almost any Cisco phone is supported. Phone support: Fully / almost fully: 7970, 7960, 7940, 7914, 7920, 7910, 7905, 7902, IP Communicator (new softphone) Basic / untested: 7936, 7935 Some features: Monitored speeddials (busy lamp field) on 7970, 7960, 7940, 7914, IPC Call forwarding Call waiting Do not disturb Park / Pickup Autoanswer (1-way or 2-way) per dial parameter Ringer control per variable Highly configurable Links: Official homepage: http://chan-sccp.berlios.de/ Unofficial homepage: http://chan-sccp.org/ Mailinglist: http://lists.berlios.de/mailman/listinfo/chan-sccp-users Webforum:http://forum.chan-sccp.org/ Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP support is making good progress
Chris Bagnall schrieb: whoever owns a Cisco phone and is unhappy about slow firmware, incomplete XML support etc... should really have a look at Sergio Chersovani's rewrite of chan-sccp! Is there a good resource out there for people who don't have a lot of experience with Cisco phones? I picked up a 7960 earlier this week to give potential clients an example of what they get when they spend a *lot* of money on IP phones, but I must confess I'm having a nightmare of a time trying to configure it. The main problem seem to be that I have nothing but a phone and a brief licence agreement/regulatory approval sheet, and nothing else. I've trawled through the numerous pages about these phones both on Cisco's website and on voip-info, but I'm still not really sure what files I need to have on the TFTP server to get the phone going in the first place, or find some up-to-date examples to work from. Even after that I'm not sure I'll be able to upgrade the firmware without a Cisco service agreement (from what I've read), which is ridiculous for a phone that's twice as expensive as many other enterprise IP phones. Any suggested reading others on the list have found helpful in this scenario? The list archives of chan-sccp-users provides a lot of information. www.voip-info.org also has. There are a number of ressources at cisco.com and if all this does not help, the people at chan-sccp-users or forum.chan-sccp.org use to friendly answer questions. There are also a number of people working at various howtos at the moment. Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP support is making good progress
Wayne schrieb: Stefan Gofferje wrote: Hi folks, whoever owns a Cisco phone and is unhappy about slow firmware, incomplete XML support etc... should really have a look at Sergio Chersovani's rewrite of chan-sccp! Ok - I'll give it a go :) - Just one problem... My phones have been converted to SIP - and I dont have skinny to put back! Can you even swap these things back again?! Well, if you have a SmartNET contract, you could download the Skinny image from cisco.com. If not, I'm afraid, you have to buy it from a Cisco reseller... Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP support is making good progress
Paul schrieb: Chris, I wrote a post that contains the information (files) you need for the asterisk tftpboot directory to load a 7960 Sip 7.5 image from the server. See this post https://sourceforge.net/forum/message.php?msg_id=3374221 As far as obtaining the SIP 7.5 software, I see it on EBay all the time for $12.00 buy it now. If you follow my directions about what files to place in the tftpboot directory, and modify your alternate boot server on the 7960, the phone will load the sip software. I personally learned by using SolarWinds TFTP server from my XP station so I could see the file name requests as they were being transferred. On a new Cisco phone, you would set the alternate tftp server to yes then set the ip address of the alternate tftp server to the server with the images to upload. If you need to unlock the phone it can be **# or cisco based on the version loaded. I also was interested in the SCCP driver. I was able to load the driver in Asterisk (AND SEE IT) but could not get it to work. I did ask for a users guide but there is not one available yet. So I think SIP will be the easiest for you to use for you client proposal. When time permits, I'll go back and look over the sccp driver and find my mistakes. It is not likely to find a legal SIP firmware for $12. Not the CD makes it legal but the license from Cisco and this is a little more expensive. Besides from this, Cisco's license agreement does not permit the license to be resold, so to have a legal license, you need to get it from a Cisco partner or at least authorized reseller. Anyway, compared to the SCCP image, the SIP image is very bad. Incomplete XML support, VERY slow, decreased sound quality... Getting chan_sccp from Sergio to work is really easy. The distro contains a well documented sample config and - as I wrote before - there are lots of info in the chan-sccp-users mailing list archive. I myself started with a SIPped 7960 but meanwhile, I run 3 7960s and 1 7905 without any problems. Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Skinny Firware
Bobby Lacey schrieb: Hello, I have just acquired my first 7960 from a business sale. It has already been preloaded with the 7.3 SIP image which works flawlessly with my Asterisk box. I want to experiment with chan_sccp and therefore I would need the skinny firmware = 7, I guess. Could someone tell me an outlet where I could purchase a Smartnet contract to download this firmware? I have been unable to find a retailer online that can help. Thanks in advance for any help. http://tools.cisco.com/WWChannels/LOCATR/jsp/partner_locator.jsp Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP support is making good progress
Chris Bagnall schrieb: Getting chan_sccp from Sergio to work is really easy. The distro contains a well documented sample config and - as I wrote before - there are lots of info in the chan-sccp-users mailing list archive. Yep, I've just tried chan_sccp with the 7960 I have here and it appears to work fine (original factory firmware, since it appears I'm going to have to find a service contract just to download the damned updated firmware which any sane manufacturer would provide with the product - unless anyone wants to contact me off-list - nudge, nudge, wink, wink) How does one go about making changes to the display on the phone? So far, my lower soft buttons hae labels like Pnbsp;, and apart from the single line I defined in sccp.conf I have an almost blank display. Is there a good guide to where to change various bits of the display (labels and so on)? The softbutton problem sounds like a really old firmware... Softbuttons are controller by the channel. They have functions like DND, newCall, EndCall, etc... The line buttons can be configured as speeddials which can use the asterisk hint system for status monitoring. The sample config shows how. I also suggest, you browse the mailinglist archives at http://lists.berlios.de/pipermail/chan-sccp-users/ Probably a stupid question, but does this thing have a backlight? 7960? Nope... 7970 has. Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Forward - 7940 Asterisk - Help
c waddy schrieb: I am looking for a simple way to forward calls unconditionally with Asterisk. We are running an Asterisk system with 10 extensions using SIP. One of our users leaves the office regulary, when she is out, she needs to be able to forward unconditionally to her mobile or collegue. I am trying to keep it as simple as possible, we use Cisco 7940's, they have a call forward option, when she uses it, all our incoming calls go to her mobile? Not just the calls to her extension. My Question: Does Call Forward on the Cisco Phones and Asterisk work? If so do I need to implement something into the dial plan. I have read on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding Is there an updated way to do this? I thought *21* was hard coded into Asterisk? If the Cisco phones wont work, i would like her to simply dial *21*mobile number#, any suggestions on this? AFAIK, the CFWDALL option of the SIP fw send a temporarily moved message back to the caller with the new address to call. This can fail for a lot of reasons. I would recommend using 7940s with the native Skinny firmware and chan_sccp by Sergio Chersovani. Chan_sccp not only supports call forwarding but a lot more and only with the Skinny image, you can use all the features, the phone have. You can read a bit more at http://chan-sccp.org/ Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Forward - 7940 Asterisk - Help
Hi Derek Derek Conniffe schrieb: Hi Stefan, Has chan_sccp matured a lot? I remember (maybe a year ago or so) that I had a lot of problems with chan_sccp and chan_skinny (one thing is that I remember with chan_sccp is that the VM button didn't work and trying to answer multiple incoming calls tended to make the phone go into a weird state where I had to power cycle it to get it back right again). I upgraded the phoen to SIP and never looked back - but then maybe I'm not getting all the fetures like call forwarding (which sounds very useful to me).. I am talking about a complete rewrite. About half a year ago, Sergio was writing patches for Julien's chan_sccp (chan-sccp.sf.net). For some reason, Sergio decided to do a project split and started his own chan_sccp. After a few weeks he stopped patching Julien's work and did a complete rewrite. Meanwhile, chan_sccp (chan-sccp.berlios.de or chan-sccp.org) is not only stable but also very feature reach! Sergio does the coding and me and some other people do very heavy testing and the project is making great progress. Highlights are: - Line status monitoring on 7960/7914 (you see what other - not only SCCP - extensions are doing) - Good hardware support (7940/7960/7914 almost 100%, 7905 almost 90%, 7920 almost 80%, 7970 currently under heavy development) - Support for call waiting, call forwarding, ... - Intercom / autoanswer - controlled by dial application and a lot more... Have a look at http://chan-sccp.org/ ... Slan go foil, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI voice one way only
Hi, Klemens Kasemaa schrieb: hi PSTN -- [Teles ISDN / Asterisk] -- SIP client When call is made through ISDN, no matter if taken from PSTN or Asterisk side, person in PSTN side can hear perfectly but in Asterisk side I only hear a very scrambled or very low quality voice, words repeated several times. Same is with echo test (call taken from PSTN) Get a CAPI module for your Teles and try chan_capi-cm from http://sourceforge.net/projects/chan-capi/ Regards, Stefan -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sccp help
Hi, On 9:04:57 August 19, 2005 stevanus [EMAIL PROTECTED] wrote: Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : I've tried different versions of chan_sccp, yet the result were still the same. Which version of chan_sccp did you use? Sourceforge or Berlios? There is a new fork of chan_sccp by Sergio Chersovani who started work some weeks ago and did an almost complete rewrite of the channel. This version supports a lot more features on various phones and has a lot less bugs. You could find it at chan-sccp.berlios.de (official site) or chan-sccp.org (unofficial site). There is a related mailinglist at berlios.de where Sergio does a hell of a lot of support (unless he is one vacation like at the moment :-) ) and gladly accepts bug reports :-). Regards, Stefan -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sccp help
On 10:10:54 August 19, 2005 stevanus [EMAIL PROTECTED] wrote: Hi, I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp. Jep, it is... If you had problems with this, your chance for a solution is higher at the chan-sccp-users list... :-) Regards, Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_skinny issue
Mark Johnson schrieb: Jason wrote: Hey all, I have set up my cisco 30vip using chan_skinny because chan_sccp wont register. The problem I am having is, everytime a call is sent to the phone Skinny/[EMAIL PROTECTED] it rings once, then asterisk segfaults. Man... Use chan_sccp from Sergio at: ftp://ftp.berlios.de/pub/chan-sccp/ He is the most helpful person I've ever met. If you find a bug, report it to him, and it's usually fixed by the next day!! I don't have the same phone, but I've used 7910/40/60 with sccp and it works! ...and he is on vacation at the moment to recover from all the hot bug fixing :-) Regards, Stefan -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MISDN callerid
Michiel van Baak schrieb: I have a cologne chip card which is connected directly to the ntba. Outgoing and incoming calls work fine, but incoming calls from ntba have the wrong callerid (first 0 is missing). I'm using current jolly misdn drivers and chan_misdn-14_04_05 with asterisk stable. I think this is normal behavior for ISDN lines. My guess it's the telco not sending the 0. The callerid is tranmitted completely different. Adding zeros or plusses is a matter of the equipment (phone). I have the same thing on AVM Fritz with chan_capi and with the QuadBRI with qozap and chan_zap. I simply add the 0 to national numbers (based on length of numbers without leading 0) or a + to international numbers. This fixed all later lookup functionality in my dialplan. For chan_capi - capi.conf: [general] nationalprefix=0 internationalprefix=00 For HFC-cards (bristuff / ZAPHFC) - zapata.conf: [channels] pridialplan=local prilocaldialplan=local For MISDN: no idea... look into the docs or use ZAPHFC :-). Changing the CID in the dialplan is really ugly :-). Besides, Michiel, I strongly recommend chan_capi-cm! It has more functions than chan_capi. You can e.g. send any hangupcause you want back to the network e.g. 021 call rejected :-) ... chan_capi-cm - http://sourceforge.net/projects/chan-capi/ cause codes - http://www.telos-systems.com/?/techtalk/cause.htm -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_skinny issue turned to chan_sccp issue.
Jason schrieb: Couple questions since I finally got chan_sccp to work. 1. Has anyone successfully gotten it to work behind NAT, i called my isp and asked them for 35 more ip's and they laughed at me + the cost of getting them would far outdo the benifits. and 2. can someone show me a config with a multi phone configuration 1.) No NAT... SCCP is NOT nat-capable. But it works fine with VPN. You anyway don't want to talk plain SCCP/RTP via the internet without encryption. Real-time splitting and copying a RTP-stream is really easy going! 2.) Multi-phone sccp.conf or extensions.conf? Here's a sccp.conf: [general] keepalive = 30 ; IMPORTANT: 5secs. lead to trouble with ; 7960 context = internal dateFormat = D.M.Y ; M-D-Y in any order (5 chars max) bindaddr = 192.168.1.200 ; asterisk box. port = 2000 ; listen on port 2000 (Skinny, default) debug = 0 [devices] type= 7905 description = Bedroom tzoffset= 0 autologin = 6004 device = SEP type= 7960 description = Office tzoffset= 0 autologin = 6000 speeddial = 6001,6001,[EMAIL PROTECTED] speeddial = 6004,6004,[EMAIL PROTECTED] device = SEP type= 7960 description = LivingRoom tzoffset= 0 autologin = 6001 speeddial = 6000,6000,[EMAIL PROTECTED] speeddial = 6004,6004,[EMAIL PROTECTED] device = SEP [lines] id= 6000 pin = 1234 label = 6000 description = Office context = client_int_unrestricted callwaiting = 1 incominglimit = 2 mailbox = 1000 vmnum = 8500 cid_name = Office cid_num = 6000 line = 6000 id= 6001 pin = 1234 label = 6001 description = LivingRoom context = client_int_unrestricted callwaiting = 1 incominglimit = 2 mailbox = 1000 vmnum = 8500 cid_name = Living Room cis_num = 6001 line = 6001 id= 6004 pin = 1234 label = 6004 description = Bedroom context = client_int_unrestricted callwaiting = 1 incominglimit = 2 mailbox = 1000 vmnum = 8500 cid_name = Bedroom cid_num = 6004 line = 6004 Regards, Stefan -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID Info from Cisco router to Asterisk
Dear Asterisk Gurus: Are Cisco routers able to pass caller ID information (from PRI T1) to Asterisk when using SIP? I've done some reasonable searching of the archives and the wiki. I've found some good examples of Cisco configurations, but most examples relate to FXO ports (and most of the FXO ports are of the variety that do not support caller ID). I was not able to find a definitive answer to this question when using PRI for inbound calls. Well, I'm not an asterisk Guru but I'm fairly good with Cisco stuff... If you have a CCM (Callmanager), you also use a router as gateway. As the IP-phones wich are managed by the CCM DO display incoming caller-id, I suppose, the router is capable of relaying that information. However, the connection between the router and the CCM is usually realized with h.323. When you take those information and look at them logically, I think, you have quite a good chance that you will get your caller-ids. Either the router relays them via SIP or you take one of the h.323 channels for asterisk. Anyway, my question is, why don't you use a T1-card for your asterisk. I suppose, a T1-card from Digium is MUCH less expensive than a big Cisco with a Cisco T1-card... Regards, Stefan -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco 7920
On 15:15:19 July 04, 2005 [EMAIL PROTECTED] wrote: Joseph [EMAIL PROTECTED] : Would Creating files mentioned on the document be enough for configuration? You will need to get the sccp channel software here: http://chan-sccp.berlios.de/ Download it and follow the instructions included. Please keep in mind that is under development. The code is not at all tested for a production server. It will soon. Expect it to crash, random hangups, etc. btw it is working :-) You are far to decent, Sergio! chan-sccp from Sergio is working fine and undergoing heavy stress testing here. Until now, I just found smaller inconsistencies, no real nasty bugs. Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Colored asterisk -R?
Hi folks, when I start asterisk directly, I get a colored CLI. When connect to a already running asterisk with asterisk -R, it's never colored, despite I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Colored asterisk -R?
[EMAIL PROTECTED] schrieb: Asterisk -gc I don't see a -R in that... Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI and Caller ID name not showing.
On 15:54:12 June 29, 2005 Armin Schindler [EMAIL PROTECTED] wrote: On Wed, 29 Jun 2005, louis g wrote: I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4 port Eicon Diva card. All works fine, but i'd like calls from the PBX to Asterisk to show the Caller ID name and not just the number. I know this information is being presented by looking through the ISDN trace for the Eicon Card. Asterisk trace show dialparties.agi: Caller ID name is '605' number is '605'. Can anyone point me in the right direction to get this sorted?. It's works with X100P cards :) What 'name' do you mean? Is it a subaddress? Please paste an example for that Eicon card trace where you see that name. His PBX probably transmits the name per UUS1. zaphfc supports this also. I have a zaphfc card as internal ISDN and connected a Siemens ISDN DECT phone to it. Now, on incoming calls, the Siemens shows the CallerIDName as set by Asterisk in the display. zaphfc also supports SendText... Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Play an announcement to the CALLING party
Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So, to avoid the employees forgetting it, an automatic announcement should be played. Besides, same rules are applicable for calls that may be recorded for quality assurance issues. At least for premium rate calls, queues won't work as the customer would strongly dislike hearing an announcement about the rate while waiting for an agent. The a() option of the dial app only works for CALLED parties and when trying to use a macro with the m() option, the Playback also goes to the called party. Anyone any hints on that? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement
On 11:29:31 June 27, 2005 Armin Schindler [EMAIL PROTECTED] wrote: I did some more testing.. On incoming calls, the caller hears the called party very much chopped. On outgoing calls, the called party hears nothing. Using *1.0.7 bristuffed... I cannot reproduce this here. Does it happen with plain 1.0.7 as well Don't know. My * is kinda productive, so I am bound to BRIstuff for use of zapHFC for internal phones. Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement
On 11:28:09 June 27, 2005 Armin Schindler [EMAIL PROTECTED] wrote: Yes, maybe you would like to implement some of those feature-requests ? ;-) I would love to if I were a bright programmer :-). --Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement
Armin Schindler schrieb: On Sat, 25 Jun 2005, Stefan Gofferje wrote: Armin Schindler schrieb: I have added busy()/congestion() support to CVS HEAD now, can you please test if it works for you? Works perfectly well! Also CallingPres(32) does work! The only thing I wonder about is a delay. I know there is a delay. Currently the calling party is 'alerted' every time by default and this is not correct for calls which shall not be accepted. I noted a number of bugs and a feature-request at SF :-). Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
