[asterisk-users] How to count Parked calls?
Hey All, I am working on a SIP Call bridging application. Every time I receive a incoming call in Asteriskserver1 my AGI should alert to AsteriskServer2 and AsteriskServer2 should callback to AsteriskServer1 and call should be bridged on specified extension. (making call in this way is customer requirement) Every time I receive a call in AsteriskServer1, I Park the call and through AGI, AsteriskServer2 callback to AsteriskServer1 with parked extension. My actual problem is, I can't maintain the record of Parked calls, View Parked Calls in dialplan. Is there any way to count or track the ParkedCalls() in the dialplan?? Through Asterisk CLI I can see the parked calls but I need to count the calls in dialplan. Muhamamd Faheem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help in oh323 Gatekeeper
Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. WARNING[8446]: chan_oh323.c:3555 cleanup_h323_connection: Call 'ip$192.168.8.96:30005/27890-f5194af7' not found (clear). Please any one give suggestions to disable the gatekeeper access in Asterisk... Thanks in Advance... Regards, Velusamy.K ___ -- 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] Help in oh323 Gatekeeper
On Tue, Jul 14, 2009 at 02:10:47PM +0530, velusamy velu wrote: Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. WARNING[8446]: chan_oh323.c:3555 cleanup_h323_connection: Call 'ip$192.168.8.96:30005/27890-f5194af7' not found (clear). What version of Asterisk is it? Any chance of using the built-in chan_h323? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and several clients behind NAT
Is it possible to have several clients behind NAT to register to an Asterisk-server with a public IP-address ? When Asterisk receives an incoming call, how will it know @ which private IP-address the client is reachable ? I guess it is impossible for Asterisk to directly contact the private client behind the NAT ?! Or to distinguish between the private clients ?! Is there an easy solution to this ? How does hosted IP-PBX services work then ?! Jonas. ___ -- 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 and several clients behind NAT
jonas kellens wrote: Is it possible to have several clients behind NAT to register to an Asterisk-server with a public IP-address ? When Asterisk receives an incoming call, how will it know @ which private IP-address the client is reachable ? I guess it is impossible for Asterisk to directly contact the private client behind the NAT ?! Or to distinguish between the private clients ?! Is there an easy solution to this ? How does hosted IP-PBX services work then ?! Yes, this problem has a solution. The NAT gateway creates a UDP state mapping between internal source ports and external source (and destination, since most user agents are symmetrical nowadays) ports. The NAT gateway then allocates different external UDP ports for different connections being tracked in this manner. Consider, for example, two phones - 192.168.1.10 and 192.168.1.11 - registering to an outside SIP UAS through a NAT gateway whose public address is 67.194.23.55. The NAT gateway maps the source ports in a random or pseudorandom manner akin to: 192.168.1.10:5060 -- 67.194.23.55:32947 192.168.1.11:5060 -- 67.194.23.55:47948 If far-end NAT traversal is enabled on the UAS (in the case of Asterisk, that's nat=yes in sip.conf), the Contact URI supplied in the REGISTER message is ignored and the actual received IP and port on the network and transport layer is used in its place. The latter is what is stored as the contact binding. Later, a call comes in and the UAS maps it back to 67.194.23.55:47948 or 32947 depending on which registrant it is destined to go to. This scenario is not without its problems. Some user agents do not behave symmetrically. Some firewall/NAT router ALGs (application layer gateways) break this process, though they mean well and try to be helpful. But by far the most pressing problem is that many NAT gateways rather quickly age the temporary state information (internal:external UDP port mapping) out after a relatively short period of inactivity. That is why many far-end NAT traversal approaches implement a policy of periodically pinging the stored (received) contact with some sort of message that causes a bidirectional exchange of communication, and therefore causes the NAT gateway to reset its expiration timer for that connection state. In Asterisk, the OPTIONS messages generated when the qualify=yes option is enabled in sip.conf fulfill this function. Hope that helps, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.26 final release - What is blocking?
Hello everybody, I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. Best Regards, -- -- Marc LEURENT lf...@leurent.eu ___ -- 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] unknown RTP codec 126 ??
2009/7/14 gergis.rasmy gergis.ra...@gmail.com: could anyone help explaining what does this error mean? i get this error when make a video/ audio call from X-lite to Bria prof. phone rtp.c:1739 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.26' Gres To quote Counterpath, 126 is normal, and nothing to be worried about. but they don't go on to explain why it is normal or what it is :) Steve ___ -- 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 and several clients behind NAT
jonas kellens schrieb: Is it possible to have several clients behind NAT to register to an Asterisk-server with a public IP-address ? When Asterisk receives an incoming call, how will it know @ which private IP-address the client is reachable ? I guess it is impossible for Asterisk to directly contact the private client behind the NAT ?! Or to distinguish between the private clients ?! Is there an easy solution to this ? How does hosted IP-PBX services work then ?! Jonas. hello, this is how NAT works, it doesnt work only with the IP it uses also the port. So one phone will register with port 1 the next with 10001 and so on, and asterisk knows this port from the registration and will send the communication to this port. But you have to keep the NAT Port open at least from the asterisk side with qualify=yes and also if possible from the client side, so your router keeps the port open. best regards steve ___ -- 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] How to count Parked calls?
You could use global variables to record when and where parks occurred. The issue I would see with this (besides perhaps being cumbersome) is that you wouldn't have a way to undo the counters when a caller hung up instead of coming off of park. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faheem Sent: Tuesday, July 14, 2009 3:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to count Parked calls? Hey All, I am working on a SIP Call bridging application. Every time I receive a incoming call in Asteriskserver1 my AGI should alert to AsteriskServer2 and AsteriskServer2 should callback to AsteriskServer1 and call should be bridged on specified extension. (making call in this way is customer requirement) Every time I receive a call in AsteriskServer1, I Park the call and through AGI, AsteriskServer2 callback to AsteriskServer1 with parked extension. My actual problem is, I can't maintain the record of Parked calls, View Parked Calls in dialplan. Is there any way to count or track the ParkedCalls() in the dialplan?? Through Asterisk CLI I can see the parked calls but I need to count the calls in dialplan. Muhamamd Faheem ___ -- 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 1.4.26 final release - What is blocking?
Marc Leurent schrieb: I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. No more issues are targeted for 1.4.26 however I guess if somebody wanted to it wouldn't hurt to test issues in ready for testing state targeted for 1.4.27. https://issues.asterisk.org/view.php?id=14309 https://issues.asterisk.org/view.php?id=15182 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is Enum safe from spammers?
Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... Is anyone using Enum? Does anyone (other than cynical old me) think that Enum is a spammers best friend? Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) I can see that Enum is good to provide another way round the PSTN, but at the same time, I'm just not convinced... What do others think? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to block inbound call with Asterisk?
Guys, How would you block inbound call's? for example person who is calling me is 212-555-1212, and I would like to do not receive the calls from this person and give them busy tone. What should I write in asterisk config files? and in to witch file should I write it??? ___ -- 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] How to block inbound call with Asterisk?
It is a simple ex-girlfriend thing to do, assuming callerid is working correctly. - exten = s,1,answer - exten = s,n/2125551212,Goto(torture|s|1) - exten = s,n,Dial. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VIP Carrier Sent: Tuesday, July 14, 2009 10:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to block inbound call with Asterisk? Guys, How would you block inbound call's? for example person who is calling me is 212-555-1212, and I would like to do not receive the calls from this person and give them busy tone. What should I write in asterisk config files? and in to witch file should I write it??? ___ -- 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] How to block inbound call with Asterisk?
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling me is
212-555-1212, and I would like to do not receive the calls from this person
and give them busy tone.
What should I write in asterisk config files?
core show function CALLERID
Verbose(1,### Inbound call from ${CALLERID(num)});
if (${CALLERID(num)} = 2125551212) {
Verbose(1,### Block this guy);
Busy(5);
Hangup();
}
Dial(...);
and in to witch file should I
write it???
extensions.ael? extensions.conf?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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[asterisk-users] Error
Does anyone have any light to shed on: c_avpair_new: unknown attribute sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597 We are getting congestion errors on a Pri to telco, and not sure what is going on. Thanks Cary Fitch ___ -- 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] Error
Cary Fitch wrote: Does anyone have any light to shed on: c_avpair_new: unknown attribute sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597 We are getting congestion errors on a Pri to telco, and not sure what is going on. Doing a google search gave an indication that it's a max connection error, but they were talking about ppdp http://osdir.com/ml/network.poptop/2004-04/msg00095.html Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Error
Thanks, we agree.. have reset PRI on telco end and rebooted here and trouble cleared... for a while anyway. Our PRI card seems to have issues. CF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, July 14, 2009 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Error Cary Fitch wrote: Does anyone have any light to shed on: c_avpair_new: unknown attribute sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597 We are getting congestion errors on a Pri to telco, and not sure what is going on. Doing a google search gave an indication that it's a max connection error, but they were talking about ppdp http://osdir.com/ml/network.poptop/2004-04/msg00095.html Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Spectralink 8002 WiFi Phones
Has anyone played with this phone? i cant seem to get it to work properly, i manged to get it registered and can make calls from it, but i havent been able to make it receive calls. Weird thing its that if you make a call from it and while you are on that call you dial its number does calls go thru in second line, but as soon as you terminate both calls it wont recieve any calls again. Heres a look from the asterisk CLI : -- Registered SIP '245' at 192.168.0.239 port 5060 expires 60 trixbox1*CLI sip show peer 245 trixbox1*CLI Name : 245 Secret : Set MD5Secret : Not set Context : from-internal Subscr.Cont. : Not set Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 2...@device VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 50 Dynamic : Yes Callerid : device 245 MaxCallBR : 384 kbps Expire : 67 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 192.168.0.239 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 245 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing: No Status : OK (124 ms) Useragent : Slnk/12 Reg. Contact : sip:2...@192.168.0.239:5060 But after a few seconds the Status goes to UNKNOWN : Auto-Framing: No Status : UNKNOWN -- Useragent : Slnk/12 Reg. Contact : sip:2...@192.168.0.239:5060 This are the config files : sip_245.cfg AUTH = 245; 123456 LINE1 = 245 LINE1_PROXY = 1 LINE1_CALLID = Wireless LINE1_AUTH = 245; 123456 LINE2 = 245 LINE2_PROXY = 1 LINE2_CALLID = Wireless LINE2_AUTH = 245; 123456 sip_allusers.cfg CODECS = g711u, g711a PROXY1_TYPE = Asterisk PROXY1_ADDR = 192.168.0.253:5060 #PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = disable PROXY1_HOLD_IP0 = disable #PROXY1_PRACK = enable PROXY1_REREG_SECS=3600 PROXY1_KEEPALIVE_SECS=14 #PROXY1_DOMAIN = 192.168.0.253 PROXY1_CALLID_PER_LINE = disable PROXY1_MAIL_ACCESS = *97 Access Points are Ubiquitki NanoStation 2 in WDS Mode with QoS enabled. One last thing is that while you're on a call you can ping the phone and soon as the call ends phone stops pinging. Any Ideas? Thanks ___ -- 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] How to block inbound call with Asterisk?
Here is what asterisk said
Sorry, but 1 syntax errors and 0 semantic errors were detected. It doesn't
make sense to compile.
On Tue, Jul 14, 2009 at 12:05 PM, Philipp Kempgen philipp.kemp...@amooma.de
wrote:
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling me
is
212-555-1212, and I would like to do not receive the calls from this
person
and give them busy tone.
What should I write in asterisk config files?
core show function CALLERID
Verbose(1,### Inbound call from ${CALLERID(num)});
if (${CALLERID(num)} = 2125551212) {
Verbose(1,### Block this guy);
Busy(5);
Hangup();
}
Dial(...);
and in to witch file should I
write it???
extensions.ael? extensions.conf?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] How to block inbound call with Asterisk?
The assumption here is that you took Phillipps AEL snippet and put into
extensions.ael. Can you post what you put in?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VIP Carrier
Sent: Tuesday, July 14, 2009 12:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to block inbound call with Asterisk?
Here is what asterisk said
Sorry, but 1 syntax errors and 0 semantic errors were detected. It doesn't
make sense to compile.
On Tue, Jul 14, 2009 at 12:05 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling me
is
212-555-1212, and I would like to do not receive the calls from this
person
and give them busy tone.
What should I write in asterisk config files?
core show function CALLERID
Verbose(1,### Inbound call from ${CALLERID(num)});
if (${CALLERID(num)} = 2125551212) {
Verbose(1,### Block this guy);
Busy(5);
Hangup();
}
Dial(...);
and in to witch file should I
write it???
extensions.ael? extensions.conf?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
--
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Re: [asterisk-users] How to block inbound call with Asterisk?
None of your stuff mentioned above is working!!!
On Tue, Jul 14, 2009 at 1:27 PM, VIP Carrier vipcarr...@gmail.com wrote:
Here is what asterisk said
Sorry, but 1 syntax errors and 0 semantic errors were detected. It doesn't
make sense to compile.
On Tue, Jul 14, 2009 at 12:05 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling me
is
212-555-1212, and I would like to do not receive the calls from this
person
and give them busy tone.
What should I write in asterisk config files?
core show function CALLERID
Verbose(1,### Inbound call from ${CALLERID(num)});
if (${CALLERID(num)} = 2125551212) {
Verbose(1,### Block this guy);
Busy(5);
Hangup();
}
Dial(...);
and in to witch file should I
write it???
extensions.ael? extensions.conf?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
--
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Re: [asterisk-users] How to block inbound call with Asterisk?
what ever he have posted there I have added it, just changed DID
On Tue, Jul 14, 2009 at 1:39 PM, Danny Nicholas da...@debsinc.com wrote:
The assumption here is that you took Phillipp’s AEL snippet and put into
extensions.ael. Can you post what you put in?
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *VIP Carrier
*Sent:* Tuesday, July 14, 2009 12:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How to block inbound call with Asterisk?
Here is what asterisk said
Sorry, but 1 syntax errors and 0 semantic errors were detected. It doesn't
make sense to compile.
On Tue, Jul 14, 2009 at 12:05 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling me
is
212-555-1212, and I would like to do not receive the calls from this
person
and give them busy tone.
What should I write in asterisk config files?
core show function CALLERID
Verbose(1,### Inbound call from ${CALLERID(num)});
if (${CALLERID(num)} = 2125551212) {
Verbose(1,### Block this guy);
Busy(5);
Hangup();
}
Dial(...);
and in to witch file should I
write it???
extensions.ael? extensions.conf?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
--
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Re: [asterisk-users] #exec in #include'd file
On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote: Tilghman Lesher schrieb: On Monday 13 July 2009 01:03:48 pm Philipp Kempgen wrote: Philipp Kempgen schrieb: Is Asterisk supposed to evaluate #exec's in an #include'd file? The directive #exec is not permitted in an AEL configuration file. I see, that would explain why it doesn't work. :-) But in that case it's a documentation issue. The extensions.conf sample says: The #exec command works on all asterisk configuration files. I guess it should read The #exec command works on all asterisk *.conf files except for asterisk.conf. Is there a specific reason not to permit #exec in AEL files? It wasn't coded that way, and it's parsed in a completely different way than any other Asterisk configuration file. I don't know the reason Murf didn't do '#exec' specifically, but I suspect it has to do with the complexity thereof. Is any *.conf file (which permits #exec) guaranteed to be read before extensions.ael? It would then be possible to (ab)use an #exec in there to trigger my generator script (which must not output anything then of course). extconfig.conf? logger.conf? modules.conf? Ugly workaround but doable. No, but you can force it by doing an explicit load of a particular module in modules.conf. Explicitly loaded modules are loaded before all automatically-loaded modules. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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] How to block inbound call with Asterisk?
The Verbose and If statements should be kosher, the Dial is not.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VIP Carrier
Sent: Tuesday, July 14, 2009 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to block inbound call with Asterisk?
what ever he have posted there I have added it, just changed DID
On Tue, Jul 14, 2009 at 1:39 PM, Danny Nicholas da...@debsinc.com wrote:
The assumption here is that you took Phillipps AEL snippet and put into
extensions.ael. Can you post what you put in?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VIP Carrier
Sent: Tuesday, July 14, 2009 12:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to block inbound call with Asterisk?
Here is what asterisk said
Sorry, but 1 syntax errors and 0 semantic errors were detected. It doesn't
make sense to compile.
On Tue, Jul 14, 2009 at 12:05 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling me
is
212-555-1212, and I would like to do not receive the calls from this
person
and give them busy tone.
What should I write in asterisk config files?
core show function CALLERID
Verbose(1,### Inbound call from ${CALLERID(num)});
if (${CALLERID(num)} = 2125551212) {
Verbose(1,### Block this guy);
Busy(5);
Hangup();
}
Dial(...);
and in to witch file should I
write it???
extensions.ael? extensions.conf?
Philipp Kempgen
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Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] How to block inbound call with Asterisk?
Un-top-posting...
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling
me is 212-555-1212, and I would like to do not receive the calls from
this person and give them busy tone. What should I write in asterisk
config files?
On Tue, Jul 14, 2009 at 12:05 PM, Philipp Kempgen philipp.kemp...@amooma.de
wrote:
core show function CALLERID
Verbose(1,### Inbound call from ${CALLERID(num)});
if (${CALLERID(num)} = 2125551212) {
Verbose(1,### Block this guy);
Busy(5);
Hangup();
}
On Tue, 14 Jul 2009, VIP Carrier wrote:
Here is what asterisk said
Sorry, but 1 syntax errors and 0 semantic errors were detected. It
doesn't make sense to compile.
I'm a 1.2 Luddite (AEL is supposedly much improved since the dark ages),
but I think you need a semi after the if's closing brace.
I'd suggest staying away from AEL until you have a much better
understanding of Asterisk and dialplans. Until your dialplans reach a
level of complexity where you need AEL's control structures for clarity,
you should stay with conf.
You should have started here:
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf
and search for girlfriend.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] How to block inbound call with Asterisk?
On Tue, Jul 14, 2009 at 12:44 PM, VIP Carriervipcarr...@gmail.com wrote: what ever he have posted there I have added it, just changed DID To help clear things up... what file did you add this to? -jonathan ___ -- 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] Call Parking timeout fails
John A. Sullivan III wrote: Hello, all. I'm having a nasty problem with call parking in Asterisk 1.6.1.1 that smells like a bug. When the call returns, it seems to be returning to a | delimited extension and failing. Here is the output from the console: Hi John. I've just run into the same problem on 1.6.0.10. Have you heard any more about this problem? TIA, Barry ___ -- 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 1.4.26 final release - What is blocking?
Marc Leurent wrote: Hello everybody, I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. There have been a few issues that keep cropping up that have stopped 1.4.26 from being fully released. Each of these issues have been put into some release candidates and are ready to be tested. Asterisk 1.4.26-rc6 was just released today. For more information, see the release announcement at http://www.asterisk.org/node/48608. Thanks! Leif Madsen. Asterisk issue marshal ___ -- 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] Help in oh323 Gatekeeper + does not know what to do when bridging the call
Actually I am facing a problem with H.323 (the standard and the ooh323) with Asterisk vesion 1.4.25 and I discover the following: 1) Using the standard h323 that come with Asterisk: The chan_h323.so it is not existed in the /usr/lib/asterisk/modules after doing the compilation and installation for (pwlib, openh323, /chanels/h323, asterisk), although make menuselect was done and the h323 channel was selected. Also, I was doing make opt for the h323, pwblib, and openh323. But the chan_h323.so is not existed. But when the h323 IP Phone originate the call, asterisk receive it (I see this in asterisk consol) and respond in the default context (even we added the [ ] in the h323.conf), but no voice at all (we do not hear the played wave file), I beleive this is because chan_h323.so is not existed under the /usr/lib/asterisk/modules/, maybe ! Why chan_h323.so is nto generated there? Why asterisk does not take in consideration that we added a friend in there h323.conf file and it should be authorized using its h323 id, and should be routed using the configured context, but all of this is not happenning, why? But ofcourse the telnet for the port 1720 is working, that means the h323 port is under listenning case. 2) Using the asterisk-addons-1.4.8: The chan_ooh323.so become existed in the /usr/lib/asterisk/modules and when we place a call from the IP Phone, it come to asterisk (I see this in asterisk consol) and the call manipulated in the default context (even if we added the endpoint configuration in the h323.conf file as a friend and determine its context), and we hear the wave file when it is played (so we hear a voice), but ofcourse it is not taking into consideration the endpoint configuration in the h323.conf file, so no authorization and actually can not do anything, because the configuration of the endpoint in the h323.conf file is not taken. Is it related to the Asterisk 1.4.25? Or we are missing something in the configuration? Regards Bilal Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. WARNING[8446]: chan_oh323.c:3555 cleanup_h323_connection: Call 'ip$192.168.8.96:30005/27890-f5194af7' not found (clear). What version of Asterisk is it? Any chance of using the built-in chan_h323? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Polycom Spectralink 8002 WiFi Phones
Search the archives - we had a big discussion about this phone about six months ago. If you make it work and want another one I will give you special price!. j On Tue, 14 Jul 2009, Cesar Gonzalez wrote: Has anyone played with this phone? i cant seem to get it to work properly, i manged to get it registered and can make calls from it, but i havent been able to make it receive calls. Weird thing its that if you make a call from it and while you are on that call you dial its number does calls go thru in second line, but as soon as you terminate both calls it wont recieve any calls again. Heres a look from the asterisk CLI : -- Registered SIP '245' at 192.168.0.239 port 5060 expires 60 trixbox1*CLI sip show peer 245 trixbox1*CLI Name : 245 Secret : Set MD5Secret : Not set Context : from-internal Subscr.Cont. : Not set Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 2...@device VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 50 Dynamic : Yes Callerid : device 245 MaxCallBR : 384 kbps Expire : 67 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 192.168.0.239 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 245 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing: No Status : OK (124 ms) Useragent : Slnk/12 Reg. Contact : sip:2...@192.168.0.239:5060 But after a few seconds the Status goes to UNKNOWN : Auto-Framing: No Status : UNKNOWN -- Useragent : Slnk/12 Reg. Contact : sip:2...@192.168.0.239:5060 This are the config files : sip_245.cfg AUTH = 245; 123456 LINE1 = 245 LINE1_PROXY = 1 LINE1_CALLID = Wireless LINE1_AUTH = 245; 123456 LINE2 = 245 LINE2_PROXY = 1 LINE2_CALLID = Wireless LINE2_AUTH = 245; 123456 sip_allusers.cfg CODECS = g711u, g711a PROXY1_TYPE = Asterisk PROXY1_ADDR = 192.168.0.253:5060 #PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = disable PROXY1_HOLD_IP0 = disable #PROXY1_PRACK = enable PROXY1_REREG_SECS=3600 PROXY1_KEEPALIVE_SECS=14 #PROXY1_DOMAIN = 192.168.0.253 PROXY1_CALLID_PER_LINE = disable PROXY1_MAIL_ACCESS = *97 Access Points are Ubiquitki NanoStation 2 in WDS Mode with QoS enabled. One last thing is that while you're on a call you can ping the phone and soon as the call ends phone stops pinging. Any Ideas? Thanks ___ -- 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 ___ -- 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] #exec in #include'd file
On Tue, Jul 14, 2009 at 11:47 AM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote: Tilghman Lesher schrieb: On Monday 13 July 2009 01:03:48 pm Philipp Kempgen wrote: Philipp Kempgen schrieb: Is Asterisk supposed to evaluate #exec's in an #include'd file? The directive #exec is not permitted in an AEL configuration file. I see, that would explain why it doesn't work. :-) But in that case it's a documentation issue. The extensions.conf sample says: The #exec command works on all asterisk configuration files. I guess it should read The #exec command works on all asterisk *.conf files except for asterisk.conf. Is there a specific reason not to permit #exec in AEL files? It wasn't coded that way, and it's parsed in a completely different way than any other Asterisk configuration file. I don't know the reason Murf didn't do '#exec' specifically, but I suspect it has to do with the complexity thereof. I didn't exclude the #exec for any particular reason. I think it just wasn't in the original AEL (1.2) code, so I missed it... (or it escaped my all-powerful eyes somehow). If someone files a bug, I might be able to code up something to handle it in future releases. (just as a reminder for me). I guess you could, for the time being, put your #exec stuff in an extensions.conf file, and use the modules.conf tricks to preload the extensions.conf file first, if that is a requirement, as previously suggested... murf Is any *.conf file (which permits #exec) guaranteed to be read before extensions.ael? It would then be possible to (ab)use an #exec in there to trigger my generator script (which must not output anything then of course). extconfig.conf? logger.conf? modules.conf? Ugly workaround but doable. No, but you can force it by doing an explicit load of a particular module in modules.conf. Explicitly loaded modules are loaded before all automatically-loaded modules. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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 -- Steve Murphy ParseTree Corp ___ -- 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] Why CDR is recording dst value = h?
It should be an easy one for many of the experts here.
On Mon, Jul 13, 2009 at 8:10 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
For a new project, I have written a dialplan and it is pretty straight
forward: The [dialout] context dials out a number, and h extension in this
context writes the CDR. But what is happening is that if the callee hangs up
first, all values in the CDR are fine, but if the caller hangs up first, the
'dst' column is always 'h'. I put a NoOp right in the begining of this macro
to verify it.
Any idea why is this happening and how can I have correct 'dst' value if
the caller hangs up first.
[dialout]
exten = _NXXNXX,s,1,Dial(SIP/XX/${EXTEN},30)
exten = h,1,Macro(hangupcall,${EXTEN},${CDR(accountcode)})
[macro-hangupcall]
NoOp(${CDR(dst)})
Set(dialout_num=${ARG1})
Set(user_id=${ARG2})
ResetCDR(vw);
NoCDR();
Hangup();
--
Zeeshan A Zakaria
--
Zeeshan A Zakaria
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Re: [asterisk-users] #exec in #include'd file
Tilghman Lesher schrieb: On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote: Is there a specific reason not to permit #exec in AEL files? It wasn't coded that way, and it's parsed in a completely different way than any other Asterisk configuration file. I don't know the reason Murf didn't do '#exec' specifically, but I suspect it has to do with the complexity thereof. Thanks for the clarification. I was under the (false) assumption that #include and #exec were some kind of preprocessor directives which would be evaluated before any parsing is done, bu that is not true, at least not for extensions.ael. Is any *.conf file (which permits #exec) guaranteed to be read before extensions.ael? It would then be possible to (ab)use an #exec in there to trigger my generator script (which must not output anything then of course). extconfig.conf? logger.conf? modules.conf? Ugly workaround but doable. No, but you can force it by doing an explicit load of a particular module in modules.conf. Explicitly loaded modules are loaded before all automatically-loaded modules. I'm thinking about the options in following: a) load = extconfig ; possible? #exec in extconfig.conf. b) #exec in modules.conf itself Need to figure out if load is enough or if I should preload the module. And that raises the question how often Asterisk will reload extconfig.conf and modules.conf. It certainly reads modules.conf twice on startup and reads extconfig.conf on startup and reload. Do any other events make Asterisk re-read these files? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Help in oh323 gatekeeper
Dear; I would like to ask: when Asterisk was registering on the gnugk, both (asterisk and gnugk) were on the same hardware machine and same IP address? Can they be on the same IP address? In case they were on the same IP address then: I am afraid the oh323 channel in asterisk will respond for the H323 endpoint (IP Phone) instead of the gnugk (specially if the IP Phone was in routed mode and not register to gnugk)? I mean, if the IP Phone was need to place call via the gnugk in the routed mode, and the call need to be send for Asterisk, so how can u avoid that oh323 channel in asterisk from responding for the IP Phone instead of the gnugk it self? Because if u let the IP Phone send the call for the IP address that asterisk running on it, then the h323 channel in the asterisk will respond as u know, so how to let the gnugk respond and not the asterisk h323 channel? Regards Bilal -- Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. WARNING[8446]: chan_oh323.c:3555 cleanup_h323_connection: Call 'ip$192.168.8.96:30005/27890-f5194af7' not found (clear). Please any one give suggestions to disable the gatekeeper access in Asterisk... Thanks in Advance... Regards, Velusamy.K ___ -- 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] Why CDR is recording dst value = h?
Zeeshan Zakaria schrieb: For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to verify it. Any idea why is this happening and how can I have correct 'dst' value if the caller hangs up first. Maybe endbeforehexten=yes in cdr.conf does what you need. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] #exec in #include'd file
On Tuesday 14 July 2009 15:35:20 Philipp Kempgen wrote: Tilghman Lesher schrieb: On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote: Is there a specific reason not to permit #exec in AEL files? It wasn't coded that way, and it's parsed in a completely different way than any other Asterisk configuration file. I don't know the reason Murf didn't do '#exec' specifically, but I suspect it has to do with the complexity thereof. Thanks for the clarification. I was under the (false) assumption that #include and #exec were some kind of preprocessor directives which would be evaluated before any parsing is done, bu that is not true, at least not for extensions.ael. Is any *.conf file (which permits #exec) guaranteed to be read before extensions.ael? It would then be possible to (ab)use an #exec in there to trigger my generator script (which must not output anything then of course). extconfig.conf? logger.conf? modules.conf? Ugly workaround but doable. No, but you can force it by doing an explicit load of a particular module in modules.conf. Explicitly loaded modules are loaded before all automatically-loaded modules. I'm thinking about the options in following: a) load = extconfig ; possible? #exec in extconfig.conf. b) #exec in modules.conf itself Need to figure out if load is enough or if I should preload the module. And that raises the question how often Asterisk will reload extconfig.conf and modules.conf. It certainly reads modules.conf twice on startup and reads extconfig.conf on startup and reload. Do any other events make Asterisk re-read these files? For modules.conf, no, but extconfig.conf is re-read upon a 'module reload extconfig' command. This permits changing realtime settings without restarting Asterisk. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] setvar and transfer (was: Re: How to Change size of CDR(accountcode) variable?)
Benny Amorsen schrieb: Last concern: Does setvar work even for transfers, like accountcode does? I can't answer your question, but transfer != transfer. Some use a feature code in Asterisk, some initiate a transfer on their phone, some use a way to call the Transfer() application. Mixing it up causes a lot of confusion. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] setvar and transfer
Philipp Kempgen schrieb: Benny Amorsen schrieb: Last concern: Does setvar work even for transfers, like accountcode does? I can't answer your question, but transfer != transfer. Some use a feature code in Asterisk, some initiate a transfer on their phone, some use a way to call the Transfer() application. Mixing it up causes a lot of confusion. Many users always call Dial() with the tT args. They want transfers to work, after all. :-) Many of them probably don't need it, because a good number of SIP phones have a Transfer key which initiates a transfer the SIP way. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QoS
Howdy, Getting ready to play with QoS settings. We have an asterisk 1.4.23 server running in a colo bunker in the US Virgin Islands under a large radio tower. That tower has multiple sector radio/antenna pairs that blanket a valley in 802.11a. The customers have directed dishes aimed at the sector antennas, mounted on their roofs. This setup has been working great for their broadband access for many years. Now we want to sell voice services on top of this infrastructure, and it works fine too, until they start some data intensive process on the customer end, like bittorrent :) We would like to avoid these problems by properly setting up packet prioritization between the customer and the sector radios, which we have control over. Any links to share to get us started? Basically from zero? :) Cheers, j ___ -- 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] Polycom Spectralink 8002 WiFi Phones
Jeff LaCoursiere wrote: Search the archives - we had a big discussion about this phone about six months ago. If you make it work and want another one I will give you special price!. j Jeff, yeah i saw the posts, i followed Bob Pierce config and had no luck, BUT it just started to work, i changed AP's, seems like theres something wrong with Ubiquiti NanoStation2 WMM implementation, i used a Linksys WRT54G2 and viola! it started to work, i guess i should've done that to begin with... :( I'll play around whit the Nanostations QoS settings and see if i can get it to work on those AP's. What AP's were you using? -Cesar On Tue, 14 Jul 2009, Cesar Gonzalez wrote: Has anyone played with this phone? i cant seem to get it to work properly, i manged to get it registered and can make calls from it, but i havent been able to make it receive calls. Weird thing its that if you make a call from it and while you are on that call you dial its number does calls go thru in second line, but as soon as you terminate both calls it wont recieve any calls again. Heres a look from the asterisk CLI : -- Registered SIP '245' at 192.168.0.239 port 5060 expires 60 trixbox1*CLI sip show peer 245 trixbox1*CLI Name : 245 Secret : Set MD5Secret : Not set Context : from-internal Subscr.Cont. : Not set Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 2...@device VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 50 Dynamic : Yes Callerid : device 245 MaxCallBR : 384 kbps Expire : 67 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 192.168.0.239 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 245 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing: No Status : OK (124 ms) Useragent : Slnk/12 Reg. Contact : sip:2...@192.168.0.239:5060 But after a few seconds the Status goes to UNKNOWN : Auto-Framing: No Status : UNKNOWN -- Useragent : Slnk/12 Reg. Contact : sip:2...@192.168.0.239:5060 This are the config files : sip_245.cfg AUTH = 245; 123456 LINE1 = 245 LINE1_PROXY = 1 LINE1_CALLID = Wireless LINE1_AUTH = 245; 123456 LINE2 = 245 LINE2_PROXY = 1 LINE2_CALLID = Wireless LINE2_AUTH = 245; 123456 sip_allusers.cfg CODECS = g711u, g711a PROXY1_TYPE = Asterisk PROXY1_ADDR = 192.168.0.253:5060 #PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = disable PROXY1_HOLD_IP0 = disable #PROXY1_PRACK = enable PROXY1_REREG_SECS=3600 PROXY1_KEEPALIVE_SECS=14 #PROXY1_DOMAIN = 192.168.0.253 PROXY1_CALLID_PER_LINE = disable PROXY1_MAIL_ACCESS = *97 Access Points are Ubiquitki NanoStation 2 in WDS Mode with QoS enabled. One last thing is that while you're on a call you can ping the phone and soon as the call ends phone stops pinging. Any Ideas? Thanks ___ -- 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 ___ -- 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 ___ -- 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] #exec in #include'd file
Tilghman Lesher schrieb: On Tuesday 14 July 2009 15:35:20 Philipp Kempgen wrote: Tilghman Lesher schrieb: On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote: Is any *.conf file (which permits #exec) guaranteed to be read before extensions.ael? It would then be possible to (ab)use an #exec in there to trigger my generator script (which must not output anything then of course). extconfig.conf? logger.conf? modules.conf? Ugly workaround but doable. No, but you can force it by doing an explicit load of a particular module in modules.conf. Explicitly loaded modules are loaded before all automatically-loaded modules. I'm thinking about the options in following: a) load = extconfig ; possible? #exec in extconfig.conf. b) #exec in modules.conf itself It certainly reads modules.conf twice on startup and reads extconfig.conf on startup and reload. Do any other events make Asterisk re-read these files? For modules.conf, no, but extconfig.conf is re-read upon a 'module reload extconfig' command. This permits changing realtime settings without restarting Asterisk. Just found out that even a simple `asterisk -r` causes extconfig.conf to be re-read. That rules extconfig.conf out for what I am trying to do since I don't want to re-generate extensions.ael every time some- body connects to the Asterisk console. Looks like I should go for modules.conf then and implement a mechanism to avoid re-generating extensions.ael on the second pass in modules.conf. E.g. I could make the script not do anything if extensions.ael was modified less than 5 seconds ago. That's not perfect but should do the trick. Alternatively I could use load = pbx_config.so and put the #exec in extensions.conf as murf suggested. Need to play around a bit. Thanks to both of you and sorry for hijacking my own thread a bit. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Call Parking timeout fails
This was fixed in the 1.6.1 SVN, and I would guess that it was also fixed in the 1.6.0. SVN log: r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines Fix call parking callback. Pipes - Commas. You will have to create a patch against the 1.6.0 source, but you could start by looking at the patch in this issue: https://issues.asterisk.org/view.php?id=15162 Please note again that that patch was against 1.6.1.0. -Jonathan On Tue, Jul 14, 2009 at 11:09 AM, Barry L. Kline blkl...@attglobal.netwrote: John A. Sullivan III wrote: Hello, all. I'm having a nasty problem with call parking in Asterisk 1.6.1.1 that smells like a bug. When the call returns, it seems to be returning to a | delimited extension and failing. Here is the output from the console: Hi John. I've just run into the same problem on 1.6.0.10. Have you heard any more about this problem? TIA, Barry ___ -- 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 ___ -- 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] Polycom Spectralink 8002 WiFi Phones
On Tue, 14 Jul 2009, Cesar Gonzalez wrote: Jeff LaCoursiere wrote: Search the archives - we had a big discussion about this phone about six months ago. If you make it work and want another one I will give you special price!. j Jeff, yeah i saw the posts, i followed Bob Pierce config and had no luck, BUT it just started to work, i changed AP's, seems like theres something wrong with Ubiquiti NanoStation2 WMM implementation, i used a Linksys WRT54G2 and viola! it started to work, i guess i should've done that to begin with... :( I'll play around whit the Nanostations QoS settings and see if i can get it to work on those AP's. What AP's were you using? Hi Cesar, I did actually get it to work as well, and was using Linksys WRT54G with dd-wrt. I *intended* for the phone to be useful at random wifi hotspots, however, and was a bit disappointed to find that that was not going to work. So it sits on a shelf gathering dust... Cheers, j -Cesar On Tue, 14 Jul 2009, Cesar Gonzalez wrote: Has anyone played with this phone? i cant seem to get it to work properly, i manged to get it registered and can make calls from it, but i havent been able to make it receive calls. Weird thing its that if you make a call from it and while you are on that call you dial its number does calls go thru in second line, but as soon as you terminate both calls it wont recieve any calls again. Heres a look from the asterisk CLI : -- Registered SIP '245' at 192.168.0.239 port 5060 expires 60 trixbox1*CLI sip show peer 245 trixbox1*CLI Name : 245 Secret : Set MD5Secret : Not set Context : from-internal Subscr.Cont. : Not set Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 2...@device VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 50 Dynamic : Yes Callerid : device 245 MaxCallBR : 384 kbps Expire : 67 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 192.168.0.239 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 245 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing: No Status : OK (124 ms) Useragent : Slnk/12 Reg. Contact : sip:2...@192.168.0.239:5060 But after a few seconds the Status goes to UNKNOWN : Auto-Framing: No Status : UNKNOWN -- Useragent : Slnk/12 Reg. Contact : sip:2...@192.168.0.239:5060 This are the config files : sip_245.cfg AUTH = 245; 123456 LINE1 = 245 LINE1_PROXY = 1 LINE1_CALLID = Wireless LINE1_AUTH = 245; 123456 LINE2 = 245 LINE2_PROXY = 1 LINE2_CALLID = Wireless LINE2_AUTH = 245; 123456 sip_allusers.cfg CODECS = g711u, g711a PROXY1_TYPE = Asterisk PROXY1_ADDR = 192.168.0.253:5060 #PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = disable PROXY1_HOLD_IP0 = disable #PROXY1_PRACK = enable PROXY1_REREG_SECS=3600 PROXY1_KEEPALIVE_SECS=14 #PROXY1_DOMAIN = 192.168.0.253 PROXY1_CALLID_PER_LINE = disable PROXY1_MAIL_ACCESS = *97 Access Points are Ubiquitki NanoStation 2 in WDS Mode with QoS enabled. One last thing is that while you're on a call you can ping the phone and soon as the call ends phone stops pinging. Any Ideas? Thanks ___ -- 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 ___ -- 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 ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE120P loosing link...
Hi guys, Hope someone can help me with this... I got an Asterisk with TE120P card... using dahdi 2.1.0.4 and asterisk 1.4.24. The problem is the link between the card and the telco... suddenly through the day looses connection... and the people on site just power cycle the PC and the problem got fixed for a while I've already replace the cable between the card and the telco. Telco had already complete their tests, and they said the problem is in our side. I ran dahdi_test: --- Results after 88 passes --- Best: 100.000 -- Worst: 99.987 -- Average: 99.995104, Difference: 99.996540 I started the wcte12xp module debug: Jul 14 17:36:19 voipidn kernel: wcte12xp0: Missed interrupt. Increasing latency to 11 ms in order to compensate. Jul 14 17:37:14 voipidn kernel: wcte12xp0: Missed interrupt. Increasing latency to 12 ms in order to compensate. Any ideas, Thanks Arturo Ochoa Electrosystems ___ -- 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] Is Enum safe from spammers?
I think an equally interesting question is whether the Federal Trade Commission (and foreign equivalents) draw a distinction between calls to E.164 numbers based on their transport technology. In other words, is there a legal difference depending on whether the call touches the PSTN vs. being looked up in an ENUM directory with Pure IP transport? If you are an attorney, please chime in. I'm not an attorney, but I suspect the answer would be that there is no distinction. I know the definition of phone call is a moving target these days, so perhaps today's legal answer will be different tomorrow. On the other hand perhaps the legal question is completely moot. The zero-cost nature of SPIT might make it like SPAM wherein the fact that it violates many laws in most countries is ultimately of no consequence. Will this ultimately come down to a technical arms race like we see with SPAM? . December 21, 2012 - Original Message - From: Gordon Henderson gordon+aster...@drogon.net To: Asterisk Users Mailing List Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 14, 2009 9:14 AM Subject: [asterisk-users] Is Enum safe from spammers? Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... Is anyone using Enum? Does anyone (other than cynical old me) think that Enum is a spammers best friend? Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) I can see that Enum is good to provide another way round the PSTN, but at the same time, I'm just not convinced... What do others think? Gordon ___ -- 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 ___ -- 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] Call Parking timeout fails
Jonathan Thurman wrote: This was fixed in the 1.6.1 SVN, and I would guess that it was also fixed in the 1.6.0. SVN log: r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines Fix call parking callback. Pipes - Commas. You will have to create a patch against the 1.6.0 source, but you could start by looking at the patch in this issue: https://issues.asterisk.org/view.php?id=15162 Please note again that that patch was against 1.6.1.0. -Jonathan Thanks very much Jonathan. I'll figure out how to make this patch against 1.6.0.10. Barry ___ -- 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] Call Parking timeout fails
Barry L. Kline wrote: I'll figure out how to make this patch against 1.6.0.10. That was a trivial fix. I hope that they permanently add that patch to the 1.6.0.x series. Thanks again Jonathan. Barry ___ -- 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] Trouble with originating a call through Asterisk Manager Interface
On 13/7/09 2:23 PM, eric weaver wrote: I am doing a little application to originate a call through Asterisk via AMI (Perl Asterisk::Manager). It logs in successfully, does an originate command with Exten: 0020 (which is set up to answer and wait for 60 then hang up) Channel: SIP/5101234...@test-host (which comes to my desktop machine also running Asterisk). At the target machine I see only a CANCEL to which it immediately responds with a No Transaction. Except for every nth try, when I see an INVITE; but only that often. It looks like AstMan is asking for a Slin-format connection and the channel is set up only for Slin or Ulaw but it says joint capabilities 0x0. Don't know if that's a red herring. Any advice welcome. The Asterisk Manager won't have any idea about codecs etc. I suggest the best way to tackle this is to make sure you can make the same call using the dialplan first, then move to the manager - settings like codecs will therefore show up in the initial testing. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Help in oh323 gatekeeper
On Tue, Jul 14, 2009 at 2:19 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; I would like to ask: when Asterisk was registering on the gnugk, both (asterisk and gnugk) were on the same hardware machine and same IP address? Can they be on the same IP address? If I understand your questions: Can Asterisk and GnuGK both run as an h323 server on the same IP Address, the answer would be no unless they are running on different ports. You can not have two processes on the same machine/ip/port combination. In case they were on the same IP address then: I am afraid the oh323 channel in asterisk will respond for the H323 endpoint (IP Phone) instead of the gnugk (specially if the IP Phone was in routed mode and not register to gnugk)? I mean, if the IP Phone was need to place call via the gnugk in the routed mode, and the call need to be send for Asterisk, so how can u avoid that oh323 channel in asterisk from responding for the IP Phone instead of the gnugk it self? Because if u let the IP Phone send the call for the IP address that asterisk running on it, then the h323 channel in the asterisk will respond as u know, so how to let the gnugk respond and not the asterisk h323 channel? Right. If they both run on the same ip/port then the one started first would win, and listen for connections (the second app should fail to bind and complain). You could change the port, or the IP that the one of the apps is listening on. Hope that helps. -Jonathan ___ -- 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] Is Enum safe from spammers?
On Tue, Jul 14, 2009 at 06:46:50PM -0500, Karl Fife wrote: [snip] missed the original message - Original Message - From: Gordon Henderson gordon+aster...@drogon.net To: Asterisk Users Mailing List Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 14, 2009 9:14 AM Subject: [asterisk-users] Is Enum safe from spammers? Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... As a Director of UKEC Ltd (the governing body of ENUM in the UK) I'd be interested in knowing more about this. Is anyone using Enum? Currently there is a need to populate the ENUM database. UKEC and Nominet are working together to try and get vendors to support ENUM. Does anyone (other than cynical old me) think that Enum is a spammers best friend? ENUM isn't just about VoIP, it allows end users to set policies on how they want to receive calls. Unfortunately not many telcos yet support ENUM (or public ENUM anyway). The most likely growth area are ITSPs populating the ENUM database with their customer's numbers. Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) If you find out, please do let me know. I can see that Enum is good to provide another way round the PSTN, but at the same time, I'm just not convinced... ENUM is the future of telephony, it's just needs mass adoption. Unfortunately there are likely to be at least 3 ENUM systems in the UK. * Public ENUM as in e164.arpa * Carrier ENUM whereby telcos use ENUM to route calls to other telcos. * Eventually a central porting database for mobiles (and also fixed lines) which uses ENUM to store the port information. It would be good if these all merged into one body. What do others think? Happy to have a chat off-line. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with originating a call through Asterisk Manager Interface
Turns out I was using the wrong units in the TIMEOUT parameter to the Manager Originate command... It was supposed to be milliseconds and I put 15. D'o Was timing out before it got started. Now it connects but odd things happen. But there are two NATting firewalls between the two Asterices. I think I need to set up some kind of UDP tunneling or use a NAT-free instance for one of them... Thanks ___ -- 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] Trouble with originating a call through Asterisk Manager Interface
On 15/7/09 1:34 PM, eric weaver wrote: Turns out I was using the wrong units in the TIMEOUT parameter to the Manager Originate command... It was supposed to be milliseconds and I put 15. D'o Was timing out before it got started. Now it connects but odd things happen. But there are two NATting firewalls between the two Asterices. I think I need to set up some kind of UDP tunneling or use a NAT-free instance for one of them... You should probably be right with the externip/localnet settings -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Is Enum safe from spammers?
The answer, quickly, is No, ENUM is not safe from spam. But there is security in obscurity at the moment. Since nobody really uses ENUM, it's not been brought to the attention of phone spammers. However, witness AOL AIM, or Skype - now that people know it exists and there are millions of endpoints, the bots move in. I get frequent connections on both services from random bots wanting to chat, though no voice connections yet. So ENUM is a target, yes. But as far as SIP URIs in ENUM, there may be some easy solutions that don't require a lot of backflips and can quickly integrate with Asterisk. The good news is that Asterisk is easily scriptable to block/squelch calls that don't meet certain criteria. Here's a post I wrote a while back on the topic, including code. https://mail.internet2.edu/wws/arc/sip.edu/2006-07/msg00012.html ...and a better-formatted version: http://forum.e164.org/index.php?topic=16.0 JT On Jul 14, 2009, at 4:46 PM, Karl Fife wrote: I think an equally interesting question is whether the Federal Trade Commission (and foreign equivalents) draw a distinction between calls to E.164 numbers based on their transport technology. In other words, is there a legal difference depending on whether the call touches the PSTN vs. being looked up in an ENUM directory with Pure IP transport? If you are an attorney, please chime in. I'm not an attorney, but I suspect the answer would be that there is no distinction. I know the definition of phone call is a moving target these days, so perhaps today's legal answer will be different tomorrow. On the other hand perhaps the legal question is completely moot. The zero-cost nature of SPIT might make it like SPAM wherein the fact that it violates many laws in most countries is ultimately of no consequence. Will this ultimately come down to a technical arms race like we see with SPAM? . December 21, 2012 - Original Message - From: Gordon Henderson gordon+aster...@drogon.net To: Asterisk Users Mailing List Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 14, 2009 9:14 AM Subject: [asterisk-users] Is Enum safe from spammers? Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... Is anyone using Enum? Does anyone (other than cynical old me) think that Enum is a spammers best friend? Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) I can see that Enum is good to provide another way round the PSTN, but at the same time, I'm just not convinced... What do others think? Gordon ___ -- 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 ___ -- 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 --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call transfer using DTMF
Is there a way to transfer a call, while in the middle of the call, using DTMF? ___ -- 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] call transfer using DTMF
On 15/7/09 3:07 PM, Michael wrote: Is there a way to transfer a call, while in the middle of the call, using DTMF? Yep, just pass the t or T options to the dial command and set it up in /etc/asterisk/features.conf -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] call transfer using DTMF
Yes, In the features.conf under featuremap you need the blindtransfer un-commented blindxfer = ## Then in your extensions.conf you need to have at least a capital T exten = example,1,Dial(ZAP/4/12345,,T) Then during the call you can press ## and asterisk will say transfer. Then dial in the extension you want to transfer too. Thank you, Brad Finberg - Original Message - From: Michael as...@nettrust.co.nz To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Date: Tuesday, July 14 2009 11:07 PM Subject: [asterisk-users] call transfer using DTMF Is there a way to transfer a call, while in the middle of the call, using DTMF? ___ -- 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___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to enable dial to a...@asterisk.blurb.com
Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex -- There is no instance of a country having benefited from prolonged warfare -- Sun Tzu - The Art of War signature.asc Description: Digital signature ___ -- 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] how to enable dial to a...@asterisk.blurb.com
Alex Samad wrote: Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex Alex, Here's a good place to start. http://www.voip-info.org/wiki/view/DNS+SRV Then you would need to enable a few things in /etc/asterisk/sip.conf [general] allowguest=yes context=yourdefaultcontext domain=yourdomain.com Then configure the default context in your extensions.conf file to include routing for your calls. There may be a few more steps, but this should get you going down the right road. Darrick ___ -- 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] how to enable dial to a...@asterisk.blurb.com
On Wed, 2009-07-15 at 14:34 +1000, Alex Samad wrote:
Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
snip
If I understand what you are seeking, you can try these URIs:
http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
http://www.blyon.com/blog/index.php/2009/06/22/p2p-sip-uri-dialing/
However, I found I changed mine substantially. I am very new to
Asterisk so if this seems like a silly idea, it probably is and I would
appreciate being told so! We generally use numeric extensions - old
habits I suppose. We found that the catch-all _. for uri dialing was
also catching mis-dialed extensions. That led us to this solution:
[dial-uri] ; Always include this last because of its broad matches
exten = _[a-zA-Z0-9].,1,GotoIf($[${SIPDOMAIN}!=pbx01.ssiservices.biz]?:_.,1)
; non-URIs will automatically append @pbx01.ssiservices.biz
; this logic separates mistyped extensions from valid URI attempts
exten = _[a-zA-Z0-9].,n,Macro(uridial,${ext...@${sipdomain})
exten = _.,1,Answer(0.5)
exten = _.,n,Playback(im-sorry)
exten = _.,n,Wait(0.0.5)
exten = _.,n,Playback(you-dialed-wrong-number)
exten = _.,n,Wait(0.4)
exten = _.,n,Playback(vm-goodbye)
exten = _.,n,Hangup()
Here is the macro:
[macro-uridial]
exten = s,1,NoOp(Calling remote SIP peer ${ARG1})
exten = s,n,Dial(SIP/${ARG1},60)
exten = s,n,Congestion()
As I think about it, I wonder if that NoOp should be replace with a
Verbose. In any event, I hope this helps.
Oh, of course, this is for outbound. For inbound, one creates explicit
entries for each SIP URI and map these to the appropriate extensions.
For example, for users, we typically map to their email address (which
is different than their internal ID; for security purposes, publicly
exposed IDs are different from internally used IDs). We also create
direct SIP extensions for things like voicemail, office numbers, world
headquarters, so that direct SIP calls can be used just like regular
calls and enter our calling tree:
[a100in] ; direct inbound SIP dialing
exten = conference,1,Goto(a100pub,6000,1)
exten = someone,1,Goto(a100pub,314,1)
exten = helpdesk,1,Goto(a100pub,302,1)
exten = someoneelse,1,Goto(a100pub,312,1)
exten = mycompany-hq,1,Goto(a100pub,9,welcome)
exten = mycompany-europe,1,Goto(a100pub,9,welcome)
exten = mycompany-us,1,Goto(a100pub,9,welcome)
exten = vmail,1,Goto(a100pub,7000,1)
Since we are a secure, multi-tenant environment, we do not place these
in the default inbound context for sip. Instead, we only allow
designated domains in our sip.conf and specify a separate inbound
context for each which lands them into these sip directories, e.g., :
autodomain=no
domain=pbx01.mycompany.com
domain=172.x.y.8
; define client domains
domain=yourcompany.com,a100in
domain=theircompany.com,a10in
domain=pbx01.theircompany.com
allowexternaldomains=yes
Hope this helps. If someone sees a better way, please say so. Thanks -
John
--
John A. Sullivan III
Open Source Development Corporation
Street Preacher: Are you SAVED?!!
Educated Skeptic: Saved from WHAT?!!
Educated Believer: From our selfishness that hurts the ones we love
and condemns us to an eternity of hurting each other.
http://www.spiritualoutreach.com
Christianity that makes sense
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