Re: [asterisk-users] a2billing
You might have to tamper the main a2billing.php or more files for that feature to work. Or it might cost around $800 in development time. On Tue, Oct 19, 2010 at 4:34 AM, Baha @ SH i...@saudihome.com wrote: Exactly, I don’t want that, it’s annoying! I just want it to run if the customer balance reach for example 1 dollar! Anyway? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, October 18, 2010 8:17 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] a2billing I don't think voucher can be triggered to announce at certain threshold ONLY but it will be run everytime at the begining after PIN is asked for. By default it's set to: Press 8 to fill up with a voucher. System Settings is the last in the menu. -Bruce On Tue, Oct 19, 2010 at 2:31 AM, Baha @ SH i...@saudihome.com wrote: I am sorry , but where is System Settings??? And what is the parameter name? And also, id like to mention that the voucher is working, only when balance is below minimum balance it does not go to voucher ivr. Thanks, awaiting,,, *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, October 18, 2010 12:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] a2billing Turn on the voucher feature in System Settings and it will tell the user right after the PIN authentication or CLID authentication that their balance is below threshold and they should pay. -Bruce On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH i...@saudihome.com wrote: Not sure if a2billing can be shared here, but ill give a shot If the credit min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ; SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00] [18/10/2010 07:01:12]:[file:a2billing.php - line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff, activated, inuse, simultaccess, typepaid, creditlimit, language, removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate), expiredays, nbused, UNIX_TIMESTAMP(firstusedate), UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname, cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign, cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON tariff=cc_tariffgroup.id WHERE username='9971524976'] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 :: active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 :: language=en]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD : prepaid-zero-balance (cardnumber:9971524976)]] [18/10/2010 07:01:14]:[file:a2billing.php - line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY : callingcard_ivr_authenticate]] [18/10/2010 07:01:14]:[file:a2billing.php - line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED (cia_res:-2)]] [18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]] -- _ -- 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 -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
[asterisk-users] How to check if Agent is logged into a specific Queue using dial-plan?
Hi,
I have this on an Aastra phone:
Button 1:Login English Queue
Button 2:Login French Queue
Button 3:Logout both English and French
I am out of buttons and using only three buttons I want my third button to
be smarter. Currently the third button does a QueueRemoveMember to both
English and French Queue at the same time. I want this button to be smarter
to and to check and see if the Agent is logged into only English to only do
a Remove on English or if the Agent is only logged into French to only log
out French. Same goes for both, if both are logged in to log out both.
Currently I have this for Third Button:
exten = 99,1,Answer
exten =
99,n,RemoveQueueMember(900|Local/${CALLERID(num)}...@from-internal/n)
exten =
99,n,RemoveQueueMember(899|Local/${CALLERID(num)}...@from-internal/n)
exten = 99,n,Hangup
900 is English and 800 is Spanish Queue numbers.
P.S. Is there a way to do exten = s rather than exten = 99 as someone
from outside might find out about the 99 extension and try to log into
it? This is for an Aastra phone with XML support.
Thanks
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Re: [asterisk-users] a2billing
Turn on the voucher feature in System Settings and it will tell the user right after the PIN authentication or CLID authentication that their balance is below threshold and they should pay. -Bruce On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH i...@saudihome.com wrote: Not sure if a2billing can be shared here, but ill give a shot If the credit min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ; SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00] [18/10/2010 07:01:12]:[file:a2billing.php - line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff, activated, inuse, simultaccess, typepaid, creditlimit, language, removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate), expiredays, nbused, UNIX_TIMESTAMP(firstusedate), UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname, cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign, cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON tariff=cc_tariffgroup.id WHERE username='9971524976'] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 :: active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 :: language=en]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD : prepaid-zero-balance (cardnumber:9971524976)]] [18/10/2010 07:01:14]:[file:a2billing.php - line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY : callingcard_ivr_authenticate]] [18/10/2010 07:01:14]:[file:a2billing.php - line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED (cia_res:-2)]] [18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]] -- _ -- 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 -- _ -- 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] a2billing
I don't think voucher can be triggered to announce at certain threshold ONLY but it will be run everytime at the begining after PIN is asked for. By default it's set to: Press 8 to fill up with a voucher. System Settings is the last in the menu. -Bruce On Tue, Oct 19, 2010 at 2:31 AM, Baha @ SH i...@saudihome.com wrote: I am sorry , but where is System Settings??? And what is the parameter name? And also, id like to mention that the voucher is working, only when balance is below minimum balance it does not go to voucher ivr. Thanks, awaiting,,, *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, October 18, 2010 12:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] a2billing Turn on the voucher feature in System Settings and it will tell the user right after the PIN authentication or CLID authentication that their balance is below threshold and they should pay. -Bruce On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH i...@saudihome.com wrote: Not sure if a2billing can be shared here, but ill give a shot If the credit min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:601]:[CallerID:10001]:[CN:]:[ get_agi_request_parameter = 10001 ; SIP/10001-0005d08b ; 1287374472.907170 ; 9971524976 ; 00] [18/10/2010 07:01:12]:[file:a2billing.php - line:138]:[CallerID:10001]:[CN:]:[[ANSWER CALL]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1653]:[CallerID:10001]:[CN:]:[ - Account code - 9971524976] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1668]:[CallerID:10001]:[CN:9971524976]:[SELECT credit, tariff, activated, inuse, simultaccess, typepaid, creditlimit, language, removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate), expiredays, nbused, UNIX_TIMESTAMP(firstusedate), UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname, cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign, cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON tariff=cc_tariffgroup.id WHERE username='9971524976'] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1742]:[CallerID:10001]:[CN:9971524976]:[[SET LANGUAGE() en]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1745]:[CallerID:10001]:[CN:9971524976]:[[credit=0.0 :: tariff=1 :: active=t :: isused=0 :: simultaccess=1 :: typepaid=0 :: creditlimit=5 :: language=en]] [18/10/2010 07:01:12]:[file:Class.A2Billing.php - line:1777]:[CallerID:10001]:[CN:9971524976]:[[ERROR CHECK CARD : prepaid-zero-balance (cardnumber:9971524976)]] [18/10/2010 07:01:14]:[file:a2billing.php - line:155]:[CallerID:10001]:[CN:9971524976]:[[TRY : callingcard_ivr_authenticate]] [18/10/2010 07:01:14]:[file:a2billing.php - line:316]:[CallerID:10001]:[CN:9971524976]:[[AUTHENTICATION FAILED (cia_res:-2)]] [18/10/2010 07:01:14]:[CallerID:10001]:[CN:9971524976]:[[exit]] -- _ -- 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 -- _ -- 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 -- _ -- 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] fraud advice
Jeff, I suggest talking to your PSTN/VoIP provider. We had a large amount going through TATA communications and have not accepted their word for payment because they had a duty to not allow traffic if our credit went down to $1k while the calls charged were actually more than that. Unfortunately, probably there is no one you can complain to. But it also sickens me at how badly Asterisk is made to not cope with situations like this and worse than that is FreePBX. I suggest checking your contract terms with your provider as they might have some sort of restrictions. At the very least PSTN providers try to bring the price per minute lowered to their buy rate which is usually less than half of the original bill. Regards, Bruce On Thu, Oct 14, 2010 at 9:10 PM, Jeff LaCoursiere j...@sunfone.com wrote: Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was taken advantage of over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN trunk, and of course there was no warning until the bill came today. Unfortunately the bill only covered the first few days of this fiasco, and was only $700. I am afraid the one that is on the way will be tens of thousands. ONE CALL on the bill that just arrived was $200 (80 minutes to Sierra Leone). I'm sure this started out as a single scan. It must have been posted, because I have at least ten IP addresses now that were placing calls via the same peer. They are from all over the world. So what is the accepted procedure? I'm in the US Virgin Islands, so do I go to the FBI? Police? Is their some telecom fraud body to report such things to? Does any one ever get any relief from such events? I'm basically sick to my stomach right now. j -- _ -- 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 -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
And that is exactly what is done on the device: Nat=yes but Asterisk still sees the SIP packet coming in to register with a local IP an so it responds to a local IP which doesn't even exist on the Asterisk network. This is what frustrates me that it's not so straight forward to Asterisk to obtain the proper public IP of the device from the IP packet headers rather than the SIP packets. Thanks On Sat, Oct 9, 2010 at 8:27 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 10/08/2010 10:16 PM, bruce bruce wrote: I said previously, Asterisk receives packets like extens...@192.168.0.10 mailto:extens...@192.168.0.10 is trying to register to it. So, Asterisk sends out to local LAN an ACK which obviously is not right. SPA-2102 should send SIP request like extens...@123.123.123.123 mailto:extens...@123.123.123.123 (public IP). If you set 'nat=yes' in the sip.conf peer entry for that device in Asterisk, Asterisk will reply to the IP address and port number the REGISTER request was received from, not the address in the Contact header provided in the request itself. It will also record this address and port number as the location of that peer for future INVITE messages to be sent to it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Glad to hear it helped you Dennison. VPN is such a confusing beast to lots of people I think and hence the responses to this thread were all sort of work around and sometimes off-topic. It's also not well documented or maybe the feature is not widely used within the Asterisk community. I think it would be very good if some standard guidelines become available from the Asterisk side on this. Good day, Bruce On Wed, Oct 6, 2010 at 7:33 PM, Dennison Williams dennison.willi...@gmail.com wrote: On 09/22/2010 08:36 AM, Carlos Chavez wrote: Do you have a localnet statement in your sip.conf? That and using nat=no will make sure Asterisk does not replace the IP address in the Invite. I just wanted to give a +1 for this response. I am using openvpn to connect road warriors and remote offices to a central asterisk server. When setting up the configuration for the road warriors I created a new subnet for them, but forgot to include their subnet as a localnet directive in sip.conf. The result was that sip clients on the road warrior network would be able to register, but then when initiating a sip call the 200 response (to the INVITE from the client) from the asterisk server would include a contact address for some external ip that I did not recognize. This hint here allowed me to fix this bug, now calls from the road warrior subnet are coming in fine. Thanks! -- _ -- 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 -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
Kyle, Got an empty response from you. Were you intending to give your feedback? Regards, Bruce On Wed, Oct 6, 2010 at 8:10 PM, Kyle Kienapfel doctor.w...@gmail.comwrote: On Wed, Oct 6, 2010 at 12:50 PM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, This is such an annoying issue whenever it comes up. The sender and receive always receive the source public IP no matter what in the IP packets but then SIP packets go out with something like 192.168.0.20. In this instance, a Bell Canada DSL modem is installed and I saw the SPA-2102 register properly but only to fail later on due to sending it's LAN IP to the Asterisk server. So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also put the device on DMZ in the Bell Canada DSL modem and still the same issue. I am wondering when would manufacturers finally fully comply the SIP RFC?! I don't have the luxury to put SIP proxy, do a VPN, install a server or anything on client site. Diagram: Asterisk Server = Internet = Bell Canada Modem = SPA2102 Please send me your suggestions on how to fix this if you have this type of experience with SPA-2102 Thanks for the input, Bruce -- _ -- 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 Are you using stun? http://en.wikipedia.org/wiki/Session_Traversal_Utilities_for_NAT -- _ -- 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 -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
Thanks for the feedback. I said previously, Asterisk receives packets like extens...@192.168.0.10 is trying to register to it. So, Asterisk sends out to local LAN an ACK which obviously is not right. SPA-2102 should send SIP request like extens...@123.123.123.123 (public IP). Thanks On Fri, Oct 8, 2010 at 3:32 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 10/06/2010 02:50 PM, bruce bruce wrote: Hi Guys, This is such an annoying issue whenever it comes up. The sender and receive always receive the source public IP no matter what in the IP packets but then SIP packets go out with something like 192.168.0.20. In this instance, a Bell Canada DSL modem is installed and I saw the SPA-2102 register properly but only to fail later on due to sending it's LAN IP to the Asterisk server. So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also put the device on DMZ in the Bell Canada DSL modem and still the same issue. I am wondering when would manufacturers finally fully comply the SIP RFC?! Exactly how is this behavior non-compliant with the (sic) SIP RFC? There is nothing in any SIP RFC that mandates that a SIP UA must be aware of multiple IP addresses over which it can be reached, and select the proper one to include in SIP requests and responses. In fact, many SIP UAs, Asterisk included, work just fine behind NAT devices without ever knowing what their external IP addresses are. If you had actually described how the device failed, we might be able to tell you what you could do to resolve the problem. In general, Asterisk works just fine with endpoints that are behind NAT devices and never send their external IP addresses in their SIP messages... there are probably millions of devices working that way every day. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
Hi Guys, This is such an annoying issue whenever it comes up. The sender and receive always receive the source public IP no matter what in the IP packets but then SIP packets go out with something like 192.168.0.20. In this instance, a Bell Canada DSL modem is installed and I saw the SPA-2102 register properly but only to fail later on due to sending it's LAN IP to the Asterisk server. So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also put the device on DMZ in the Bell Canada DSL modem and still the same issue. I am wondering when would manufacturers finally fully comply the SIP RFC?! I don't have the luxury to put SIP proxy, do a VPN, install a server or anything on client site. Diagram: Asterisk Server = Internet = Bell Canada Modem = SPA2102 Please send me your suggestions on how to fix this if you have this type of experience with SPA-2102 Thanks for the input, Bruce -- _ -- 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] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
Thanks for the input guys. So, the IP is resolved only when IPTABLES is loaded or reloaded. Therefore, the best approach would be to ping the hostname every let's say 3 seconds and see if the IP is still the same and if it is then move on, otherwise update the iptables with the new IP address. This sounds it would work but I am not sure how fast DynDns can resolve the IP for me (delay) and I am looking to connect 40 PAP2T to this system. So, all in all that is 40 queries to DynDNS each 3 seconds. As I mentioned earlier, wouldn't it be more solid if I run my own Dynamic DNS server on the same box as Asterisk (is that even possible?) and what sort of other security holes would I be exposing doing that? Thanks again for all the great input. -Bruce On Sun, Oct 3, 2010 at 8:01 AM, Steve Edwards asterisk@sedwards.comwrote: On Sat, 2 Oct 2010, Kyle Kienapfel wrote: You're not going to be able to put a dns hostname in the iptables, but you could have a script that runs at times and gets the ip address for your dynamic hostname and allows that. Almost. You can put a host name in iptables, but it is resolved when loaded. You could restart iptables when your dynamic host name changes and it will be resolved correctly with the new IP address. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- _ -- 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] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support VPN and I don't have the luxury of putting a VPN client on the PAP2T side to connect back to the server. Is there any way I can DynDNS on the PAP2T to somehow notify the Asterisk Server that it's a safe device coming in? I do use fail2ban but that is not what I am looking for at this moment. And since the IP is dynamic on the PAP2T, I can't just use the iptables to let it in as it might change all a sudden. Thanks -- _ -- 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] Attempts to hack Asterisk - What do these lines means
Hi Everyone, Like always, here are IPs from China that try to hack an Asterisk server. Can someone please explain what is happening or what the hacker is trying to reach: 02/10/2010 11:10 SIP/113.105.152.51-00fb sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00fe sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00fc sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00fd sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-00ff sip sip sip s ANSWERED 13 02/10/2010 11:10 SIP/113.105.152.51-0100 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0101 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0102 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0103 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0104 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0105 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0106 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0107 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0108 sip sip sip s ANSWERED 13 02/10/2010 11:17 SIP/222.73.204.198-0109 sip sip sip s ANSWERED 13 Thanks -- _ -- 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] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
Hi, Can you please explain the DynDNS part. How would I put that in my Asterisk server as an identified party? Usually it comes to me with IP address (dynamic). Or do add something like this in sip_nat.conf: externip=mybox.dyndns.org localnet=192.168.0.0/255.255.255.0 ??? Thansk again, On Sat, Oct 2, 2010 at 2:59 PM, jon pounder j...@inline.net wrote: On 10/02/2010 02:56 PM, bruce bruce wrote: Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support VPN and I don't have the luxury of putting a VPN client on the PAP2T side to connect back to the server. Is there any way I can DynDNS on the PAP2T to somehow notify the Asterisk Server that it's a safe device coming in? I do use fail2ban but that is not what I am looking for at this moment. And since the IP is dynamic on the PAP2T, I can't just use the iptables to let it in as it might change all a sudden. Thanks do the dyndns on whatever router is in front of the pap2t or get some other box that supports it. other than that you are looking for some sort of magic bullet -- _ -- 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 -- _ -- 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] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
I was confusing the asterisk server side of sip_nat with the PAP2T. So, PAP2T can only register to DynDNS and that's all. What sort of a script would I be looking for? something to query DynDNS for the new IP of the device to add to firewall? This might however bring down time if inquiry is not successful. Or can I setup my own Dyndns server on the Asterisk server and have those PAP2T units registered to it and then work it from there when their IPs change? Thanks On Sat, Oct 2, 2010 at 3:32 PM, jon pounder j...@inline.net wrote: On 10/02/2010 03:31 PM, bruce bruce wrote: Hi, Can you please explain the DynDNS part. How would I put that in my Asterisk server as an identified party? Usually it comes to me with IP address (dynamic). Or do add something like this in sip_nat.conf: externip=mybox.dyndns.org localnet=192.168.0.0/255.255.255.0 every time the address changes you have to have some script to make the change in your firewall. ??? Thansk again, On Sat, Oct 2, 2010 at 2:59 PM, jon pounder j...@inline.net wrote: On 10/02/2010 02:56 PM, bruce bruce wrote: Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support VPN and I don't have the luxury of putting a VPN client on the PAP2T side to connect back to the server. Is there any way I can DynDNS on the PAP2T to somehow notify the Asterisk Server that it's a safe device coming in? I do use fail2ban but that is not what I am looking for at this moment. And since the IP is dynamic on the PAP2T, I can't just use the iptables to let it in as it might change all a sudden. Thanks do the dyndns on whatever router is in front of the pap2t or get some other box that supports it. other than that you are looking for some sort of magic bullet -- _ -- 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 -- _ -- 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 -- _ -- 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] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
Can't I in my ip tables just accept the pap2t.dyndns.org if that is bind to the PAP2T? do you think the devices comes in with it's external IP rather than the dyndns domain? Thanks On Sat, Oct 2, 2010 at 3:43 PM, bruce bruce bruceb...@gmail.com wrote: I was confusing the asterisk server side of sip_nat with the PAP2T. So, PAP2T can only register to DynDNS and that's all. What sort of a script would I be looking for? something to query DynDNS for the new IP of the device to add to firewall? This might however bring down time if inquiry is not successful. Or can I setup my own Dyndns server on the Asterisk server and have those PAP2T units registered to it and then work it from there when their IPs change? Thanks On Sat, Oct 2, 2010 at 3:32 PM, jon pounder j...@inline.net wrote: On 10/02/2010 03:31 PM, bruce bruce wrote: Hi, Can you please explain the DynDNS part. How would I put that in my Asterisk server as an identified party? Usually it comes to me with IP address (dynamic). Or do add something like this in sip_nat.conf: externip=mybox.dyndns.org localnet=192.168.0.0/255.255.255.0 every time the address changes you have to have some script to make the change in your firewall. ??? Thansk again, On Sat, Oct 2, 2010 at 2:59 PM, jon pounder j...@inline.net wrote: On 10/02/2010 02:56 PM, bruce bruce wrote: Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support VPN and I don't have the luxury of putting a VPN client on the PAP2T side to connect back to the server. Is there any way I can DynDNS on the PAP2T to somehow notify the Asterisk Server that it's a safe device coming in? I do use fail2ban but that is not what I am looking for at this moment. And since the IP is dynamic on the PAP2T, I can't just use the iptables to let it in as it might change all a sudden. Thanks do the dyndns on whatever router is in front of the pap2t or get some other box that supports it. other than that you are looking for some sort of magic bullet -- _ -- 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 -- _ -- 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 -- _ -- 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] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
Yeah, you are missing all :-) Sorry, read the thread again. On Sat, Oct 2, 2010 at 5:05 PM, sean darcy seandar...@gmail.com wrote: On 10/02/2010 04:09 PM, bruce bruce wrote: Can't I in my ip tables just accept the pap2t.dyndns.org http://pap2t.dyndns.org if that is bind to the PAP2T? do you think the devices comes in with it's external IP rather than the dyndns domain? Thanks On Sat, Oct 2, 2010 at 3:43 PM, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: I was confusing the asterisk server side of sip_nat with the PAP2T. So, PAP2T can only register to DynDNS and that's all. What sort of a script would I be looking for? something to query DynDNS for the new IP of the device to add to firewall? This might however bring down time if inquiry is not successful. Or can I setup my own Dyndns server on the Asterisk server and have those PAP2T units registered to it and then work it from there when their IPs change? Thanks On Sat, Oct 2, 2010 at 3:32 PM, jon pounder j...@inline.net mailto:j...@inline.net wrote: On 10/02/2010 03:31 PM, bruce bruce wrote: Hi, Can you please explain the DynDNS part. How would I put that in my Asterisk server as an identified party? Usually it comes to me with IP address (dynamic). Or do add something like this in sip_nat.conf: externip=mybox.dyndns.org http://mybox.dyndns.org localnet=192.168.0.0/255.255.255.0 http://192.168.0.0/255.255.255.0 every time the address changes you have to have some script to make the change in your firewall. ??? Thansk again, On Sat, Oct 2, 2010 at 2:59 PM, jon pounder j...@inline.net mailto:j...@inline.net wrote: On 10/02/2010 02:56 PM, bruce bruce wrote: Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support VPN and I don't have the luxury of putting a VPN client on the PAP2T side to connect back to the server. Is there any way I can DynDNS on the PAP2T to somehow notify the Asterisk Server that it's a safe device coming in? I do use fail2ban but that is not what I am looking for at this moment. And since the IP is dynamic on the PAP2T, I can't just use the iptables to let it in as it might change all a sudden. Thanks do the dyndns on whatever router is in front of the pap2t or get some other box that supports it. other than that you are looking for some sort of magic bullet -- _ -- 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 -- _ -- 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 I'm puzzled. Do you want the pap2t to connect directly to the internet? If so, then what does this have to do with asterisk or your box? If you want the pap2t to be connected to asterisk on your box, then the box has two interfaces. One is internal and open to a static address on pap2t, the other on the internet and subject to iptables. You can port forward to the pap2t. Or am I missing something? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
Thanks Roger. I will be trying this box to see what I can do. Otherwise, I'd probably have to find a list of all of the Rogers (The ISP providing internet to these boxes) IPs to at least limit the attacks to Rogers ISP. hmmm Or maybe secure is using DNS like this: sdlfjds...@$523k4j98sd7fkjh324#@$832.dyndns.org isn't that a security feature in itself? Thanks On Sat, Oct 2, 2010 at 4:32 PM, Roger Burton West ro...@firedrake.orgwrote: On Sat, Oct 02, 2010 at 04:09:33PM -0400, bruce bruce wrote: Can't I in my ip tables just accept the pap2t.dyndns.org if that is bind to the PAP2T? do you think the devices comes in with it's external IP rather than the dyndns domain? Yes. An IP datagram carries only the source and destination IP addresses, not the DNS names associated with them. Your firewall _may_ be able to accept a DNS name to block or allow rather than an IP address, but most don't, and doing so makes you vulnerable to DNS spoofing attacks. To go further would be thoroughly off-topic for this list. Roger -- _ -- 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 -- _ -- 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] Need to pick your brain for recommendation on using 2.5 or 3.5 HDDs for Asterisk server...
Thanks guys. Amazing feedback. Sounds like 2.5 is a better choice for being less in size (easier access for voicemail for example), as fast as 3.5 HDD in RPM, and allows 6 HDDs per Node which allows more RAID choice. However, it does come to be more expensive. Thanks again, On Mon, Sep 27, 2010 at 10:57 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: bruce bruce bruceb...@gmail.com writes: Other than the price difference (2.5 is more expensive and can't find many of the 1TB or so) is there any preference, advantage, or disadvatage of chosing 2.5 HDD or 3.5 when it comes to the server operations or Asterisk operation? There is no difference. Pick the server which offers the disk bandwidth and I/O's per second which you need. Do you really need 1TB disks? If you do, be careful what you place on those disks. Reading e.g. a voice mail or a speak off a large slow platter which is busy writing CDR's does not sound good at all. /Benny -- _ -- 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 -- _ -- 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] Need to pick your brain for recommendation on using 2.5 or 3.5 HDDs for Asterisk server...
Hi Everyone, I am stack between two identical systems (2U Twin2, 4 nodes, SuperMicro) servers that have the same exact specs except for HDDs. These nodes will all either have Asterisk installed with CentOS or will have Asterisk install in virtual environment. Option 1: *12* x 3.5 HDD (3 HDDs per node) Option 2: *24* x 2.5 HDD (6 HDDs per node) **both options come to the same price. Other than the price difference (2.5 is more expensive and can't find many of the 1TB or so) is there any preference, advantage, or disadvatage of chosing 2.5 HDD or 3.5 when it comes to the server operations or Asterisk operation? Each node of this server will be running CentOS 5.5 either in 64 or 32 bit + Asterisk or they will be used for virtual environment where multiple instance of Asterisk will be installed within CentOS XEN. Your input is much appreciated. Thanks -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce)
Thanks for the detailed info. Problem was solved by including Server B subnet as the localnet of the Server A (OpenVPN server) and setting each extension NAT=NO. Your points are good guides for future problem diagnoses. Thanks again, Bruce On Thu, Sep 23, 2010 at 1:56 PM, Dave Platt dpl...@radagast.org wrote: I don't think it's an endpoint issue. I think the SIP packet headers get over-written by the tunnel (openvpn) protocol. I'd be rather astonished if OpenVPN itself were responsible for this. As far as I know, OpenVPN doesn't do higher-level-protocol rewriting of any sort. It just provides the bit pipe through the tunnel. I'd suggest several other possible culprits: (1) Server B might be doing higher-level protocol rewriting (i.e. SIP border gateway stuff) prior to routing the SIP packets through the OpenVPN tunnel. This might happen if Server B were configured to use the Linux iptables features, with a SIP protocol module and Network Address Translation features. The fix would be to disable NAT and boundary processing in Server B's routing functions. (2) The SIP endpoints (phones) might be configured to discover their external address, via STUN or a similar mechanism. The fix would be to change the endpoint device configuration. I think you'll need to use Wireshark or a similar sniffer, to see what the SIP traffic looks like at several points along the path, so you can locate the earliest point at which the wrong address is in the SIP packet payload. Several examination points come to mind: - Right after the packet leaves the endpoint device. I'd suggest using a laptop running Wireshark as a passive packet sniffer... connect the endpoint device and the laptop to an Ethernet hub (not a switch!) and sniff the packets before any router gets its hands on them. - As the packets enter Server B - use Wireshark on Server B and have it tap into the incoming Ethernet interface. - As the packets are pushed out of Server B's routing layer into the OpenVPN tunnel. Use Wireshark to monitor the tap or tun virtual interface, to which the kernel transmits the packets that OpenVPN is to convey. - As the packets come out of the tap/tun device on Server A. In scenario (1) I described above, you'd see the packets be correct at the first and second Wireshark sniffing points, and incorrect at the third and fourth (i.e. the modifications are being performed in Server B's routing/NAT'ing layer). In Scenario (2), they'd be incorrect at every point, including just after they come out of the IP-phone. In the scenario you described, they'd be correct at the first, second, and third points, and wrong at the fourth. -- _ -- 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 -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Thanks for the feedback. I thought about that but it's not an option for me right now. Any other ways folks? Thanks On Wed, Sep 22, 2010 at 4:06 AM, Roger Burton West ro...@firedrake.orgwrote: On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote: I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs. Have you considered running Asterisk on Server B as well, and using IAX to trunk between them? This is working well for me. Roger -- _ -- 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 -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
I don't think it's an endpoint issue. I think the SIP packet headers get over-written by the tunnel (openvpn) protocol. Thanks, Bruce On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce bruceb...@gmail.com wrote: Any feed back is appreciated. Then configure you endpoints to use the 192.168.100.0/24 network. This is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is sending the INVITE message. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Thanks for that Carlos. I am playing with that right now. What do you suggest localnet should say? Server A = OpenVPN Server: localnet=127.0.01 localnet=192.168.100.0/255.255.255.0 Where 192.168.100.0 is the DHCPd subnet of Server B (the openvpn client) Server A doesn't have any localnet other than the loop back and then a Vnet to internet (public ip address). Thanks, Bruce On Wed, Sep 22, 2010 at 11:36 AM, Carlos Chavez cur...@telecomabmex.comwrote: Do you have a localnet statement in your sip.conf? That and using nat=no will make sure Asterisk does not replace the IP address in the Invite. On Wed, 2010-09-22 at 01:27 -0400, bruce bruce wrote: Hi Everyone, I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs. So, the tunnel is established nicely and everyone can ping others. sip show peers shows the local subnet of the SIP Phones registered (192.168.100.0/24). But there is the old bad one-way audio. Calls also drop after few seconds. In the SIP debug I can see that asterisk uses it's external public IP address to communicate to endpoints that are known to it as the 192.168.100.0/24 endpoints and the endpoints identify themselves with the OpenVPN tunnel IP address scheme in one part of the sip handshake. How can this be fixed? After all, with the OpenVPN this should all look like an internal network to Asterisk. I have added my comments followed by # to lines below that are problematic. --- SIP read from UDP:192.168.100.5:5060 ---#This line is good as it uses the local DHCP supplied network address scheme INVITE sip:2...@172.16.0.1:5060 SIP/2.0 #This line is BAD. Why are we inviting Ext. 203 with it's OpenVPN IP while it's on the same network of 192.168.50.0/24 as 202? Via: SIP/2.0/UDP 192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6 Max-Forwards: 70 From: SIP Phone - Ext. 202 sip:2...@172.16.0.1:5060;tag=6d6f8c4226 #BAD line again. Should be SIP:2...@192.168.100.6sip%3a...@192.168.100.6 To: 203 sip:2...@172.16.0.1:5060 #Bad again Call-ID: 43af67a634e06e75 CSeq: 32058 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: SIP Phone - Ext. 202 sip:2...@192.168.50.5:5060;transport=udp; +sip.instance=urn:uuid:--1000-8000-00085D25B72F Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.5.2.1500 Content-Type: application/sdp Content-Length: 594 Basically the phones should only send with FROM their local 192.168.100.0/24 address and Asterisk should only send ANSWER and ACK back to 192.168.100.0/24 rather than sending it to 172.16.0.0/24 (which is the openvpn client ip). Once above is fixed, I think all the audio and call cut will go away. I hate to use a sip proxy in this situation since I already have an openvpn connection. Any feed back is appreciated. Thanks, -- _ -- 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 -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- 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 -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Thanks, but Carlos Chavez was right on point. This fixed the problem: externip=123.123.123.123 localnet=192.168.100.0/255.255.255.0 nat=no in each extension. Maybe combination of both or only the localnet just fixed it. Thanks, Bruce On Wed, Sep 22, 2010 at 1:35 PM, Steve Edwards asterisk@sedwards.comwrote: Un-top-posting... On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce bruceb...@gmail.com wrote: Any feed back is appreciated. On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger paul.belan...@polybeacon.com wrote: Then configure you endpoints to use the 192.168.100.0/24 network. This is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is sending the INVITE message. On Wed, 22 Sep 2010, bruce bruce wrote: I don't think it's an endpoint issue. I think the SIP packet headers get over-written by the tunnel (openvpn) protocol. Would wireshark shed some light? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Calls are not going outside of the network. I had to setup up the subnet of the other side (openvpn client) as the localnet of the Asterisk server for Asterisk to not handle it with NAT or hand shake it with external IP. Thanks, -Bruce On Wed, Sep 22, 2010 at 1:58 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Sep 22, 2010 at 1:46 PM, bruce bruce bruceb...@gmail.com wrote: Thanks, but Carlos Chavez was right on point. This fixed the problem: externip=123.123.123.123 localnet=192.168.100.0/255.255.255.0 nat=no in each extension. So now I am confused, If you have a VPN setup between sites, why are calls going outside the VPN? Or do you have remote agents that are not using a VPN? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- _ -- 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] Installing Asterisk + FreePBX from Repsitory spits out some warnings and errors for ever
Hello, This is what what I see after a Yum install asterisk16 asterisk16-config freepbx: Use of uninitialized value in string ne at /var/www/html/panel/op_server.plline 4997. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5439. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5440. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5441. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5442. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 5444. Usually these error stay on the /var/log/messages for ever. I mean they repeat. Is there a problem with these? Thanks -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Hi Everyone, I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs. So, the tunnel is established nicely and everyone can ping others. sip show peers shows the local subnet of the SIP Phones registered (192.168.100.0/24 ). But there is the old bad one-way audio. Calls also drop after few seconds. In the SIP debug I can see that asterisk uses it's external public IP address to communicate to endpoints that are known to it as the 192.168.100.0/24 endpoints and the endpoints identify themselves with the OpenVPN tunnel IP address scheme in one part of the sip handshake. How can this be fixed? After all, with the OpenVPN this should all look like an internal network to Asterisk. I have added my comments followed by # to lines below that are problematic. --- SIP read from UDP:192.168.100.5:5060 ---#This line is good as it uses the local DHCP supplied network address scheme INVITE sip:2...@172.16.0.1:5060 SIP/2.0 #This line is BAD. Why are we inviting Ext. 203 with it's OpenVPN IP while it's on the same network of 192.168.50.0/24 as 202? Via: SIP/2.0/UDP 192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6 Max-Forwards: 70 From: SIP Phone - Ext. 202 sip:2...@172.16.0.1:5060;tag=6d6f8c4226 #BAD line again. Should be SIP:2...@192.168.100.6 sip%3a...@192.168.100.6 To: 203 sip:2...@172.16.0.1:5060 #Bad again Call-ID: 43af67a634e06e75 CSeq: 32058 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: SIP Phone - Ext. 202 sip:2...@192.168.50.5:5060 ;transport=udp;+sip.instance=urn:uuid:--1000-8000-00085D25B72F Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.5.2.1500 Content-Type: application/sdp Content-Length: 594 Basically the phones should only send with FROM their local 192.168.100.0/24address and Asterisk should only send ANSWER and ACK back to 192.168.100.0/24 rather than sending it to 172.16.0.0/24 (which is the openvpn client ip). Once above is fixed, I think all the audio and call cut will go away. I hate to use a sip proxy in this situation since I already have an openvpn connection. Any feed back is appreciated. Thanks, -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Thanks guys. I wasn't able to collect enough SIP debug as the problem was resolved as I was testing different configuration for the trunk. Probably a change on the provider side. John Novack: Unfortunately, it seems that this list has a non-stop list of people who like to stir up things or try to censor people who bring legit questions without the consideration that they are not moderators of the list at any level. They forget to remember that AsteriskNow uses FreePBX as well and that Asterisk IS the underlying technology for all the flavours. Thanks for the feedback. Most I was able to collect was that: - if the trunk configuration even included context=from-pstn, the CLI would show Executing .@from-sip-external. - if SIP Anonymous was set to YES then the [from-sip-external] context would match the peer to the right trunk defined as that is what is expected of that context from the code. If SIP Anonymous was set to off @from-sip-external is set to go to ss-noservice. - Later on when the calls resumed and the problem was fixed, calls were coming in with Executing @from-pstn which should have always been the case regardless of the SIP Anonymous or not. I was puzzled because the FISRT line of the CLI was the Executing @from-pstn or Executing .@from-sip-external and that made a world of difference. The latter one not working. I just couldn't pinpoint where FreePBX failed to read the context=from-pstn. If it was something to do with the MySQL database or of parsing the _custom.conf files as the problem was fixed all a sudden. I guess now I have to wait and see if it comes back. Thanks, On Tue, Sep 14, 2010 at 10:47 AM, Zeeshan Zakaria zisha...@gmail.comwrote: This might help to answer poster's question. It tells how the allow anonymous sip connections work in FreePBX, and shows the code. http://www.geekzone.co.nz/sbiddle/7183 http://www.geekzone.co.nz/sbiddle/7183-- Zeeshan On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Poster is having problem when he disallows anonymous sip peers. Do you know at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet seen the dialplan for FreePBX. It's very simple to find the actually issue, if the OP does the following: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt The attached the debug log to thread. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- Zeeshan A Zakaria -- _ -- 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 -- _ -- 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] Anyone can share their experience about Thomson TG784 wireless router/ATA?
Hi Everyone, Wondering if any of you folks ever had trouble using *Thomson TG784http://www.w7forums.com/thomson-tg784-t1199.html *DSL/Wireless Router/FXS ATA combo? I am looking to use this to connect users from home to a hosted Asterisk PBX. Any and all inputs are appreciated. Thanks -- _ -- 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] Cisco or Linksys WRP400 reliability?
Hi Everyone, I see one long post on Cisco community forum where everyone including ISPs are complaining about silence on FXS port, reboots, frozen state, etcof WRP400. This is the a wireless router + 2 FXS combo box. I am looking to use this for home user to connect to hosted Asterisk PBX. I am looking for some feedback from the community as to how stable the unit is - or if it is stable at all? Thanks for your input. -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks -- _ -- 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] VoIP friendly Internet providers in Dallas and Philadelphia
Hi Everyone, My experience is only with the Canadian providers. What options/providers are there in Dallas and Philadelphia other than Verizon when it comes to internet? Something in the order of at least 10mbps down and up - I understand that and higher bandwidths are easily available in USA due to vast fiber networks? The connection will be replacing a T1 and will be support VPN connections to connect office for VoIP. Thanks -- _ -- 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] openvz
1- I am interested in this as well. Looking into Proxmox as it provides a nice interface (do you guys know of any other good one?) 2- Would the conference calls be fine as well? I understanding Asterisk 1.6.x uses a kernel timing source now a days so that ztdummy is not needed anymore? 3- Would installing from yum repository be just fine? Thanks On Fri, Sep 3, 2010 at 10:31 AM, mattias m...@mjw.se wrote: Outlook? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 03, 2010 3:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] openvz From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias Subject: [asterisk-users] openvz Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi Blind Answer - you should be able to; Asterisk doesn't rebuild the kernel. You might have to get some kernel source using ZYPPER (in caps so Outlook express doesn't change it to zipper). -- _ -- 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 -- _ -- 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 -- _ -- 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] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?
Hi Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Server B can place calls to Server A but when trying to place calls from Server A to Server B this is what I am getting: pbx*CLI originate iax2/mel/14161234567 extension s...@null Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3ms SCall: 16389 DCall: 0 [72.72.72.72:4569] VERSION : 2 CALLED NUMBER : 14161234567 CODEC_PREFS : (gsm) CALLING NUMBER : CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: LANGUAGE: en USERNAME: mel FORMAT : 2 CAPABILITY : 57346 ADSICPE : 2 DATE TIME : 2010-09-02 02:14:50 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 3ms SCall: 1 DCall: 16389 [72.72.72.72:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 16389 DCall: 1 [72.72.72.72:4569] -- Hungup 'IAX2/mel-16389' As you can see above, Subclass: REJECT comes back wtih no cause code. Usually there is a cause code to debug but in this case there is no Cause code. Trunks on both sides in the context=from-internal so it's not an Inbound Route issue. Any pointers are much appreciated. -Bruce -- _ -- 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] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?
I'd rather find the problem than upgrade blindly. Upgrading not always solves the problem and has the potential to break other things. Thanks for the offer though. Bug # 16753 applies. call token not required was set in the trunk and problem solved. There is not warning for this in iax2 debug but there is in core set debug. That's a petty. -Bruce On Thu, Sep 2, 2010 at 3:19 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Sep 02, 2010 at 02:21:11AM -0400, bruce bruce wrote: Hi Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Any chance you could upgrade that? Elastix has newer versions of Asterisk, for starters. Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Is it configured to talk to old IAX2 peers? http://downloads.asterisk.org/pub/security/IAX2-security.html http://downloads.asterisk.org/pub/security/AST-2009-006.html -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- 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 -- _ -- 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] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?
Maybe dvossel can re-open issue # 16753 and fix the warning to show on iax2 debug as well along with core set debug like all other warnings. That way it's straight forward. That ticket shouldn't have been closed without a fix. On Thu, Sep 2, 2010 at 4:11 AM, bruce bruce bruceb...@gmail.com wrote: I'd rather find the problem than upgrade blindly. Upgrading not always solves the problem and has the potential to break other things. Thanks for the offer though. Bug # 16753 applies. call token not required was set in the trunk and problem solved. There is not warning for this in iax2 debug but there is in core set debug. That's a petty. -Bruce On Thu, Sep 2, 2010 at 3:19 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Sep 02, 2010 at 02:21:11AM -0400, bruce bruce wrote: Hi Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Any chance you could upgrade that? Elastix has newer versions of Asterisk, for starters. Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Is it configured to talk to old IAX2 peers? http://downloads.asterisk.org/pub/security/IAX2-security.html http://downloads.asterisk.org/pub/security/AST-2009-006.html -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- 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 -- _ -- 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] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?
I am not interested in open source solutions. I want to know how much the propriety systems cost in terms of licensing. Specially Avaya now a days per extension. Exclusive or Inclusive of the hardware for 10 agents as noted. Thanks On Thu, Sep 2, 2010 at 8:18 AM, Muhammad Shomail Haider msh0...@gmail.comwrote: Hi Bruce, It all depends what exactly you are in need of. A basic call center solution will only cost $500 exclusive of hardware, depending on your need you will have to decide what type of servers you need or weather you would have handsets or softphones, type of headgears you want, kind of workstation you will need. I work for a company that provides open source call center solution. You can visit the website www.crystalconsulting.pk or if you want more detail you can email the detail of the requirements on sa...@crystalconsulting.pkor you can email me and I can revert back to you with detail. Regards, Shomail On Fri, Aug 27, 2010 at 2:03 PM, justmun...@gmail.com wrote: Hi Everyone, Just a quick estimate of what Call Center Software/Hardware providers charge now a days for a 10 seat and 20 seat with upfront costs and monthly licensing cost? Thanks, -- Muhammad Shomail Haider www.shomail.blogspot.com www.facebook.com/shomail www.twitter.com/shomail www.linkedin.com/in/shomail -- _ -- 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] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost?
Thanks Don for clarification. There are lots of people on this list that hastily decide to answer without even reading a post properly. I am sure they won't even read the follow-ups. They just talk for the sake of talking. Sickens me! Please note the subject line in my original post: To compete with Avaya - What are their current cost? My question is specifically related to Avaya and other propriety call centers because I want to compete with them with Asterisk. If you know recent prices please post back. If not, don't bother. Also, please do not private message me. I will move this post to Biz List as I just noted I posted to wrong listing. Thanks to those who tried meaningful posts. On Thu, Sep 2, 2010 at 3:06 PM, Don Kelly d...@donkelly.biz wrote: It could be that I'm entirely confused, but I think he asked what people are paying for Avaya solutions--so he'd know what competitive pricing would be for the open source solution he's prepared to offer. When someone replied with open-source suggestions, he pointed out that that was not the information he was looking for. He did not say that he's not interested in providing open source solutions for his clients. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, September 02, 2010 1:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost? He doesn't deserve the responses, but it seems that boundaries are being pushed in both sides of the response. If he thinks he's on the biz list, that's one thing, but in the purely open discussion, don't be dissing open source either. -- _ -- 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 -- _ -- 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 -- _ -- 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] Quick Question - Jabra Headset and Aastra 53i - Where is the speaker/headset enable setting on Aastra UI?
Thanks. That is one thing I really HATE about AASTRA - them confusing the user with providing different setting level on the WEB UI and the PHONE UI - very stupid. But thank you and it works just fine. -Bruce On Thu, Aug 26, 2010 at 4:09 AM, Gareth Blades list-aster...@skycomuk.comwrote: bruce bruce wrote: Hi Everyone, I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i receiver port and I get a tone. But when I connect it to the headset port there is no tone. I am running firmware 2.4 and I can't seem to find that DHSG, EHS or whatever the setting maybe called to enable to get this headset work with the phone. Can anyone quickly tell me where the audio options are on this phone? Thanks, Bruce Press the tools button. Press 2 (Preferences) Press 5 (Set Audio) You now have 3 options to set the handsfree/headset mode, mic volume and DHSG -- _ -- 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 -- _ -- 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] OrderlyStats or QueueMetrics
Hi Everyone, There are a few things I like in OrderlyStats, specially some graph presentations and the fact that if agent puts someone on HOLD or PAUSE it shows fine. 1 -But I see a lot of similarities in pricing, descriptions, wording on both sites. Were these same projects forked out? or is it still both owned by the same company? 2- What are the main differences between the two? - so that we can make a better choice. 3- What part of these two products are open source? and what part is not opensource? - just the license part? 4- Any better alternatives? Thanks -- _ -- 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] Quick Question - Jabra Headset and Aastra 53i - Where is the speaker/headset enable setting on Aastra UI?
Hi Everyone, I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i receiver port and I get a tone. But when I connect it to the headset port there is no tone. I am running firmware 2.4 and I can't seem to find that DHSG, EHS or whatever the setting maybe called to enable to get this headset work with the phone. Can anyone quickly tell me where the audio options are on this phone? Thanks, Bruce -- _ -- 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] Opensource Speech recognition for Asterisk
Bob, Both ZanziIVR and Speechforge have similar look web pages. I guess you have used one of those to get the speech going as this link: http://scribblej.com/svn/ probably is not the full thing. These seem like practical project. Thanks for pointing out. This is what I was looking for. Now starts the try to get these installed and tested. Thanks, Bruce On Tue, Aug 24, 2010 at 7:30 AM, Bob Kleiner bob.klei...@gmail.com wrote: Thanks guys. A lot of info here :-) I am wondering if anyone followed this and it was working for them: http://scribblej.com/svn/ ??? Hello Bruce We successfully deployed it and now saving thousands on commercial ASR ports. It seems users are rather happy with it. The recognition seems pretty accurate. Of course it has it's own limitations but so any other technology. It will not hurt if some of your users will benefit from ASR. I am not looking for anything fancy. The basic yes, no, dialing a number, asking for agent, etc...out of which probably the hardest is a 10 digit number to be asked to be dialed. Yes, that should work. It also supports JSGF grammars, so you should be able to recognize digit strings easily. And if you want something serious, there are at least two open source products providing ASR over standard MRCP protocol. They also use CMUSphinx, so provide the same accuracy Zanzibar http://www.spokentech.org/writing-speechlets.html Cairo http://www.speechforge.org/ Though Cairo is a bit dated. -- _ -- 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 -- _ -- 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] Opensource Speech recognition for Asterisk
Thanks guys. A lot of info here :-) I am wondering if anyone followed this and it was working for them: http://scribblej.com/svn/ ??? I am not looking for anything fancy. The basic yes, no, dialing a number, asking for agent, etc...out of which probably the hardest is a 10 digit number to be asked to be dialed. Thanks On Sun, Aug 22, 2010 at 2:30 AM, Nickolay V. Shmyrev nshmy...@nexiwave.comwrote: Hi Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the theoretically should work ones! Hi I definitely suggest you to try CMU Sphinx connector for Asterisk. You can find all required information here http://scribblej.com/svn/ If you need any help with setup, just ask. -- Nexiwave - Speech Indexing Solution For Call Centers http://nexiwave.com -- _ -- 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 -- _ -- 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] Opensource Speech recognition for Asterisk
Hi Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the theoretically should work ones! Thanks -- _ -- 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] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?
That is set and here is what I get: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 01217 DCall: 0 [44.55.66.77:4569] USERNAME: 9988 REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 01217 DCall: 1 [44.55.66.77:4569] Any other suggestions. Anyone with a working pfsense configuration that can share with me? Thanks, Bruce On Wed, Aug 18, 2010 at 3:42 AM, Nasir Iqbal na...@ictinnovations.comwrote: Hi, Use requirecalltoken=no in your peer configuration Regards On Wed, Aug 11, 2010 at 4:28 AM, bruce bruce bruceb...@gmail.com wrote: Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00130 DCall: 0 [64.229.229.111:64823] * * USERNAME: 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK* * Timestamp: 3ms SCall: 00130 DCall: 1 [64.229.229.111:64823] * So, all the packets are coming in, but there is no Tx response. Is that normal and is that how IAX2 works according to RFC to not respond back? I have checked my firewall and all is set fine. I have any WAN address to come in through port 4569 to map to the server and it worked last week but now it doesn't. Any suggestions? Thanks -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- 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 -- _ -- 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] Asterisk with Motorola Canopy
Hi Everyone, Can anyone share their experience with Motorola Canopy solution deployment and Asterisk? Is this a good fit? Thanks, Bruce -- _ -- 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] Pfsense and IAX2 - What is the proper firewall NAT setup?
Hi Everyone, Just trying to connect the Zoiper Communicator to connect to Asterisk which is behind Pfsense. Here is what I get at debug and it doesn't register. Error code 16. Can someone please let me know their firewall, NAT, outbound 1-to-1 pfsense settings as it seems to me I am doing something wrong on the firewall? *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00426 DCall: 0 [44.55.66.77:4569]* * USERNAME: 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00427 DCall: 0 [44.55.66.77:4569]* * USERNAME: 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK * * Timestamp: 3ms SCall: 00427 DCall: 1 [44.55.66.77:4569]* 44.55.66.77 is my client IP. I see no Tx packets. What is happening? Thanks for sharing, Bruce -- _ -- 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 determine which party hangup th e call? cause of Hang-up needed.
Sorry, I am not following:
***read the value of var ${HANGUPCAUSE} next line to dial command.*
*
*
*Where is that value? Next to dial you mean right when the call was placed?
or check next few lines to find HANGUP cause?*
*
*
*Note: This is using ZAP (analogue) and not PRI.*
*
*
*Thanks,*
*Bruce
*
On Wed, Aug 11, 2010 at 12:33 AM, Faisal Hanif fai...@vopium.com wrote:
read the value of var ${HANGUPCAUSE} next line to dial command.
Regards,
Faisal Hanif
*VoIP Manager
***Vopium A/S
On 8/10/2010 9:51 PM, bruce bruce wrote:
Hi Everyone
Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell
Canada.
User claims that call hangup without any interferance to the phone set.
Is there ANYWAY to find out which party hang-up the call or if the call was
cut-off due to other reasons?
I checked the *asteriskcdrb* table and it's pretty much useless in this
case as it only logs the duration and other properties but not cause of the
Hangup.
/var/log/asterisk/full
[Jul 10 10:37:02] VERBOSE[29366] logger.c: == Manager 'admin' logged off
from 127.0.0.1
[Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing
[...@macro-dialout-trunk:1] Macro(SIP/1007-069a, hangupcall|) in new
stack
[Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing
[...@macro-hangupcall:1] GotoIf(SIP/1007-069a, 1?skiprg) in new
stack
[Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Goto
(macro-hangupcall,s,4)
Thanks,
Bruce
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[asterisk-users] How to determine which party hangup th e call? cause of Hang-up needed.
Hi Everyone Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell Canada. User claims that call hangup without any interferance to the phone set. Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? I checked the *asteriskcdrb* table and it's pretty much useless in this case as it only logs the duration and other properties but not cause of the Hangup. /var/log/asterisk/full [Jul 10 10:37:02] VERBOSE[29366] logger.c: == Manager 'admin' logged off from 127.0.0.1 [Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing [...@macro-dialout-trunk:1] Macro(SIP/1007-069a, hangupcall|) in new stack [Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing [...@macro-hangupcall:1] GotoIf(SIP/1007-069a, 1?skiprg) in new stack [Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Goto (macro-hangupcall,s,4) Thanks, Bruce -- _ -- 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] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?
Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00130 DCall: 0 [64.229.229.111:64823]* * USERNAME: 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK * * Timestamp: 3ms SCall: 00130 DCall: 1 [64.229.229.111:64823]* So, all the packets are coming in, but there is no Tx response. Is that normal and is that how IAX2 works according to RFC to not respond back? I have checked my firewall and all is set fine. I have any WAN address to come in through port 4569 to map to the server and it worked last week but now it doesn't. Any suggestions? Thanks -- _ -- 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] What do you use for Invoicing?
I agree but the mentioned software is not opensource. My conditions clearly included opensource. On Tue, Aug 3, 2010 at 12:35 AM, Nick Brown n...@ipera.com.au wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Tuesday, 3 August 2010 1:58 PM *To:* j...@sunfone.com; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] What do you use for Invoicing? Maybe good but the first look brought me to a Pay version. Doesn't satisfy the opensource condition. thanks, Open Source software does not necessarily mean free software. Nick. -- _ -- 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 -- _ -- 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] What do you use for Invoicing?
Oh, you seem to be right on. It's actually an install of Elastix. I will be testing this for sure. Hope it doesn't do any damages though. I guess the installation material is inside the tar ball? Thanks On Tue, Aug 3, 2010 at 2:01 AM, Nasir Iqbal na...@ictinnovations.comwrote: Hi Bruce, We have build an Invoicing module (ICTInovice) for Elastix. It is Free, Open Source, Generate PDF Invoices, and can mail invoices to clients! You can download it from http://sourceforge.net/projects/ictinvoice/ http://sourceforge.net/projects/ictinvoice/Note: Currently ICTInvoice only work with Elastix 1.6 Regards On Mon, Aug 2, 2010 at 11:26 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Thanks -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- 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 -- _ -- 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] What do you use for Invoicing?
Yep, I seen that. That is probably the closet thing but looking at he interface it makes me not try to install it. Maybe too complicated. I wouldn't want to send customer the whole CDRs but rather a nice Bill like the telco sends out. I am currently toying with NCH Invoicing. Those guys make a software for anything and everything. Thanks, Bruce On Tue, Aug 3, 2010 at 9:17 AM, Zeeshan Zakaria zisha...@gmail.com wrote: I know someone who uses a billing solution called 'freeside', and is happy with it. Personally I developed my own solution because none could satisfy my needs. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-03 2:34 AM, ad...@3a.hu wrote: Hi, On 08-02-2010 20:55, Gordon Henderson wrote: I generated invoices with PHP code - it uses a LaTe... well, then i must be a geek too, because i also decided to throw some php code together to generate PDFs from sql. It was just quicker this way rather than looking and trying a buch of other software. I'm not sure how other (real) softwares work but since then i'm not spending even a minute with invoices, it's all in crontab. regards adam -- _ -- Bandwidth and Colocati... -- _ -- 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 -- _ -- 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] Aastra phones occasionally show No Service - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
Hi Mike, I am putting the phones on AC Adapter now as I am suspecting the Linksys POE switch. Once that test is done and if problem still presists, I will be enabling DHCPMasq and also set the SIP registration time to 1 second on the phone UI. -Bruce On Tue, Aug 3, 2010 at 1:13 PM, Mike l...@net-wall.com wrote: Hi Bruce, Did you ever get a working solution and confirm the underlying issue ? I am having the same issue on a set of phones, my next step is replacing the router, but I was wondering if you found something else. Regards, Mike *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Thursday, July 29, 2010 22:36 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Aastra phones occasionally show No Service - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones? Hi Everyone, I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones occasionally go into No Service mode. The POE switch doesn't seem to be the problem as it's tested fine. I think the router sometimes gives up and comes back quickly. Or something of that nature. However, the connections are maintained if a call is going on because there are peer to peer connections between the phones in a network. Anyhow, if the phones are restarted they work fine. So, I was looking around the Aastra Admin UI to find any timer to lower it to 1 second to check and make sure the device always has an ip but I can't seem to find anything other than LLDP which is set at 30 and I don't think that will be of any help. I did a test where I would disconnect the router from the switch and after a while phones go into No Service but if I plug it back into the switch the phones do not come back right away. Maybe something should be dialed on the phone or wait long time or restart it to work again. Any work around? Thanks a lot -- _ -- 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 -- _ -- 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] What do you use for Invoicing?
Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Thanks -- _ -- 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] What do you use for Invoicing?
Maybe good but the first look brought me to a Pay version. Doesn't satisfy the opensource condition. thanks, On Mon, Aug 2, 2010 at 2:39 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Its partially open source (you get the source to everything but the financial routines), and it runs on Unix rather than Windows, though you do have a web interface. Checkout BillMax: www.billmax.com They have some extensions that create PDF invoices in telecom style. Its pretty powerful otherwise for doing any kind of recurring billing. I wrote the initial version, but I am not associated with the company anymore. j Thanks -- _ -- 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 -- _ -- 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 -- _ -- 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] What do you use for Invoicing?
Sorry, I am not familiar with them. Wondering if any full package system out there does the job. Thanks On Mon, Aug 2, 2010 at 2:55 PM, Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net wrote: On Mon, 2 Aug 2010, bruce bruce wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. I generated invoices with PHP code - it uses a LaTeX template which it fills in the gaps, then feeds it through LaTeX and dvi2pdf to generate PDFs. Bit of a geek solution though. Gordon -- _ -- 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 -- _ -- 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] Aastra phones occasionally show No Service - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
Now that I check again, I see that DNSMasq for DHCP and DNSMasq for DNS is NOT enabled. Which one should I enable and also can you please detail what DNSMasq really does? With my situation, only some of the phones go off randomly. But it is a few specific extensions I think. Not the server itself. At best, I can think of a cable or two jacked improperly into the patch panel and that's all which MAYBE the cause for failing of DNS. Thanks, Bruce On Fri, Jul 30, 2010 at 12:03 PM, bruce bruce bruceb...@gmail.com wrote: DNSMasq has always been enabled. It's only one check box and if I didn't have it enabled phones won't work. So, that is set. Any other suggestions? including things regarding DNSMasq? Thanks On Fri, Jul 30, 2010 at 11:04 AM, Dave Cotton dcot...@linuxautrement.comwrote: On 30/07/10 16:15, bruce bruce wrote: Adria, How can I build a dns cache into my lan? I am using a Linksys 48 port POE switch and running a micro DD-WRT firmware on a linksys router. DD-WRT supports DNSMasq which would do exactly what you need. DC -- _ -- 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 -- _ -- 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] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
2 users. So, it's probably never used as a free version as probably there are no 2 seat call centers that can survive this economy. But, it should great for testing. On Sat, Jul 31, 2010 at 10:28 AM, Leif Madsen leif.mad...@asteriskdocs.orgwrote: On 7/30/2010 5:49 AM, Lenz Emilitri wrote: QueueMetrics is actually free (as in beer) for very small call centers and individual hackers. Oh really! I didn't know that! Very nice. What is considered a small call centre? Are we talking up to around 5 agents or something? Is there a limit on the number of queues? (I'm sure there is a page on the website that answers most of these questions, heh :)) Leif Madsen. -- _ -- 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 -- _ -- 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] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Thank Martin, That makes absolute sense. I have turned busy detect off for now and haven't heard complains or lines remaining open for a Day. I am in Canada. I just checked chan_dahdi.conf and I don't see callprogress there at all. So, I guess the lines are fine for hanging up by themselves. Hope this doesn't give me probs in future. Thanks, Bruce On Fri, Jul 30, 2010 at 6:18 AM, Martin asteriskl...@callthem.info wrote: Either turn off busydetect or increase the busycount to 5-7 or even more ... (10 should be conservative) busydetect looks for cadence or patterns of the same length ... beep on [X ms] beep off [Y ms] so you can afford to increase busycount and have a few second longer calls / the line is kept longer offhook but you don't get false busy detections Also in US/Canada callprogress will do a better job then busydetect since it looks for specific frequencies of the busy signal and not just noise/beep then silence ... If you're somewhere else then you can hire a coder to tweak callprogress algorithm to your country's busy signal frequencies ... Just record the busy signal with ztmonitor and send to someone for code patch... regards Martin On Wed, Jul 28, 2010 at 4:54 PM, bruce bruce bruceb...@gmail.com wrote: Hmmwhat about call waiting? You mean, when a call comes in on that specific line, it generate two beep tones and hence the system hangs up thinking it's end of the call? Interesting!!! If it is call-waiting do I have to set all of the following off for it to not give me problem again: callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes busydetect=yes busycount=3 Please elaborate a bit if I am off-topic. Regards, Bruce On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Subject: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem? I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: Could be callwaiting? -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
Is it easy to install along with FreePBX as well? Thanks On Fri, Jul 30, 2010 at 5:49 AM, Lenz Emilitri lenz.lo...@gmail.com wrote: QueueMetrics is actually free (as in beer) for very small call centers and individual hackers. l. 2010/7/28 Zeeshan Zakaria zisha...@gmail.com There is none for free. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-27 6:12 PM, bruce bruce bruceb...@gmail.com wrote: :-) I knew someone would bring up FreePBX. I have FreePBX installed and it's not good for Queues at all. It's using the reporting tool from Areski and Areski has recently released an upgrade to it which again is not what I want. There are few other programs that do this but really none that are neat in interface or useful in features. I guess no one else has any thoughts on this? Maybe there is none available? Thanks, Bruce On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg dbackeb...@gmail.com wrote: On Mon, Jul 26... -- _ -- 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 -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- 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 -- _ -- 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] Aastra phones occasionally show No Service - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
Adria, How can I build a dns cache into my lan? I am using a Linksys 48 port POE switch and running a micro DD-WRT firmware on a linksys router. Gareth, I think the registration time is part of the reason. I have lowered it less than 10 seconds. Thanks On Fri, Jul 30, 2010 at 8:21 AM, Adrià Vidal adriavi...@gmail.com wrote: try to have a dns cache into your LAN, Aastra phone are prone to fail when have any little DNS error. -- -- Adrià Vidal -- _ -- 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 -- _ -- 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] Aastra phones occasionally show No Service - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
DNSMasq has always been enabled. It's only one check box and if I didn't have it enabled phones won't work. So, that is set. Any other suggestions? including things regarding DNSMasq? Thanks On Fri, Jul 30, 2010 at 11:04 AM, Dave Cotton dcot...@linuxautrement.comwrote: On 30/07/10 16:15, bruce bruce wrote: Adria, How can I build a dns cache into my lan? I am using a Linksys 48 port POE switch and running a micro DD-WRT firmware on a linksys router. DD-WRT supports DNSMasq which would do exactly what you need. DC -- _ -- 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 -- _ -- 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] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1
I am not sure why it would be sleeping. I have never dealt with putting a linux server to sleep. It is connected to a UPS, but I don't think it has been put to sleep by the UPS as the USB cable from UPS is not connected to it. Can you please elaborate on what you mean by AMI:Ping? Is there a service that you recommand that does this or are there any opensource monitoring tools out there that I can use? But my main question remains why there are no activities on 24th and 25th? This is what I see in the /var/log/messages.1: Jul 23 17:11:55 elastix last message repeated 20 times Jul 23 17:22:51 elastix last message repeated 38 times Jul 23 17:30:39 elastix last message repeated 26 times Jul 23 17:30:39 elastix last message repeated 45 times Jul 23 19:09:42 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum 2 Jul 23 20:17:44 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum 2 Jul 23 21:29:16 elastix dhclient: DHCPREQUEST on eth0 to 192.168.1.254 port 67 Jul 23 21:29:16 elastix dhclient: DHCPACK from 192.168.1.254 Jul 23 21:29:16 elastix dhclient: bound to 192.168.1.100 -- renewal in 37640 seconds. Jul 26 09:22:37 elastix syslogd 1.4.1: restart. Jul 26 09:22:37 elastix kernel: klogd 1.4.1, log source = /proc/kmsg started. Jul 26 09:22:37 elastix kernel: Linux version 2.6.18-164.el5 ( mockbu...@builder16.centos.org) (gcc version 4.1.2 20080704 (Red Hat 4.1.2-46)) #1 SMP Thu Se$ Jul 26 09:22:37 elastix kernel: BIOS-provided physical RAM map: Morning of the 26th at 9:22 the server was restarted because it was un-reachable from outside and hence the restart log but where is the 24th, and 25th? Thanks, Bruce On Thu, Jul 29, 2010 at 9:10 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Jul 28, 2010 at 9:06 PM, bruce bruce bruceb...@gmail.com wrote: See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk being down? No, it just means there was no logger activity for those days. You need to add a monitoring solution to your Asterisk box (IE: AMI: Ping). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- _ -- 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] Aastra phones occasionally show No Service - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
Hi Everyone, I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones occasionally go into No Service mode. The POE switch doesn't seem to be the problem as it's tested fine. I think the router sometimes gives up and comes back quickly. Or something of that nature. However, the connections are maintained if a call is going on because there are peer to peer connections between the phones in a network. Anyhow, if the phones are restarted they work fine. So, I was looking around the Aastra Admin UI to find any timer to lower it to 1 second to check and make sure the device always has an ip but I can't seem to find anything other than LLDP which is set at 30 and I don't think that will be of any help. I did a test where I would disconnect the router from the switch and after a while phones go into No Service but if I plug it back into the switch the phones do not come back right away. Maybe something should be dialed on the phone or wait long time or restart it to work again. Any work around? Thanks a lot -- _ -- 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] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1
This was a static IP. Further checks into the server prevails that there are no logs of what happened on the 24th and 25th even in /var/log/messages. This makes me believe that a hardware lockup has happened and according to people on CentOS forum this is VERY HARD to diagnose as there will be no logs. Even MS Blue Screed Of Death does a better job of logging at instances like this :-( On Thu, Jul 29, 2010 at 10:13 PM, Lyle Giese l...@lcrcomputer.net wrote: Lyle Giese wrote: bruce bruce wrote: I am not sure why it would be sleeping. I have never dealt with putting a linux server to sleep. It is connected to a UPS, but I don't think it has been put to sleep by the UPS as the USB cable from UPS is not connected to it. Can you please elaborate on what you mean by AMI:Ping? Is there a service that you recommand that does this or are there any opensource monitoring tools out there that I can use? But my main question remains why there are no activities on 24th and 25th? This is what I see in the /var/log/messages.1: Jul 23 17:11:55 elastix last message repeated 20 times Jul 23 17:22:51 elastix last message repeated 38 times Jul 23 17:30:39 elastix last message repeated 26 times Jul 23 17:30:39 elastix last message repeated 45 times Jul 23 19:09:42 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum 2 Jul 23 20:17:44 elastix ntpd[3113]: synchronized to 216.216.216.216, stratum 2 Jul 23 21:29:16 elastix dhclient: DHCPREQUEST on eth0 to 192.168.1.254 port 67 Jul 23 21:29:16 elastix dhclient: DHCPACK from 192.168.1.254 Jul 23 21:29:16 elastix dhclient: bound to 192.168.1.100 -- renewal in 37640 seconds. Jul 26 09:22:37 elastix syslogd 1.4.1: restart. Jul 26 09:22:37 elastix kernel: klogd 1.4.1, log source = /proc/kmsg started. Jul 26 09:22:37 elastix kernel: Linux version 2.6.18-164.el5 ( mockbu...@builder16.centos.org) (gcc version 4.1.2 20080704 (Red Hat 4.1.2-46)) #1 SMP Thu Se$ Jul 26 09:22:37 elastix kernel: BIOS-provided physical RAM map: Morning of the 26th at 9:22 the server was restarted because it was un-reachable from outside and hence the restart log but where is the 24th, and 25th? Thanks, Bruce On Thu, Jul 29, 2010 at 9:10 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Jul 28, 2010 at 9:06 PM, bruce bruce bruceb...@gmail.com wrote: See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk being down? No, it just means there was no logger activity for those days. You need to add a monitoring solution to your Asterisk box (IE: AMI: Ping). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- It's 'well known' that Asterisk gets confused and runs around in a very tight loop when DNS resolution is failing. Asterisk does a lot of DNS queries and when the Internet goes down, that puts Asterisk into a loop. Depending on your machine, I am guessing that Asterisk locked up or dropped out on the 23rd and the restart on the 26th brought it back to life. Nagios is a good choice for monitoring servers and services. I use it here to monitor all the servers and SIP on my Asterisk box. Lyle Giese LCR Computer Services, Inc. While the above comment about DNS holds, I also realized that most likely your Asterisk machine lost it's only ip address when the DSL went down. That may also have caused Asterisk to exit. I think most(if not all) admins here would never have a dynamic ip address on an Asterisk server. Lyle Giese LCR Computer Services, Inc. -- _ -- 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 -- _ -- 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] Why do Zaptel calls drop all of a sudden? Could busy detect be the problem?
Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [...@macro-dialout-trunk:1] Macro(SIP/2111-b6a400b0, hangupcall|) in new stack [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [...@macro-hangupcall:1] GotoIf(SIP/2111-b6a400b0, 1?skiprg) in new stack [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Goto (macro-hangupcall,s,4) This is running 1.4.26.1 (Elastix) Should I turn of busy detect in chan_dahdi.conf? or is this a known bug and has a workaround? Thanks, Bruce -- _ -- 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] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Hmmwhat about call waiting? You mean, when a call comes in on that specific line, it generate two beep tones and hence the system hangs up thinking it's end of the call? Interesting!!! If it is call-waiting do I have to set all of the following off for it to not give me problem again: *callwaiting=yes* *usecallingpres=yes* *callwaitingcallerid=yes* *threewaycalling=yes* *transfer=yes* *canpark=yes* *cancallforward=yes* *busydetect=yes* *busycount=3* Please elaborate a bit if I am off-topic. Regards, Bruce On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Subject:* [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem? I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: Could be callwaiting? -- _ -- 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 -- _ -- 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] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Furthermore, these are lines in Hunt, so, I am not sure if Call-Waiting is turned ON on these lines at all. But it's definitely an interesting idea. On Wed, Jul 28, 2010 at 5:54 PM, bruce bruce bruceb...@gmail.com wrote: Hmmwhat about call waiting? You mean, when a call comes in on that specific line, it generate two beep tones and hence the system hangs up thinking it's end of the call? Interesting!!! If it is call-waiting do I have to set all of the following off for it to not give me problem again: *callwaiting=yes* *usecallingpres=yes* *callwaitingcallerid=yes* *threewaycalling=yes* *transfer=yes* *canpark=yes* *cancallforward=yes* *busydetect=yes* *busycount=3* Please elaborate a bit if I am off-topic. Regards, Bruce On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Subject:* [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem? I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: Could be callwaiting? -- _ -- 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 -- _ -- 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] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1
Hi Everyone, This is probably more related to Linux than to Asterisk. Analogue channels on a system were un-responsive on Monday morning. Apparently something happened over the weekend and the router went off or it lost it's DSL connection. [Jul 23 22:50:01] VERBOSE[12437] logger.c: -- Remote UNIX connection [Jul 23 22:50:01] VERBOSE[27087] logger.c: -- Remote UNIX connection disconnected [Jul 23 22:55:01] VERBOSE[12437] logger.c: -- Remote UNIX connection [Jul 23 22:55:01] VERBOSE[27093] logger.c: -- Remote UNIX connection disconnected [Jul 23 23:00:01] VERBOSE[12437] logger.c: -- Remote UNIX connection [Jul 23 23:00:02] VERBOSE[27099] logger.c: -- Remote UNIX connection disconnected [Jul 26 09:22:59] VERBOSE[3529] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Jul 26 09:22:59] VERBOSE[3529] logger.c: Asterisk Dynamic Loader Starting: [Jul 26 09:22:59] VERBOSE[3529] logger.c: == Parsing '/etc/asterisk/modules.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found [Jul 26 09:22:59] VERBOSE[3529] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found [Jul 26 09:22:59] VERBOSE[3529] logger.c: == Parsing '/etc/asterisk/http.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk being down? I don't see any of that but yet no calls are on the report for July 24th and 25th indicating to me that Analogue channels, or Asterisk, or the server was down during this time as this office always receives calls on the weekend to the IVR. Where are the logs for eth0 so that I can check to see why this happened and if indeed it was a drop in internet connection. If so, and this is the known bug for Asterisk stop working due to internet drop, why is it not listed in the log file posted above? Thanks, -- _ -- 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] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
:-) I knew someone would bring up FreePBX. I have FreePBX installed and it's not good for Queues at all. It's using the reporting tool from Areski and Areski has recently released an upgrade to it which again is not what I want. There are few other programs that do this but really none that are neat in interface or useful in features. I guess no one else has any thoughts on this? Maybe there is none available? Thanks, Bruce On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg dbackeb...@gmail.comwrote: On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce bruceb...@gmail.com wrote: I seem to not be able to find any good open source Asterisk Queue Analyzer and Asterisk Log Analyzer on the web. google 'freepbx' It does some of what you want. For the rest of what you want, strongly consider paying a professional consultant. -- _ -- 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 -- _ -- 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] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
Hi Guys, I seem to not be able to find any good open source Asterisk Queue Analyzer and Asterisk Log Analyzer on the web. The Asterisk Queue Analyzer is to serve as the graphic tool for call center or pbx admins. It will pull the info in queue.log and in MySQL asterisk CDR to create a graphic bar or to report on each extension that received the queue calls, etc... The Asterisk Log Analyzer is to analyze the log and to show any serious errors or as bonus maybe send out e-mails to admin and to e-mail any downtime during the day. Please note that I am not talking about making my own scripts to analyze and output this data as I know it exists in the system but rather am looking for a project that has done it already. Thanks, Bruce -- _ -- 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] Why does a bridged channel stay open for 4 hours?
This is running Elastix (FreePBX), so I am pretty sure there is Hangup() requested. At least this doesn't happen ALL THE TIME. So, something is getting stuck. Thanks, Bruce On Fri, Jul 23, 2010 at 9:10 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Jul 23, 2010 at 1:16 AM, bruce bruce bruceb...@gmail.com wrote: Any help is appreciated. Are you explicitly calling Hangup() within your dial-plans? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- _ -- 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] POE Splitters
You can also use Ethernet Over Power Lines solution or wireless :-) On Fri, Jul 23, 2010 at 8:55 AM, David Backeberg dbackeb...@gmail.comwrote: On Fri, Jul 23, 2010 at 8:46 AM, Matt mhop...@gmail.com wrote: It's not necessarily this simple. There is an approximately 50-75foot cable run through ceilings and walls (CAT5) to the location where the phones will be. At the phone location there is no power. You always have options. You just have to decide what is more difficult: * moving the phone/devices somewhere else. Easiest solution. * having an electrician pull AC power to the location, then use DC power bricks or PoE switch * having a data cable person pull more ethernet to the location If you already have one ethernet cable that managed to make that 50-75 foot run, then clearly it can be done, and a professional could even use that cable to yank three more along the same run, and then you're all set. -- _ -- 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 -- _ -- 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] Why does a bridged channel stay open for 4 hours?
Well, what about PRI? Why should this stay on? Isn't the native bridge just a bridge channel that should go down automatically if the actually Dahdi/ZAP channel is down and there are no SIP channels on either? Thanks, Bruce On Fri, Jul 23, 2010 at 5:09 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Fri, Jul 23, 2010 at 05:37:44PM -0300, Maurizio Faccio adinet wrote: I guess same trouble with Elastix 1.5.2-2.3 dahdi 2.1.0.4 19 Asterisk 1.4.25.1 Digium TDM 2400 That's an analog card. With an analog trunk, you're not guaranteed to know if the remote CO has hung up the line. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- 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 -- _ -- 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] Why does a bridged channel stay open for 4 hours?
I am having this issue with PRI. But I do not use conference rooms. Our system is a simple queue and extensions. -Bruce On Fri, Jul 23, 2010 at 6:13 PM, Maurizio Faccio adinet mauf...@adinet.com.uy wrote: You're right but it do not detect that I hungs on my side of the line. I think that in some way we are going into a conference in some unwanted way with the two dadhi channels and when i hang up both lines stay bridged. I think that the trouble appears when i dial a number in an analog phone, hook quickly (seems like a flash), and dial again. I am wondering that if I change my lines to a pri I solve this trouble but now I do not see clear at all. (my analog telco cannot bring me polarity reversal on hang up for signaling Thank you in advance Maurizio El 23/07/2010 06:09 p.m., Tzafrir Cohen escribió: On Fri, Jul 23, 2010 at 05:37:44PM -0300, Maurizio Faccio adinet wrote: I guess same trouble with Elastix 1.5.2-2.3 dahdi 2.1.0.4 19 Asterisk 1.4.25.1 Digium TDM 2400 That's an analog card. With an analog trunk, you're not guaranteed to know if the remote CO has hung up the line. -- _ -- 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 -- _ -- 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] POE Splitters
The Aastra 53i draws only 2 Watts from a Linksys 24 port POE switch. 25 phones is around 55 Watts. -Bruce On Thu, Jul 22, 2010 at 5:16 PM, Andrew Latham lath...@gmail.com wrote: The Snom 360 phone in front of me draws 4w... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Jul 22, 2010 at 2:58 PM, David Gibbons d...@videon-central.com wrote: There is no such device -- it's outside of the POE spec. Class 3 devices are allowed to consume at max 15.4W. Most phones are class 3 devices. The math just doesn't work out. Even if you used the draft standard for class 4 (~30W), you could still power max 2 devices at 15W/ea. -Dave On Thu, Jul 22, 2010 at 2:46 PM, Matt mhop...@gmail.com wrote: I've got an interesting situation where I have one cable run from the feed area to the service area. I have three devices that I need to power at the service area. Is anyone aware of a device that will take the POE from the cable run and then allow me to split it to two or three devices at the service end? When I search for splitter all I get are the injectors, but I figure someone has to make something I realize I'll need a power adapter with enough amps to power the full load at the end. -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] Why does a bridged channel stay open for 4 hours?
Hi Everyone, Using a PRI with Sangoma A101D and Asterisk 1.4.2.x. I notice that occasionally after a call is disconnected and both the phone devices and the the channel is down but the bridge stays open for hours. Channel Location State Application(Data) Local/9054445...@fro (None) Up Bridged Call(Zap/4-1) 905444 is the inbound DID on the PRI. core show channels show Zap 4 in use but if try to save the stream using ztmonitor -f it doesn't give me anything because the channel is actually down. It's just the bridge that stayed open for more than 4 hours. Any help is appreciated. Thanks, Bruce -- _ -- 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] Does Flash Operator Panel allow for dragging a call into a parking lot?
It's doable with a work around. Create a misc extension with followme set to ##70# which point to your parking lots and failed destination to Misc parking extension. Regards, Bruce On Sun, Jul 18, 2010 at 3:38 PM, Doug Lytle supp...@drdos.info wrote: bruce bruce wrote: Hi Everyone, If I receive a call on a ZAP line and pickup the call and drag and drop it (by mouse) into a Parking Lot through FOP, it just hangs up. Is this feature supported by FOP? I don't believe so, how would Asterisk know what phone to ring on timeout? 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 -- 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 -- _ -- 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 calculate number of speakers needed for PAGING and INTERCOM coverage area?
Yes, thanks. I think lots of manufacturers just boost the number of speakers really needed but again this really depends on the environment noise level. Regards, Bruce On Wed, Jul 14, 2010 at 11:50 PM, C F shma...@gmail.com wrote: I'm happy to hear it worked out so well with so little. :) On Wed, Jul 14, 2010 at 12:39 AM, bruce bruce bruceb...@gmail.com wrote: Thanks for the input guys. For other refrence, a CyberData Voip Amplifier which supplies 10 Watt to each of the two bogen 30 Watt speakers did the job for a 35, square feet warehouse with environmental noise level of slightly higher than standard but not those of industrial. Only two speakers and done deal. Though I know that three speaker would have been the perfect solution but 4 would cover every single little corner and be an overkill. -Bruce On Tue, Jul 13, 2010 at 8:47 PM, C F shma...@gmail.com wrote: I agree with horns you'll usually get better coverage. I have done this in the past with 5 speakers for a 30k sq ft warehouse very good coverage. Using bogen horns. This was for a 300ft by 100ft warehouse. Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft side I installed a horn every 60ft alternating facing one north and the other south, which ended up 3 facing one way and 2 the other. You can get double horn speakers which will face 2 sides. I wouldn't mount them on the wall specifically not so low as fork lifts and what not will damage them. On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce bruceb...@gmail.com wrote: Well, these are horn speakers with 30 Watt which will receive 10 Watt only from Amplifer. I am not connecting them to ceiling so maybe 10 feet off the ground. I guess my coverage would be better??? Based on your calculations for for 40k sqfeet that would be 33 speakers. I think that's way too much of an overkill. thanks, Bruce On Mon, Jul 12, 2010 at 1:05 AM, C F shma...@gmail.com wrote: In my experience using height for radius works, for example if you have a 20 ft high ceiling then the coverage for one speaker would be 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft has never killed anyone, but this really depends on the power of the speaker, I usually deal with 70v speakers tapped at 16 or 8 watts depending on how many speakers I put on one amplifier and the output wattage of that amplifier. On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so I am not sure if the full capacity of the speaker (30 watt) will be used. I appreciate your advice. Thanks, Bruce -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
[asterisk-users] Does Flash Operator Panel allow for dragging a call into a parking lot?
Hi Everyone, If I receive a call on a ZAP line and pickup the call and drag and drop it (by mouse) into a Parking Lot through FOP, it just hangs up. Is this feature supported by FOP? Thanks, Bruce -- _ -- 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] Can't compile DAHDI - wrong kernel source
I am stuck with the same problem but I have used asterisk yum repository and it worked by itself without me worrying for kernel stuff. However, I need to install speex codec and now I am stuck as it doesn't get picked up by the yum asterisk install somehow. I have lib speex and speex already installed and when doing yum install asterisk16 I don't see speex in core show translation Is there anything specific I have to do? Do I have to build from source as well? -Sorry, didn't mean to hijack the thread. Thanks, Bruce On Wed, Jul 14, 2010 at 5:08 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hi If you install rpm from any location it goes to its default location. You just go for above steps. For kernel you can go for http://kernel.org -- Regards, Chandrakant Solanki On Wed, Jul 14, 2010 at 2:06 PM, liuxin nyliuxin...@gmail.com wrote: Hi. The best easy way is: copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm 2010/7/14 Gareth Blades list-aster...@skycomuk.com Thermal Wetland wrote: I have a virtual server with godaddy but can not compile DAHDI as it complains that I do not have the correct kernel source. The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest version Nothing to do uname -a returns: Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.nethttp://ip-xxx-xxx-xxx-xxx.ip.secureserver.net/ http://ip-XXX-XXX-XXX-XXX.ip.secureserver.nethttp://ip-xxx-xxx-xxx-xxx.ip.secureserver.net/ 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux When I try to compile DAHDI it fails with: make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-028stab064.7 kernel installed. Is there a way to trick DAHDI to use the installed kernel? Thanks for the help! -- -Thermal What kernel versions do you have installed? If you are currently running an older kernel but installed a newer kernel and sources but havent rebooted to activate the new one yet then it may still be trying to locate the source for the older running kernel. -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] Where should I look for MWI settings if Aastra phones don't do it?
Hi Guys, Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and 6730i, but none of them indicate the voic-email. Where should I look for trouble to find the root issue for MWI? Thanks, -- _ -- 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] Where should I look for MWI settings if Aastra phones don't do it?
Thanks for the input guys. I don't use .xml files for Aastra. Everything is done on the UI. #voicemail show users: *ContextMbox User Zone NewMsg* *|default007 Alex 2* *default2100 Peter1* * * This system is using FreePBX, so I checked the Device and Users in asterisk tables and they have default for voicemail setup which I think is right. I can also see the msg.txt in a folder that has new voicemail waiting. However, I am not sure about privileges though. Here is it: [r...@elastix INBOX]# ls -la total 284 drwxrwxr-x 2 asterisk asterisk 4096 Jul 14 09:57 . drwxrwxr-x 8 asterisk asterisk 4096 Jul 13 11:57 .. *-rw-rw-r-- 1 asterisk asterisk282 Jul 12 18:46 msg.txt* -rwxrwxr-x 1 asterisk asterisk 83244 Jul 12 18:46 msg.wav -rwxrwxr-x 1 asterisk asterisk 8510 Jul 12 18:46 msg.WAV *-rw--- 1 asterisk asterisk261 Jul 14 09:57 msg0001.txt* -rwx-- 1 asterisk asterisk 150124 Jul 14 09:57 msg0001.wav -rwx-- 1 asterisk asterisk 15270 Jul 14 09:57 msg0001.WAV Thanks, Bruce On Wed, Jul 14, 2010 at 12:25 PM, Steve Johnson stevej...@gmail.com wrote: On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and 6730i, but none of them indicate the voic-email. Where should I look for trouble to find the root issue for MWI? (1) Check from the CLI voicemail show users to ensure that the proper mailboxes have been set up and there is new mail in them. If this is not right, check the voicemail.conf entry for this mailbox. (2) Check the phone device configuration (in sip.conf) to ensure that the phone has a mailbox=xxx entry. for example: ;entry in sip.conf for extension 115 [115] context=yourcontext mailbox=115 ... Restart asterisk if you've made changes and re-test. -- _ -- 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 -- _ -- 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] Distinctive ring for INTERNAL calls only? How to do it?
Hi Everyone, Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Even though FreePBX Inbound has an option for Alert_INFO but that doesn't work when the call comes into an IVR or Queue. The calls has to go directly to extension for external ringtone to be different. So, I am looking for internal calls ringtones to be different rather than external call ringtones. Anyone has got this working? Thanks, Burce -- _ -- 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] Distinctive ring for INTERNAL calls only? How to do it?
Thanks for the input but that won't be good because people are not going to remember two extensions for one person. The sip header should be able to carry alert_info to internal extensions really easily. Anyone else got a thought? Thanks again, On Wed, Jul 14, 2010 at 5:44 PM, Ira i...@extrasensory.com wrote: At 11:44 AM 7/14/2010, you wrote: Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? It's ugly, but you could give the phone two different SIP IDs and give those different ringtones. Ira -- _ -- 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 -- _ -- 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] How to install speex codec for Asterisk that is downloaded from Digium Yum Repository?
Hi Everyone, I have done yum install speex libspeex-devel speex-devel and it was succesful on CentOS. I then tried yum install asterisk16 asterisk16-addons asterisk16-configs but core show translation doesn't show speex loaded. Is there a way to or an option that I can append to the asterisk install to make sure it compiles with speex in mind? Thanks, Bruce -- _ -- 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 calculate number of speakers needed for PAGING and INTERCOM coverage area?
Thanks for the input guys. For other refrence, a CyberData Voip Amplifier which supplies 10 Watt to each of the two bogen 30 Watt speakers did the job for a 35, square feet warehouse with environmental noise level of slightly higher than standard but not those of industrial. Only two speakers and done deal. Though I know that three speaker would have been the perfect solution but 4 would cover every single little corner and be an overkill. -Bruce On Tue, Jul 13, 2010 at 8:47 PM, C F shma...@gmail.com wrote: I agree with horns you'll usually get better coverage. I have done this in the past with 5 speakers for a 30k sq ft warehouse very good coverage. Using bogen horns. This was for a 300ft by 100ft warehouse. Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft side I installed a horn every 60ft alternating facing one north and the other south, which ended up 3 facing one way and 2 the other. You can get double horn speakers which will face 2 sides. I wouldn't mount them on the wall specifically not so low as fork lifts and what not will damage them. On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce bruceb...@gmail.com wrote: Well, these are horn speakers with 30 Watt which will receive 10 Watt only from Amplifer. I am not connecting them to ceiling so maybe 10 feet off the ground. I guess my coverage would be better??? Based on your calculations for for 40k sqfeet that would be 33 speakers. I think that's way too much of an overkill. thanks, Bruce On Mon, Jul 12, 2010 at 1:05 AM, C F shma...@gmail.com wrote: In my experience using height for radius works, for example if you have a 20 ft high ceiling then the coverage for one speaker would be 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft has never killed anyone, but this really depends on the power of the speaker, I usually deal with 70v speakers tapped at 16 or 8 watts depending on how many speakers I put on one amplifier and the output wattage of that amplifier. On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so I am not sure if the full capacity of the speaker (30 watt) will be used. I appreciate your advice. Thanks, Bruce -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] My own FreePBX FollowME module - Stuck at Reload - Anyone else had experience with this?
Hi Everyone,
I have done some php coding to come up with my own FollowME module for
FreePBX. The need for this has some security considerations behind it.
This is what my code does at core:
$sql=REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id, ringing,
pre_ring) VALUES
('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]','$grplist','0','$postdest','','','0','0','Ring','$_POST[pre_ring]');
This all conforms with the fields that are filled up by FreePBX followme
module but it seems that this is not all becuase the followme doesn't work
when I do it this way. It only works if I press submit and confirm the
orange bar.
For one thing, I think the Orange Reload bar does something that I can't
seem to find and that my php code doesn't do. I tried doing a manual
reload and dialplan reload but that wouldn't do the job.
Can someone please shed some light if you know where I am stuck and had to
tackle the issue yourself?
Thanks,
Bruce
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Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?
Well, these are horn speakers with 30 Watt which will receive 10 Watt only from Amplifer. I am not connecting them to ceiling so maybe 10 feet off the ground. I guess my coverage would be better??? Based on your calculations for for 40k sqfeet that would be 33 speakers. I think that's way too much of an overkill. thanks, Bruce On Mon, Jul 12, 2010 at 1:05 AM, C F shma...@gmail.com wrote: In my experience using height for radius works, for example if you have a 20 ft high ceiling then the coverage for one speaker would be 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft has never killed anyone, but this really depends on the power of the speaker, I usually deal with 70v speakers tapped at 16 or 8 watts depending on how many speakers I put on one amplifier and the output wattage of that amplifier. On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so I am not sure if the full capacity of the speaker (30 watt) will be used. I appreciate your advice. Thanks, Bruce -- _ -- 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 -- _ -- 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 -- _ -- 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] My own FreePBX FollowME module - Stuck at Reload - Anyone else had experience with this?
Thanks for that Tim. *Wondering how I can trigger that reload?* I have tried
dialplan reload and reload but that doesn't work. Obviously amportal reload
wouldn't be doable in this case even if it works because the system will go
down.
Thanks,
Bruce
On Mon, Jul 12, 2010 at 2:13 PM, Tim Nelson tnel...@rockbochs.com wrote:
- bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
I have done some php coding to come up with my own FollowME module for
FreePBX. The need for this has some security considerations behind it.
This is what my code does at core:
$sql=REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre,
grplist, annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id,
ringing, pre_ring) VALUES
('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]','$grplist','0','$postdest','','','0','0','Ring','$_POST[pre_ring]');
This all conforms with the fields that are filled up by FreePBX followme
module but it seems that this is not all becuase the followme doesn't work
when I do it this way. It only works if I press submit and confirm the
orange bar.
For one thing, I think the Orange Reload bar does something that I can't
seem to find and that my php code doesn't do. I tried doing a manual
reload and dialplan reload but that wouldn't do the job.
Can someone please shed some light if you know where I am stuck and had to
tackle the issue yourself?
Thanks,
Bruce
--
FreePBX development may be best discussed on the FreePBX forums [1].
As an aside, the 'Orange Reload Bar' takes all of the information in the
FreePBX MySQL database, generates the dialplan code, and then replaces your
extensions.conf, sip.conf, iax.conf, etc with the appropriate information.
[1] http://www.freepbx.org/forums
http://www.freepbx.org/forums
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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Re: [asterisk-users] PHP can't insert - Can someone please help
Thank you for the amazing reply. First few lines of your e-mail was EXACTLY
getting me to where I made a mistake. I guess I didn't take the () and ' '
at their face value and was looking somewhere else for the problem.
For sanatizing you mean checking the numbers to make sure they are valid
numbers and not alphabet or other charecters? or, are you pointing the fact
that I am keeping mysql root password in plain .php file? I have done an
include of a php file which has mysql root password and that is insert as an
#incldue in the html file. So, if someone checks source for html can't see
mysql root password. Even though root is user on mysql is to accept only
from localhost.
I would really appreciate it if you can weigh in on it a bit.
Thanks,
Bruce
On Sat, Jul 10, 2010 at 7:42 AM, Gerald A geraldabli...@gmail.com wrote:
Hi Bruce,
First, your problem isn't PHP, it seems to be SQL and I'm guessing MySQL at
that.
Next, you seem to be accepting user input and not sanatizing it. DANGER
WILL ROBINSON!!!
This is bad, because it leaves you open to something known as a SQL
injection attack.
Now, as to syntax:
On Sat, Jul 10, 2010 at 12:07 AM, bruce bruce bruceb...@gmail.com wrote:
I am making another module for Voicemail. I have three fields in a POST
form that have to be connected together to make it a single 10 digit number
but there is something wrong in my syntax probably.
$npaa = ('$_POST[anpa]');
$nxxa = ('$_POST[anxx]');
$blocka = ('$_POST[ablock]');
*$grplist = $npaa.$nxxa.$blocka;*
Ok, so suppose arpa=111, anxx=222 and ablock=.
grplist would then be ('111')('333')('').
$sql=INSERT INTO findmefollow(grpnum, strategy, grptime, grppre,
grplist, annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id,
ringing, pre_ring)
VALUES
('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]',$grplist,'0','$_POST[postdest]','','','0','0','Ring','$_POST[pre_ring]');
It seems that $grplist is the problem. Can someone please point what is
wrong?
Error:
Error: You have an error in your SQL syntax; check the manual that
corresponds to your MySQL server version for the right syntax to use near
'('333')(''),'0','ext-local,vmb2000,1','','','0','0','Ring','0')' at
line 3
Look closesly, grasshopper. See it? (Does the hint above help?) Hmmm, ok.
Let's write the line as SQL:
INSERT INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id, ringing,
pre_ring)
VALUES
('0','ringall','0','0',('111')('333')(''),'0','0','','','0','0','Ring','0');
Clear now? You are trying to insert the raw value --
('111')('333')('') -- into your database. This can't make any sense
except as string, And this isn't one.
I think what you might have meant is to quote the _whole thing_ as a
string, and not the individual pieces. Then:
$grplist = '(.$npaa.$nxxa.$blocka.)';
and
$blocka = ($_POST[ablock]); # and for all of them above
This would make the value '(111)(333)()', which should work fine.
Now, if you really meant to add in the quotes, you'll have to quote the
quotes, which can be hard to do in good times.
Hope this helps,
Gerald.
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Re: [asterisk-users] PHP can't insert - Can someone please help
Further to my last post, I added this to santize. I also created a new mysql
user with access to only findmefollow portion of the asterisk table for
limited access and assigned only two simultaneous connections with only 10
changes queries per hour (as I know that no more queries will be put through
probably)
if ($npaa=200 $nxxa=200 $npaa!=900 $npaa!=911)
Should that suffice against SQL injections? The if condition changes the
string to number so it removes the chance of people adding
other characters and it also sticks to format NPAN or 2XX2.
Thanks
On Sat, Jul 10, 2010 at 10:21 AM, bruce bruce bruceb...@gmail.com wrote:
Thank you for the amazing reply. First few lines of your e-mail was EXACTLY
getting me to where I made a mistake. I guess I didn't take the () and ' '
at their face value and was looking somewhere else for the problem.
For sanatizing you mean checking the numbers to make sure they are valid
numbers and not alphabet or other charecters? or, are you pointing the fact
that I am keeping mysql root password in plain .php file? I have done an
include of a php file which has mysql root password and that is insert as an
#incldue in the html file. So, if someone checks source for html can't see
mysql root password. Even though root is user on mysql is to accept only
from localhost.
I would really appreciate it if you can weigh in on it a bit.
Thanks,
Bruce
On Sat, Jul 10, 2010 at 7:42 AM, Gerald A geraldabli...@gmail.com wrote:
Hi Bruce,
First, your problem isn't PHP, it seems to be SQL and I'm guessing MySQL
at that.
Next, you seem to be accepting user input and not sanatizing it. DANGER
WILL ROBINSON!!!
This is bad, because it leaves you open to something known as a SQL
injection attack.
Now, as to syntax:
On Sat, Jul 10, 2010 at 12:07 AM, bruce bruce bruceb...@gmail.comwrote:
I am making another module for Voicemail. I have three fields in a POST
form that have to be connected together to make it a single 10 digit number
but there is something wrong in my syntax probably.
$npaa = ('$_POST[anpa]');
$nxxa = ('$_POST[anxx]');
$blocka = ('$_POST[ablock]');
*$grplist = $npaa.$nxxa.$blocka;*
Ok, so suppose arpa=111, anxx=222 and ablock=.
grplist would then be ('111')('333')('').
$sql=INSERT INTO findmefollow(grpnum, strategy, grptime, grppre,
grplist, annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id,
ringing, pre_ring)
VALUES
('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]',$grplist,'0','$_POST[postdest]','','','0','0','Ring','$_POST[pre_ring]');
It seems that $grplist is the problem. Can someone please point what is
wrong?
Error:
Error: You have an error in your SQL syntax; check the manual that
corresponds to your MySQL server version for the right syntax to use near
'('333')(''),'0','ext-local,vmb2000,1','','','0','0','Ring','0')' at
line 3
Look closesly, grasshopper. See it? (Does the hint above help?) Hmmm, ok.
Let's write the line as SQL:
INSERT INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id, ringing,
pre_ring)
VALUES
('0','ringall','0','0',('111')('333')(''),'0','0','','','0','0','Ring','0');
Clear now? You are trying to insert the raw value --
('111')('333')('') -- into your database. This can't make any sense
except as string, And this isn't one.
I think what you might have meant is to quote the _whole thing_ as a
string, and not the individual pieces. Then:
$grplist = '(.$npaa.$nxxa.$blocka.)';
and
$blocka = ($_POST[ablock]); # and for all of them above
This would make the value '(111)(333)()', which should work fine.
Now, if you really meant to add in the quotes, you'll have to quote the
quotes, which can be hard to do in good times.
Hope this helps,
Gerald.
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Re: [asterisk-users] PHP can't insert - Can someone please help
Thanks again. Apparently all POST variables come through as strings. The function you pointed out is I think already built in php as is_numeric() http://www.php.net/manual/en/function.is-numeric.php. http://php.net/manual/en/function.is-int.php http://www.php.net/manual/en/function.is-numeric.php http://www.php.net/manual/en/function.is-numeric.phpIf that runs TRUE and if I keep my =200 and !=911 or !900 I should be safe from SQL injections. And along with dial-out routes rules, I think I can make this stronger. I have my html/php file set so that the input field only takes 3 digit 3 digit 4 digit (NPA, NXX, Block) so your purposal of: *'201,0); drop database YOUR_DATABASE'; *would fail due to big length and also I tested with inputing letters and my IF function caught it and exited. Further more, everything else (other than phone input fields) is drop down boxes with specific numbers or letters inserted in them. I should be 100% safe with those right? By using form POST there should be no other loop holes left opened right? It's not like php $_GET so people can't try typing to the browser in this format: http://www.w3schools.com/welcome.php?fname=Peterage=37 Thanks a lot, Bruce On Sat, Jul 10, 2010 at 1:41 PM, Gerald A geraldabli...@gmail.com wrote: Hi Bruce, On Sat, Jul 10, 2010 at 11:12 AM, bruce bruce bruceb...@gmail.com wrote: Further to my last post, I added this to santize. I also created a new mysql user with access to only findmefollow portion of the asterisk table for limited access and assigned only two simultaneous connections with only 10 changes queries per hour (as I know that no more queries will be put through probably) if ($npaa=200 $nxxa=200 $npaa!=900 $npaa!=911) Should that suffice against SQL injections? The if condition changes the string to number so it removes the chance of people adding other characters and it also sticks to format NPAN or 2XX2. There are two things -- the first is, who call this script? If it's something you control 100%, you can mitigate the risk a bit. I don't really like this tact, because if the script gets repurposed, you end up with something that could be very dangerous. The second thing is simple -- most people think small here, but you have to think big and know a bit about how PHP works. PHP strings are pretty amazing things, and one of the pesky things is that you can put all kinds of things in it. Now, if that string variable is created as a result of a form input, then that string can be anything. For a moment, think about if it $npaa = '201,0); drop database YOUR_DATABASE'; Now, that is pretty nasty, and it would muck up further SQL injections, but now you get the idea. You should always check to make sure the data you are getting is what you are expecting, and exclude what you aren't. So, are your tests sufficient? I can't remember off the top of my head if the string - integer only considers the first number, or it considers the whole string. (PHP usually errs on the side of ease of use, so I think my snippet above would still pass your test). If your expecting only numbers, I'd write a function that ensures that only numbers are parts of the input. (And not just for the 3 above variables). Really, you should never see $_POST(var) (or any PHP CGI variable) that derives directly from user input. It takes a few minutes extra, but it'll save hours of sorting later if you get hit by a SQL injection. Hope this helps, Gerald -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can get user inputs from called party after dial?
You need read():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
http://www.voip-info.org/wiki/view/Asterisk+cmd+ReadIt's as easy as:
exten = s,n,Read(variable,,11)
exten = s,n,NoOp(${variable})
Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.
-Bruce
On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman egoltz...@gmail.com wrote:
Hi,
I want to dial a party, play him a message and wait for his input, i.e.
DTMF digits and use them to control the rest of the dial plan.
How do I do it?
If I use Dial it will not return until the end of the call, isn't it?
Thanks,
Eyal
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Re: [asterisk-users] PHP can't insert - Can someone please help
Here is the steel strong sanitizer:
$npaa = $_POST[anpa];
$nxxa = $_POST[anxx];
$blocka = $_POST[ablock];
# Sanitize
$blocka_san = strspn($blocka, 0123456789);
*if ($blocka_san==4 is_numeric($npaa) is_numeric($nxxa)
is_numeric($blocka) $npaa=200 $nxxa=200 $npaa!=900
$npaa!=911) *
*
*
* {*
echo Number passed sanitization;
}
What do you think? :-)
-Bruce
On Sat, Jul 10, 2010 at 2:17 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks again. Apparently all POST variables come through as strings. The
function you pointed out is I think already built in php as
is_numeric() http://www.php.net/manual/en/function.is-numeric.php.
http://php.net/manual/en/function.is-int.php
http://www.php.net/manual/en/function.is-numeric.php
http://www.php.net/manual/en/function.is-numeric.phpIf that runs TRUE
and if I keep my =200 and !=911 or !900 I should be safe from SQL
injections. And along with dial-out routes rules, I think I can make this
stronger.
I have my html/php file set so that the input field only takes 3 digit 3
digit 4 digit (NPA, NXX, Block) so your purposal of: *'201,0); drop
database YOUR_DATABASE'; *would fail due to big length and also I tested
with inputing letters and my IF function caught it and exited.
Further more, everything else (other than phone input fields) is drop down
boxes with specific numbers or letters inserted in them. I should be 100%
safe with those right?
By using form POST there should be no other loop holes left opened right?
It's not like php $_GET so people can't try typing to the browser in this
format:
http://www.w3schools.com/welcome.php?fname=Peterage=37
Thanks a lot,
Bruce
On Sat, Jul 10, 2010 at 1:41 PM, Gerald A geraldabli...@gmail.com wrote:
Hi Bruce,
On Sat, Jul 10, 2010 at 11:12 AM, bruce bruce bruceb...@gmail.comwrote:
Further to my last post, I added this to santize. I also created a new
mysql user with access to only findmefollow portion of the asterisk table
for limited access and assigned only two simultaneous connections with only
10 changes queries per hour (as I know that no more queries will be put
through probably)
if ($npaa=200 $nxxa=200 $npaa!=900 $npaa!=911)
Should that suffice against SQL injections? The if condition changes the
string to number so it removes the chance of people adding
other characters and it also sticks to format NPAN or 2XX2.
There are two things -- the first is, who call this script? If it's
something you control 100%, you can mitigate the risk a bit. I don't really
like this tact, because if the script gets repurposed, you end up with
something that could be very dangerous.
The second thing is simple -- most people think small here, but you have
to think big and know a bit about how PHP works. PHP strings are pretty
amazing things, and one of the pesky things is that you can put all kinds of
things in it. Now, if that string variable is created as a result of a form
input, then that string can be anything. For a moment, think about if it
$npaa = '201,0); drop database YOUR_DATABASE'; Now, that is pretty nasty,
and it would muck up further SQL injections, but now you get the idea. You
should always check to make sure the data you are getting is what you are
expecting, and exclude what you aren't.
So, are your tests sufficient? I can't remember off the top of my head if
the string - integer only considers the first number, or it considers the
whole string. (PHP usually errs on the side of ease of use, so I think my
snippet above would still pass your test). If your expecting only numbers,
I'd write a function that ensures that only numbers are parts of the input.
(And not just for the 3 above variables).
Really, you should never see $_POST(var) (or any PHP CGI variable) that
derives directly from user input.
It takes a few minutes extra, but it'll save hours of sorting later if you
get hit by a SQL injection.
Hope this helps,
Gerald
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Re: [asterisk-users] How can get user inputs from called party after dial?
You need to do some reading :-)
I will give you a quick teach here. At the end of file
/etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in
/etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add
this:
[first-Dialplan]
exten = s,1,Answer
exten = s,n,Playback(Welcome)
exten = s,n,Read(numb,,10)
exten = s,n,NoOp(${numb})
And send your inbound route to context first-Dialplan so that it's triggered
when a call comes in. Then on terminal do a asterisk -r and you
will see the NoOp show the DTMF number entered. From there on you can do
anything you want with the variable ${numb}
If any part of above is unclear to you, you must consult your friend,
google, for examples of Asterisk dialplan.
-Bruce
On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman egoltz...@gmail.com wrote:
Thanks, but I'm missing something here, the dial command is where?
I need to do something like:
Dial(1234)
Read(1 digit)
DoSomthing(based on digit from 1234)
And as far as I understand the Dial start the call and only come back (ig
you use the g option) after call finished.
Eyal
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Sent:* Saturday, July 10, 2010 9:30 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How can get user inputs from called party
after dial?
You need read():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
It's as easy as:
exten = s,n,Read(variable,,11)
exten = s,n,NoOp(${variable})
Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.
-Bruce
On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman egoltz...@gmail.com
wrote:
Hi,
I want to dial a party, play him a message and wait for his input, i.e.
DTMF digits and use them to control the rest of the dial plan.
How do I do it?
If I use Dial it will not return until the end of the call, isn't it?
Thanks,
Eyal
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No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00
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Re: [asterisk-users] How can get user inputs from called party after dial?
For dial you do this:
[first-Dialplan]
exten = s,1,Answer
exten = s,n,Dial(SIP/provider/111222)
exten = s,n,Playback(Welcome)
exten = s,n,Read(numb,,10)
exten = s,n,NoOp(${numb})
-Bruce
On Sat, Jul 10, 2010 at 2:51 PM, bruce bruce bruceb...@gmail.com wrote:
You need to do some reading :-)
I will give you a quick teach here. At the end of file
/etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in
/etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add
this:
[first-Dialplan]
exten = s,1,Answer
exten = s,n,Playback(Welcome)
exten = s,n,Read(numb,,10)
exten = s,n,NoOp(${numb})
And send your inbound route to context first-Dialplan so that it's
triggered when a call comes in. Then on terminal do a asterisk -r
and you will see the NoOp show the DTMF number entered. From there on you
can do anything you want with the variable ${numb}
If any part of above is unclear to you, you must consult your friend,
google, for examples of Asterisk dialplan.
-Bruce
On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman egoltz...@gmail.comwrote:
Thanks, but I'm missing something here, the dial command is where?
I need to do something like:
Dial(1234)
Read(1 digit)
DoSomthing(based on digit from 1234)
And as far as I understand the Dial start the call and only come back (ig
you use the g option) after call finished.
Eyal
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Sent:* Saturday, July 10, 2010 9:30 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How can get user inputs from called party
after dial?
You need read():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
It's as easy as:
exten = s,n,Read(variable,,11)
exten = s,n,NoOp(${variable})
Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.
-Bruce
On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman egoltz...@gmail.com
wrote:
Hi,
I want to dial a party, play him a message and wait for his input, i.e.
DTMF digits and use them to control the rest of the dial plan.
How do I do it?
If I use Dial it will not return until the end of the call, isn't it?
Thanks,
Eyal
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No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00
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Re: [asterisk-users] How can get user inputs from called party after dial?
I was under the impression that he is new to Asterisk. No need to fuss. Hence the :-) On Sat, Jul 10, 2010 at 3:35 PM, Steve Edwards asterisk@sedwards.comwrote: On Sat, 10 Jul 2010, bruce bruce wrote: You need to do some reading :-) Now that is funny -- maybe you could take your own advice and look at http://www.php.net/docs.php instead of posting please help me debug code I'm too lazy to even see if PHP says it is syntactically correct and the only relevance it has to Asterisk is I'm trying to concatenate some strings and make sure it could be a phone number requests. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- _ -- 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
