Re: [asterisk-users] Asterisk Queue's
Date: Fri, 29 Aug 2008 09:12:12 -0500 From: Mark Michelson [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Queue's To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Tobias Ahlander wrote: Hello Philipp, Yes, I have autofill set in queues.conf. I suspect that this behaviour is because the Polycom phones I use have 2 lines. Has anyone used this function with polycom phones before? Also, my agents are Dynamic, perhaps this works better with Static agents? Here's my queues.conf (with commented lines deleted for easier reading): [general] autofill = yes monitor-type = MixMonitor [sales] strategy = rrmemory wrapuptime=15 Depending on which Asterisk version you are using, there was a bug in the queue application for some 1.4 releases where the autofill option would only be set properly if it were placed inside a queue. In other words, you may want to try putting autofill=yes inside the [sales] queue in your configuration. Also, if you're using a version of Asterisk 1.2, autofill is not a valid option and you'll be stuck with the behavior you're seeing. Mark Michelson Mark, Unfortunately this didn't help at all... Anyone else has any tips? Is there a way to limit the polycom phones to only take one call from the Queue at the same time? Asterisk version running is 1.4.13 Best regards, Tobias ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway errors
Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue's
Tobias Ahlander schrieb: From: Mark Michelson [EMAIL PROTECTED] Tobias Ahlander wrote: Yes, I have autofill set in queues.conf. I suspect that this behaviour is because the Polycom phones I use have 2 lines. Has anyone used this function with polycom phones before? Also, my agents are Dynamic, perhaps this works better with Static agents? Here's my queues.conf (with commented lines deleted for easier reading): [general] autofill = yes monitor-type = MixMonitor [sales] strategy = rrmemory wrapuptime=15 Depending on which Asterisk version you are using, there was a bug in the queue application for some 1.4 releases where the autofill option would only be set properly if it were placed inside a queue. In other words, you may want to try putting autofill=yes inside the [sales] queue in your configuration. Also, if you're using a version of Asterisk 1.2, autofill is not a valid option and you'll be stuck with the behavior you're seeing. Unfortunately this didn't help at all... Anyone else has any tips? Is there a way to limit the polycom phones to only take one call from the Queue at the same time? Asterisk version running is 1.4.13 Maybe the phones have call-waiting enabled? Does it work if you remove the second line? Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Ciao VoipCrazy, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? SIP locks if it tries to do DNS queries and doesn't get an answer. Try using a local caching DNS server. HTH -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Asterisk is looking for a SIP trunk if you have recorded the usage of SIP trunks all it need is to find 1 SIP trunk, To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1 and make sure that it is the first one in sip.conf file. OR you can make a sip trunk to ATA in the same lan and also be sure that it is the first trunk in sip.conf . On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Igor, From asterisk, when internet is down I can ping all extensions. The same occurs in others instalations, when the internet is down, my lical extensions log off from asterisk. VoipCrazy 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Penalties for agents
Hello again list. I have gotten some indications that the penalties for agents in Asterisk might not work as intended. Can I get peoples view on this that uses the penalty function in Asterisk Queues? Thanks in advance Best regards, Tobias ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] not able to make call to landline no...to mobile works fine
hi all I have a PRI line which i have connected to my asterisk server. I am able to make calls to mobile no through my asterisk server, while i am not able to make calls to land line nos. This is strange. Where do u think the? problem is , is it from the service provider or? mis configuration of my asterisk. I am from India and using airtel pri lines. Below i am pasting you my configuration file which works to call land line not but doesn't work for land line no. zaptel.conf span=1,1,0,ccs,hdb3 span=2,1,0,ccs,hdb3 bchan = 1-15,17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone = in defaultzone = in zapata.conf [channels] context=test group=1 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echotraining=yes rxgain=0.0 txgain=0.0 callprogress=no callerid=asreceived pickupgroup=1 ;pridialplan=unknown immediate=no ;signalling=pri_cpe_ptmp ;switchtype=ni1 channel = 1-15,17-31 I have commented the pridailplan , signalling and switchtype because when i enable this, asterisk give me warning Ignoring pridialplan same with other parameter. When i disable those options still i am able to call mobile no. when i make call , asterisk give me following log when debug mode is enabled.. asterisk*CLI ??? -- Executing [EMAIL PROTECTED]:1] Goto(SIP/2002-b76144d0, trunkdial|947521744|1) in new stack ??? -- Goto (trunkdial,947521744,1) ??? -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2002-b76144d0, Zap/r1/47521744) in new stack -- Making new call for cr 32854 ??? -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8)? len=45 Call Ref: len= 2 (reference 86/0x56) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1? Q.931 Std: 0? Info transfer capability: Speech (0) ? Ext: 1? Trans mode/rate: 64kbps, circuit-mode (16) ??? User information layer 1: A-Law (35) [18 03 a9 83 8c] Channel ID (len= 5) [ Ext: 1? IntID: Implicit? PRI? Spare: 0? Exclusive? Dchan: 0 ??? ChanSel: As indicated in following octets ?? Ext: 1? Coding: 0? Number Specified? Channel Type: 3 ?? Ext: 1? Channel: 12 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1? Coding: CCITT (ITU) standard (0)? 0: 0? Location: User (0) ?? Ext: 1? Progress Description: Calling equipment is non-ISDN. (3) ] [28 05 b1 32 30 30 32] Display (len= 5) Charset: 31 [ 2002 ] [6c 06 21 81 32 30 30 32] Calling Number (len= 8) [ Ext: 0? TON: National Number (2)? NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) ?? Presentation: Presentation permitted, user number passed network screening (1)? '2002' ] [70 09 a1 34 37 35 32 31 37 34 34] Called Number (len=11) [ Ext: 1? TON: National Number (2)? NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)? '47521744' ] q931.c:3087 q931_setup: call 32854 on channel 12 enters state 1 (Call Initiated) ??? -- Called r1/47521744 Protocol Discriminator: Q.931 (8)? len=14 Call Ref: len= 2 (reference 86/0x56) (Terminator) Message type: STATUS (125) [08 04 82 e3 98 28] Cause (len= 6) [ Ext: 1? Coding: CCITT (ITU) standard (0)? Spare: 0? Location: Public network serving the local user (2) ? Ext: 1? Cause: Info. element nonexist or not implemented (99), class = Protocol Error (e.g. unknown message) (6) ] ? Cause data 1: 98 (152, Non-Locking Shift To Codeset 0 IE) ? Cause data 2: 28 (40, Display IE) [14 01 01] Call State (len= 3) [ Ext: 0? Coding: CCITT (ITU) standard (0)? Call state: Call Initiated (1) -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) Protocol Discriminator: Q.931 (8)? len=10 Call Ref: len= 2 (reference 86/0x56) (Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 03 a9 83 8c] Channel ID (len= 5) [ Ext: 1? IntID: Implicit? PRI? Spare: 0? Exclusive? Dchan: 0 ??? ChanSel: As indicated in following octets ?? Ext: 1? Coding: 0? Number Specified? Channel Type: 3 ?? Ext: 1? Channel: 12 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3844 q931_receive: call 32854 on channel 12 enters state 2 (Overlap sending) Protocol Discriminator: Q.931 (8)? len=13 Call Ref: len= 2 (reference 86/0x56) (Terminator) Message type: DISCONNECT (69) [08 02 82 9c] Cause (len= 4) [ Ext: 1? Coding: CCITT (ITU) standard (0)? Spare: 0? Location: Public network serving the local user (2) ? Ext: 1? Cause: Invalid number format (28), class = Normal Event (1) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1? Coding: CCITT (ITU) standard (0)? 0: 0? Location: Public network serving the local user (2) ?? Ext: 1? Progress Description: Inband information or appropriate pattern now available. (8) ] --
[asterisk-users] Problematic Trunk SIP: Got SIP response 405 Method not allowed
Hi guys, I need to create a SIP trunk between my * (trixbox) and a legacy Samsung pbx. I create the SIP trunk as usual: the calls from my * to the Samsung pbx worked immediately, but I can not place any calls from the Samsung pbx to the *. On the * CLI I see several errors of this type: -- Got SIP response 405 Method not allowed back from 10.1.1.11 (Samsung Pbx IP); in RFC3261 (pag. 185) it's written: 405 Method Not Allowed: The method specified in the Request-Line is understood, but not allowed for the address identified by the Request-URI. The response MUST include an Allow header field containing a list of valid methods for the indicated address. What does it mean? How can i fix this problem? Thanks - Kind Regards Daniele ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
2008/8/31 Olivier [EMAIL PROTECTED] What happens if the PC supporting this card is powered off ? It is powered over USB from the main (internal USB) and backup (external USB) server. If one of the power fails it will switch to the other server. If both servers power fail you have a problem anyway. ;-) Do you have an idea of its price ? Approx. US$ 700 Regards Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway errors
Hatem, I cannot understan exactly what you told me. Could you try to explain that in other words. Better if you could post an example of this SIP trunk. thanks in advance. Voip Crazy 2008/9/1 hatem moiz [EMAIL PROTECTED]: Asterisk is looking for a SIP trunk if you have recorded the usage of SIP trunks all it need is to find 1 SIP trunk, To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1 and make sure that it is the first one in sip.conf file. OR you can make a sip trunk to ATA in the same lan and also be sure that it is the first trunk in sip.conf . On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Thats strange, have you checked that you're not having issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote: Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet connection is down asterisk stops working correctly? ¿How could I solve that? Thansk. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problematic Trunk SIP: Got SIP response 405 Method not allowed
Sip debug please. On 1 Sep 2008, at 10:07, daniele visaggio wrote: Hi guys, I need to create a SIP trunk between my * (trixbox) and a legacy Samsung pbx. I create the SIP trunk as usual: the calls from my * to the Samsung pbx worked immediately, but I can not place any calls from the Samsung pbx to the *. On the * CLI I see several errors of this type: -- Got SIP response 405 Method not allowed back from 10.1.1.11 (Samsung Pbx IP); in RFC3261 (pag. 185) it's written: 405 Method Not Allowed: The method specified in the Request-Line is understood, but not allowed for the address identified by the Request-URI. The response MUST include an Allow header field containing a list of valid methods for the indicated address. What does it mean? How can i fix this problem? Thanks - Kind Regards Daniele ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problematic Trunk SIP: Got SIP response 405 Method not allowed
2008/9/1 Steven Howes [EMAIL PROTECTED] Sip debug please. --- Sep 1 11:53:42 VERBOSE[3599] logger.c: -- SIP read from 10.1.1.11:5060: SIP/2.0 405 Method not allowed From: ;tag=as411269a4 To: ;tag=a01010b-13c4-64622-1881f814-4122656d Call-ID: [EMAIL PROTECTED] CSeq: 146 REGISTER Via: SIP/2.0/UDP 10.1.1.36:5060;rport=5060;branch=z9hG4bK01f3b5c6 Supported: 100rel,replaces Content-Length: 0 Sep 1 11:53:42 VERBOSE[3599] logger.c: --- (8 headers 0 lines) --- Sep 1 11:53:42 DEBUG[3599] chan_sip.c: Stopping retransmission on ' [EMAIL PROTECTED]' of Request 146: Match Found Sep 1 11:53:42 VERBOSE[3599] logger.c: -- Got SIP response 405 Method not allowed back from 10.1.1.11 Sep 1 11:53:42 VERBOSE[3599] logger.c: Destroying call ' [EMAIL PROTECTED]' Sep 1 11:53:45 DEBUG[10671] manager.c: Manager received command 'login' Sep 1 11:53:45 VERBOSE[10671] logger.c: == Parsing '/etc/asterisk/manager.conf': Sep 1 11:53:45 VERBOSE[10671] logger.c: == Parsing '/etc/asterisk/manager.conf': Found Sep 1 11:53:45 VERBOSE[10671] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': Sep 1 11:53:45 VERBOSE[10671] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': Found Sep 1 11:53:45 VERBOSE[10671] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Sep 1 11:53:45 VERBOSE[10671] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Found Sep 1 11:53:45 DEBUG[10671] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer Sep 1 11:53:45 DEBUG[10671] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0appended to acl for peer Sep 1 11:53:45 DEBUG[10671] acl.c: # Testing 127.0.0.1 with 0.0.0.0 Sep 1 11:53:45 DEBUG[10671] acl.c: # Testing 127.0.0.1 with 127.0.0.0 Sep 1 11:53:45 VERBOSE[10671] logger.c: == Manager 'admin' logged on from 127.0.0.1 Sep 1 11:53:47 DEBUG[10671] manager.c: Manager received command 'Command' Sep 1 11:53:47 DEBUG[10671] manager.c: Manager received command 'Command' Sep 1 11:53:47 DEBUG[10671] manager.c: Manager received command 'Command' Sep 1 11:53:47 DEBUG[10671] manager.c: Manager received command 'Command' Sep 1 11:53:47 DEBUG[10671] manager.c: Manager received command 'Command' Sep 1 11:53:47 VERBOSE[10671] logger.c: == Manager 'admin' logged off from 127.0.0.1 Sep 1 11:53:47 DEBUG[10679] manager.c: Manager received command 'login' Sep 1 11:53:47 VERBOSE[10679] logger.c: == Parsing '/etc/asterisk/manager.conf': Sep 1 11:53:47 VERBOSE[10679] logger.c: == Parsing '/etc/asterisk/manager.conf': Found Sep 1 11:53:47 VERBOSE[10679] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': Sep 1 11:53:47 VERBOSE[10679] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': Found Sep 1 11:53:47 VERBOSE[10679] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Sep 1 11:53:47 VERBOSE[10679] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Found Sep 1 11:53:47 DEBUG[10679] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer Sep 1 11:53:47 DEBUG[10679] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0appended to acl for peer Sep 1 11:53:47 DEBUG[10679] acl.c: # Testing 127.0.0.1 with 0.0.0.0 Sep 1 11:53:47 DEBUG[10679] acl.c: # Testing 127.0.0.1 with 127.0.0.0 Sep 1 11:53:47 VERBOSE[10679] logger.c: == Manager 'admin' logged on from 127.0.0.1 Sep 1 11:53:47 VERBOSE[10679] logger.c: == Manager 'admin' logged off from 127.0.0.1 Sep 1 11:53:49 DEBUG[10681] manager.c: Manager received command 'login' Sep 1 11:53:49 VERBOSE[10681] logger.c: == Parsing '/etc/asterisk/manager.conf': Sep 1 11:53:49 VERBOSE[10681] logger.c: == Parsing '/etc/asterisk/manager.conf': Found Sep 1 11:53:49 VERBOSE[10681] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': Sep 1 11:53:49 VERBOSE[10681] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': Found Sep 1 11:53:49 VERBOSE[10681] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Sep 1 11:53:49 VERBOSE[10681] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Found Sep 1 11:53:49 DEBUG[10681] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer Sep 1 11:53:49 DEBUG[10681] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0appended to acl for peer Sep 1 11:53:49 DEBUG[10681] acl.c: # Testing 127.0.0.1 with 0.0.0.0 Sep 1 11:53:49 DEBUG[10681] acl.c: # Testing 127.0.0.1 with 127.0.0.0 Sep 1 11:53:49 VERBOSE[10681] logger.c: == Manager 'admin' logged on from 127.0.0.1 Sep 1 11:53:49 VERBOSE[10681] logger.c: == Manager 'admin' logged on from 127.0.0.1 Thanks for your prompt reply ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] lists.digium.com monthly reminders
I am subscribed to several of the mailing lists hosted at lists.digium.com. However, the memberships reminder that I receive on the first of each month only lists asterisk-biz, and none of the others. Just curious whether this was intentional or a mis-configuration. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
On 10:43, Mon 01 Sep 08, Tony Mountifield wrote: I am subscribed to several of the mailing lists hosted at lists.digium.com. However, the memberships reminder that I receive on the first of each month only lists asterisk-biz, and none of the others. Just curious whether this was intentional or a mis-configuration. Just a confirmation from here. I'm not on the -biz list, and I did not get any reminder. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
On Mon, Sep 1, 2008 at 1:04 PM, Michiel van Baak [EMAIL PROTECTED] wrote: Just curious whether this was intentional or a mis-configuration. Just a confirmation from here. I'm not on the -biz list, and I did not get any reminder. The lists seem to be configured differently, one has a post ack the other does not, etc. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
On Mon, Sep 01, 2008 at 10:43:51AM +, Tony Mountifield wrote: I am subscribed to several of the mailing lists hosted at lists.digium.com. However, the memberships reminder that I receive on the first of each month only lists asterisk-biz, and none of the others. Just curious whether this was intentional or a mis-configuration. I suppose it's not the first list on an alphabetic order, as I'm subscribed to -devel and some others, and yet get the info message only for -i18n . Maybe it's the first one you subscribe to?? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
In article [EMAIL PROTECTED], Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 01, 2008 at 10:43:51AM +, Tony Mountifield wrote: I am subscribed to several of the mailing lists hosted at lists.digium.com. However, the memberships reminder that I receive on the first of each month only lists asterisk-biz, and none of the others. Just curious whether this was intentional or a mis-configuration. I suppose it's not the first list on an alphabetic order, as I'm subscribed to -devel and some others, and yet get the info message only for -i18n . Maybe it's the first one you subscribe to?? Interesting... I would have expected a single email from the site, listing all the groups I am subscribed to. I'm sure I've had that from other mailman-driven list sites in the past, when I've been subscribed to more than one list. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
Tony Mountifield wrote: Interesting... I would have expected a single email from the site, listing all the groups I am subscribed to. I'm sure I've had that from other mailman-driven list sites in the past, when I've been subscribed to more than one list. The 'send_reminders' option was not set for all the lists on lists.digium.com, but it is now. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
In article [EMAIL PROTECTED], Kevin P. Fleming [EMAIL PROTECTED] wrote: Tony Mountifield wrote: Interesting... I would have expected a single email from the site, listing all the groups I am subscribed to. I'm sure I've had that from other mailman-driven list sites in the past, when I've been subscribed to more than one list. The 'send_reminders' option was not set for all the lists on lists.digium.com, but it is now. Cool, thanks! Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 beta
Hello users, Can you share with me your experiences with Asterisk 1.6? Is it stable enough for commercial service? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
Look at this brand new failover device: http://www.rhinoequipment.com/1portfail.html http://www.rhinoequipment.com/Single%20Port%20Failover%20Datasheet%201-22-2008.pdf Interesting ! I didn't know this one. What happens if the PC supporting this card is powered off ? Do you have an idea of its price ? This is an important point: Not only does the loss of PC power NOT present a problem, is the main DESIGN element. If the PC/Server powers off, the relays naturally de-energize, which by design, PASSES the service to the failover port (to the port connected to your spare server or analog devices). While by itself that would be a pretty good design, there would be a vulnerability, because of course it's possible to have Asterisk crash while the server is still happily powered on. Therefore the Rhino failover device incorporates another design element: If you choose, you can enable its watchdog system. If asterisk stops responding to a periodic message, It knows that Asterisk is down, and fails over your services to the other server or analog devices EVEN IF the power is still on at the main server! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
On Mon, 01 Sep 2008 09:20:32 -0500, Karl Fife wrote: Look at this brand new failover device: http://www.rhinoequipment.com/1portfail.html http://www.rhinoequipment.com/Single%20Port%20Failover%20Datasheet%201-22-2008.pdf Interesting ! I didn't know this one. What happens if the PC supporting this card is powered off ? Do you have an idea of its price ? This is an important point: Not only does the loss of PC power NOT present a problem, is the main DESIGN element. If the PC/Server powers off, the relays naturally de-energize, which by design, PASSES the service to the failover port (to the port connected to your spare server or analog devices). While by itself that would be a pretty good design, there would be a vulnerability, because of course it's possible to have Asterisk crash while the server is still happily powered on. Therefore the Rhino failover device incorporates another design element: If you choose, you can enable its watchdog system. If asterisk stops responding to a periodic message, It knows that Asterisk is down, and fails over your services to the other server or analog devices EVEN IF the power is still on at the main server! -Karl In the broadcast business we call this two-stage fault tolerant bypass. Hardware relays trip in the event of loss of power. A software based watchdog trips if the application under test becomes unresponsive. Often the watchdog doesn't trip the relays as being mechanical that creates a switching glitch. They trip a logic switch so that the signal through the device stays clean into bypass. It's also worth testing to see how the card boots up. Is it clean on restart? That is, can you reset it without interrupting your traffic? Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get call status and hangup
Thanks, that might put me on the right track. Best regards, Loic Didelot. On Tue, 2008-08-26 at 14:19 +0200, Andrea Spadaccini wrote: Ciao Loic, Hello, I am looking for a way to check if a call could be established with the destination (SIP,IAX,ZAP). So I thought about an application like DIAL but instead it should return a variable and hangup immediately as soon as it gets something that could lead to a valid connection ringing... Is there something like this that could be used in Asterisk or can anyone recommend a different/better solution? I don't know if it's exactly what you want, but you can monitor the status of a given user using the Asterisk Manager Interface (AMI), through the events NewState, PeerStatus, Hangup. More info: http://www.voip-info.org/wiki-Asterisk+manager+API http://www.voip-info.org/wiki/view/asterisk+manager+events HTH, -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
Christian Victor [EMAIL PROTECTED] said: It is powered over USB from the main (internal USB) and backup (external USB) server. If one of the power fails it will switch to the other server. If both servers power fail you have a problem anyway. ;-) This is incorrect. According to Jim Rhodes at Rhino, there is no NEED for 'backup' power from another server via USB: As I described in the last post: Not only does the loss of PC power NOT present a problem, is the main DESIGN element. If the PC/Server powers off, the relays naturally de-energize, which by design, PASSES the service [mechanically, as tne natural result of no power] to the failover port (to the port connected to your spare server or analog devices) Christian Victor [EMAIL PROTECTED] said: Do you have an idea of its price ? Approx. US$ 700 This is incorrect. Again, according to Jim Rhodes, the FULL nobody-ever-actually-pays-this-much LIST price is US $350. In my estimation, the street price will be between US $220 and $299 depending on your reseller's markup. I don't know if Rhino has M.A.P. rules. Karl Fife said: Therefore the Rhino failover device incorporates another design element: If you choose, you can enable its watchdog system. If asterisk stops responding to a periodic message, It knows that Asterisk is down, and fails over your services to the other server or analog devices EVEN IF the power is still on at the main server! I would have to ask the guys at Rhino for confirmation on this point but: My current understanding of the EXTERNAL usb connection is so that in the event that the secondary server NEEDS TO TAKE OVER the service for any reason, it can pre-empt the main server without the main server failing. Essentially the secondary server can 'ask' for it. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
Michael Graves [EMAIL PROTECTED] said: Often the watchdog doesn't trip the relays as being mechanical that creates a switching glitch. They trip a logic switch so that the signal through the device stays clean into bypass. It's also worth testing to see how the card boots up. Is it clean on restart? That is, can you reset it without interrupting your traffic? Good distinction. What you're describing sounds more like the RedFone device which is actively parsing, processing, and routing the digital media stream of a T1/E1, using a digital switching system not a mechanical one. I imagine that in your field of broadcasting, there would be little tolerance for artificats introduced a mechanical switch. http://www.red-fone.com/Products/fonebridge2/ The Rhino is just a passive (normally closed) mechanical switch with an active monitoring system. The advantage is that it can be used to do failover on switch Ethernet, Analog or ISDN. You couldn't get that kind of flexibility otherwise. Thanks! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 beta
VoIP Cyprus wrote: Can you share with me your experiences with Asterisk 1.6? Is it stable enough for commercial service? No. No matter how good some people may tell you it is, 1.6 is still beta software and software is rarely beta for no good reason. Don't even THINK about running 1.6 until it leaves beta and RC stage unless you are truly desperate for the features and are willing to accept random crashes, unusual behaviour and the possibility of things changing before the final release. The company I worked for up until June this year was still selling 1.2 systems until late April because we hadn't worked through all the changes and tested things fully. If your company will depend on your phone system for customer service, don't take the risk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 beta
On 1 Sep 2008, at 17:34, Rob Hillis wrote: VoIP Cyprus wrote: Can you share with me your experiences with Asterisk 1.6? Is it stable enough for commercial service? No. No matter how good some people may tell you it is, 1.6 is still beta software and software is rarely beta for no good reason. Don't even THINK about running 1.6 until it leaves beta and RC stage unless you are truly desperate for the features and are willing to accept random crashes, unusual behaviour and the possibility of things changing before the final release. The company I worked for up until June this year was still selling 1.2 systems until late April because we hadn't worked through all the changes and tested things fully. If your company will depend on your phone system for customer service, don't take the risk. I agree with the advice (i.e. don't use a Beta for commercial service.) But Rob is mixing 2 issues - porting an existing set up to a new version (of anything) is always a pain, there are always unexpected gotchas so once you have service running there is a _huge_ disincentive to upgrading. On the other hand, if you are building a new service, you should go for as new and crispy a version as you dare. The question you to ask is - will it be stable by the time I want to launch?. Doing this can save you at least one round of upgrades in the next year or so. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Documentation of users.conf
Hello, does anybody know where is documented every parameter of the users.conf file in the asterisk distribucion tarball ? thanks a lot. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
2008/9/1 Karl Fife [EMAIL PROTECTED] Christian Victor [EMAIL PROTECTED] said: Do you have an idea of its price ? Approx. US$ 700 This is incorrect. Again, according to Jim Rhodes, the FULL nobody-ever-actually-pays-this-much LIST price is US $350. In my estimation, the street price will be between US $220 and $299 depending on your reseller's markup. I don't know if Rhino has M.A.P. rules. It's nice to know as, for $800 or $900, I think you can get a 4 digital ports standalone Fail-over-switch. So this card has interesting price position, the main drawback being, IMHO, it's eating a slot, which can be a rare resource in rackable servers. Anyway, it's worth knowing this exist. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
At 06:59 AM 9/1/2008, you wrote: The 'send_reminders' option was not set for all the lists on lists.digium.com, but it is now. Cool, thanks! Is this really necessary for a list that generates 50 messages a day. I get reminded plenty and I don't really need the additional noise. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Documentation of users.conf
On Mon, Sep 01, 2008 at 12:11:31PM -0500, Nestor A. Diaz wrote: Hello, does anybody know where is documented every parameter of the users.conf file in the asterisk distribucion tarball ? I believe that this is the same format as sip.conf and it's included from sip.conf in asterisknow setups, but it has a mix of settings from the other files. Entries that you define manually should probably be in some other file (sip.conf etc?) Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
So this card has interesting price position, the main drawback being, IMHO, it's eating a slot, which can be a rare resource in rackable servers. You raise a very important point. This device uses a BRACKET, but not a motherboard SLOT. In other words, it hangs free in one of the chassis slots that do not have a corresponding slot on the motherboard. If you do not have a bracket slot, you could mount it externally, but you'd have to engineer a way to hold it. FYI, As an analog failover device it can fail over as many as 4 analog lines. I think it's a really attractive value proposition. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
In article [EMAIL PROTECTED], Ira [EMAIL PROTECTED] wrote: At 06:59 AM 9/1/2008, you wrote: The 'send_reminders' option was not set for all the lists on lists.digium.com, but it is now. Cool, thanks! Is this really necessary for a list that generates 50 messages a day. I get reminded plenty and I don't really need the additional noise. You mean like ONE extra message a month? For 50 messages a day, that is approximately a 0.066% increase! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 beta
Rob Hillis [EMAIL PROTECTED] writes: No. No matter how good some people may tell you it is, 1.6 is still beta software and software is rarely beta for no good reason. Tell that to Google. So far, for us, 1.6 beta is running better than any of the early 1.2 releases. Perhaps even better than early 1.4. If you need the new features in 1.6, you can go with 1.6 or you can backport. Either has disadvantages. If you can do without the 1.6 features, go with 1.4. We deploy new customers on 1.4 today, but we have one 1.6 in production and two in testing. Anyway, all that is just a long-winded way of saying: Test, test, test, and have a plan B if 1.6 fails. Oh and a thank you to Asterisk developers at Digium and elsewhere: The Asterisk development process is a lot better now than it was 2 years ago. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
[EMAIL PROTECTED] (Tony Mountifield) writes: You mean like ONE extra message a month? For 50 messages a day, that is approximately a 0.066% increase! If you normally read this list via gmane and only keep a subscription to be able to post, it's an infinite increase. Yes, I could write a rule to junk those mails, but they aren't really consistent enough between mailing lists. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
On Mon, 01 Sep 2008 10:18:17 -0500, Karl Fife I would have to ask the guys at Rhino for confirmation on this point but: My current understanding of the EXTERNAL usb connection is so that in the event that the secondary server NEEDS TO TAKE OVER the service for any reason, it can pre-empt the main server without the main server failing. Essentially the secondary server can 'ask' for it. I'm glad I checked with Rhino on this point. My original understanding was incorrect. Bryce Chidester at Rhino said The external port is to be used *instead* of the internal header in the event there is no internal header, or if you just want to mount the [failover card]externally. I also learned that the name of this device is the Single Port T1/E1/J1, Ethernet and Analog Failover Card sometimes referred to as the SPF (single port failover) card. I suppose it's so named because even though it will failover 4 analog circuits, it will only failover 1 Ethernet or T1/E1/J1 circuit. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
2008/9/1 Karl Fife [EMAIL PROTECTED] It is powered over USB from the main (internal USB) and backup (external USB) server. If one of the power fails it will switch to the other server. If both servers power fail you have a problem anyway. ;-) This is incorrect. According to Jim Rhodes at Rhino, there is no NEED for 'backup' power from another server via USB: Did I write you NEED a second power supply? I was just refering to the fact that when both servers power fail you have a problem no matter if the failover switch ist still working or not. Do you have an idea of its price ? Approx. US$ 700 This is incorrect. Again, according to Jim Rhodes, the FULL nobody-ever-actually-pays-this-much LIST price is US $350. In my estimation, the street price will be between US $220 and $299 depending on your reseller's markup. I don't know if Rhino has M.A.P. rules. I hope you are right. Maybe this guy should share his information with the world. According to http://store.variantdistribution.com/category-s/49.htmVariant - one of Rhinos distributors and the only source I was able to find - quotes the card for US$ 700. Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Still badly in need for mISDN help!
Hello folks! I still need someone's help! Badly! I tried even more researching, posting to the isdn4linux list. But nothing. and besides the isdn4linux guys, who obviously are VERY quiet, you are the only ones I can think of for help. Here's my hardware, basic linux setup: Linux Debian with custom kernel version 2.6.24-rt1 ISDN-card: AVM A1 pci (Fritz) mISDN version: git-tree from one week ago. mISDNUser version: git-tree from the same day. How I built mISDN: misdn # ./std2kern linux-2.6.24-rt1 # gmake menuconfig Enabled capi2.0 Enabled Modular ISDN, the fritz card, also tried enable/disable audio DSP option. Had to fix mISDN to compile: replaced struct semaphore with: struct compat_semaphoe Files: stack.c and core.c (did this after a bit of reading about my type of compilation error. Seems to be standard practice.) Here's my mISDN startup messages from dmesg (I saw an error belatedly! SORRY!!!) mISDNd: kernel daemon started (current:f7c6eb90) mISDNd: test event done ISDN L2 driver version 1.32 X25 DTE modul version 1.13 DTMF modul version 1.18 AVM Fritz PCI/PnP driver Rev. 1.43 ACPI: PCI Interrupt Link [APC3] enabled at IRQ 18 ACPI: PCI Interrupt :01:08.0[A] - Link [APC3] - GSI 18 (level, high) - IR Q 20 mISDN_fcpcipnp: found adapter Fritz!Card PCI at :01:08.0 fritz card f7de9800 dch f7de9898 bch1 f7de99e8 bch2 f7de9b38 AVM PCI: stat 0x2020a AVM PCI: Class A Rev 2 AVM PnP: HDLC version 1 mISDN: AVM Fritz!PCI config irq:20 base:0xD000 AVM PCI/PnP: reset mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:1173 AVM PCI/PnP: S0/S1 6/0 Fritz1 ISAC STAR 4a Fritz1 ISAC MODE 0 Fritz1 ISAC ADF2 0 Fritz1 ISAC ISTA 4 Fritz1 ISAC CIR0 52 mISDN_isac_init: ISAC version (0): 2086/2186 V1.1 Fritz1 HDLC 1 STA 8100 Fritz1 HDLC 2 STA 0 AVM Fritz!PCI: IRQ 20 count 5 fritz 1 cards installed [...] mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range Can anyone make head or tails of this? I'm out of it. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
On Tue, 2 Sep 2008 00:22:45 +0200, Christian Victor said: that when both servers power fail you have a problem no matter if the failover switch ist still working or not. You've got that right my friend! :-) On Tue, 2 Sep 2008 00:22:45 +0200, Christian Victor said: http://store.variantdistribution.com/category-s/49.htmVariant - one of Rhinos distributors and the only source I was able to find - quotes the card for US$ 700. Strange. I've seen this happen before where retailers will list outrageously high prices for soon-to-be-released products. For example the SNOM KlarVoice handset. MSRP is $32, but I've seen it advertised for $200! http://www.8774e4voip.com/SearchResults.asp?Search=klarvoice I can say with confidence that the LIST price is US $350. The street price will be considerably lower. Frankly, if I were Snom or Rhino I'd be pretty cheezed off about this phenomenon. After hearing the 'buzz' about a new product such as this, I'd hate for customers to *decide* against it mistkenly believing this incorrect price. I'd turn my nose at either of these two products for the incorrect prices I've seen advertised. We're pretty stoked to have stumbled onto this product because it's brand new, and we've been looking for something like it for some time. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redundant PSTN PRI Gateways using Asterisk
I currently have two T1 PRI lines feeding our company's legacy PBX. All
our numbers are DIDs and can pass over both PRI. Currently, if one PRI
line (or T1 interface card in the PBX) is down, communication continues
to function as normal (with the exception of the reduction of channels
available for in- and outgoing calls).
I am looking to implement Asterisk in front of the legacy PBX as the new
gateway to the PSTN for the company in an effort to begin a slow
transition to VoIP. While I am familiar with Asterisk when it comes to
implementing it as a single box solution for PRI, SIP, voicemail, IVR,
etc, I get a little confused when you start to separate the functions to
different boxes for redundancy and scalability.
I believe that the first phase of what I am asking probably just
involves setting up two asterisk boxes with dual port T1 cards (one port
as incoming from PSTN and second as feed to legacy PBX), and including a
simple dial plan with regex extension capturing all my DIDs passing
incoming calls out the PRI and allowing from the PBX to go out the PRI.
This way if a PRI line goes down, one of the Digium T1 cards fail, or
one of the asterisk servers fail, everything continues to work through
the other one. This essentially give me the exact same functionality
and reliability I have currently plus the ability to start a migration
to VoIP internally. I don't believe any fancy load balancing /
redundancy techniques are need for this part as that is handled on the
providers end which routes the DIDs over both PRIs. Is there something
I am missing about this part? I know alternatively I could use
something like redfone bridge but then I feel that device becomes a
single point of failure which I do not want to introduce into the
environment.
The next step I would like to take is to move the voicemail from the
legacy PBX to another pair of asterisk servers acting as redundant
voicemail servers. This is where I get a bit lost. Here is the outline
of what I believe I need to do.
1. build asterisk server to act as voicemail server
2. setup mailboxes in voicemail.conf for each extension/user that is to
have voicemail
3. connect both gateways to the voicemail server via IAX
4. edit the dial plans on both gateway asterisk servers to include an
extension that points to voicemail() and voicemailmain() extension on
the voicemail server
5. edit legacy pbx to direct voicemail (message button on phones) to
extension on asterisk server for voicemailmain app
6. edit legacy pbx to direct unanswered call to extension on asterisk
server for voicemail app
7. build second asterisk server to act as redundant voicemail server and
integrate it somehow
Questions on this include:
- Best way to have active/active redundant voicemail nodes (iSCSI
SAN is available) to accomplish load balancing and failure redundancy
(LVS, Linux-HA, ultramonkey, etc)
- If balanced via LVS, then there is only one IAX connection to
the virtual IP address of the cluster and the director handles balancing
which server processes the VM call, correct? In other solutions, how
would this work?
- Is it appropriate for the gateway server to directly link to the
voicemail servers when distributing the services like this or should
something like DUNDi or SER be used?
- How do I route from the legacy PBX back to one of the gateways
in order to pass to the VM servers across the two T1 connections (again,
so that in the event one is down, voicemail routing still works) - this
may be more of a question for the PBX guys.
- Will be interested in Unified Communications, so storage of
voicemail in our email server (exchange) using IMAP may be the route
taken as opposed to using SAN in which case csync2 or similar may be
used to just keep conf files the same on cluster nodes instead of shared
storage.
The eventual next step would be to implement another cluster of asterisk
servers to act as media servers for call recording, moh, ivr menus,
conferences, and agi. Setup a SER/openSER cluster for registering new
IP phones and somehow routing all the traffic between the IP phones,
media servers, gateways, voicemail servers, and legacy pbx. Then,
eventual phasing out the pbx. I will pose more specific question on
this when the time comes but I mention it in case anyone sees issue with
what I do in the early phases that would negatively impact where I
ultimately want to go.
PBX PRI A connection
|
PRI A Asterisk Gateway A -IAX\
Asterisk VM Servers x2
(LVS Cluster with iSCSI SAN shared storage ?)
PRI B Asterisk Gateway B -IAX/
|
PBX PRI B connection
Any thoughts, comments, suggestion would be appreciated as I don't feel
there is sufficient information out there on redundant enterprise type
setups. Again the initial focus is on the gateways and voicemail.
___
--
[asterisk-users] Dialplan terminates when the caller hangs up
Hi,
I've tried the following toy dialplan:
[sipcalls]
exten = _X.,1,NoOp()
exten = _X.,n,Dial(SIP/${EXTEN},,g)
exten = _X.,n,Playback(good-bye)
exten = _X.,n,Hangup()
With the above dialplan, when the callee hangs up, Asterisk does play
good-bye to the caller. However, when the caller hangs up, the dialplan seems
to be terminated immediately without playing good-bye to the callee.
Is it possible to do somethings so that the dialplan continues after the caller
hangs up?
Cong-Van
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
