Re: [asterisk-users] Asterisk/Cisco AS5300 = Two problems in incoming (extension not found)
Phibee Network Operation Center a écrit : Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. (042600 = my phone number) .. I have put a debug: [Kvoip*CLI --- SIP read from UDP://192.168.50.125:59124 --- INVITE sip:0426000...@192.168.50.130:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.125:5060 From: sip:477000...@192.168.50.125;tag=6950F0-25C7 To: sip:0426000...@192.168.50.130 Date: Wed, 28 Oct 2009 05:16:26 GMT Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 3761097657-3266777566-2192416711-2957366127 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: sip:477000...@192.168.50.125;party=calling;screen=yes;privacy=off Timestamp: 1256706986 Contact: sip:477000...@192.168.50.125:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 8642 2741 IN IP4 192.168.50.125 s=SIP Call c=IN IP4 192.168.50.125 t=0 0 m=audio 18726 RTP/AVP 8 101 c=IN IP4 192.168.50.125 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 - [Kvoip*CLI --- (20 headers 11 lines) --- [Kvoip*CLI Sending to 192.168.50.125 : 5060 (no NAT) [Kvoip*CLI Using INVITE request as basis request - e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 [Kvoip*CLI No matching peer for '47700' from '192.168.50.125:59124' [Kvoip*CLI Found RTP audio format 8 [Kvoip*CLI Found RTP audio format 101 [Kvoip*CLI Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI Found audio description format PCMA for ID 8 [Kvoip*CLI Found audio description format telephone-event for ID 101 [Kvoip*CLI Got unsupported a:fmtp in SDP offer [Kvoip*CLI Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Kvoip*CLI Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Kvoip*CLI Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI Looking for 042600 in default (domain 192.168.50.130) [Kvoip*CLI --- Reliably Transmitting (no NAT) to 192.168.50.125:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.50.125:5060;received=192.168.50.125 From: sip:477000...@192.168.50.125;tag=6950F0-25C7 To: sip:0426000...@192.168.50.130;tag=as25696e60 Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 Ok, i see that: 1- Cisco sent the phone number of the caller (47700) 2- I have a To: sip:0426000...@192.168.50.130 192.168.50.130 = My Asterisk Server 192.168.50.125 = My Cisco AS5300 3- i have a No matching peer for '47700' from '192.168.50.125:59124' why he search a peer with 47700 ?? bye Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing out a T1
On Oct 27, 2009, at 10:50 PM, trebaum wrote: Ok, so this might seem like a stupid question, but I don't quite understand how to dial out to the pstn though my T1 from a specific number. Maybe i'm missing something, but everything I'm reading has you dial a number from the group but that's not what i'm looking for. If someone can just point me into the right direction, I would greatly appreciate it. Thanks ~T Ok, so I was able to get my answer from the IRC channel. Just in case anyone is curious in the future, you just need to set the CID to the number you want to call from as you are making the outgoing call. ~T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP client MAC address.
hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client MAC address.
From Linux you could use arp | grep 192.168.0.1 substituting the IP address of the SIP device. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client MAC address. hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client MAC address.
hello david, what in case of sip client is behind NAT, and i want SIP client IP address. not from system from which client registered. if it is a SIP phone then what? if you have any idea then tell me. regards dhaval On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C david.klavers...@intergraph.com wrote: From Linux you could use arp | grep 192.168.0.1 substituting the IP address of the SIP device. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* Wednesday, 28 October 2009 4:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] SIP client MAC address. hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client MAC address.
If there is more than one SIP devices operating from the same NAT device then I'm not sure what you could do as it would always show the same IP for all SIP devices behind the same NAT. If there is only one device behind that NAT making a connection to your server then that is easy, if not I think your screwed. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP client MAC address. hello david, what in case of sip client is behind NAT, and i want SIP client IP address. not from system from which client registered. if it is a SIP phone then what? if you have any idea then tell me. regards dhaval On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C david.klavers...@intergraph.com wrote: From Linux you could use arp | grep 192.168.0.1 substituting the IP address of the SIP device. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client MAC address. hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client MAC address.
hi, though , the SIP client is behinf the NAT cannot we get MAC address of that client , from SIP headers. or do you suggest any alternate method . regards dhaval On Wed, Oct 28, 2009 at 12:20 PM, Klaverstyn, David C david.klavers...@intergraph.com wrote: If there is more than one SIP devices operating from the same NAT device then I’m not sure what you could do as it would always show the same IP for all SIP devices behind the same NAT. If there is only one device behind that NAT making a connection to your server then that is easy, if not I think your screwed. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* Wednesday, 28 October 2009 4:47 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SIP client MAC address. hello david, what in case of sip client is behind NAT, and i want SIP client IP address. not from system from which client registered. if it is a SIP phone then what? if you have any idea then tell me. regards dhaval On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C david.klavers...@intergraph.com wrote: From Linux you could use arp | grep 192.168.0.1 substituting the IP address of the SIP device. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* Wednesday, 28 October 2009 4:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] SIP client MAC address. hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP timestamps
Hi, One more interesting fact, i see correlation with DTMF features, after i disabled corresponding options on dial commands (like htw) the timestamps on rtp are constantly growing and no more one way audio problems after call transfer, hold, parking etc. So it seems there is a bug related to rtp, rfc2833 and timestamp calculation. Or maybe some misconfigured features ? Has anyone seen this behaviour before ? Greetings, Liivo 27.10.2009 16:53, Liivo Vöörmann kirjutas: Hi Alex, Yes, it's almost the same, except the fact that in my case timestamps sometimes decrease drastically. In internal network I have Snom 3xx phones with upgraded firmware, internal leg has no issues, i captured both legs and phones-asterisk part is ok, the other part, asterisk-provider has these issues which are mentioned above. Greetings, Liivo 27.10.2009 15:28, Alex Balashov kirjutas: Liivo, I wonder if you are dealing with this general class of issues: https://issues.asterisk.org/view.php?id=11491 -- Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_echolink
On Sun, 2009-10-25 at 17:10 -0400, Matt wrote: Greetings, Where can I get the chan_echolink channel driver from? I've seen reference to it, but have yet to find a place to download/compile it. It is part of the app_rpt.so module... I am told, but do not see the source with app_rpt. http://qrvc.com/viewvc/projects/allstar/?root=svn http://qrvc.com/viewvc/projects/allstar/astsrc-1.4.23-pre/trunk/asterisk/channels/chan_echolink.c?revision=311root=svn -- Cheers, Michael E-Mail: metalmic...@gmail.com Web:www.mikey.webhop.org signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Server with Panasonic PBX
Hello I have a scenerio to integrate an Existing Panasonic PBX with a new PBX that will be Asterisk system. I know that Asterisk can be integrated with existing Panasonic TDA 100 PBX to recieve calls (ie PSTN lines to Panasonic PBX and out lines of Panasaonic to in of Asterisk PBX). --But i am in doubt if i can make Asterisk to make calls outside from the existing PBX ?(ie usually press nine and then wait for a line. In Asterisk system we will dail 9 first then wait then dail the number). Please share your ideas and experience. All the calls will be recieved by existing Panasonic PBX and an Operator will forward calls to Asterisk PBX ... this is requirement. Please also let me know which type of hardware will be required at Asterisk end to handle lines from a PBX. -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Syncronizing files on different Asterisk servers
Thanks all Robin Drop Box looks cool but I have developed my own code in JAVA that will use Sockets to syncronize files across different servers. Thanks Arjan for the link. @ li...@torrenga.com yeah i do have considered but finally developed my own code for sysncronization. thanks :) if Any One need to know Technical Details regarding JAVA code to handle syncronization of files i am here to explain. On Wed, Oct 21, 2009 at 10:10 PM, li...@torrenga.com wrote: Have you considered rsync? We use it to synchronize voicemail between offices connected through a VPN. Of course you need to run rsync somehow, which is easy with an external command every time someone checks their voice mail, but no reason it couldn't be done with a cron job. Sincerely, Brent A. Torrenga Sorry For the wording actually i need to send to a central server. then a central server to all others. Because all servers have VPN To central Server only. The Drive Mount Option seems cool to me but I dont have any Idea About it . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Cisco AS5300 = Two problems in incoming (extension not found)
Double-check the IP and port associated with the AS5300 peer. The messages below indicate that calls coming in from it are not being matched to the right peer, and as a result, not routed to the correct dial plan context. Phibee Network Operation Center wrote: Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. (042600 = my phone number) First problems: Why he don't see the extension ? sip.conf: [AS5300] host=192.168.50.125 context=as5300-incoming type=peer dtmf=rfc2833 nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp [as5300-incoming] exten = 042600,1,Ringing exten = 042600,2,Answer exten = 042600,3,Dial(SIP/Jpc,25,m) exten = 042600,4,Hangup And second problems: Call from '' to, AS5300 don't sent the number of the caller ? Thanks for your help Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client MAC address.
This is a very strange discussion. MAC addresses can only be discovered for peers that are on the same broadcast segment - which is the realm within which ARP lookups participate. Any peers not on the same logical Layer 2 network are reached through a Layer 3 hop. MAC addresses behind that routing hop cannot be found out because the nodes are in a different MAC domain. NAT has absolutely nothing to do with this, and thus is irrelevant one way or another. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Cisco AS5300 = Two problems in incoming (extension not found)
Try throw the following options into your sip.conf peer: port=5060 insecure=invite,port Phibee Network Operation Center wrote: Phibee Network Operation Center a écrit : Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '042600' rejected because extension not found. (042600 = my phone number) .. I have put a debug: [Kvoip*CLI --- SIP read from UDP://192.168.50.125:59124 --- INVITE sip:0426000...@192.168.50.130:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.125:5060 From: sip:477000...@192.168.50.125;tag=6950F0-25C7 To: sip:0426000...@192.168.50.130 Date: Wed, 28 Oct 2009 05:16:26 GMT Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 3761097657-3266777566-2192416711-2957366127 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: sip:477000...@192.168.50.125;party=calling;screen=yes;privacy=off Timestamp: 1256706986 Contact: sip:477000...@192.168.50.125:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 8642 2741 IN IP4 192.168.50.125 s=SIP Call c=IN IP4 192.168.50.125 t=0 0 m=audio 18726 RTP/AVP 8 101 c=IN IP4 192.168.50.125 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 - [Kvoip*CLI --- (20 headers 11 lines) --- [Kvoip*CLI Sending to 192.168.50.125 : 5060 (no NAT) [Kvoip*CLI Using INVITE request as basis request - e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 [Kvoip*CLI No matching peer for '47700' from '192.168.50.125:59124' [Kvoip*CLI Found RTP audio format 8 [Kvoip*CLI Found RTP audio format 101 [Kvoip*CLI Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI Found audio description format PCMA for ID 8 [Kvoip*CLI Found audio description format telephone-event for ID 101 [Kvoip*CLI Got unsupported a:fmtp in SDP offer [Kvoip*CLI Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Kvoip*CLI Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Kvoip*CLI Peer audio RTP is at port 192.168.50.125:18726 [Kvoip*CLI Looking for 042600 in default (domain 192.168.50.130) [Kvoip*CLI --- Reliably Transmitting (no NAT) to 192.168.50.125:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.50.125:5060;received=192.168.50.125 From: sip:477000...@192.168.50.125;tag=6950F0-25C7 To: sip:0426000...@192.168.50.130;tag=as25696e60 Call-ID: e02f04a1-c2b711de-82b09fc7-b045d...@192.168.50.125 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 Ok, i see that: 1- Cisco sent the phone number of the caller (47700) 2- I have a To: sip:0426000...@192.168.50.130 192.168.50.130 = My Asterisk Server 192.168.50.125 = My Cisco AS5300 3- i have a No matching peer for '47700' from '192.168.50.125:59124' why he search a peer with 47700 ?? bye Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The SIP in the Mobile Phones are not able to register on asterisk
I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to install fring on the mobiles, as it come already with SIP client). The problem that I am facing it: I tried to let my nokia mobile to register on Asterisk, but I am always failing, while I can register to other service provider. I did alot of trouble shooting but did not succeed. Is there any special thing need to do in asterisk configuration to let Nokia (or any other type) of the mobiles that has built in SIP to register on my Asterisk? What could be? Regards Bilal - Try this link http://www.voip-info.org/wiki/view/Asterisk+Connecting+to+the+Cellular+Netwo rk -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, October 27, 2009 3:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] The Mobile devices are not able to register on myasterisk Dear All; I am facing a problem that all the mobile devices that support SIP and are able to register with a lot of providers, they are not able to register on my asterisk. What could be the reason? Any specific thing I have to do? The used port is UDP 5060 Actually, any SIP Phone can register with my asterisk, but when I try from the mobile devices, it does not !! (While these mobiles are able to register with other SIP service provider). Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once likein aring group and put them in conference?
Hi Matt, That is exactly what I am doing now and it has solved my problem. Now all the calls originate instantly with no noticeable delay. -- Zeeshan A Zakaria On Wed, Oct 28, 2009 at 12:18 AM, Matt Riddell li...@venturevoip.comwrote: On 28/10/09 3:52 AM, Danny Nicholas wrote: This might be a better application of a call file than an AMI originate. The AMI originate in this case has to operate in a threaded fashion, whereas if you created a call file for each extension and dumped them into /var/spool/asterisk/outgoing, pbx.c would call all of them at once without the “first pickup” problem. Not true - you can use Async mode in an Asterisk Manager originate command to create a call and return instantly. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Peers still ping with SIP OPTIONS on a reload
Hello, when I remove a peer from my sip.conf and just do a reload, the peer is still ping with SIP OPTIONS until I restart Asterisk, I use Asterisk 1.4.27-rc2. Is it normal? Thanks As an example, I have added and after removed this lines and ;[sip_trk_vm] ;host=88.191.80.8 ;type=peer ;context=default ;dtmfmode=info ;insecure=port,invite ;nat=never ;sendrpid=yes ;disallow=all ;allow=alaw -- -- -- Marc LEURENT lf...@leurent.eu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The SIP in the Mobile Phones are not able to register on asterisk
Have you set the realm in the sip settings in the mobile? Default one is asterisk . It's important too, defining Registration to Always on, because if not, it doesn't enable the wifi connection. Finally, don't enable compression and security --- El mié, 28/10/09, bilal ghayyad bilmar...@yahoo.com escribió: De: bilal ghayyad bilmar...@yahoo.com Asunto: Re: [asterisk-users] The SIP in the Mobile Phones are not able to register on asterisk Para: asterisk-users@lists.digium.com Fecha: miércoles, 28 octubre, 2009 11:53 I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to install fring on the mobiles, as it come already with SIP client). The problem that I am facing it: I tried to let my nokia mobile to register on Asterisk, but I am always failing, while I can register to other service provider. I did alot of trouble shooting but did not succeed. Is there any special thing need to do in asterisk configuration to let Nokia (or any other type) of the mobiles that has built in SIP to register on my Asterisk? What could be? Regards Bilal - Try this link http://www.voip-info.org/wiki/view/Asterisk+Connecting+to+the+Cellular+Netwo rk -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, October 27, 2009 3:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] The Mobile devices are not able to register on myasterisk Dear All; I am facing a problem that all the mobile devices that support SIP and are able to register with a lot of providers, they are not able to register on my asterisk. What could be the reason? Any specific thing I have to do? The used port is UDP 5060 Actually, any SIP Phone can register with my asterisk, but when I try from the mobile devices, it does not !! (While these mobiles are able to register with other SIP service provider). Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0... but if
after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can I get
the call duration in the h extension?
Thanks,
Anahi Ludueña
_
Infórmate, mantente en contacto y encuéntralo todo, a la vez. Con la nueva
Toolbar de MSN nunca has tenido tantas ventajas en tan poco espacio.
http://toolbar.es.msn.com/___
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[asterisk-users] sip fullcontact and port values
Hi We're using asterisk 1.4.17 with RealTime so our port and fullcontact values in out DB get updated dynamically. We use snom handsets and always set the network identity (port) in each phone to something in the 1 range, so that each phone in a single location has a different port. When we look in the DB the location always has the port we set but the port value is often something else? What do these 2 values actually do and where is the port value being generated? Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software for PC-PC voice comunication
If it is SIP use following softphones: 1) X-lite http://counterpath.com/x-lite.htmlactive=4 2) SJPhone http://www.sjlabs.com/sjp.html 3) Snom http://www.snomindia.com/snomsoftphone.htm On Wed, Oct 28, 2009 at 3:36 AM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 27 Oct 2009, giancarlo lombardo wrote: I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. If your PC is running Windows, DIAX is the smallest and easiest soft phone -- no installation, uses IAX instead of SIP. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ramu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ravindra K (ravi...@gmail.com) has sent you a private message
Title: Private Message from Ravindra Ravindra K has sent you a private message Click to read messagePlease read it or Ravindra will think you ignored this :( This message has been forwarded at the request of ravi...@gmail.com. To block all emails from FanIQ, please click here. FanIQ is located at 604 mission St, Suite 600, San Francisco, CA 94105, USA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ravindra K (ravi...@gmail.com) has sent you a private message
Fail. Ravindra K wrote: FanIQ http://FanIQ.com/user/ravibth/connect/334259105/?etid=207 Ravindra K has sent you a private message Click to read message http://FanIQ.com/user/ravibth/connect/334259105/?etid=207 Read private message http://FanIQ.com/user/ravibth/connect/334259105/?etid=207 Please read it or Ravindra will think you ignored this :( This message has been forwarded at the request of ravi...@gmail.com mailto:ravi...@gmail.com. To block all emails from FanIQ, please click here http://www.faniq.com/unsubscribe.php?invite_id=334259105stkn=d2ac70d7c26103ad532d6850dbdee7e2. FanIQ is located at 604 mission St, Suite 600, San Francisco, CA 94105, USA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software for PC-PC voice comunication
Give zoiper a try, http://www.zoiper.com (I'm working for them) Works with SIP and IAX, and should be pretty easy to setup. Zoa giancarlo lombardo wrote: I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR(billsec)
Since CDR(billsec) is a live variable until the Hangup command is issued
(actually until the CDR is written), the only way to get the value (IMO)
would be after the call was completed. You could do a DeadAGI or System
call using CDR(uniqueid) to report the value from the CDR back to another
call or web page.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Wednesday, October 28, 2009 6:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0... but if
after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can I get
the call duration in the h extension?
Thanks,
_
Anahi Ludueña
_
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en
Windows Live Fotos. ¡Pruébalo! http://www.vivelive.com/compartirfotos/
___
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Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?
Mea Culpa?? Since I've only been dabbling with AMI for about 6 weeks, I hadn't stumbled upon the Async parameter. A more correct dissertation of the sentence would be The AMI originate by default operates in a synchronous or threaded fashion, unless you specify Asynchronous mode using Async: true. Guess I'll never be as smart as you, Matt. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Wednesday, October 28, 2009 5:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference? Hi Matt, That is exactly what I am doing now and it has solved my problem. Now all the calls originate instantly with no noticeable delay. -- Zeeshan A Zakaria On Wed, Oct 28, 2009 at 12:18 AM, Matt Riddell li...@venturevoip.com wrote: On 28/10/09 3:52 AM, Danny Nicholas wrote: This might be a better application of a call file than an AMI originate. The AMI originate in this case has to operate in a threaded fashion, whereas if you created a call file for each extension and dumped them into /var/spool/asterisk/outgoing, pbx.c would call all of them at once without the first pickup problem. Not true - you can use Async mode in an Asterisk Manager originate command to create a call and return instantly. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need to find firmware for cisco ata-188
Actually no. But i cannot get a smartnet on an ATA-188. At least not in latinamerica. Actually, all ata-188/186 come with sccp, i just reflashed mine to sip and now i want it back to sccp. it was very dissapointing to learn that i cannot download *any* sccp firmware, not even the original one. Any other suggestions? On Tue, Oct 27, 2009 at 6:53 PM, Steve Howes st...@geekinter.net wrote: On 27 Oct 2009, at 23:29, Erick Perez wrote: any links beside cisco to download the firmware? i do not have a valid contract, so cisco does not allow me to download it. So you want to pirate it instead? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software for PC-PC voice comunication
Hi, If you want an online option to make calls right from webpage, you can use doddle online SIP webphone: http://widget.doddlephone.com/ Sergio On Wed, Oct 28, 2009 at 11:11 AM, Zoa zoach...@securax.org wrote: Give zoiper a try, http://www.zoiper.com (I'm working for them) Works with SIP and IAX, and should be pretty easy to setup. Zoa giancarlo lombardo wrote: I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] R: CDR(billsec)
Hello Anahi,
I've encountered issues with CDR function when I was using the
1.4 version and was trying to get ${CDR(duration)} in extension h.
Passing to 1.6.X.X resolved it.
I hope this helps.
Alex
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Wednesday, October 28, 2009 6:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0... but if
after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can I get
the call duration in the h extension?
Thanks,
Anahi Ludueña
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en Windows
Live Fotos. ¡Pruébalo! http://www.vivelive.com/compartirfotos/
___
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[asterisk-users] deploying asterisk
hello all, friends i am new in asterisk. i had just finished the installation requirment of asterisk. i am using Centos 5.3 in which ill be installing asterisk now guys plz guide me my requirment for deploying asterisk is, i am having a client, (HR Consultancy) where 40 executives work and on each 40 desk, phone is there. i want confrencing facility,hold facility,extention nos,music. when ever call comes to the no it should be routed to phones which ever phone is free. guys plz forgive me if i am not able to make it clear. your support n guidance will be highly appreciated. thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] deploying asterisk
Go to www.asterisk.org http://www.asterisk.org/ and read the install from YUM repo section. This will make the install pretty much automatic. You will then want to set up a queue to route your incoming calls to your 40 extensions. You do not state what technology (SIP/DAHDI) you want to use to connect to the outside world. We aren't here to do this for you, just to offer encouragement. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of aster...@opensourcesolution.in Sent: Wednesday, October 28, 2009 9:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] deploying asterisk hello all, friends i am new in asterisk. i had just finished the installation requirment of asterisk. i am using Centos 5.3 in which ill be installing asterisk now guys plz guide me my requirment for deploying asterisk is, i am having a client, (HR Consultancy) where 40 executives work and on each 40 desk, phone is there. i want confrencing facility,hold facility,extention nos,music. when ever call comes to the no it should be routed to phones which ever phone is free. guys plz forgive me if i am not able to make it clear. your support n guidance will be highly appreciated. thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Clear pending SIP channels
Hi all, I have a question regarding pending (zombie) SIP sessions: on Asterisk CLI, with command 'sip show channels' , I see two channels in use with callID and other infos detailed; also 'sip show inuse' give me same result (in terms of channels usage): PeerUser/ANRCall ID Seq (Tx/Rx) Format Hold Last Message xx.xx.xx.79 209 1745914a212 00102/0 0x8 (alaw) No Tx: ACK xx.xx.xx.34 217 3c515bbb7c8 00101/2 0x8 (alaw) No Rx: ACK * Peer name In use Limit 997 0/0 2 (...) 217 1/0 2 216 0/0 2 215 0/0 2 214 0/0 2 213 0/0 2 212 0/0 2 211 0/0 2 210 0/0 2 209 1/0 2 208 0/0 2 (...) 200 0/0 2 is there a way to drop these calls throughout the CLI or I have to restart asterisk? Many thanks in advance and regards, Alberto Aggio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clear pending SIP channels
Hi, You could use: soft hangup [channel name] Note: You can write the first letter of the channel name, and use [Tab] key to autocomplete. Regards, Aggio Alberto wrote: Hi all, I have a question regarding pending (zombie) SIP sessions: on Asterisk CLI, with command ‘sip show channels’ , I see two channels in use with callID and other infos detailed; also ‘sip show inuse’ give me same result (in terms of channels usage): PeerUser/ANRCall ID Seq (Tx/Rx) Format Hold Last Message xx.xx.xx.79 209 1745914a212 00102/0 0x8 (alaw) No Tx: ACK xx.xx.xx.34 217 3c515bbb7c8 00101/2 0x8 (alaw) No Rx: ACK * Peer name In use Limit 997 0/0 2 (…) 217 1/0 2 216 0/0 2 215 0/0 2 214 0/0 2 213 0/0 2 212 0/0 2 211 0/0 2 210 0/0 2 209 1/0 2 208 0/0 2 (…) 200 0/0 2 is there a way to drop these calls throughout the CLI or I have to restart asterisk? Many thanks in advance and regards, Alberto Aggio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] need a local tech
I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can figure out a way to talk privately about the details (pay, the problems, etc). If someone knows of a company in the area i am open to that to. _ Windows 7: Simplify your PC. Learn more. http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need a local tech
Might want to try these guys http://www.bluegrassnetvoice.com/services/customerpremisePBX.html _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Wednesday, October 28, 2009 9:59 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] need a local tech I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can figure out a way to talk privately about the details (pay, the problems, etc). If someone knows of a company in the area i am open to that to. _ Windows 7: Simplify your PC. Learn more. http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WL MTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: CDR(billsec)
Does this mean its a bug in 1.4 or an enhancement in 1.6? If the latter,
can the change be back-ported?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandru
Oniciuc
Sent: Wednesday, October 28, 2009 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] R: CDR(billsec)
Hello Anahi,
Ive encountered issues with CDR function when I was using
the 1.4 version and was trying to get ${CDR(duration)} in extension h.
Passing to 1.6.X.X resolved it.
I hope this helps.
Alex
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Wednesday, October 28, 2009 6:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0... but if
after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can I get
the call duration in the h extension?
Thanks,
_
Anahi Ludueña
_
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en
Windows Live Fotos. ¡Pruébalo! http://www.vivelive.com/compartirfotos/
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Re: [asterisk-users] deploying asterisk
aster...@opensourcesolution.in wrote: hello all, friends i am new in asterisk. i had just finished the installation requirment of asterisk. i am using Centos 5.3 in which ill be installing asterisk now guys plz guide me my requirment for deploying asterisk is, i am having a client, (HR Consultancy) where 40 executives work and on each 40 desk, phone is there. i want confrencing facility,hold facility,extention nos,music. when ever call comes to the no it should be routed to phones which ever phone is free. guys plz forgive me if i am not able to make it clear. your support n guidance will be highly appreciated. thx Let's be realistic here. You need to 'drink the koolaid' before you install it for a client. What I'm saying is you really need to install Asterisk for yourself and get a good understanding of how it works before attempting to sell and install it for a client. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] deploying asterisk
On Wed, Oct 28, 2009 at 5:05 PM, Darrick Hartman dhart...@djhsolutions.com wrote: Let's be realistic here. You need to 'drink the koolaid' before you install it for a client. What I'm saying is you really need to install Darrick, No, he already drank the koolaid by believing in asterisk. Now he needs to install it in order to eat his own dog food. Keep that straight! /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] deploying asterisk
aster...@opensourcesolution.in wrote: friends i am new in asterisk. i had just finished the installation requirment of asterisk. i am using Centos 5.3 in which ill be installing asterisk now guys plz guide me my requirment for deploying asterisk is, i am having a client, (HR Consultancy) where 40 executives work and on each 40 desk, phone is there. i want confrencing facility,hold facility,extention nos,music. when ever call comes to the no it should be routed to phones which ever phone is free. guys plz forgive me if i am not able to make it clear. your support n guidance will be highly appreciated. thx On Wed, 28 Oct 2009, Darrick Hartman wrote: Let's be realistic here. You need to 'drink the koolaid' before you install it for a client. What I'm saying is you really need to install Asterisk for yourself and get a good understanding of how it works before attempting to sell and install it for a client. 40 angry executives sound like a great way to start a new career. 1) Find someone clueful to do this project for you and let you ride on their coattails. 2) Find someone willing to mentor you through this project. 3) Break out the phonebook and start looking for a good lawyer. Being on the wrong side of a lawsuit is not pleasant. 4) Pick up a copy local paper and check out the want-ads. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: CDR(billsec)
I have used ${CDR(billsec)} in asterisk 1.4.17
How I used it was
h,1,SET(BILLTIME=${CDR(billsec)})
h,2,DeadAGI(hangup.php)
My DeadAGI script could use my BILLSEC variable and it was always
consistent with the CDR too.
Danny Nicholas wrote:
Does this mean it’s a bug in 1.4 or an enhancement in 1.6? If the
latter, can the change be back-ported?
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Alexandru Oniciuc
*Sent:* Wednesday, October 28, 2009 8:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] R: CDR(billsec)
Hello Anahi,
I’ve encountered issues with CDR function when I was using the 1.4
version and was trying to get ${CDR(duration)} in extension h.
Passing to 1.6.X.X resolved it.
I hope this helps.
Alex
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Anahi
Ludueña
*Sent:* Wednesday, October 28, 2009 6:35 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0...
but if after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can
I get the call duration in the h extension?
Thanks,
**
**
**Anahi Ludueña**
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en
Windows Live Fotos. ¡Pruébalo! http://www.vivelive.com/compartirfotos/
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--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] R: CDR(billsec)
Something seems to be missing here- you don't pass ${BILLTIME} to hangup.php
(as far as I can see), so it seems that hangup.php is operating (at least
somewhat) independently of the dialplan. The OP seemed to want in-line
knowledge of his billable seconds.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Wednesday, October 28, 2009 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] R: CDR(billsec)
I have used ${CDR(billsec)} in asterisk 1.4.17
How I used it was
h,1,SET(BILLTIME=${CDR(billsec)})
h,2,DeadAGI(hangup.php)
My DeadAGI script could use my BILLSEC variable and it was always
consistent with the CDR too.
Danny Nicholas wrote:
Does this mean its a bug in 1.4 or an enhancement in 1.6? If the
latter, can the change be back-ported?
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Alexandru Oniciuc
*Sent:* Wednesday, October 28, 2009 8:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] R: CDR(billsec)
Hello Anahi,
Ive encountered issues with CDR function when I was using the 1.4
version and was trying to get ${CDR(duration)} in extension h.
Passing to 1.6.X.X resolved it.
I hope this helps.
Alex
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Anahi
Ludueña
*Sent:* Wednesday, October 28, 2009 6:35 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0...
but if after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can
I get the call duration in the h extension?
Thanks,
**
**
**Anahi Ludueña**
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en
Windows Live Fotos. ¡Pruébalo! http://www.vivelive.com/compartirfotos/
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--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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[asterisk-users] R: R: CDR(billsec)
I used 1.4.21 and this(${CDR(duration)}) didn't work:
exten = h,1,Verbose( (${CDR(dst)}) # Call from ${CDR(clid)} ended at
${STRFTIME(${EPOCH},,%d/%m/%Y %H:%M:%S)}. Duration(sec): ${CDR(duration)}.)
Alex
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Ishfaq Malik
Inviato: mercoledì 28 ottobre 2009 17.32
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] R: CDR(billsec)
I have used ${CDR(billsec)} in asterisk 1.4.17
How I used it was
h,1,SET(BILLTIME=${CDR(billsec)})
h,2,DeadAGI(hangup.php)
My DeadAGI script could use my BILLSEC variable and it was always
consistent with the CDR too.
Danny Nicholas wrote:
Does this mean it's a bug in 1.4 or an enhancement in 1.6? If the
latter, can the change be back-ported?
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Alexandru Oniciuc
*Sent:* Wednesday, October 28, 2009 8:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] R: CDR(billsec)
Hello Anahi,
I've encountered issues with CDR function when I was using the 1.4
version and was trying to get ${CDR(duration)} in extension h.
Passing to 1.6.X.X resolved it.
I hope this helps.
Alex
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Anahi
Ludueña
*Sent:* Wednesday, October 28, 2009 6:35 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0...
but if after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can
I get the call duration in the h extension?
Thanks,
**
**
**Anahi Ludueña**
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en
Windows Live Fotos. ¡Pruébalo! http://www.vivelive.com/compartirfotos/
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--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Software for PC-PC voice comunication
Thanks, it sounds good. 2009/10/27 giancarlo lombardo gianclomba...@gmail.com I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help -- Giancarlo Lombardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] deploying asterisk
I would second Steve's advice very strongly. Steve Edwards wrote: aster...@opensourcesolution.in wrote: friends i am new in asterisk. i had just finished the installation requirment of asterisk. i am using Centos 5.3 in which ill be installing asterisk now guys plz guide me my requirment for deploying asterisk is, i am having a client, (HR Consultancy) where 40 executives work and on each 40 desk, phone is there. i want confrencing facility,hold facility,extention nos,music. when ever call comes to the no it should be routed to phones which ever phone is free. guys plz forgive me if i am not able to make it clear. your support n guidance will be highly appreciated. thx On Wed, 28 Oct 2009, Darrick Hartman wrote: Let's be realistic here. You need to 'drink the koolaid' before you install it for a client. What I'm saying is you really need to install Asterisk for yourself and get a good understanding of how it works before attempting to sell and install it for a client. 40 angry executives sound like a great way to start a new career. 1) Find someone clueful to do this project for you and let you ride on their coattails. 2) Find someone willing to mentor you through this project. 3) Break out the phonebook and start looking for a good lawyer. Being on the wrong side of a lawsuit is not pleasant. 4) Pick up a copy local paper and check out the want-ads. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Having a heck of a time
This has been a rollercoaster ride Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers) Where I stand right now, I have a PRI on the gateway and circuit is working I can make calls through the gateway Here is my problem: DAHDI_TEST is not returning anything and DAHDI_MONITOR doesn't work [r...@lin-vgw1 asterisk]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... --- Results after 0 passes --- Best: 0.000 -- Worst: 100.000 -- Average: 100.00, Difference: 100.00 Also if I establish a call and run dahdi_monitor it doesnt look quite like it is supposed to: [r...@lin-vgw1 asterisk]# dahdi_monitor 121 -vv Visual Audio Levels. Use chan_dahdi.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) I'm looking for some ideas here? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP 18x Messages
When I make an outbound call I hear a half of a ring and than silence until the call opens up. It seems asterisk is sending a 183 after the 180 message. My CPE device does not support multiple 18x messages in the same call setup. When we receive the 180 we present ring back to the phone, but when we receive the 183 we get confused and stop the ring back tone, but do not open up the early media path for the ring back to be played from the network. In Metaswitch the configuration knob to correct this is “Superfluous 18x messages”, I don’t know what it takes to configure Asterisk that way. Can anyone help with this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having a heck of a time
On Wed, Oct 28, 2009 at 2:09 PM, Robert Grignon rgrig...@fleetone.com wrote: This has been a rollercoaster ride Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers) Where I stand right now, I have a PRI on the gateway and circuit is working I can make calls through the gateway Here is my problem: DAHDI_TEST is not returning anything and DAHDI_MONITOR doesn't work [r...@lin-vgw1 asterisk]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... --- Results after 0 passes --- Best: 0.000 -- Worst: 100.000 -- Average: 100.00, Difference: 100.00 Also if I establish a call and run dahdi_monitor it doesnt look quite like it is supposed to: [r...@lin-vgw1 asterisk]# dahdi_monitor 121 -vv Visual Audio Levels. Use chan_dahdi.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) I'm looking for some ideas here? I could be way off, but I think the fact that it says it's opening the psuedo interface implies that it doesn't see your Sangoma card. You might want to try and check with Sangoma support to see what they have to say about that. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having a heck of a time
Did you use ./Setup dahdi when installing the wanpipe drivers? http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having a heck of a time
That was a good thought. I have 3 other gateways in production and I ran dahdi_test and zttest (older gateways) and they all said they were opening a psedu device -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BJ Weschke Sent: Wednesday, October 28, 2009 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Having a heck of a time On Wed, Oct 28, 2009 at 2:09 PM, Robert Grignon rgrig...@fleetone.com wrote: This has been a rollercoaster ride Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers) Where I stand right now, I have a PRI on the gateway and circuit is working I can make calls through the gateway Here is my problem: DAHDI_TEST is not returning anything and DAHDI_MONITOR doesn't work [r...@lin-vgw1 asterisk]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... --- Results after 0 passes --- Best: 0.000 -- Worst: 100.000 -- Average: 100.00, Difference: 100.00 Also if I establish a call and run dahdi_monitor it doesnt look quite like it is supposed to: [r...@lin-vgw1 asterisk]# dahdi_monitor 121 -vv Visual Audio Levels. Use chan_dahdi.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) I'm looking for some ideas here? I could be way off, but I think the fact that it says it's opening the psuedo interface implies that it doesn't see your Sangoma card. You might want to try and check with Sangoma support to see what they have to say about that. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having a heck of a time
Yes I did that... I even recompiled dahdi-linux and tools after wanpipe install... Once I did that it recognized the card and said I could run dahdi_genconf modules which in turn would only load the cards that it seeing. I had the PRI running in slot 6. Once I unplugged the PRI I was able to get a response from dahdi_test I then wondered if it was looking for a circuit on channel 1 (this didnt make much sense because the PRI is getting timing from the telco and the port location should not matter) I then moved the PRI to channel 1 and dahdi_test returned the following: [r...@lin-vgw1 asterisk]# dahdi_test -vc 10 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8190.808 system clock sample intervals (99.985%) 8192 samples in 8190.288 system clock sample intervals (99.979%) 8192 samples in 8190.776 system clock sample intervals (99.985%) 8192 samples in 8190.872 system clock sample intervals (99.986%) 8192 samples in 8190.720 system clock sample intervals (99.984%) 8192 samples in 8190.833 system clock sample intervals (99.986%) 8192 samples in 8190.960 system clock sample intervals (99.987%) 8192 samples in 8190.864 system clock sample intervals (99.986%) 8192 samples in 8190.744 system clock sample intervals (99.985%) 8192 samples in 8190.800 system clock sample intervals (99.985%) --- Results after 10 passes --- Best: 99.987 -- Worst: 99.979 -- Average: 99.984940, Difference: 99.984940 GO figure... I did notice this in the logs and am not sure what to make of the CT_C8_A clock behavior does not conform to the H.100 spec reference: Oct 28 13:26:19 lin-vgw1 kernel: wanpipe6: Lost of Signal is detected! (Unplugged circuit and moved to channel 1) Oct 28 13:26:20 lin-vgw1 kernel: wanpipe2: Lost of Signal is detected! Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6:LOF : ON Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6:RED : ON Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: T1 disconnected! Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: Enable transmit RAI alarm Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: AFT communications disabled! (Dev Cnt: 6 Cause: Link Down) Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: TDM Free Run Timing Enabled 1 ms Oct 28 13:26:23 lin-vgw1 kernel: wanpipe6:LOS : ON Oct 28 13:26:24 lin-vgw1 kernel: wanpipe2:LOS : ON Oct 28 13:27:06 lin-vgw1 kernel: wanec1: The H100 slave has lost its framing on the bus! Oct 28 13:27:06 lin-vgw1 kernel: wanec1: The CT_C8_A clock behavior does not conform to the H.100 spec! Oct 28 13:27:06 lin-vgw1 kernel: wanpipe5: Lost of Signal is cleared! Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1:RAI : ON Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1: Open Circuit is cleared! Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1: Lost of Signal is cleared! Oct 28 13:27:17 lin-vgw1 kernel: wanpipe1:RAI : OFF Oct 28 13:27:17 lin-vgw1 kernel: wanpipe5:LOS : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:LOS : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:LOF : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:RED : OFF Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: T1 connected! Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: AFT communications enabled! Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: AFT Global TDM Intr Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: Global TDM Ring Resync TDM = 0x1 Oct 28 13:27:21 lin-vgw1 kernel: ADDRCONF(NETDEV_CHANGE): w1g1: link becomes ready From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King Sent: Wednesday, October 28, 2009 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Having a heck of a time Did you use ./Setup dahdi when installing the wanpipe drivers? http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Having a heck of a time
Upon further research I kind of answered my own question.. But I will share... If you are seeing multiple H.100 errors in your system log then the hardware echo canceler does not have a good clock source. On our more recent drivers 3.3.12 and up the first port that starts up will be the clocking source. So if your wanpipe1 is not connected then please configure your card to only start the first port that connects. If you have an older driver then the timing source is the first physical port on the card. So if you are not using the first physical port then please follow the steps below to set another port as a timing source. Please note that only one port can act as timing source for HWEC in a particular AFT102/104/108 card, in other words you can only set HWEC_CLCKSRC = YES for only one port for a card! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Wednesday, October 28, 2009 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Having a heck of a time Yes I did that... I even recompiled dahdi-linux and tools after wanpipe install... Once I did that it recognized the card and said I could run dahdi_genconf modules which in turn would only load the cards that it seeing. I had the PRI running in slot 6. Once I unplugged the PRI I was able to get a response from dahdi_test I then wondered if it was looking for a circuit on channel 1 (this didnt make much sense because the PRI is getting timing from the telco and the port location should not matter) I then moved the PRI to channel 1 and dahdi_test returned the following: [r...@lin-vgw1 asterisk]# dahdi_test -vc 10 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8190.808 system clock sample intervals (99.985%) 8192 samples in 8190.288 system clock sample intervals (99.979%) 8192 samples in 8190.776 system clock sample intervals (99.985%) 8192 samples in 8190.872 system clock sample intervals (99.986%) 8192 samples in 8190.720 system clock sample intervals (99.984%) 8192 samples in 8190.833 system clock sample intervals (99.986%) 8192 samples in 8190.960 system clock sample intervals (99.987%) 8192 samples in 8190.864 system clock sample intervals (99.986%) 8192 samples in 8190.744 system clock sample intervals (99.985%) 8192 samples in 8190.800 system clock sample intervals (99.985%) --- Results after 10 passes --- Best: 99.987 -- Worst: 99.979 -- Average: 99.984940, Difference: 99.984940 GO figure... I did notice this in the logs and am not sure what to make of the CT_C8_A clock behavior does not conform to the H.100 spec reference: Oct 28 13:26:19 lin-vgw1 kernel: wanpipe6: Lost of Signal is detected! (Unplugged circuit and moved to channel 1) Oct 28 13:26:20 lin-vgw1 kernel: wanpipe2: Lost of Signal is detected! Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6:LOF : ON Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6:RED : ON Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: T1 disconnected! Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: Enable transmit RAI alarm Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: AFT communications disabled! (Dev Cnt: 6 Cause: Link Down) Oct 28 13:26:22 lin-vgw1 kernel: wanpipe6: TDM Free Run Timing Enabled 1 ms Oct 28 13:26:23 lin-vgw1 kernel: wanpipe6:LOS : ON Oct 28 13:26:24 lin-vgw1 kernel: wanpipe2:LOS : ON Oct 28 13:27:06 lin-vgw1 kernel: wanec1: The H100 slave has lost its framing on the bus! Oct 28 13:27:06 lin-vgw1 kernel: wanec1: The CT_C8_A clock behavior does not conform to the H.100 spec! Oct 28 13:27:06 lin-vgw1 kernel: wanpipe5: Lost of Signal is cleared! Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1:RAI : ON Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1: Open Circuit is cleared! Oct 28 13:27:07 lin-vgw1 kernel: wanpipe1: Lost of Signal is cleared! Oct 28 13:27:17 lin-vgw1 kernel: wanpipe1:RAI : OFF Oct 28 13:27:17 lin-vgw1 kernel: wanpipe5:LOS : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:LOS : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:LOF : OFF Oct 28 13:27:18 lin-vgw1 kernel: wanpipe1:RED : OFF Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: T1 connected! Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: AFT communications enabled! Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: AFT Global TDM Intr Oct 28 13:27:21 lin-vgw1 kernel: wanpipe1: Global TDM Ring Resync TDM = 0x1 Oct 28 13:27:21 lin-vgw1 kernel: ADDRCONF(NETDEV_CHANGE): w1g1: link becomes ready From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King Sent: Wednesday, October 28, 2009 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Having a heck of a time Did you use ./Setup dahdi when installing the wanpipe drivers? http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi
Re: [asterisk-users] SIP 18x Messages
Tim King wrote: When I make an outbound call I hear a half of a ring and than silence until the call opens up. It seems asterisk is sending a 183 after the 180 message. My CPE device does not support multiple 18x messages in the same call setup. When we receive the 180 we present ring back to the phone, but when we receive the 183 we get confused and stop the ring back tone, but do not open up the early media path for the ring back to be played from the network. In Metaswitch the configuration knob to correct this is “Superfluous 18x messages”, I don’t know what it takes to configure Asterisk that way. Can anyone help with this. Check out the 'progressinband' configuration option. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 18x Messages
I thought that was it and tried each setting and did not see any change on the line. On Wed, Oct 28, 2009 at 3:58 PM, Kevin P. Fleming kpflem...@digium.comwrote: Tim King wrote: When I make an outbound call I hear a half of a ring and than silence until the call opens up. It seems asterisk is sending a 183 after the 180 message. My CPE device does not support multiple 18x messages in the same call setup. When we receive the 180 we present ring back to the phone, but when we receive the 183 we get confused and stop the ring back tone, but do not open up the early media path for the ring back to be played from the network. In Metaswitch the configuration knob to correct this is “Superfluous 18x messages”, I don’t know what it takes to configure Asterisk that way. Can anyone help with this. Check out the 'progressinband' configuration option. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH
I am having a strange problem with MOH. Say I have two users, A and B. I can set MOH in the extension for B and if A calls B and B hits hold, A will hear B's hold music. If however A hits hold, it goes to the default music. If I pull the setmusiconhold from extensions.conf and use musicclass in sip.conf under the peer A, I get the same thing. Peer A has musicclass set and A calls B and B hits hold, A hears their own music that was specified and B hears the default. I can't find any way to call someone, put them on hold and have them hear my music, I can hear what they specify, but I can't specify what they hear (this is all assuming calls are within the same * box). Any ideas how to set that? Peder ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH
Peder wrote: I am having a strange problem with MOH. Say I have two users, A and B. I can set MOH in the extension for B and if A calls B and B hits hold, A will hear B's hold music. If however A hits hold, it goes to the default music. If I pull the setmusiconhold from extensions.conf and use musicclass in sip.conf under the peer A, I get the same thing. Peer A has musicclass set and A calls B and B hits hold, A hears their own music that was specified and B hears the default. I can't find any way to call someone, put them on hold and have them hear my music, I can hear what they specify, but I can't specify what they hear (this is all assuming calls are within the same * box). Any ideas how to set that? This is what 'mohsuggest' is for; if you don't specify musicclass for any of your endpoints, but instead specify mohsuggest, then when that endpoint places another endpoint on hold, the endpoint that is placed on hold will hear the MOH 'suggested' by the endpoint that put them on hold. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need a local tech
On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote: I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can figure out a way to talk privately about the details (pay, the problems, etc). If someone knows of a company in the area i am open to that to. __ Windows 7: Simplify your PC. Learn more. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Ott, Why do you put that URL in your sig? A) This is the non-commercial list B) We rather be refrained from such trash C) Instead of waving a white flag, do a rm -rf / or the M$-equivalent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Cisco AS5300 = Two problems in
Please post your dial peer configurations. We have as5400 (5) working with asterisk servers also. The cisco routers are at the edge of the network (connected to PSTN via E1) and send calls to asterisk over SIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need a local tech
On Wed, Oct 28, 2009 at 10:16:16PM +0100, Hans Witvliet wrote: On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote: __ Windows 7: Simplify your PC. Learn more. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Ott, Why do you put that URL in your sig? It's not mim. It's his email provider (the 'windows 7' part) and this list's provider (the rest). He's clearly not advertising himself. So I don't see as issue here. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to announce the agent answering in a queue to the caller
I've tested and confirm that the AGI script can do that. i had to enable setinterfacevar=yes in the queue conf and then can read the MEMBERINTERFACE channel variable. Just because it can be useful for someone else. On Fri, Oct 23, 2009 at 9:44 PM, nik600 nik...@gmail.com wrote: Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the call? After the AGI script the call is linked with the operator even if there is an Answer into the AGI? Thanks to all -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 302 Moved Temporarily
Hello, I have an * installation that sometimes receives a 302 "Moved Temporarily" response to an INVITE. * sends the ACK but takes about 30 seconds to start the new INVITE to the new destination (from Contact Header). I have set core debugging to 20 but do not see any abnormal message. [provider] type=friend host=a.reachable.host.ip context=incoming_context dtmfmode=rfc2833 canreinvite=yes qualify=yes The call is generated from a PHP AGI script with the Dial "RrCL" options. Does any one have an idea why it could be lasting about 30 seconds to start the new call?? Regards, Juan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID
I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons
1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except
callerid are recorded properly after every call. I have both a clid
and callerid field in the database but both fields are empty. In
cdr_mysql.conf I have this alias in the [columns] section:
alias start = calldate
alias callerid = clid
Is anyone using this version of Asterisk and Addons that can confirm if
they are getting the CLID? If I do a NoOp(CLID: ${CDR(clid)}) I see the
callerid in the CLI. The CSV file also records the callerid. Maybe a
bug in cdr_addon_mysql ?
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Re: [asterisk-users] Asterisk Server with Panasonic PBX
Any simple legacy integration will work. Search on voip-info.org Here are some problems that I know exist with panasonic systems on their SLT (analog) ports: 1. No CPC, Asterisk if connected using station ports on the TDA to FXO on asterisk, will not detect hangups since the TDA will not send them. 2. BLF and the like will not work. 3. There are different ways of making sure that asterisk users should be able to use the lines on the TDA depending on how you chose to connect them both. On a side note, may I ask why you are integrating asterisk with the TDA? What is the functionality you plan on gaining? On Wed, Oct 28, 2009 at 4:50 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello I have a scenerio to integrate an Existing Panasonic PBX with a new PBX that will be Asterisk system. I know that Asterisk can be integrated with existing Panasonic TDA 100 PBX to recieve calls (ie PSTN lines to Panasonic PBX and out lines of Panasaonic to in of Asterisk PBX). --But i am in doubt if i can make Asterisk to make calls outside from the existing PBX ?(ie usually press nine and then wait for a line. In Asterisk system we will dail 9 first then wait then dail the number). Please share your ideas and experience. All the calls will be recieved by existing Panasonic PBX and an Operator will forward calls to Asterisk PBX ... this is requirement. Please also let me know which type of hardware will be required at Asterisk end to handle lines from a PBX. -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GUI for hunt groups?
Hi, all. I've got an Asterisk box installed that I'd really like to leverage -- and installing a GUI for hunt groups would be awesome. So long as I can have a trial copy, I could even pay money. It would have to be able to make use of both SIP and ZAP extensions. Suggestions? (Note: I wouldn't much care about the GUI, myself, but my boss is all over one.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for hunt groups?
Freepbx comes with setup of ring groups and queues with different hunt strategies Also it has Flash Operator Panel which gives you the state of the system in real time graphical format No money - just a small bit of installation time and learning how to use it Cheers Duncan Ken D'Ambrosio wrote: Hi, all. I've got an Asterisk box installed that I'd really like to leverage -- and installing a GUI for hunt groups would be awesome. So long as I can have a trial copy, I could even pay money. It would have to be able to make use of both SIP and ZAP extensions. Suggestions? (Note: I wouldn't much care about the GUI, myself, but my boss is all over one.) Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic DNS trunk
I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Server with Panasonic PBX
C F thankyou very much. when i make a call to Asterisk server recieves and works fine. But as to make external calls we have to press nine so supposed a logic to dial 9 first then wait and then dail other number. But as i dail 9 asterisk show the call as connected with alot of noise. Please help in how to handle this On a side note, may I ask why you are integrating asterisk with the TDA? What is the functionality you plan on gaining? Nothing very important logical its a client who don't want to trash its existing system. So we need to do that. I know Asterisk is far more better to use and handle his requirements but On Thu, Oct 29, 2009 at 5:25 AM, C F shma...@gmail.com wrote: Any simple legacy integration will work. Search on voip-info.org Here are some problems that I know exist with panasonic systems on their SLT (analog) ports: 1. No CPC, Asterisk if connected using station ports on the TDA to FXO on asterisk, will not detect hangups since the TDA will not send them. 2. BLF and the like will not work. 3. There are different ways of making sure that asterisk users should be able to use the lines on the TDA depending on how you chose to connect them both. On a side note, may I ask why you are integrating asterisk with the TDA? What is the functionality you plan on gaining? On Wed, Oct 28, 2009 at 4:50 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello I have a scenerio to integrate an Existing Panasonic PBX with a new PBX that will be Asterisk system. I know that Asterisk can be integrated with existing Panasonic TDA 100 PBX to recieve calls (ie PSTN lines to Panasonic PBX and out lines of Panasaonic to in of Asterisk PBX). --But i am in doubt if i can make Asterisk to make calls outside from the existing PBX ?(ie usually press nine and then wait for a line. In Asterisk system we will dail 9 first then wait then dail the number). Please share your ideas and experience. All the calls will be recieved by existing Panasonic PBX and an Operator will forward calls to Asterisk PBX ... this is requirement. Please also let me know which type of hardware will be required at Asterisk end to handle lines from a PBX. -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DNS trunk
If the trunk is a dynamic IP you need the other end to register to Asterisk, so letting Asterisk know the new IP. Regards, Juan B.Masoud @ SH wrote: I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID
On Wednesday 28 October 2009 17:57:49 Carlos Chavez wrote: I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons 1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except callerid are recorded properly after every call. I have both a clid and callerid field in the database but both fields are empty. In cdr_mysql.conf I have this alias in the [columns] section: alias start = calldate alias callerid = clid Get rid of this alias callerid = clid line. What it does is to tell the driver to put the CDR variable called callerid into the clid column in the database, overriding the builtin clid mapping. Then reload. If you want the Caller*ID information in the callerid column, then your mapping is backwards and should be alias clid = callerid. Remember, the arrow points in the direction that the information flows: FROM the cdr TO the database. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
