Re: [asterisk-users] Use of 603 Declined
Agree that the 603 is wrong. It hasn't caused me issues but I see where it could. And it goes against what I have been teaching in my classes, which is irritating ;-) In Asterisk, it's only used when we have no other hangup cause - and is propably an indication that there is a code path that doesn't set the proper hangup cause. I'm willing to implement a bug fix for this and have it configurable in released code and make it default in trunk. I don't want to force changed behaviour in released code. Now, what would be a 4xx class generic error message on the same level? /O -- _ -- 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: rtp.c:883 ast_rtcp_read: RTCP Read too short
Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 21:41 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short I would very much like to get a wireshark trace (pcap file) of that session to try to understand this message. I don't think the RTCP read to short affects your communication - that is propably another issue. But since I've become a bit occupied with RTCP lately, I would like to see what causes this message. If you have the oppurtunity, or someone else that sees this message in your Aterisk, please send me the packet trace off list, directly to my personal e-mail o...@edvina.net. Thanks for the assistance! /O -- _ -- 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] Use of 603 Declined
I don't know about 4xx, but 503 would be more benign for general/ miscellaneous errors than 603. -- Sent from mobile device On Jan 29, 2010, at 3:54 AM, Olle E. Johansson o...@edvina.net wrote: Agree that the 603 is wrong. It hasn't caused me issues but I see where it could. And it goes against what I have been teaching in my classes, which is irritating ;-) In Asterisk, it's only used when we have no other hangup cause - and is propably an indication that there is a code path that doesn't set the proper hangup cause. I'm willing to implement a bug fix for this and have it configurable in released code and make it default in trunk. I don't want to force changed behaviour in released code. Now, what would be a 4xx class generic error message on the same level? /O -- _ -- 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] Use of 603 Declined
29 jan 2010 kl. 10.25 skrev Alex Balashov: I don't know about 4xx, but 503 would be more benign for general/ miscellaneous errors than 603. 503 indicates that there's a problem with the server, so that's not a good replacement. We're sending this when there's a failed call, in most cases because of the outbound channel failure without a proper hangup cause being set in that channel driver (may very well be chan_sip :-)) No, we need something in the 4xx class. I haven't had time to go through and consider all the options in the massive set of RFCs we have to work with, but will try to do that tonight after a Friday night dinner - if no one on the list comes up with the solution before that. Isn't that a hacker way of spending Friday night - enjoying the wonderful prose of RFC3261 and companions? /O -- Sent from mobile device On Jan 29, 2010, at 3:54 AM, Olle E. Johansson o...@edvina.net wrote: Agree that the 603 is wrong. It hasn't caused me issues but I see where it could. And it goes against what I have been teaching in my classes, which is irritating ;-) In Asterisk, it's only used when we have no other hangup cause - and is propably an indication that there is a code path that doesn't set the proper hangup cause. I'm willing to implement a bug fix for this and have it configurable in released code and make it default in trunk. I don't want to force changed behaviour in released code. Now, what would be a 4xx class generic error message on the same level? /O -- _ -- 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 --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- 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] chan_mobile problem with audio (distorted)
Hello to all. I have installed asterisk-1.6.2.1 + asterisk-addons-1.6.2.0 (for chan_mobile) + bluez-4.60. Bluetooth Dongle: Canyon CN-BTU4 (0a12:0001 Cambridge Silicon Radio, Ltd Bluetooth Dongle (HCI mode)) Device Descriptor: bLength18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 224 Wireless bDeviceSubClass 1 Radio Frequency bDeviceProtocol 1 Bluetooth bMaxPacketSize064 idVendor 0x0a12 Cambridge Silicon Radio, Ltd idProduct 0x0001 Bluetooth Dongle (HCI mode) bcdDevice 31.64 iManufacturer 0 iProduct2 EDRClassone iSerial 0 bNumConfigurations 1 Mobile Phone: Nokia E51 Firmware: 400.34.011 26.05.2009 RM-244 Nokia E51 (05) OS: Slackware Linux (current) Kernel: Linux 2.6.32.5-smp #2 SMP Sat Jan 23 01:37:47 CST 2010 i686 AMD Athlon(tm) 64 X2 Dual Core Processor 4200+ AuthenticAMD GNU/Linux I've got everything working, but after a few call audio got distorted similar to this: https://issues.asterisk.org/view.php?id=13282 Here is some errors in syslog: bluetoothd[2588]: Bluetooth daemon 4.60 bluetoothd[2589]: Starting SDP server crc16: exports duplicate symbol crc16 (owned by kernel) Bluetooth: L2CAP ver 2.14 Bluetooth: L2CAP socket layer initialized Bluetooth: SCO (Voice Link) ver 0.6 Bluetooth: SCO socket layer initialized bluetoothd[2589]: HCI dev 0 registered bluetoothd[2589]: HCI dev 0 up bluetoothd[2589]: Starting security manager 0 bluetoothd[2589]: Parsing /etc/bluetooth/serial.conf failed: No such file or directory crc16: exports duplicate symbol crc16 (owned by kernel) Bluetooth: RFCOMM TTY layer initialized Bluetooth: RFCOMM socket layer initialized Bluetooth: RFCOMM ver 1.11 bluetoothd[2589]: Adapter /org/bluez/2588/hci0 has been enabled bluetoothd[2589]: link_key_request (sba=00:1A:7D:11:6E:3D, dba=00:24:03:BC:4F:DD) . btusb_isoc_complete: hci0 corrupted SCO packet btusb_isoc_complete: hci0 corrupted SCO packet hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 btusb_isoc_complete: hci0 corrupted SCO packet btusb_isoc_complete: hci0 corrupted SCO packet hci_scodata_packet: hci0 SCO packet for unknown connection handle 46 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 hci_scodata_packet: hci0 SCO packet for unknown connection handle 0 Asterisk show no errors... -- Marian Zahariev -- _ -- 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] VUC Today at 1 PM EST: Counterpath/Bria
Hi, In the aftermath of Digium's and Counterpath's Bria for Asterisk announcement, we're happy to chat with Todd Carothers, Counterpath Product Manager today at 1 PM EST. For more info, http://vuc.me Join us on IRC #vuc on Freenode.net or use the web client at http://vuc.me/irc Call in starting at around 12 Noon EST: sip:200...@login.zipdx.com Hear you there! /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
[asterisk-users] Address family not supported by protocol
After an upgrade to asterisk 1.6.2.1 I'm unable to make outgoing calls via Vitelity. I get lots of these on my asterisk console: [Jan 27 08:58:41] WARNING[25653]: chan_sip.c:3581 __sip_xmit: sip_xmit of 0x834ae08 (len 927) to 64.2.142.18:0 returned -1: Address family not supported by protocol There's a bug report that seems to address this but it's categorized as minor and hasn't been acted on. https://issues.asterisk.org/view.php?id=15827 The problem seems to be related to dnsmgr. Calls go out when asterisk is restarted but then begin to fail once dnsmgr has updated the outgoing hosts's IP address. Calls fail silently. There is no indication to the caller that the call is not going through. Has anyone else seen this? I'd like to help debug this if I can, although I'm not sure where to start. -- Chris -- _ -- 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] disable comfort noise
Hi, How can I disable comfort noise on Asterisk? Szabolcs Szasz -- _ -- 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] disable comfort noise
Szasz Szabolcs wrote: How can I disable comfort noise on Asterisk? Asterisk does not have a comfort noise generator, so there is nothing to disable. You'll have to be more specific about what you are trying to accomplish. -- 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
[asterisk-users] 1 Asterisk server, multiple registrations to ITSP
Hello list ! Having troubles with multiple registrations to one and the same ITSP. This sip.conf : register = user1:pass...@sip.itsp register = user2:pass...@sip.itsp ; outgoing conversations [user1-out] type=peer host=sip.ITSP username=user1 secret=secret1 fromuser=user1 dtmfmode=rfc2833 canreinvite=no qualify=yes disallow=all allow=alaw allow=gsm amaflags=documentation ; incoming conversations [user1] type=peer host=sip.ITSP context=user1incoming disallow=all allow=alaw allow=gsm qualify=yes canreinvite=no dtmfmode=rfc2833 amaflags=documentation ; outgoing conversations [user2-out] type=peer host=sip.ITSP username=user2 secret=secret2 fromuser=user2 dtmfmode=rfc2833 canreinvite=no qualify=yes disallow=all allow=alaw allow=gsm amaflags=documentation ; incoming conversations [user2] type=peer host=sip.ITSP context=user2incoming disallow=all allow=alaw allow=gsm qualify=yes canreinvite=no dtmfmode=rfc2833 amaflags=documentation If user1 is the only account, there is no problem. But if user2 is also active, then the CLI shows : [Jan 29 11:25:01] WARNING[1473]: chan_sip.c:8711 check_auth: username mismatch, have user2-out, digest has anonymous [Jan 29 11:25:01] NOTICE[1473]: chan_sip.c:14636 handle_request_invite: Failed to authenticate user 329990102 sip:329990...@ipitsp;tag=f395ca81fc053ae2fd9b1c62 329990102 is the number trying to contact me... Is there something I need to do to distinguish between the 2 accounts for incoming calls ?? Or does my ITSP need to send information with the invite for the incoming call so I can distinguish ?? Thank you for your feedback. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 card FXS problems
I don't know what it showed at that time, I didn't know to look for that. I'll try to get the customer's permission to re-create the symptoms this weekend and post back with the lsdahdi output. Thank you, Noah Engelberth Direct Link Computer Systems - Message: 12 Date: Thu, 28 Jan 2010 23:13:56 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com Message-ID: 20100128211356.gx3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- 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] Cell Phone dialing
Apparently it is a telco issue. As I stated, when I do the dial from my Polycom 501 handset, the landline call indicates ringing at 0:01 or 0:02 on the timer; the cell call doesn't indicate ringing until 0:04 or 0:05. Manual calls excluding Asterisk from the process produced suitably similar results (didn't have the fancy timer of the Polycom phone to verify the times). Therefore, my conclusion is that it is NOT an Asterisk problem (Yeah!), but something I'll have to live with (Boo! :) ). Nice to chase a problem for once and not have the answer be something that the guru's fix in 3 seconds once they get to it in their readings... -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kyle Kienapfel Sent: Thursday, January 28, 2010 6:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cell Phone dialing What happens when you dial with a handset? Is this delay caused by the asterisk or is the telco doing it? On Thu, Jan 28, 2010 at 2:57 PM, Danny Nicholas da...@debsinc.com wrote: Greetings all, This was most likely covered in one or more of the 15K emails I tried to categorize today. Im running * 1.4.26.2 with TDM400P. When I call number 205-555-1212 (a land line), Asterisk indicates ringing after about 2-3 seconds. When I call 205-555-1313 (a cell phone), it takes 4-5 seconds to indicate. Is this a known problem and/or something I have to live with? Regards, Danny Nicholas -- _ -- 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] Use of 603 Declined
On Jan 29, 2010, at 5:59 AM, Olle E. Johansson wrote: 29 jan 2010 kl. 10.25 skrev Alex Balashov: I don't know about 4xx, but 503 would be more benign for general/ miscellaneous errors than 603. 503 indicates that there's a problem with the server, so that's not a good replacement. We're sending this when there's a failed call, in most cases because of the outbound channel failure without a proper hangup cause being set in that channel driver (may very well be chan_sip :-)) No, we need something in the 4xx class. I haven't had time to go through and consider all the options in the massive set of RFCs we have to work with, but will try to do that tonight after a Friday night dinner - if no one on the list comes up with the solution before that. Isn't that a hacker way of spending Friday night - enjoying the wonderful prose of RFC3261 and companions? /O What about a 481 if the 4xx is going to be used? ---fred http://qxork.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] TDM2400 card FXS problems
Hi Noah, IMHO replace the FXS module with new one or working module. That could be proved where is the root cause. Mostly, the trouble issue is caused by FXS modules in my experience. Good Luck, Johnson On Fri, Jan 29, 2010 at 10:10 PM, Noah I. Engelberth n...@directlinkcomputers.com wrote: I don't know what it showed at that time, I didn't know to look for that. I'll try to get the customer's permission to re-create the symptoms this weekend and post back with the lsdahdi output. Thank you, Noah Engelberth Direct Link Computer Systems - Message: 12 Date: Thu, 28 Jan 2010 23:13:56 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com Message-ID: 20100128211356.gx3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] disable comfort noise
Something else that is flaky, missing or otherwise irritatingly broken in the piece of shit that is 'Asterisk'. [Cary Fitch] It is an open source project. When can we count on your contribution of a comfort noise generator that will not be a piece of s--t? Can you have that by Monday? CF -- _ -- 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] disable comfort noise
On Fri, 2010-01-29 at 07:29 -0600, Kevin P. Fleming wrote: Szasz Szabolcs wrote: How can I disable comfort noise on Asterisk? Asterisk does not have a comfort noise generator, so there is nothing to disable. You'll have to be more specific about what you are trying to accomplish. -- 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 I expect he means this: rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Something else that is flaky, missing or otherwise irritatingly broken in the piece of shit that is 'Asterisk'. -- _ -- 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] disable comfort noise
Just read RFC 3389; Guess the solution is going to be to sell Asterisk to someone, make us pay for it and make everyone run a g.711 codec? If you don't like it, don't use it! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Friday, January 29, 2010 8:57 AM To: listu...@spamomania.co.uk; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] disable comfort noise Something else that is flaky, missing or otherwise irritatingly broken in the piece of shit that is 'Asterisk'. [Cary Fitch] It is an open source project. When can we count on your contribution of a comfort noise generator that will not be a piece of s--t? Can you have that by Monday? CF -- _ -- 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] disable comfort noise
On 29 Jan 2010, at 14:53, listu...@spamomania.co.uk wrote: Something else that is flaky, missing or otherwise irritatingly broken in the piece of shit that is 'Asterisk'. It's open source. Fix it yourself, no one else is going to fix it for you with that attitude. -- _ -- 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] Use of 603 Declined
On Fri, Jan 29, 2010 at 5:59 AM, Olle E. Johansson o...@edvina.net wrote: 29 jan 2010 kl. 10.25 skrev Alex Balashov: I don't know about 4xx, but 503 would be more benign for general/ miscellaneous errors than 603. 503 indicates that there's a problem with the server, so that's not a good replacement. We're sending this when there's a failed call, in most cases because of the outbound channel failure without a proper hangup cause being set in that channel driver (may very well be chan_sip :-)) No, we need something in the 4xx class. I haven't had time to go through and consider all the options in the massive set of RFCs we have to work with, but will try to do that tonight after a Friday night dinner - if no one on the list comes up with the solution before that. Isn't that a hacker way of spending Friday night - enjoying the wonderful prose of RFC3261 and companions? /O Olle, I don't want to ruin your plans for tonight (RFC3261 is a lot of fun) but how about 403: 21.4.4 403 Forbidden The server understood the request, but is refusing to fulfill it. Authorization will not help, and the request SHOULD NOT be repeated. I like this because the most reliable way to get Asterisk to send a 603 at the moment is with something like this: sip.conf: [general] context=nocrackers extensions.conf: [nocrackers] exten = i,1,Hangup exten = s,1,Hangup exten = t,1,Hangup Wouldn't a 403 be perfect in this scenario? It looks like there are certainly other cases where it wouldn't fit quite as well (I haven't even looked at those involving REFER) but it looks perfect to me. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cell phone redialer?
I have an Asterisk 1.4.2 system installed at our office, and have a few 'on the road' sales people that want to make calls from their cell phones in transit, but they are complaining that people returning calls that they make from their cell phones are simply just using the CID that is coming from the cell phone which is causing them to get phone calls outside of business hours. What I was hoping was that either there was an existing Add-On to Asterisk out there for this, or that I could construct something. Basically I wanted to see if I could get them to call our phone number on Asterisk, enter some special extension and/or enter a passcode, and then enter the phone number that they wanted to call in which our phone system would route the call to the party but with our caller ID. Is something like this possible? I know there are companies out there charging for this sort of thing (like SpoofID, etc.) but since we may already have the technology in place for this, I was hoping I could roll my own 'home grown' version of it. Thanks in advance for any comments. Myles - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA www.techsolusa.com Phone +1-480-451-7440 -- _ -- 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] Cell phone redialer?
On 1/29/2010 10:13 AM, Myles Wakeham wrote: Basically I wanted to see if I could get them to call our phone number on Asterisk, enter some special extension and/or enter a passcode, and then enter the phone number that they wanted to call in which our phone system would route the call to the party but with our caller ID. you're looking for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA Example 2 should slip right into your extensions.conf. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell phone redialer?
Try this
- exten = 393,1,noop(forward this call)
- exten = 393,n,authenticate(7277,a)
- exten = 393,n,Read(custno,customerphone,11,skip,5,1)
- exten = 393,n,Dial(DAHDI/1,w${custno},30,mKkg)
- exten = 393,n,Playback(vm-goodbye)
- exten = 393,n,Hangup
This will require the caller to enter a password to be able to dial, then
enter the number to call. Then Asterisk will call the number he entered and
present the callerid as your DAHDI/1 ID.
The main drawback to this approach is that DAHDI/1 would have to be
available; you could fix this with your telco.
Untested - YMMV
--
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Myles Wakeham
Sent: Friday, January 29, 2010 9:13 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Cell phone redialer?
I have an Asterisk 1.4.2 system installed at our office, and have a few
'on the road' sales people that want to make calls from their cell
phones in transit, but they are complaining that people returning calls
that they make from their cell phones are simply just using the CID that
is coming from the cell phone which is causing them to get phone calls
outside of business hours.
What I was hoping was that either there was an existing Add-On to
Asterisk out there for this, or that I could construct something.
Basically I wanted to see if I could get them to call our phone number
on Asterisk, enter some special extension and/or enter a passcode, and
then enter the phone number that they wanted to call in which our phone
system would route the call to the party but with our caller ID.
Is something like this possible? I know there are companies out there
charging for this sort of thing (like SpoofID, etc.) but since we may
already have the technology in place for this, I was hoping I could roll
my own 'home grown' version of it.
Thanks in advance for any comments.
Myles
-
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
Scottsdale, Arizona USA
www.techsolusa.com
Phone +1-480-451-7440
--
_
-- 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] Problem with ringing (or absence of...) with Polycom forwarding
Hi, I`m having a problem I cannot explain. When dialing 555-555- (for example), I get a ringing sound until the person answers. When I have my Polycom forwarded to 555-555-, I do not get the ringing, but it dials fine and eventually when the person answers everything works fine. Where could be the difference? Both are using the same context to dial out. Mike -- _ -- 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] TDM2400 card FXS problems
Hello Noah, Just shifting your TDM2400 and system to other computer, it might be figured out your issue. As my opinion, the issue is involved with computer reset and TDM2400 epld programming. Would figuring out the issue completely, it is better to replace the TDM2400 card with Digium's. Garry On Fri, Jan 29, 2010 at 10:10 PM, Noah I. Engelberth n...@directlinkcomputers.com wrote: I don't know what it showed at that time, I didn't know to look for that. I'll try to get the customer's permission to re-create the symptoms this weekend and post back with the lsdahdi output. Thank you, Noah Engelberth Direct Link Computer Systems - Message: 12 Date: Thu, 28 Jan 2010 23:13:56 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com Message-ID: 20100128211356.gx3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Use of 603 Declined
Kristian Kielhofner wrote: I don't want to ruin your plans for tonight (RFC3261 is a lot of fun) but how about 403: 21.4.4 403 Forbidden The server understood the request, but is refusing to fulfill it. Authorization will not help, and the request SHOULD NOT be repeated. Well, that's the problem, and it's the reason why 603 is so commonly used. This is a situation where the current request has failed, but there is no indication that repeating the request will also fail. 403 means that the request should not be repeated without either changing it or authenticating as a different entity, which is a different scenario. It is very likely that there is no standard-defined 4xx code for 'cannot process this call right now', only the 5xx and 6xx variants. -- 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] Cell phone redialer?
Here are snippets from my OP that I tested and the DISA that Jeremy
suggested;
; authenticate and dial
exten = 393,1,noop(forward this call)
exten = 393,n,authenticate(7277,a)
exten = 393,n,Read(custno,enter-phone-number10,7,skip,5,1)
exten = 393,n,Dial(DAHDI/1/w${custno},30,mKkg)
exten = 393,n,Playback(vm-goodbye)
exten = 393,n,Hangup
; authenticate and use DISA
exten = 3472,1,Answer
exten = 3472,2,Set(TIMEOUT(digit)=3)
exten = 3472,3,Set(TIMEOUT(response)=5)
exten = 3472,4,Authenticate(6887433)
exten = 3472,5,DISA(no-password,default)
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Kister
Sent: Friday, January 29, 2010 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cell phone redialer?
On 1/29/2010 10:13 AM, Myles Wakeham wrote:
Basically I wanted to see if I could get them to call our phone number
on Asterisk, enter some special extension and/or enter a passcode, and
then enter the phone number that they wanted to call in which our phone
system would route the call to the party but with our caller ID.
you're looking for DISA.
http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA
Example 2 should slip right into your extensions.conf.
--
Jeremy Kister
http://jeremy.kister.net./
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- 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] Problem with ringing (or absence of...) withPolycom forwarding
Please post CLI output from the 2 calls with the number xxx'ed out. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, January 29, 2010 9:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Problem with ringing (or absence of...) withPolycom forwarding Hi, I`m having a problem I cannot explain. When dialing 555-555- (for example), I get a ringing sound until the person answers. When I have my Polycom forwarded to 555-555-, I do not get the ringing, but it dials fine and eventually when the person answers everything works fine. Where could be the difference? Both are using the same context to dial out. Mike -- _ -- 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] Use of 603 Declined
On Fri, Jan 29, 2010 at 10:31 AM, Kevin P. Fleming kpflem...@digium.com wrote: Well, that's the problem, and it's the reason why 603 is so commonly used. This is a situation where the current request has failed, but there is no indication that repeating the request will also fail. 403 means that the request should not be repeated without either changing it or authenticating as a different entity, which is a different scenario. This is true... Authenticating as a different entity would/could potentially match other peers (causing a 407) and probably isn't technically correct. However, if they didn't match an existing peer (to be challenged or not) using Asterisk's standard peer matching, how did they end up in the nocrackers context anyway? Either way I wasn't considering 5xx responses because of Olle's request. It is very likely that there is no standard-defined 4xx code for 'cannot process this call right now', only the 5xx and 6xx variants. Asterisk has certainly bent standards (which real world implementation hasn't) before. It seems to me that the best reply is the one that's most likely to encourage correct behavior from the far end... 603 almost certainly doesn't do that. In this scenario any forking proxy faced with a 603 coming from Asterisk has to break RFC compliance just to successfully complete the request on another host. Nasty. Are we back to the next-most-generic SIP error, 503 (as originally suggested by Alex)? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] 1 Asterisk server, multiple registrations to ITSP
On Fri, 2010-01-29 at 15:09 +0100, jonas kellens wrote: Hello list ! Having troubles with multiple registrations to one and the same ITSP. This sip.conf : register = user1:pass...@sip.itsp register = user2:pass...@sip.itsp ; outgoing conversations [user1-out] type=peer host=sip.ITSP Try setting type=friend instead of peer for these and see what happens. -- Robert Lister - email/sip: r...@lentil.org - http://www.lentil.org tel: 020 7043 7996 -- _ -- 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] Dial cellphone from one PBX1 to PBX2? is it possible?
Hi William,
I appreciate your answer, though can you make things more clear for me:
1- i am not using extensions when registering PBX boxes in IAX files.
2- is inbounx context in the call sender PBX (pbx1) and outbound context is
in the call receiver (or dialer) PBX (pbx2)?
3- i am using two identical dialplan's is this gonna confuse
the communication process (contextes's name are duplicated over the two
servers)
thank you very much for making it clear for me!
2010/1/28 William Stillwell (Lists) william.stillwell-li...@ablebody.net
Your inbound context needs to have access to your outbound context.
[iax-inbound]
Include = outbound-conext
[outbound-context]
Exten = _1NXXNXX,1,Dial(DAHDI\g1\${EXTEN})
Something like that.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Thursday, January 28, 2010 3:29 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it
possible?
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that
way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to
a cellphone)
my questions now is : am i gonna be able to dial from an IPphone registered
within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
anybody know
IPphone-PBX1-IAXPBX2PRI
line---cellphone???
thank you for you help guys!!
--
Abdullah
--
_
-- 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
--
Abdullah
--
_
-- 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] smsq command
Is it possible to use the smsq command in asterisk to send SMS messages to a aggregator. So If I have an IP address, password and port for my connection can I use smsq to send SMS messages? I dont see how to set that up? I am looking: http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms Thanks, Jerry -- _ -- 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] Questions about asterisk and spa2102
Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a normal phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very appreciated. I'm and absolute newbie on asterisk, btw. Thanx! Kosa - Un mundo mejor es posible - -- _ -- 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] 1 Asterisk server, multiple registrations to ITSP
When setting type=friend for the incoming calls : ; outgoing conversations [user1-out] type=peer host=sip.ITSP username=user1 secret=secret1 fromuser=user1 ; incoming conversations [user1] type=friend host=sip.ITSP context=user1incoming ; outgoing conversations [user2-out] type=peer host=sip.ITSP username=user2 secret=secret2 fromuser=user2 ; incoming conversations [user2] type=friend host=sip.ITSP context=user2incoming I get the following message : [Jan 29 18:49:07] NOTICE[6314]: chan_sip.c:14703 handle_request_invite: Call from 'user2' to extension 329990102 rejected because extension not found. The call in fact needs to come from user1 in stead of from user2. Of course the extension is not found as it is defined in the context for incoming calls of user 1. In sip.conf [user2] is defined after [user1] and I have the impression that the last definition of a user is always taken. [user2] is always taken for incoming calls. How can I identify calls that are destined for user1 as defined in : register = user1:pass...@sip.itsp [user1] type=friend host=sip.ITSP context=user1incoming It must be possible to have several accounts with the same ITSP on the same Asterisk-server ?! Jonas. On Fri, 2010-01-29 at 16:51 +, Robert Lister wrote: On Fri, 2010-01-29 at 15:09 +0100, jonas kellens wrote: Hello list ! Having troubles with multiple registrations to one and the same ITSP. This sip.conf : register = user1:pass...@sip.itsp register = user2:pass...@sip.itsp ; outgoing conversations [user1-out] type=peer host=sip.ITSP Try setting type=friend instead of peer for these and see what happens. -- _ -- 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] Dial cellphone from one PBX1 to PBX2? is it possible?
You are using contexts..
Look @ destination pbx, you should see something like this:
Rejected connect attempt from ip of source pbx, request 'ext@incoming
conext' does not exist
If you didn't put a context under the peer, it uses the default one in the
iax.conf file which is normally [default]
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Friday, January 29, 2010 11:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it
possible?
Hi William,
I appreciate your answer, though can you make things more clear for me:
1- i am not using extensions when registering PBX boxes in IAX files.
2- is inbounx context in the call sender PBX (pbx1) and outbound context is
in the call receiver (or dialer) PBX (pbx2)?
3- i am using two identical dialplan's is this gonna confuse the
communication process (contextes's name are duplicated over the two servers)
thank you very much for making it clear for me!
2010/1/28 William Stillwell (Lists) william.stillwell-li...@ablebody.net
Your inbound context needs to have access to your outbound context.
[iax-inbound]
Include = outbound-conext
[outbound-context]
Exten = _1NXXNXX,1,Dial(DAHDI\g1\${EXTEN})
Something like that.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, January 28, 2010 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it
possible?
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a
cellphone)
my questions now is : am i gonna be able to dial from an IPphone registered
within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
anybody know
IPphone-PBX1-IAXPBX2PRI
line---cellphone???
thank you for you help guys!!
--
Abdullah
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
http://www.api-digital.com/ --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Abdullah
--
_
-- 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] microphone on Polycom 550/650
I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -- _ -- 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] disable comfort noise
To get back to the original poster's possible situation, i've seen this with my first IP phone, which was a cisco 7912 (SIP image). With that phone, asterisk sometimes gave me this same error. I'm quite sure i've asked the very same question here back then (probably i was a bit more specific :). Since it is related to only this type of phone, i've gone to different ip phone products. regards adam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help configuring Audiocodes MP-104 FXO
Just if it is helps someone, based on information at the blog: http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.htmlI've summarized the following steps: *Step 1:* Configure audiocodes to have registration account with asterisk, this can be done easily with Protocol Management - Protocol Definition - ProxyRegistration, fill on Proxy IP Address, Enable Registration : Yes, Username, Password, and Authentication Mode : Per Endpoint. *Step 2:* Configuring Protocol Management - Endpoint Phone Number, this is important part for make each FXO port on audiocodes registered with asterisk, in here, under Channel, you can fill with either 1, 1-2, 1-8, 3-4, or whatever you want to have, this means that port 1, or port 1-2, etc will registered on astersik with userid/username filled on Phone Number, yes, that is correct, Phone Number on this configuration page is AlphaNumeric, the password is using global Password on First step. next, on same page configure Hunt Group ID, this is another important configuration which make audiocodes forward incoming call from asterisk to any available FXO. Hunt Group ID is number from 0 to any, I put 1. *Step 3:* to make audiocodes forward call from FXO to asterisk, configure Endpoint Settings - Automatic Dialing, I have 777 number on asterisk to handle all incoming call, so I put Destination Phone Number as 777 so every incoming call on FXO will be forwarded to 777 on my Astersik. *Step 4:* this is the last configuration that everyone need, forward call from asterisk to any available FXO. in Routing Tables - IP to Hunt Group Routing Table configure under Dest. Phone Prefix with * (or any prefix that you might have), Source Phone Prefix with *, Source IP Address with *, Hunt Group ID with any number you configure on Step 2, in my case, 1. *I add here addiiotnal steps needed for me to get ready**: Step 5:* Add port by port authentication at Protocol Management - Endoint Settings - Authentication *Step 6:* Choosing Channel Selection Mode: Protocol Management - Hunt Group Settings, choose the hunt group number and the way you prefer. *Step 7:* Choosing Dialing Mode: Protocol Management - FXO Settings, I select One Stage. Hope it helps. Elder Daniel On Wed, Dec 2, 2009 at 2:08 PM, Daniel - Asterisk earohua...@gmail.comwrote: I've set at Protocol Management FXO Settings Dialing Mode == One Stage and everything is fine now Thank you very much John, EDA On Wed, Dec 2, 2009 at 1:43 PM, John Balogh j...@psu.edu wrote: I want to do single-stage dialing. I've just realized I have the two-stage running now (I get dial tone and then, when i introduce the number, the call get through). Right. According to the SIP User's Manual LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf page 67/294 Enable Digit Delivery to Tel [EnableDigitDelivery] Disable [0] = Disabled (default). Enable [1] = Enable Digit Delivery feature for MediaPack/FXO FXS. The digit delivery feature enables sending of DTMF digits to the gateway’s port after the line is offhooked (FXS) or seized (FXO). For IP-Tel calls, after the line is offhooked / seized, the MediaPack plays the DTMF digits (of the called number) towards the phone line. [...] To use this feature with FXO gateways, configure the gateway to work in one stage dialing mode. You probably need to set the above. The FXO parameter (from page 107/294): Dialing Mode [IsTwoStageDial] One Stage [0] = One-stage dialing. Two Stage [1] = Two-stage dialing (default). Used for IP-FXO gateways calls. If two-stage dialing is enabled, then the FXO gateway seizes one of the PSTN/PBX lines without performing any dial, the remote user is connected over IP to PSTN/PBX, and all further signaling (dialing and Call Progress Tones) is performed directly with the PBX without the gateway’s intervention. If one-stage dialing is enabled, then the FXO gateway seizes one of the available lines (according to Channel Select Mode parameter), and dials the destination phone number received in INVITE message. Use the ‘Waiting For Dial Tone’ parameter to specify whether the dialing should come after detection of dial tone, or immediately after seizing of the line. So you probably need to clear that parameter (it is not configured in your .INI file now, so you need to add it, or change the web interface drop-down control). Let us know if this helps. JDB *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Daniel - Asterisk *Sent:* Wednesday, December 02, 2009 12:33 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Help configuring Audiocodes MP-104 FXO Hi list, I'm trying to get ready the MP-104 FXO to use qith my box, but when I send calls I hear only dial tone and after a few seconds I get busy signal. I very appreciate your
Re: [asterisk-users] Help configuring Audiocodes MP-104 FXO
Damn, where were you 6 months ago? ;) Daniel - Asterisk wrote: Just if it is helps someone, based on information at the blog: http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.html I've summarized the following steps: *Step 1:* Configure audiocodes to have registration account with asterisk, this can be done easily with Protocol Management - Protocol Definition - ProxyRegistration, fill on Proxy IP Address, Enable Registration : Yes, Username, Password, and Authentication Mode : Per Endpoint. *Step 2:* Configuring Protocol Management - Endpoint Phone Number, this is important part for make each FXO port on audiocodes registered with asterisk, in here, under Channel, you can fill with either 1, 1-2, 1-8, 3-4, or whatever you want to have, this means that port 1, or port 1-2, etc will registered on astersik with userid/username filled on Phone Number, yes, that is correct, Phone Number on this configuration page is AlphaNumeric, the password is using global Password on First step. next, on same page configure Hunt Group ID, this is another important configuration which make audiocodes forward incoming call from asterisk to any available FXO. Hunt Group ID is number from 0 to any, I put 1. *Step 3:* to make audiocodes forward call from FXO to asterisk, configure Endpoint Settings - Automatic Dialing, I have 777 number on asterisk to handle all incoming call, so I put Destination Phone Number as 777 so every incoming call on FXO will be forwarded to 777 on my Astersik. *Step 4:* this is the last configuration that everyone need, forward call from asterisk to any available FXO. in Routing Tables - IP to Hunt Group Routing Table configure under Dest. Phone Prefix with * (or any prefix that you might have), Source Phone Prefix with *, Source IP Address with *, Hunt Group ID with any number you configure on Step 2, in my case, 1. /I add here addiiotnal steps needed for me to get ready/*: Step 5:* Add port by port authentication at Protocol Management - Endoint Settings - Authentication *Step 6:* Choosing Channel Selection Mode: Protocol Management - Hunt Group Settings, choose the hunt group number and the way you prefer. *Step 7:* Choosing Dialing Mode: Protocol Management - FXO Settings, I select One Stage. Hope it helps. Elder Daniel On Wed, Dec 2, 2009 at 2:08 PM, Daniel - Asterisk earohua...@gmail.com mailto:earohua...@gmail.com wrote: I've set at Protocol Management FXO Settings Dialing Mode == One Stage and everything is fine now Thank you very much John, EDA On Wed, Dec 2, 2009 at 1:43 PM, John Balogh j...@psu.edu mailto:j...@psu.edu wrote: I want to do single-stage dialing. I've just realized I have the two-stage running now (I get dial tone and then, when i introduce the number, the call get through). Right. According to the SIP User's Manual LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf page 67/294 Enable Digit Delivery to Tel [EnableDigitDelivery] Disable [0] = Disabled (default). Enable [1] = Enable Digit Delivery feature for MediaPack/FXO FXS. The digit delivery feature enables sending of DTMF digits to the gateway’s port after the line is offhooked (FXS) or seized (FXO). For IP-Tel calls, after the line is offhooked / seized, the MediaPack plays the DTMF digits (of the called number) towards the phone line. [...] To use this feature with FXO gateways, configure the gateway to work in one stage dialing mode. You probably need to set the above. The FXO parameter (from page 107/294): Dialing Mode [IsTwoStageDial] One Stage [0] = One-stage dialing. Two Stage [1] = Two-stage dialing (default). Used for IP-FXO gateways calls. If two-stage dialing is enabled, then the FXO gateway seizes one of the PSTN/PBX lines without performing any dial, the remote user is connected over IP to PSTN/PBX, and all further signaling (dialing and Call Progress Tones) is performed directly with the PBX without the gateway’s intervention. If one-stage dialing is enabled, then the FXO gateway seizes one of the available lines (according to Channel Select Mode parameter), and dials the destination phone number received in INVITE message. Use the ‘Waiting For Dial Tone’ parameter to specify whether the dialing should come after detection of dial tone, or immediately after seizing of the line. So you probably need to clear that parameter (it is not
Re: [asterisk-users] microphone on Polycom 550/650
You don't state this, but the assumption would be that your external calls are DAHDI based, so you might need to tweak txgain in dahdi.conf. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee Sent: Friday, January 29, 2010 12:08 PM To: Asterisk Users Subject: [asterisk-users] microphone on Polycom 550/650 I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -- _ -- 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] 911, location
Den 28-01-2010 20:15, Danny Nicholas skrev:
Here's one solution:
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4,Background(emergencyin${IMAT})
Where you would record /var/lib/asterisk/sound/emergencyin100 for extension
100, etc.
I see two problems:
1: Doesn't asterisk see a pots-call as answered as soon as it has
pressed the last digit and therefore will speak into the ring signal?
2: Often callers are answered with an automated message This is 911,
please hold, just to give pranksters/misdiallers a chance to hang up
before disturbing the operator. Unless 911 records the incoming call
right from the start, they will never hear the im-at message. And even
if they do, they have to know the message is there to seek on the recording.
An option of the operator receiving a loop of This is a call from the
Mickey Mouse building room 123, please press * to receive the call
would require the operator to be able to press *, not sure I'd depend
my life on that...
Leif
--
_
-- 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] TDM2400 card FXS problems
HI: I had this problem before with TDM2400P but with fxo modules and VPMADT032 (echo canceller),there was no audio at all.but then i unpulgged the ehco canceller module (VPMADT032) from the TDM2400P board and started the server and then i didnt face this issue any more. In your case first check the output of #dmesg if it shows repeated message of Unable to set SW Companding on channel ,then your problem is with the echocanceler module (VPMADT032) and do same what i did,dont talk to digium cause they themselves dont know about this error and dont have solution for it ,i tried before without any success. best regards; --- On Fri, 1/29/10, Noah I. Engelberth n...@directlinkcomputers.com wrote: From: Noah I. Engelberth n...@directlinkcomputers.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Date: Friday, January 29, 2010, 2:10 PM I don't know what it showed at that time, I didn't know to look for that. I'll try to get the customer's permission to re-create the symptoms this weekend and post back with the lsdahdi output. Thank you, Noah Engelberth Direct Link Computer Systems - Message: 12 Date: Thu, 28 Jan 2010 23:13:56 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] TDM2400 card FXS problems To: asterisk-users@lists.digium.com Message-ID: 20100128211356.gx3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- 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 -- _ -- 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] 911, location
This might help
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4(keepup),Background(emergencyin${IMAT})
- exten = _911,5,wait(10)
- exten = _911,6,Goto(keepup)
This would repeat the message every 10 seconds...
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Neland
Sent: Friday, January 29, 2010 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911, location
Den 28-01-2010 20:15, Danny Nicholas skrev:
Here's one solution:
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4,Background(emergencyin${IMAT})
Where you would record /var/lib/asterisk/sound/emergencyin100 for
extension
100, etc.
I see two problems:
1: Doesn't asterisk see a pots-call as answered as soon as it has
pressed the last digit and therefore will speak into the ring signal?
2: Often callers are answered with an automated message This is 911,
please hold, just to give pranksters/misdiallers a chance to hang up
before disturbing the operator. Unless 911 records the incoming call
right from the start, they will never hear the im-at message. And even
if they do, they have to know the message is there to seek on the recording.
An option of the operator receiving a loop of This is a call from the
Mickey Mouse building room 123, please press * to receive the call
would require the operator to be able to press *, not sure I'd depend
my life on that...
Leif
--
_
-- 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] New feature: Asterisk Manager Interface commands for DeviceState
Hi, I've uploaded a new patch at https://issues.asterisk.org/view.php?id=16732which adds two new AMI commands, called DeviceStateSet and DeviceStateGet. These commands let you update Custom device states, and read all devicestates from AMI. It would be very nice if someone could help me test this feature, and report back to the issue tracker. To test, log into AMI as usual, and then issue something like the following (please also test to Get device state from real devices too): -- Action: DeviceStateSet DeviceName: Custom:lamp1 DeviceState: INUSE -- Which should yield: -- Response: Success Message: Success -- Then you can check the state by doing: -- Action: DeviceStateGet DeviceName: Custom:lamp1 -- And you should receive the following: -- Response: Success Message: Result will follow Event: DeviceStateGetResponse DeviceName: Custom:lamp1 DeviceState: INUSE -- Regards, Håkon Nessjøen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help configuring Audiocodes MP-104 FXO
It was a pending draft I forgot to send.. sorry. On Fri, Jan 29, 2010 at 1:23 PM, Matt Collins mcoll...@ccdservice.netwrote: Damn, where were you 6 months ago? ;) Daniel - Asterisk wrote: Just if it is helps someone, based on information at the blog: http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.html I've summarized the following steps: *Step 1:* Configure audiocodes to have registration account with asterisk, this can be done easily with Protocol Management - Protocol Definition - ProxyRegistration, fill on Proxy IP Address, Enable Registration : Yes, Username, Password, and Authentication Mode : Per Endpoint. *Step 2:* Configuring Protocol Management - Endpoint Phone Number, this is important part for make each FXO port on audiocodes registered with asterisk, in here, under Channel, you can fill with either 1, 1-2, 1-8, 3-4, or whatever you want to have, this means that port 1, or port 1-2, etc will registered on astersik with userid/username filled on Phone Number, yes, that is correct, Phone Number on this configuration page is AlphaNumeric, the password is using global Password on First step. next, on same page configure Hunt Group ID, this is another important configuration which make audiocodes forward incoming call from asterisk to any available FXO. Hunt Group ID is number from 0 to any, I put 1. *Step 3:* to make audiocodes forward call from FXO to asterisk, configure Endpoint Settings - Automatic Dialing, I have 777 number on asterisk to handle all incoming call, so I put Destination Phone Number as 777 so every incoming call on FXO will be forwarded to 777 on my Astersik. *Step 4:* this is the last configuration that everyone need, forward call from asterisk to any available FXO. in Routing Tables - IP to Hunt Group Routing Table configure under Dest. Phone Prefix with * (or any prefix that you might have), Source Phone Prefix with *, Source IP Address with *, Hunt Group ID with any number you configure on Step 2, in my case, 1. /I add here addiiotnal steps needed for me to get ready/*: Step 5:* Add port by port authentication at Protocol Management - Endoint Settings - Authentication *Step 6:* Choosing Channel Selection Mode: Protocol Management - Hunt Group Settings, choose the hunt group number and the way you prefer. *Step 7:* Choosing Dialing Mode: Protocol Management - FXO Settings, I select One Stage. Hope it helps. Elder Daniel On Wed, Dec 2, 2009 at 2:08 PM, Daniel - Asterisk earohua...@gmail.com mailto:earohua...@gmail.com wrote: I've set at Protocol Management FXO Settings Dialing Mode == One Stage and everything is fine now Thank you very much John, EDA On Wed, Dec 2, 2009 at 1:43 PM, John Balogh j...@psu.edu mailto:j...@psu.edu wrote: I want to do single-stage dialing. I've just realized I have the two-stage running now (I get dial tone and then, when i introduce the number, the call get through). Right. According to the SIP User's Manual LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf page 67/294 Enable Digit Delivery to Tel [EnableDigitDelivery] Disable [0] = Disabled (default). Enable [1] = Enable Digit Delivery feature for MediaPack/FXO FXS. The digit delivery feature enables sending of DTMF digits to the gateway’s port after the line is offhooked (FXS) or seized (FXO). For IP-Tel calls, after the line is offhooked / seized, the MediaPack plays the DTMF digits (of the called number) towards the phone line. [...] To use this feature with FXO gateways, configure the gateway to work in one stage dialing mode. You probably need to set the above. The FXO parameter (from page 107/294): Dialing Mode [IsTwoStageDial] One Stage [0] = One-stage dialing. Two Stage [1] = Two-stage dialing (default). Used for IP-FXO gateways calls. If two-stage dialing is enabled, then the FXO gateway seizes one of the PSTN/PBX lines without performing any dial, the remote user is connected over IP to PSTN/PBX, and all further signaling (dialing and Call Progress Tones) is performed directly with the PBX without the gateway’s intervention. If one-stage dialing is enabled, then the FXO gateway seizes one of the available lines (according to Channel Select Mode parameter), and dials the destination phone number received in INVITE message. Use the ‘Waiting For Dial Tone’ parameter to specify whether the dialing should come
[asterisk-users] Digium fax - sending fax call file vs manager originate
Hello,
I have Asterisk 1.6.1.12 with
FAX For Asterisk Components:
Applications: 1.6.1.5_1.1.6
Digium FAX Driver: 1.6.1.5_1.1.6 (optimized for core2_32)
If I use call file with spool
Channel: SIP/IP/DEst No
MaxRetries: 0
RetryTime: 10
WaitTime: 50
Application:SendFAX
Data:/var/spool/asterisk/test.tif
Fax is send but if I use manager
Action: Originate
Channel: SIP/IP/dest NO
Context: fax-tx
Exten: send
Priority: 1
Callerid: Asterisk Automatic Wardial
I get
ERROR[16796]: res_fax.c:696 generic_fax_exec: channel 'SIP/IP-0015'
is in an unsupported T.38 negotiation state, cannot continue.
Here is context
-
[fax-tx]
exten = send,1,NoOp( SENDING FAX )
exten = send,n,Wait(6)
;exten = send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten = send,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten = send,n,Set(FAXFILE=faxout.tif)
; Set FAXOPTs
exten = send,n,NoOp( SETTING FAXOPT )
;exten = send,n,Set(FAXOPT(ecm)=yes)
;exten = send,n,Set(FAXOPT(headerinfo)=Fax from
${GLOBAL(LASTFAXCALLERNAME)} at ${GLOBAL(LASTFAXCALLERNUM)} was
received.)
exten = send,n,Set(FAXOPT(localstationid)=1234567890)
;exten = send,n,Set(FAXOPT(maxrate)=14400)
;exten = send,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten = send,n,NoOp( SENDING FAX : ${FAXFILE} )
exten = send,n,SendFAX(/var/spool/asterisk/fax/${FAXFILE},d)
; Hangup! Print FAXOPTs
exten = h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = h,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = h,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = h,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = h,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = h,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = h,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = h,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = h,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)})
exten = h,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = h,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = h,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = h,n,NoOp(FAXOPT(error) : ${FAXOPT(error)})
---
Any suggestions?
Thanks,
Hristo Benev
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broker lines on a T1 : Signaling convention?
I've been running Asterisk with a standard PRI for regular telecoms. This is also connected to our Nortel PBX for 'ordinary users'. The system has been working nicely (including Cisco 7970 phones that are connecting via SIP). But now I'm going 'on net' with broker lines (for a trading room environment). The telecoms people at the other end of the connection tell me that each line is just a standard ARD circuit - and terms such as 'loopstart', 'groundstart' or 'EM' don't have any resonance with them. So : Has anyone got any hints from installing trader turrets (for instance) about what dahdi config I need for this dedicated type of T1? Thanks Martin :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
Den 29-01-2010 19:38, Danny Nicholas skrev:
This might help
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4(keepup),Background(emergencyin${IMAT})
- exten = _911,5,wait(10)
- exten = _911,6,Goto(keepup)
This would repeat the message every 10 seconds...
--
This would prevent the caller talking to the 911-operator, wouldn't it?
Leif
--
_
-- 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] 911, location
The idea behind the OP was that the caller was a man down who couldn't
speak to 911, just dial the number. You could always change wait to
waitexten and make an exten to break the loop if you were able to talk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Neland
Sent: Friday, January 29, 2010 2:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911, location
Den 29-01-2010 19:38, Danny Nicholas skrev:
This might help
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4(keepup),Background(emergencyin${IMAT})
- exten = _911,5,wait(10)
- exten = _911,6,Goto(keepup)
This would repeat the message every 10 seconds...
--
This would prevent the caller talking to the 911-operator, wouldn't it?
Leif
--
_
-- 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] R: rtp.c:883 ast_rtcp_read: RTCP Read too short
Hi: i did set the rtp ports in rtp.conf to rtpstart=5000 , rtpend=31000 ,and i used canreinvite=no and the problem still exists ,however i did the rtp debug and here is the output : = Spawn extension (direct, 9613070741, 2) exited non-zero on 'SIP/03070741-083b9da0' -- Executing [9613070...@direct:1] Set(SIP/03070741-083b9da0, CALLERID(number)=96170707070) in new stack -- Executing [9613070...@direct:2] Dial(SIP/03070741-083b9da0, SIP/usa/9613070741) in new stack -- Called usa/9613070741 Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042234, ts 001920, len 24) [Jan 29 22:54:35] WARNING[21595]: rtp.c:883 ast_rtcp_read: RTCP Read too short Got RTP packet from 192.168.1.64:16392 (type 04, seq 010658, ts 423540309, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042235, ts 002160, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042236, ts 002400, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010659, ts 423540549, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042237, ts 002640, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010660, ts 423540789, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042238, ts 002880, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010661, ts 423541029, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010662, ts 423541269, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042239, ts 003120, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042240, ts 003360, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010663, ts 423541509, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010664, ts 423541749, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042241, ts 003600, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010665, ts 423541989, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042242, ts 003840, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010666, ts 423542229, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042243, ts 004080, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010667, ts 423542469, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042244, ts 004320, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010668, ts 423542709, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042245, ts 004560, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042246, ts 004800, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010669, ts 423542949, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010670, ts 423543189, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042247, ts 005040, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042248, ts 005280, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010671, ts 423543429, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042249, ts 005520, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042250, ts 005760, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010672, ts 423543669, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042251, ts 006000, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010673, ts 423543909, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010674, ts 423544149, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042252, ts 006240, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010675, ts 423544389, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042253, ts 006480, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010676, ts 423544629, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042254, ts 006720, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042255, ts 006960, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010677, ts 423544869, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042256, ts 007200, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010678, ts 423545109, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042257, ts 007440, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010679, ts 423545349, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010680, ts 423545589, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042258, ts 007680, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010681,
Re: [asterisk-users] 911, location
Leif Neland wrote: 2: Often callers are answered with an automated message This is 911, please hold, just to give pranksters/misdiallers a chance to hang up before disturbing the operator. Unless 911 records the incoming call right from the start, they will never hear the im-at message. And even if they do, they have to know the message is there to seek on the recording. In the US at least, calls to PSAPs are recorded from the instant the last digit is dialed, before the call is even routed and ringing (on wireline networks where this is possible, anyway). -- 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
[asterisk-users] Help for MOH - sounding scratchy/static on hold
Hi All, I tried using some music on hold (music) files, when I test it with normal SIP phone its clear and good, but when I call from my cell phone or POTS line it sounds a bit scratchy/static and not clear at all, is there any software that i need to install in the asterisk system to make this music on hold clear when using music files? (Where as the commercial that we record from the phone and use it as message on hold then its clear when the call is on hold, since its recording is compatible with asterisk: 8000Hz, 16 bits PCM encoded). My versions of asterisk: 1.4.18.1. I appreciate your advices. Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold
Mpg123 works well for us. You have to get your files into mp3 format, but LAME does this simply. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of das sandesh Sent: Friday, January 29, 2010 4:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help for MOH - sounding scratchy/static on hold Hi All, I tried using some music on hold (music) files, when I test it with normal SIP phone its clear and good, but when I call from my cell phone or POTS line it sounds a bit scratchy/static and not clear at all, is there any software that i need to install in the asterisk system to make this music on hold clear when using music files? (Where as the commercial that we record from the phone and use it as message on hold then its clear when the call is on hold, since its recording is compatible with asterisk: 8000Hz, 16 bits PCM encoded). My versions of asterisk: 1.4.18.1. I appreciate your advices. Thank you very much Regards Sandesh -- _ -- 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] Unable to create channel of type 'DAHDI'(cause 0- Unknown)
listu...@spamomania.co.uk wrote: On Thu, 2010-01-28 at 23:11 -0600, Karl Fife wrote: Appears completely resolved! No more home-spun patches! Thanks! -K It's *not* fixed here: DAHDI Version: 2.2.1 Echo Canceller: MG2 But as is depressingly the 'norm' for Asterisk it comes back to bitching about hardware 'buy an expensive Digium echo machine instead of a cheap one' rather than the fact that the core of Asterisk is rotten, buggy and the fix usually comes in the form of a developer arguing that it's somebody else's issue. Really - if Asterisk is 'The future of telephony' I can only assume that statement comes from the late 1800's. If you like echo, flaky connections, intermittent service and partially working DTMF coupled with a hefty hardware price tag then hey ho - Asterisk is your man Nice try, be great when it's finished. Sigh. OK you don't like asterisk - sorry. Obviously some other software works better for you. I'm glad. For at least some of us, asterisk works extremely well in demanding environments. But not perfectly. So the collegial help from the mailing list and bug spotting is quite important. Sorry you don't want to participate. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold
Sandesh- I tried using some music on hold (music) files, when I test it with normal SIP phone its clear and good, but when I call from my cell phone or POTS line it sounds a bit scratchy/static and not clear at all, is there any software that i need to install in the asterisk system to make this music on hold clear when using music files? (Where as the commercial that we record from the phone and use it as message on hold then its clear when the call is on hold, since its recording is compatible with asterisk: 8000Hz, 16 bits PCM encoded). My versions of asterisk: 1.4.18.1. Is there any difference in your POTS line vs. cell quality? As you may know, cell phones typically use some type of voice codec (GSM-AMR, EVRC, etc) that can reduce quality of non-speech audio (e.g. music). But the POTS line should sound fine, unless you sent it first through a carrier IP route using compression (such as G729). -Jeff -- _ -- 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] callerid not working over sip
Calling from my home using Asterisk 1.6.2.1 to an office extension (Asterisk 1.6.1.13) the callerid is not honored: Home: -- Starting simple switch on 'DAHDI/1-1' -- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@internal:2] NoOp(DAHDI/1-1, Context: office-extensions) in new stack -- Executing [...@internal:3] Set(DAHDI/1-1, CALLERID=Test 447) in new stack -- Executing [...@internal:4] Dial(DAHDI/1-1, SIP/office-home-sip/170) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Called office-home-sip/170 On the office asterisk: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Executing [...@default:1] Macro(SIP/xxx.yyy.zzz.aaa-0176, stdexten,170,SIP/170) in new stack -- Executing [...@macro-stdexten:1] NoOp(SIP/xxx.yyy.zzz.aaa-0176, CallerID is: asterisk asterisk) in new stack -- Executing [...@macro-stdexten:2] Dial(SIP/xxx.yyy.zzz.aaa-0176, SIP/170,18,rtT) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Called 170 Why isn't the office asterisk picking up the callerid from the home asterisk? sean -- _ -- 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] Caller ID not working properly on some phones...
I have a strange problem with CallerID that only affects some phones. The problem is that whenever I receive a call the Callerid Name is correct but the Callerid number is always my own extension. It does not matter if the call is internal or external. So far only Aastra phones and Linksys PAP2T adapters seem to have this problem. Other phones like Snom and Cisco SPA525 display the correct number. I am using Asterisk 1.6.2.1 on two different servers that have the same problem. I guess there is a setting on Asterisk that the phones do not like. One of the servers was upgraded from 1.4.28 last week and we never had that problem. If I do a NoOP on the Dialplan I can see that the correct CallerID info is set but the phone will always say the number is my own extension no matter what. This is a problem because I cannot call back from the call history on the phone. CDR is correct. Any ideas what may be happening? Why would this only affect some phones and not others? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 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
Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold
On Fri, 29 Jan 2010, Danny Nicholas wrote: Mpg123 works well for us. You have to get your files into mp3 format, but LAME does this simply. Why would you want to compress files when you will have to decompress them again every single time the are used? I'd rather use the CPU cycles to process more calls. Are you in a severely storage challenged environment? You should store all of your audio encoded to match the codec used by the channel. -- 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
[asterisk-users] Asterisk status 488 Not acceptable here on receiving fax
Hello, I have been trying to setup asterisk 1.6.1.1 to receive fax. Whenever a SIP peer (zoiper soft phones) tries to send a fax message asterisk responds by sending a 488 Not acceptable here and the sending fails. I tried changing a few sip settings like canreinvite and codec preferences, but it did not help. The same sip peer is able to make normal calls. The same settings works on on asterisk 1.6.2.0 and I am able to receive fax successfully in asterisk. I would like to get this working in 1.6.1.1 as It is not possible for me to upgrade asterisk on my production servers. Can someone please help. Thanks, Deepesh -- _ -- 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] Unable to create channel of type 'DAHDI'(cause 0- Unknown)
On 30/01/10 11:48 AM, sean darcy wrote: Sigh. OK you don't like asterisk - sorry. Obviously some other software works better for you. I'm glad. Don't worry, he/she's trolling, second post like that for the day :) Obviously has an issue with something, but rather than try and get it sorted he/she'd rather just bitch. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Questions about asterisk and spa2102
Kosa wrote: Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a normal phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. In simple terms: FXS ports provide battery to analog phones, provide ringing to analog phones when so instructed, and provide dialtone to analog phones, then forward the dialed number as data to a server. FXO ports expect to see battery from analog exchange lines, supply a loop closure to request service from an exchange, in some cases will pulse dial a string of digits, in all cases send a string of DTMF digits, and detect a ringing voltage from an exchange, forwarding received information as data to a server. Some devices have both types of connections. If you want an external device to do both, it will need both types of ports, one cannot be both. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very appreciated. Why waste your ( valuable? ) time?? New unlocked similar devices are available from multiple sources. Many of these are born locked to a specific service, and cannot be changed. I'm and absolute newbie on asterisk, btw. Thanx! Kosa - Un mundo mejor es posible - -- Dog is my co-pilot -- _ -- 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] Questions about asterisk and spa2102
On Fri, 29 Jan 2010, Kosa wrote: 1.- is it possible to use an spa2102 to make and revice calls from a normal phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. The 2102 is an FXS (station) device. It connects to things like a telephone or a fax machine. The 3102 is an FXS and FXO (office) device. You can plug in a telephone and the wire coming out of the wall. I have the predecessor, the 3000. It has the neat feature that if it loses power it will bridge the FXS and FXO ports so the telephone can still be used. I don't know if current models still have this feature. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very appreciated. I did this many years ago with some PAP2s that were locked to Vonage. Definitely not worth the effort it took to configure my name servers to pretend they were Vonage's so they could resolve Vonage's names to my local IP addresses and setting up the Ethernet interface aliases to Vonage's IP addresses. -- 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
[asterisk-users] Astribank problem
H all... I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final. My problem is, every time i unplug the astribank power supply, and reconnect it, astribank cannot work again (lsusb result is 11x0)... but, after reinstall the asterisk and dahdi, astribank will detected (lsusb result is 11x2)... any suggestion? Regard, frank. _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-id:SI_SB_3:092010-- _ -- 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] disable comfort noise
On Fri, Jan 29, 2010 at 1:14 PM, ad...@3a.hu wrote: To get back to the original poster's possible situation, i've seen this with my first IP phone, which was a cisco 7912 (SIP image). With that phone, asterisk sometimes gave me this same error. I'm quite sure i've asked the very same question here back then (probably i was a bit more specific :). Since it is related to only this type of phone, i've gone to different ip phone products. regards adam -- _ -- 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 Please correct me if I'm wrong As the error says, Please turn off on client if possible. Comfort noise (aka silent suppression, or Voice Activity Detection (VAD)) is not supported by Asterisk. It needs to be turned off on the user (client) end. This may be a phone or another switch/PBX. See http://www.voip-info.org/wiki/view/RTP+Silence+Suppression for more details -- _ -- 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
