[asterisk-users] Queues and RingInUse
Hi, I have some problems with a Queue and state interface. In the queue we have 7 deskphones and 7 cordless. Every user have one deskphone and one cordless phone. I have set up every user with hints, eg: exten = 802,hint,SIP/9144-802SIP/9144-902 exten = 902,hint,SIP/9144-902SIP/9144-802 Queue_members: Queue_name Interface state_interface 9144-vmi SIP/9144-802 hint:802@hints 9144-vmi SIP/9144-902 hint:902@hints 802 = is a cordless phone 902 = is a deskphone Both are in use by same user. To the problem, when we have RingInUse = 0, we only get calls on one of the devices (because the other one gets in RINGING state). If the user is not on phone we want both phones to call, but if the user is in phone the other phone should not call. Any idées... Med vänlig hälsning / Best regards Magnus Löfqvist VMI Internet Services AB Hantverksvägen 15 760 40 VÄDDÖ, SWEDEN Tel +46 176 20 89 00 (02) Fax +46 176 20 89 19 E-mail: m...@vmi.se Homepage: www.vmi.sehttp://www.vmi.se/ Facebook: www.vmi.se/facebookhttp://www.vmi.se/facebook *** VMI E-mail disclaimer *** The information in this e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. Any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in the VMI General terms and conditions -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calender and EWS with shared calenders
Hi all, We want to use res_calender_ews to close users extensions if there are busy in there exchange calenders. It is possible to use shared calenders ? Ie, I have a resource user that have access to the users calenders, so I dont need to maintain every users password/username in asterisk. We are today running 1.8 but are going to upgrade to 11... Best regards / Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Spam] Re: Calender and EWS with shared calenders
4 jan 2013 kl. 23:59 skrev Danny Nicholas da...@debsinc.commailto:da...@debsinc.com: From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus Löfqvist Sent: Friday, January 04, 2013 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Calender and EWS with shared calenders Hi all, We want to use res_calender_ews to close users extensions if there are busy in there exchange calenders. It is possible to use shared calenders ? Ie, I have a resource user that have access to the users calenders, so I dont need to maintain every users password/username in asterisk. We are today running 1.8 but are going to upgrade to 11... Best regards / Magnus If it works for you in 1.8 it should work in 11.X – the only change is the $Revision number. Dont know if it workes in 1.8. Havent set it up yet. My question is if it works in 1.8 (or if it will work in 11) / Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with AGI and existing channel
Hi, Asterisk 1.8.10.0-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2012-03-08 23:05:09 UTC We have some problem when running a AGI script (build with PHP), existing channels (all of them) gets a hickup and then continues. We are using AGI to lookup incoming calls in directory. It is kinda annoying, and I don't understand how it can be related. Best regards Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Spam] Re: Problems with AGI and existing channel
Hi, The script is not essentials in this. I have tried with a empty script... It stills generate problems with existing channels. But if a try with a empty perl script, it dosent... seems to be something that php is doing when starting up. Best regards Magnus -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För A J Stiles Skickat: den 18 oktober 2012 13:57 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: [Spam] Re: [asterisk-users] Problems with AGI and existing channel On Thursday 18 October 2012, Magnus Löfqvist wrote: We have some problem when running a AGI script (build with PHP), existing channels (all of them) gets a hickup and then continues. We are using AGI to lookup incoming calls in directory. It is kinda annoying, and I don't understand how it can be related. I think you forgot to attach the actual script. Without that, we can't help you. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] EWS Calendar support
Hi, We are running Asterisk 1.8.10.0 and are testing calendar functions. I have it connected to our Exchange 2010. As we are using queues I have added Calendar:calendarname to the users hints, to arrange that the phone not will ring if the user are in a meeting. That works fine, but if the user removed the task from exchange calendar, then the user will still be busy on the hint, but not on calendar show calendars. Any thoughts ? // Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended Windows client to display CID?
Gilles skrev: Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple clients can connect simultaneously and each be notified of incoming calls. There may be yet other ways to do what I want. Are there open-source solutions you could recommend? Thank you. Hi Gilles, If you want someting really light weight there is always the old winpopup protocoll. For example: smbclient -M NETBIOSNAME text.txt (on linux) I am unsure about support in later versions of Windows but up to at least win2000 this pops up a simple message box on the windows machine remotely. /Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adit 600 over MGCP.
Hi, Anybody out there running Adit600s? I have in my care an Adit600 channel bank connected to an old (version 1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk (1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail. I have attempted to add the slowsequence = yes line to mgcp.conf. (It seemed to be the only likely candidate in the example files I found online.) No improvement. A working configuration or any other tips and ideas regarding this would be most welcome. Thank you in advance. Magnus Persson Log output: Jul 21 13:59:18 crabbofix asterisk[2689]: NOTICE[6220]: chan_mgcp.c:3564 in mgcp_request: Asked to get a channel of unsupported format '0' Jul 21 13:59:18 crabbofix asterisk[2689]: NOTICE[6220]: channel.c:2901 in __ast_request_and_dial_uniqueid: Unable to request channel MGCP/aal n/3...@adit.westel.nt and Jul 21 16:11:01 crabbofix asterisk[2689]: NOTICE[6263]: rtp.c:788 in process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.30.10 mgcp.conf (relevant parts) [general] port = 2427 ;port = 2727 bindaddr = 192.168.30.1 disallow = all allow = alaw [adit.westel.nt] host = 192.168.30.10 context = westel callgroup = 0 pickupgroup = 0 transfer = no cancallforward = no canreinvite = no dtmfmode = inband ;dtmfmode = rfc2833 wcardep=* ; define the internal lines callerid = Westel 069022130 line = aaln/1 line = aaln/2 line = aaln/3 line = aaln/4 ; remove all features transfer = no cancallforward = no canreinvite = no context = newmrg ;dtmfmode = rfc2833 dtmfmode = inband ; define old TeleVaxel callerid = MRG 2150 line = aaln/5 line = aaln/6 line = aaln/7 (snip)... Adit configuration (relevant parts): adit print config - -Adit 600 configuration file -Created on 01/24/2002 at 02:32:23 -This file is valid for the following configuration only: - -CardType - -SLOT A ITE SW Version: 9.0.0 -SLOT 1 FXS8Ax8 -SLOT 2 FXS8Ax8 -SLOT 3 FXS8Ax8 -SLOT 4 FXS8Ax8 -SLOT 5 FXS8Ax8 -SLOT 6 CMGx1 (snip)... -Setting slot 6 CMG. (snip)... set 6:1 up set 6 snmp name unknown set 6 snmp contact unknown set 6 snmp location unknown set 6 ntp server 192.168.30.1 set 6 ntp timezone 1 set 6 ntp enable set 6 cdr enable set 6 hookflash 0 set 6 mgcp addressformat nobrackets set 6 mgcp callagent address 192.168.30.1 set 6 mgcp callagent port 2427 set 6 mgcp up set 6 mgcp rsipwildcard enable set 6 voip ptime g711mu 10 set 6 voip ptime g711a 10 set 6 voip rtcp cname adit set 6 compander alaw set 6 voip sdpaddress gatewayid set 6:1:1:1 log start mgcp set 6:1:1:1-48 algorithm preference g711mu g711a g726_16 g726_24 g726_32 \ g726_40 set 6:1:1:4 echo cancellation disable set 6:1:1:4 fax bypass set 6:1:1:4 modem bypass reset 6 -Turning verification on. set verification on adit -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] confbridge not working?
Hi!
Did a quick test, worked as a clock:
exten = 0317998959,1,Set(CHANNEL(language)=se)
exten = 0317998959,n,Answer()
exten = 0317998959,n,ConfBridge(1001,s) 0317998959,n,Hangup()
On Thu, 18 Mar 2010 20:20:35 -0700, Kelvin Chan wrote:
Hi guys,
I'm trying to move away from meetme to loose the dependency on dahdi.
ConfBridge seems to be a good fit but I can't get it going. The document
sounds like an easy to use app. Am I missing any bridge_ modules?
Asterisk 1.6.2.0~rc2-0ubuntu1.2
-- Executing [...@outbound:1] Answer(SIP/109-b877a8c8, ) in new
stack
-- Executing [...@outbound:2] ConfBridge(SIP/109-b877a8c8,
conf) in new stack
[Mar 19 03:16:33] ERROR[2294]: app_confbridge.c:434
join_conference_bridge: Conference bridge '521' could not be created.
dial plan:
exten = _52X,1,Answer()
exten = _52X,n,ConfBridge(${EXTEN})
Thanks,
Kelvin
--
_
-- Bandwidth and Colocation Provided by
http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding agent with 2 phones to a queue
I tried, [agents] exten = 1,hint,SIP/0317998975SIP/0317998985 exten = 1,1,Dial(SIP/0317998975SIP/0317998985) and queue add member Local/1...@agents to 0317998989 sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 109s talktime), W:0, C:1, A:0, SL:0.0% within 0s Members: Local/1...@agents (dynamic) (Not in use) has taken no calls yet No Callers did call out from 0317998975 sip*CLI core show hints 0317998...@inputinterior.se : SIP/0317998975 State:InUse Watchers 0 0317998...@inputinterior.se : SIP/0317998985 State:Idle Watchers 0 1...@agents : SIP/0317998975 padding-left: 5px; margin-left: 5px; width: 100%; Your best option is likely to be to create a separate context that calls both numbers, like so... [agents] exten = 1,Dial(SIP/0317998975SIP/0317998985) ...then add Local/1...@agents to the queue. On 03/14/10 00:03, Magnus Benngård wrote: Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975 (dynamic) (Not in use) has taken no calls yet But what i would like to do is something like: queue add member SIP/0317998975SIP/0317998985 to 0317998989 But that doesnt work. :( sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975SIP/0317998985 (dynamic) (Unknown) has taken no calls yet Did try to add a hint: exten = kalle,hint,SIP/0317998975SIP/0317998985 just for testing purposes, that did work: ka...@inputinterior.se : SIP/0317998975SIP/0 State:Idle Watchers 0 But I cant figure out howto connect the queue with kalle, or maybe it is not possible? /Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding agent with 2 phones to a queue - SOLVED
queue add member Local/1...@agents to 0317998989 penalty 1 as Magnus Benngard state_interface hint:1...@agents 1,hint,SIP/0317998975SIP/0317998985 exten = 1,1,Dial(SIP/0317998975SIP/0317998985) and queue add member Local/1...@agents to 0317998989 sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 109s talktime), W:0, C:1, A:0, SL:0.0% within 0s Members: Local/1...@agents (dynamic) (Not in use) has taken no calls yet No Callers did call out from 0317998975 sip*CLI core show hints 0317998...@inputinterior.se : SIP/0317998975 State:InUse Watchers 0 0317998...@inputinterior.se : SIP/0317998985 State:Idle Watchers 0 1...@agents : SIP/0317998975 padding-left: 5px; margin-left: 5px; width: 100%; Your best option is likely to be to create a separate context that calls both numbers, like so... [agents] exten = 1,Dial(SIP/0317998975SIP/0317998985) ...then add Local/1...@agents to the queue. On 03/14/10 00:03, Magnus Benngård wrote: Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975 (dynamic) (Not in use) has taken no calls yet But what i would like to do is something like: queue add member SIP/0317998975SIP/0317998985 to 0317998989 But that doesnt work. :( sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975SIP/0317998985 (dynamic) (Unknown) has taken no calls yet Did try to add a hint: exten = kalle,hint,SIP/0317998975SIP/0317998985 just for testing purposes, that did work: ka...@inputinterior.se : SIP/0317998975SIP/0 State:Idle Watchers 0 But I cant figure out howto connect the queue with kalle, or maybe it is not possible? /Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DECT phone wont stop ringing
Hi, Did a test with Local, exten = 1234,1,Dial(Local/1...@agents) [agents] exten = 1,hint,SIP/0317998975SIP/0317998985 exten = 1,1,Dial(SIP/0317998975SIP/0317998985) When calling 1234, both 0317998975 and 0317998985 rings when answering in 0317998985, 0317998975 stops ringing, all fine but when answering in 0317998975, 0317998985 keeps on ringing... 0317998975 is an Avaya 9650 0317998985 is a Siemens S685IP (DECT) Shouldn't above scenario work or did I miss anything?-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding agent with 2 phones to a queue - SOLVED
Thx Rob! On Mon, 15 Mar 2010 00:53:06 +1100, Rob Hillis wrote: Glad to see I was able to point you in the right direction. On 03/14/10 23:56, Magnus Benngård wrote: queue add member Local/1...@agents to 0317998989 penalty 1 as Magnus Benngard state_interface hint:1...@agents On Sun, 14 Mar 2010 11:38:13 +0100, Magnus Benngård wrote: I tried, [agents] exten = 1,hint,SIP/0317998975SIP/0317998985 exten = 1,1,Dial(SIP/0317998975SIP/0317998985) and queue add member Local/1...@agents to 0317998989 sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 109s talktime), W:0, C:1, A:0, SL:0.0% within 0s Members: Local/1...@agents (dynamic) (Not in use) has taken no calls yet No Callers did call out from 0317998975 sip*CLI core show hints 0317998...@inputinterior.se : SIP/0317998975 State:InUse Watchers 0 0317998...@inputinterior.se : SIP/0317998985 State:Idle Watchers 0 1...@agents : SIP/0317998975SIP/0 State:InUse Watchers 0 Looks correct to me... but: sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 109s talktime), W:0, C:1, A:0, SL:0.0% within 0s Members: Local/1...@agents (dynamic) (Not in use) has taken no calls yet No Callers Do understand that I have missed something here, shouldn't it be InUse?, Calling the queue (both phone are ringing) and answer gives: sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (1s holdtime, 109s talktime), W:0, C:1, A:0, SL:0.0% within 0s Members: Local/1...@agents (dynamic) (Not in use) has taken no calls yet No Callers I am completly lost. :( On Sun, 14 Mar 2010 01:08:53 +1100, Rob Hillis wrote: Your best option is likely to be to create a separate context that calls both numbers, like so... [agents] exten = 1,Dial(SIP/0317998975SIP/0317998985) ...then add Local/1...@agents to the queue. On 03/14/10 00:03, Magnus Benngård wrote: Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975 (dynamic) (Not in use) has taken no calls yet But what i would like to do is something like: queue add member SIP/0317998975SIP/0317998985 to 0317998989 But that doesnt work. :( sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975SIP/0317998985 (dynamic) (Unknown) has taken no calls yet Did try to add a hint: exten = kalle,hint,SIP/0317998975SIP/0317998985 just for testing purposes, that did work: ka...@inputinterior.se : SIP/0317998975SIP/0 State:Idle Watchers 0 But I cant figure out howto connect the queue with kalle, or maybe it is not possible? /Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding agent with 2 phones to a queue
Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975 (dynamic) (Not in use) has taken no calls yet But what i would like to do is something like: queue add member SIP/0317998975SIP/0317998985 to 0317998989 But that doesnt work. :( sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975SIP/0317998985 (dynamic) (Unknown) has taken no calls yet Did try to add a hint: exten = kalle,hint,SIP/0317998975SIP/0317998985 just for testing purposes, that did work: ka...@inputinterior.se : SIP/0317998975SIP/0 State:Idle Watchers 0 But I cant figure out howto connect the queue with kalle, or maybe it is not possible? /Magnus-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 00h323 cant get gatekeeper to connect
I am running Asterisk trunk with ooh323 towards an Avaya CM 3.1 without
any problems. I need your ooh323.conf and all relevant CM config
(signal-group, trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis wrote: I'm trying to
connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN). When
chan_ooh323 first loads it tries to establish a connection with the gk but
I it fails. I have the following extract from the ooh323 log. Can anyone
give some insight? Thanks! MD 23:02:59:045 Sent GRQ message
23:02:59:045 GkClient Received RAS Message
23:02:59:045 Received RAS Message = {
23:02:59:045 gatekeeperReject = {
23:02:59:045 requestSeqNum = {
23:02:59:045 1
23:02:59:046 }
23:02:59:046 protocolIdentifier = {
23:02:59:046 {
23:02:59:046 0 0 8 2250 0 5 }
23:02:59:046 }
23:02:59:047 rejectReason = {
23:02:59:047 resourceUnavailable = {
23:02:59:048 NULL
23:02:59:048 }
23:02:59:048 }
23:02:59:048 }
23:02:59:048 }
23:02:59:048
Gatekeeper Reject (GRJ) message received
23:02:59:048 Deleted GRQ Timer.
23:02:59:048 Error: Gatekeeper Reject - Resource Unavailable
23:02:59:049 Error: Gatekeeper error. Either Gk not responding or Gk
sending invalid messages
23:02:59:049 Error: Gatekeeper error detected. Closing GkClient as Gk mode
is UseSpecifcGatekeeper
23:02:59:049 Destroying Gatekeep
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 00h323 cant get gatekeeper to connect
hmmm... will be hard to help u without u having access... will do my best. Here is my ooh323.conf anyway... sip:/etc/asterisk# cat ooh323.conf [general] bindaddr=213.88.138.183 port=5088 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Trunk with multiple remote ip-addresses
Hi! Did a setup of 2 peers as Klaus suggested, it worked thx! Has anyone thought about the possibility to add multiple ip/hosts to host=? I my case: host=130.244.190.42,130.244.190.46 or host=sip-corporate1.tele2.se,sip-corporate2.tele2.se Step 1 could be to send to the first ip/host and accept from both. Step 2 could be round-robin send if both are up and alive... Btw, did try trunk version, no support for multiple SRV records there. Am 02.03.2010 08:50, schrieb Magnus Benngård: Hi, Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem to get outgoing calls to work but i have some problems with incoming. Did set srvlookup=yes in sip.conf. Sending all outgoing calls to sip-corporate.tele2.se which is either sip-corporate1.tele2.se (130.244.190.42) or sip-corporate1.tele2.se (130.244.190.46). If i do a sip show peer Tele2, I see that Asterisk has chosen one of them: ToHost : sip-corporate.tele2.se Addr-IP : 130.244.190.46 Port 5060 Now my problems starts, when Tele2 sends a call to my Asterisk, the call can come frome any of those two ip-adresses. If it comes from 130.244.190.46 everything if fine, but if it comes from 130.244.190.42: [Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167 handle_request_invite: Failed to authenticate! I thought srvlookup=yes should take care about that, but then i read a little bit more and found: Note: Asterisk only uses the first host in SRV records. :( Hi Magnus! Asterisk does not support multiple SRV records (expcet there were some recent changes which I missed) - it takes one of the most priors and use it all the time. Thus, in your scenario you have to specify the possible inbound sources manually as peers: [tele2-1] type=peer host=130.244.190.42 context=fromTele2 ... [tele2-2] type=peer host=130.244.190.46 context=fromTele2 ... regards klaus Can anyone plz give me some hint howto solve my problem? Regards, Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Trunk with multiple remote ip-addresses
Hi, Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem to get outgoing calls to work but i have some problems with incoming. Did set srvlookup=yes in sip.conf. Sending all outgoing calls to sip-corporate.tele2.se which is either sip-corporate1.tele2.se (130.244.190.42) or sip-corporate1.tele2.se (130.244.190.46). If i do a sip show peer Tele2, I see that Asterisk has chosen one of them: ToHost : sip-corporate.tele2.se Addr-IP : 130.244.190.46 Port 5060 Now my problems starts, when Tele2 sends a call to my Asterisk, the call can come frome any of those two ip-adresses. If it comes from 130.244.190.46 everything if fine, but if it comes from 130.244.190.42: [Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167 handle_request_invite: Failed to authenticate! I thought srvlookup=yes should take care about that, but then i read a little bit more and found: Note: Asterisk only uses the first host in SRV records. :( Can anyone plz give me some hint howto solve my problem? Regards, Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax, T38 and NAT
Yes, when I added t38pt_usertpsource=yes to the NAT'ed fax everything works! Big thanks Johann! On Sun, 21 Feb 2010 17:22:40 +0100, Magnus Benngård wrote: t38pt_usertpsource=yes seems to do the trick, switches to T38 and fax seems to go through (cant be 100% sure, the fax i am sending to is 500 km avay from me, but i dont get any errors and my fax thinks everything is ok, so I cross my fingers),,, On Sun, 21 Feb 2010 16:36:42 +0100, Johann Steinwendtner wrote: Magnus Benngård wrote: Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38 on 0197673581, changing t38pt_udptl=no to t38pt_udptl=yes,fec and try to send from 0197673581 to 0851711201, it is not working, switches to T38 sendimg a lot of UDPTL packages but it looks like (at least for me) that addresses are wrong. UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) 90.230.92.67 is WAN ip of 0197673581's router. 10.242.20.149 is ip of 0851711201's ATA (SPA2102). Shouldn't the UDPTL stream go through Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=yes context=inputinterior.se directmedia=yes dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=no qualify=yes sendrpid=pai t38pt_udptl=yes,fec transport=udp trustrpid=yes type=friend videosupport=no [0851711290] secret=xyz callerid=Input Interior Sundbyberg (fax) ... rest is the same as [0851711201] Regards, Magnus Maybe you should give t38pt_usertpsource=yes a try. Regards Hans -- _ -- 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] [OT] Asterisk 1.6 and DECT Phones
Running Asterisk trunk with Siemens Gigaset S685IP, no normal problems, just some with connected-line, probaly me, who is not smart enough. :( Sound is great, use them both at our WAN and NAT'et at my home, DTMF working as a clock... what more can I say? On Mon, 22 Feb 2010 16:43:04 -, Chris Bagnall wrote: looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. We've been using the Siemens Gigaset range for a few years now (specifically C475IP and S685IP). Not had any major problems with them. Regards, Chris -- For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons -- _ -- 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] Fax, T38 and NAT
t38pt_usertpsource=yes seems to do the trick, switches to T38 and fax seems to go through (cant be 100% sure, the fax i am sending to is 500 km avay from me, but i dont get any errors and my fax thinks everything is ok, so I cross my fingers),,, On Sun, 21 Feb 2010 16:36:42 +0100, Johann Steinwendtner wrote: Magnus Benngård wrote: Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38 on 0197673581, changing t38pt_udptl=no to t38pt_udptl=yes,fec and try to send from 0197673581 to 0851711201, it is not working, switches to T38 sendimg a lot of UDPTL packages but it looks like (at least for me) that addresses are wrong. UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) 90.230.92.67 is WAN ip of 0197673581's router. 10.242.20.149 is ip of 0851711201's ATA (SPA2102). Shouldn't the UDPTL stream go through Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=yes context=inputinterior.se directmedia=yes dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=no qualify=yes sendrpid=pai t38pt_udptl=yes,fec transport=udp trustrpid=yes type=friend videosupport=no [0851711290] secret=xyz callerid=Input Interior Sundbyberg (fax) ... rest is the same as [0851711201] Regards, Magnus Maybe you should give t38pt_usertpsource=yes a try. Regards Hans -- _ -- 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] Fax, T38 and NAT
Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38 on 0197673581, changing t38pt_udptl=no to t38pt_udptl=yes,fec and try to send from 0197673581 to 0851711201, it is not working, switches to T38 sendimg a lot of UDPTL packages but it looks like (at least for me) that addresses are wrong. UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) 90.230.92.67 is WAN ip of 0197673581's router. 10.242.20.149 is ip of 0851711201's ATA (SPA2102). Shouldn't the UDPTL stream go through Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=yes context=inputinterior.se directmedia=yes dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=no qualify=yes sendrpid=pai t38pt_udptl=yes,fec transport=udp trustrpid=yes type=friend videosupport=no [0851711290] secret=xyz callerid=Input Interior Sundbyberg (fax) ... rest is the same as [0851711201] Regards, Magnus-- _ -- 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] CONNECTEDLINE
Gentlemen, Did tryout CONNECTEDLINE function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and stupid extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from 955 to Connected Line 955 when my call is answered, shouldn't the display on my phone change to Connected Line 0317998955? exten = 956,1,Goto(0317998956,1) exten = 0317998956,1,Set(CONNECTEDLINE(all)=Connected Line ) exten = 0317998956,n,Answer() exten = 0317998956,n,Wait(2) exten = 0317998956,n,Hangup() -- Executing [...@inputinterior.se:1] Goto(SIP/0317998975-0004, 0317998956,1) in new stack -- Goto (inputinterior.se,0317998956,1) -- Executing [0317998...@inputinterior.se:1] Set(SIP/0317998975-0004, CONNECTEDLINE(all)=Connected Line ) in new stack -- Executing [0317998...@inputinterior.se:2] Answer(SIP/0317998975-0004, ) in new stack -- Executing [0317998...@inputinterior.se:3] Wait(SIP/0317998975-0004, 2) in new stack -- Executing [0317998...@inputinterior.se:4] Hangup(SIP/0317998975-0004, ) in new stack == Spawn extension (inputinterior.se, 0317998956, 4) exited non-zero on 'SIP/0317998975-0004' Asterisk SVN-trunk-r245147M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC /Magnus-- _ -- 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] Aastra RFP-32 and CLID
Gentlemen, I did borrow an Aastra RFP 32 for some tests that i wanted to do. Everything seems to be working except CLID. Setup as below: DECT handset - GAP - Aastra RFP-32 - SIP - Asterisk - SIP Phone When SIP Phone calls DECT handset, the display on the DECT handset only shows the number of SIP phone, not the name. Did unplug the Aastra RFP-32 and register X-Lite to the same extension. Ofc X-Lite showed both number and name, so it must be me that have missded something in the Aastra RFP-32. :( Any suggestions? Med vänliga hälsningar MAGNUS BENNGRD Direktnr 031-799 89 75 Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: 7-- _ -- 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] Attended Transfer with REFER
checkout ${BLINDTRANSFER}
On Tue, 26 Jan 2010 15:48:51 +, Örn Arnarson wrote: Hi guys,
I am wondering (and have been unable to find out thus far) whether
Asterisk sets some special channel variables or something when a call is
transfered with the REFER method. Basically, I'm trying to figure out if it
is possible to somehow get a transferred call back to the transferrer (as
it is done with the built-in atxfer) after X seconds (or an unsuccessful
attempt).
Using a timeout in the Dial command is not suitable unless I am able to
tell somehow that the call in question is being forwarded (which is of
course not the case, as the Dial command is called befer the REFER is
sent).
Can anyone think of a way to get the call back to the transferrer after
this timeout?
Best regards, Örn Arnarson
--
_
-- 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] ReceiveFAX and SendFAX questions
Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten = 101,1,Answer() exten = 101,2,Wait(3) exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten = 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff /var/spool/asterisk/tmp/fax.pdf) exten = 101,5,System(mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf magnu...@inputinterior.se /dev/null) I do receive the fax, the fax got converted to a pdf but 101,5 never get executed, when i look in cli, last line is 101,4... can any1 se why? 2:nd example: exten = 101,1,Answer() exten = 101,2,Wait(3) exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten = 101,4,System(fax.sh) cat /usr/bin/fax.sh tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff /var/spool/asterisk/tmp/fax.pdf mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf bo...@inputinterior.se /dev/null That works, i receive the fax as an attachment, but as I asked before why is not example 1 working? SendFAX question: exten = 101,1,Answer() exten = 101,2,Wait(3) exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten = 101,4,Some magical way to setup the channel to: SIP/033211101 exten = 101,5,SendFAX(/var/spool/asterisk/tmp/fax.tiff) 033211101 is an ATA (SPA2102) registered to *. I wonder if it is possible to do something like my example or not? Any suggestions? I was looking at: http://www.evilspurv.net/blog/2010/01/sending-pdfs-as-fax-with-asterisk/ I could do something like that but i would prefer to have all in the dialplan without need for an external program. /Magnus -- _ -- 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] DTMF Issue?
This is the setting i am using for Avaya CM to Aseterisk. (and pinf code is working when dialing from Avaya to Asterisk conference)sip:/etc/asterisk# cat ooh323.conf [general] bindaddr=213.88.138.183 port=5088 context=inputinterior.se dtmfmode=rfc2833 ;h323id=may day ;callerid=may day disallow=all allow=alaw:40 ;allow=alaw tracelevel=6 [Avaya] type=friend context=inputinterior.se dtmfcodec=127 dtmfmode=rfc2833 ip=10.242.14.11 port=5088 I am using H.323 to create a trunk between Asterisk and Avaya IP Office system. Everything is working correctly, Asterisk can call Avaya and vise versa. Now I create a conference room with a user pin in Asterisk. Avaya can call into the conference room, but can enter the pin number. The error message they are getting is the invalid pin number. However the pin number works if they use it over the phone line, but not via H.323 trunk. I have a feeling it's the DTMF setting, but I don't know what I should set it to. Got any ideas? From the Avaya H.323 trunk: Jan 19 09:59:43 VERBOSE [7168] logger.c: -- Playing 'enter-conf-pin-number' (language 'en') Jan 19 09:59:56 VERBOSE [7168] logger.c: -- USER ENTERED NOTHING. Jan 19 09:59:56 VERBOSE [7168] logger.c: -- Executing [1...@from-internal:8] GotoIf(OOH323/denver-0ca5, 0?USER) in new stack Jan 19 09:59:56 VERBOSE [7168] logger.c: -- Executing [1...@from-internal:11] Playback(OOH323/denver-0ca5, conf-invalidpin) in new stack Jan 19 09:59:56 VERBOSE [7168] logger.c: -- Playing 'conf-invalidpin' (language 'en') From the landline using Avaya: Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Executing [1...@ext-meetme:7] Read(DAHDI/2-1, PIN|enter-conf-pin-number) in new stack Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Playing 'enter-conf-pin-number' (language 'en') Jan 19 10:00:43 VERBOSE [7177] logger.c: -- USER ENTERED 'THE PIN NUMBER' Jan 19 10:00:43 VERBOSE [7177] logger.c: -- Executing [1...@ext-meetme:8] GotoIf(DAHDI/2-1, 1?USER) in new stack My H323.conf file: [general] port=1720 bindaddr=ip address of asterisk gateway=no faststart=yes h245tunneling=yes h323id=ObjSysAsterisk e164=100 callerid=asterisk gatekeeper = DISABLE context=default disallow=all allow=ulaw dtmfmode=rfc2833 progress_setup = 8 progress_alert = 8 [denver] type=friend port=1720 ip=ip address of avaya context=from-internal disallow=all allow=ulaw rtptimeout=90 -- _ -- 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] DTMF Issue?
Make sure u have the correct DTMF over IP (or what it is named in IP Office, thats the CM name) setting on the signal-group. In my case: DTMF over IP: rtp-payload On Wed, 20 Jan 2010 16:11:58 -0800 (PST), hin lee wrote: Beside the port number and the alaw, the only difference is the dtmf. I added this into my ooh323.conf and it still didn't work. dtmfcodec=127 dtmfmode=rfc2833 I also tried: dtmfmode=h245signal This is to an Avaya IP Office 500. -- _ -- 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] Avaya 96xx handset with SIP 2.5, no name in display
Hi! Any familiar with Avaya handsets? Did convert a 9650 handset to SIP. Cant get the name just the number on the Avaya display. Did put: SET DISPLAY_NAME_NUMBER 1 in 46xxsettings.txt When I call from 0317998985 (Siemens DECT) to 0317998975 (Avaya 9650) i just se 0317998985 in the Avaya display. :( For me the invite looks ok, both name and number. From: Cecilia Benngard ;tag=as46c6bdf6 To: ;tag=49facf544b3897974dc0bcec_T192.168.0.115 Call-ID: 4245be4130adf8727b9bcd9b1575d...@213.88.138.183 CSeq: 102 INVITE Via: sip/2.0/udp 213.88.138.183:5060;branch=z9hG4bK75e3128c;rport Contact: Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Accept-Language: en User-Agent: Avaya one-X Deskphone Content-Length: 0 I am loost, I wonder if i ran into an Avaya bug or am I doing something totally wrong? If I call from Avaya to Siemens, I see both name and number!___ -- 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] Showing name of extension when calling
Is it in the trunk version or will it be added there? On Tue, 22 Dec 2009 08:12:40 -0600, Kevin P. Fleming wrote: Magnus Benngård wrote: Is it possible, when placing a call that u see the name of the extension in your diplay? For example, 2 sip.conf entries: [971] callerid=Stefan [975] callerid=Magnus 975 calls 971 today 975 sees 971 in the display but would like to se: Stefan or just Stefan or... This is called Connected Party information display, and it will be in Asterisk 1.8. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Showing name of extension when calling
Hi! Is it possible, when placing a call that u see the name of the extension in your diplay? For example, 2 sip.conf entries: [971] callerid=Stefan [975] callerid=Magnus 975 calls 971 today 975 sees 971 in the display but would like to se: Stefan or just Stefan or... /Magnus ___ -- 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] Manager command that equal to database show CFIM
Hi! Probably me that cannot read the manual... I am trying to get all Keys that belongs to a certain Family from the manager interface. Can just get single values for example: Action: DBGet Family: CFIM Key: 0317998975 I was looking for something like Action: DBShow Family: CFIM. Any one has some smart way to implement it or did I just miss some stuff... /Magnus ___ -- 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] Rewrite of calling number for all extensions
Hi!
I am trying to figure out how to rewrite calling number for all
extensions.
What I am trying to do is:
1) Have a block of rewriting rules that will apply to all calls:
Something like...
(???),ExecIf($[${CALLERID(num)}=702221448]?Set(CALLERID(all)=Cecilia
Benngard))
(???),ExecIf($[${CALLERID(num)}=705886363]?Set(CALLERID(all)=Magnus
Benngard))
...
2) Standard dialing rules...
971,1,Dial(SIP/0317998971)
975,1,Dial(SIP/0317998975)
...
I know hot to do it per extension...
971,1,ExecIf($[${CALLERID(num)}=702221448]?Set(CALLERID(all)=Cecilia
Benngard)
971,2,Dial(SIP/0317998971)
But there must be a better way to do it, or?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rewrite of calling number for all extensions
Did found a way to do it:
exten =
975,1,ExecIf($[${DB_EXISTS(CFIM/0317998975)}]?Goto(${DB(CFIM/0317998975)},1)
exten = 975,2,Goto(set-caller-id,s,1)
exten = 975,3,Goto(975-${DEVICE_STATE(SIP/0317998975)},1)
exten = 975-INUSE,1,VoiceMail(0317998...@inputinterior.se,bs)
..
[set-caller-id]
;
exten =
s,1,ExecIf($[${CALLERID(num)}=702221448]?Set(CALLERID(all)=Cecilia
Benngard))
exten = s,1,ExecIf($[${CALLERID(num)}=705886363]?Set(CALLERID(all)=Magnus
Benngard))
exten = s,2,Goto(inputinterior.se,${CALLERID(dnid)},3)
Hi!
I am trying to figure out how to rewrite calling number for all
extensions.
What I am trying to do is:
1) Have a block of rewriting rules that will apply to all calls:
Something like...
(???),ExecIf($[${CALLERID(num)}=702221448]?Set(CALLERID(all)=Cecilia
Benngard))
(???),ExecIf($[${CALLERID(num)}=705886363]?Set(CALLERID(all)=Magnus
Benngard))
...
2) Standard dialing rules...
971,1,Dial(SIP/0317998971)
975,1,Dial(SIP/0317998975)
...
I
know hot to do it per extension...
971,1,ExecIf($[${CALLERID(num)}=702221448]?Set(CALLERID(all)=Cecilia
Benngard)
971,2,Dial(SIP/0317998971)
But there must be a better way to do it, or?
___
-- 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] wrapuptime?
Hi! Trying to understand how wrapuptime is working... I have written a small php script that let agents log in/out off a queue. That part is working as a clock but wrapuptime is not doing what I expect. Input Interiör - Queue Manager 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (5s holdtime, 94s talktime), W:0, C:8, A:1, SL:0.0% within 0s Members: SIP/0317998971 with penalty 2 (dynamic) (Not in use) has taken no calls yet SIP/0317998975 with penalty 2 (dynamic) (Not in use) has taken no calls yet SIP/0317998972 with penalty 1 (dynamic) (Not in use) has taken 8 calls (last was 2 secs ago) No Callers SIP/0317998972 did hang up 8 seconds ago, but if someone calls the queue at this moment, the call will start ringing on SIP/0317998972 again. I thought the wrapuptime should cause the call to go to one of the other agents. Did I miss anything in the configs or is it that we have different penalties or...? cat queues.conf [general] ; autofill=yes keepstats=yes ; ; [0317998989] retry=5 strategy=rrmemory timeout=20 wrapuptime=120 cat agents.conf [general] ; persistentagents=yes ; ; [agents] ; agent = 0317998971,1234,Stefan Andersson agent = 0317998972,1234,Kerem Tubluk agent = 0317998975,1234,Magnus Benngard agent = 0317998976,1234,Jimmy Beckman ___ -- 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] Rewrite calling number of incoming call
Thx!
Worked as a clock!
I did modify it to:
exten =
977,1,ExecIf($[${CALLERID(num)}=733025975]?Set(CALLERID(all)=Magnus
Benngard))
Even better! :)
On Tue, 15 Dec 2009 11:32:44 -0600, Steve Johnson wrote:
How about:
exten = 977,1,ExecIf($[${CALLERID(num)} =
733025975]?Set(CALLERID(num)=0317998975))
exten = 977,n,ExecIf($[${CALLERID(num)} =
1234]?Set(CALLERID(num)=317998977))
exten = 977,n,ExecIf($[${CALLERID(num)} =
5678]?Set(CALLERID(num)=317998978))
[..]
exten = 977,n,Dial(SIP/0317998977)
On Mon, Dec 14, 2009 at 12:21 PM, Magnus Benngård
wrote:
Hi!
Trying to figure out how to rewrite calling number of an incoming
call...
A cell phone (0733025975) dials a X-Lite (977).
X-Lite shows 733025975 at the display, but I want it to be 0317998975.
I thought i could do something like:
exten = 977/733025975,1,Set(CALLERID(number)=0317998975)
exten = 977,n,Dial(SIP/0317998977)
[Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272 answer_call:
Dropping
call
because extensions '977', 's' and 'i' doesn't exists in context
[inputinterior.se]
Rewriting of outgoing is working... snip
exten = _0X!/0317998975,1,Set(CALLERID(number)=317998975)
exten = _0X!/0317998977,1,Set(CALLERID(number)=317998977)
exten = _0X!/0317998978,1,Set(CALLERID(number)=317998978)
exten = _0X!/0317998985,1,Set(CALLERID(number)=317998985)
exten = _0X!/0317998987,1,Set(CALLERID(number)=317998987)
exten = _0X!,n,Dial(H323/0${ext...@avaya)
Can someone guide me on the correct track?
___
-- 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] DEVICE_STATE
Thx! Did try callcounter=yes and it worked the way u told me! It might have solved another problem 2, need to do some more tests... On Sun, 13 Dec 2009 15:14:22 -0500, Leif Madsen wrote: Philipp Kempgen wrote: Magnus Benngård schrieb: Set call-limit=10 (or any other value 0) Actually, I believe call-limit is deprecated, and you can instead use callcounter=yes Leif Madsen. ___ -- 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] Dial with timeout don't end call
Did move 0317998975 phone from my home to my office, didnt need any:
nat=yes in sip.conf, everything worked.
I did also add callcounter=yes in sip.conf so I am not sure how it
will work when I move the phone to my home and need nat=yes again.
Will do some tests later tonight when I am at home.
On Sun, 13 Dec 2009 14:25:39 +0100, Magnus Benngård wrote: Hi!
Trying to figure out what I am doing wrong...
1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256)
1 Cell phone 00733025975 attached through H323.
extensions.conf
exten = 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1)
exten = 975-INUSE,1,VoiceMail(0317998...@inputinterior.se,bs)
exten = 975-INUSE,2,Hangup()
exten = 975-NOANSWER,1,VoiceMail(0317998...@inputinterior.se,us)
exten = 975-NOANSWER,2,Hangup()
exten = 975-NOT_INUSE,1,Dial(SIP/0317998975H323/00733025...@avaya,20)
exten = 975-NOT_INUSE,2,Goto(975-${DIALSTATUS},1)
exten = 975-NOT_INUSE,3,Hangup()
When calling 975, both SIP and cell
phone starts to ring.
Answering on the SIP phone, cell phone stop ringing.
Answering on the cell phone, SIP phone keeps ringing.
If not answering any, cell phone stops ringing after 20 sec but
SIP phone just keeps ringing.
== Using UDPTL CoS mark 5
-- Executing [...@inputinterior.se:1] Goto(SIP/0317998985-005e,
975-NOT_INUSE,1) in new stack
-- Goto (inputinterior.se,975-NOT_INUSE,1)
-- Executing [975-not_in...@inputinterior.se:1]
Dial(SIP/0317998985-005e, SIP/0317998975H323/00733025...@avaya,20)
in new stack
== Using UDPTL CoS mark 5
-- Called 0317998975
-- Requested transfer capability: 0x00 - SPEECH
-- Called 00733025...@avaya
-- SIP/0317998975-005f is ringing
-- H323/Avaya-16 is making progress passing it to SIP/0317998985-005e
-- H323/Avaya-16 is making progress passing it to SIP/0317998985-005e
-- H323/Avaya-16 is ringing
-- Nobody picked up in 2 ms
-- Executing [975-not_in...@inputinterior.se:2]
Goto(SIP/0317998985-005e,
975-NOANSWER,1) in new stack
-- Goto (inputinterior.se,975-NOANSWER,1)
-- Executing [975-noans...@inputinterior.se:1]
VoiceMail(SIP/0317998985-005e, 0317998...@inputinterior.se,us) in
new stack
-- Playing
'/var/spool/asterisk/voicemail/inputinterior.se/0317998975/unavail.slin'
(language 'se')
-- Playing 'beep.gsm' (language 'se')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/inputinterior.se/0317998975/tmp/EKTi4P
format: wav, 0x8c448d0
-- User hung up
== Spawn extension (inputinterior.se, 975-NOANSWER, 1) exited non-zero on
'SIP/0317998985-005e'
___
-- 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] Rewrite calling number of incoming call
Hi!
Trying to figure out how to rewrite calling number of an incoming call...
A cell phone (0733025975) dials a X-Lite (977).
X-Lite shows 733025975 at the display, but I want it to be 0317998975.
I thought i could do something like:
exten = 977/733025975,1,Set(CALLERID(number)=0317998975)
exten = 977,n,Dial(SIP/0317998977)
[Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272 answer_call: Dropping
call because extensions '977', 's' and 'i' doesn't exists in context
[inputinterior.se]
Rewriting of outgoing is working... snip
exten = _0X!/0317998975,1,Set(CALLERID(number)=317998975)
exten = _0X!/0317998977,1,Set(CALLERID(number)=317998977)
exten = _0X!/0317998978,1,Set(CALLERID(number)=317998978)
exten = _0X!/0317998985,1,Set(CALLERID(number)=317998985)
exten = _0X!/0317998987,1,Set(CALLERID(number)=317998987)
exten = _0X!,n,Dial(H323/0${ext...@avaya)
Can someone guide me on the correct track?
___
-- 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] DEVICE_STATE - Solved
Thx, that did the trick!
On Sat, 12 Dec 2009 17:34:19 +0100, Philipp Kempgen wrote:
Magnus Benngård schrieb:
I am trying to figure out how DEVICE_STATE is working, no luck so far.
sip.conf
[0317998975]
Set
call-limit=10
(or any other value 0)
extensions.conf
exten = 0317998975,hint,SIP/0317998975
exten = 0317998975,1,NoOp(0317998...@inputinterior.se has state
${DEVICE_STATE(SIP/0317998975)})
exten = 0317998975,2,Dial(SIP/0317998975)
It doesn't matter if I have a call on 0317998975 or not. i always get:
-- Executing [0317998...@inputinterior.se:1]
NoOp(SIP/0317998985-0011, 0317998...@inputinterior.se has state
NOT_INUSE) in new stack
So I figure out that I have missed something but cant figure out what.
:(
Any ideeas?
sip.conf:
[general]
allowsubscribe = yes
notifyringing = yes
notifyhold = yes
limitonpeers = yes
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer:
Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
--
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avaya 9650 SIP phone and dial timeout
Hi!
Have a weired problem with Avaya 9650 phones:
extensions.conf
exten = 0317998975,hint,SIP/0317998975
exten = 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1)
exten = 0317998975,2,Hangup()
exten = 0317998975-INUSE,1,VoiceMail(0317998...@inputinterior.se,bs)
exten = 0317998975-INUSE,2,Hangup()
exten = 0317998975-NOANSWER,1,VoiceMail(0317998...@inputinterior.se,us)
exten = 0317998975-NOANSWER,2,Hangup()
exten = 0317998975-NOT_INUSE,1,Dial(SIP/0317998975,2)
exten = 0317998975-NOT_INUSE,2,Goto(0317998975-${DIALSTATUS},1)
exten = 0317998975-NOT_INUSE,3,Hangup()
I know that I have a very short dial timeout, just for testing purposes.
If i call 0317998975 and that extension is free:
The 9650 phones rings for 2 seconds.
== Using UDPTL CoS mark 5
-- Executing [0317998...@inputinterior.se:1]
Goto(SIP/0317998985-0031, 0317998975-NOT_INUSE,1) in new stack
-- Goto (inputinterior.se,0317998975-NOT_INUSE,1)
-- Executing
[0317998975-not_in...@inputinterior.se:1]
Dial(SIP/0317998985-0031, SIP/0317998975,2) in new stack
== Using UDPTL CoS mark 5
-- Called 0317998975
-- SIP/0317998975-0032 is ringing
-- Nobody picked up in 2000 ms
-- Executing [0317998975-not_in...@inputinterior.se:2]
Goto(SIP/0317998985-0031, 0317998975-NOANSWER,1) in new stack
-- Goto (inputinterior.se,0317998975-NOANSWER,1)
-- Executing [0317998975-noans...@inputinterior.se:1]
VoiceMail(SIP/0317998985-0031, 0317998...@inputinterior.se,us) in
new stack
-- Playing
'/var/spool/asterisk/voicemail/inputinterior.se/0317998975/unavail.slin'
(language 'se')
And as u can see the systems plays my unavailable message but,
the 9650 phones keep ringing, forever, or at least until I lift
and put down the handset.
Any ideas how i cant stop the ringing?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial with timeout don't end call
Hi!
Trying to figure out what I am doing wrong...
1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256)
1 Cell phone 00733025975 attached through H323.
extensions.conf
exten = 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1)
exten = 975-INUSE,1,VoiceMail(0317998...@inputinterior.se,bs)
exten = 975-INUSE,2,Hangup()
exten = 975-NOANSWER,1,VoiceMail(0317998...@inputinterior.se,us)
exten = 975-NOANSWER,2,Hangup()
exten = 975-NOT_INUSE,1,Dial(SIP/0317998975H323/00733025...@avaya,20)
exten = 975-NOT_INUSE,2,Goto(975-${DIALSTATUS},1)
exten = 975-NOT_INUSE,3,Hangup()
When calling 975, both SIP and cell phone starts to ring.
Answering on the SIP phone, cell phone stop ringing.
Answering on the cell phone, SIP phone keeps ringing.
If not answering any, cell phone stops ringing after 20 sec but
SIP phone just keeps ringing.
== Using UDPTL CoS mark 5
-- Executing [...@inputinterior.se:1] Goto(SIP/0317998985-005e,
975-NOT_INUSE,1) in new stack
--
Goto (inputinterior.se,975-NOT_INUSE,1)
-- Executing [975-not_in...@inputinterior.se:1]
Dial(SIP/0317998985-005e, SIP/0317998975H323/00733025...@avaya,20)
in new stack
== Using UDPTL CoS mark 5
-- Called 0317998975
-- Requested transfer capability: 0x00 - SPEECH
-- Called 00733025...@avaya
-- SIP/0317998975-005f is ringing
-- H323/Avaya-16 is making progress passing it to SIP/0317998985-005e
-- H323/Avaya-16 is making progress passing it to SIP/0317998985-005e
-- H323/Avaya-16 is ringing
-- Nobody picked up in 2 ms
-- Executing [975-not_in...@inputinterior.se:2]
Goto(SIP/0317998985-005e, 975-NOANSWER,1) in new stack
-- Goto (inputinterior.se,975-NOANSWER,1)
-- Executing [975-noans...@inputinterior.se:1]
VoiceMail(SIP/0317998985-005e, 0317998...@inputinterior.se,us) in
new stack
-- Playing
'/var/spool/asterisk/voicemail/inputinterior.se/0317998975/unavail.slin'
(language 'se')
-- Playing 'beep.gsm' (language 'se')
-- Recording
the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/inputinterior.se/0317998975/tmp/EKTi4P
format: wav, 0x8c448d0
-- User hung up
== Spawn extension (inputinterior.se, 975-NOANSWER, 1) exited non-zero on
'SIP/0317998985-005e'
___
-- 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] DEVICE_STATE
Hi all!
I am trying to figure out how DEVICE_STATE is working, no luck so far.
sip.conf
[0317998975]
type=friend
regexten=0317998975
secret=
username=0317998975
callerid=Magnus Benngard
mailbox=0317998...@inputinterior.se
host=dynamic
canreinvite=yes
dtmfmode=rfc2833
nat=yes
disallow=all
allow=alaw
extensions.conf
exten = 0317998975,hint,SIP/0317998975
exten = 0317998975,1,NoOp(0317998...@inputinterior.se has state
${DEVICE_STATE(SIP/0317998975)})
exten = 0317998975,2,Dial(SIP/0317998975)
It doesn't matter if I have a call on 0317998975 or not. i always get:
-- Executing [0317998...@inputinterior.se:1]
NoOp(SIP/0317998985-0011, 0317998...@inputinterior.se has state
NOT_INUSE) in new stack
So I figure out that I have missed something but cant figure out what. :(
Any ideeas?
Asterisk SVN-trunk-r234256 built by root @ sip on a i686 running Linux on
2009-12-11 11:07:02 UTC
Med vänliga hälsningar
MAGNUS BENNGRD
Direktnr 031-799 89 75
Fältspatsgatan 2
421 30
Västra Frölunda
Tel. 031-799 89 00
Fax 031-799 89 01
www.inputinterior.se [1]
Links:
--
[1] http://www.inputinterior.se
inline: 7___
-- 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] Avaya 950 one-X Deskphone
Hi! Avaya has just released SIP 2.5 which supports 9650 so i did convert one from H.323 to SIP and would like to share what I have to do to get basic stuff working. sip.conf [0317998977] type=friend regexten=0317998977 secret=1234 username=0317998977 callerid=Stefan Andersson mailbox=0317998...@inputinterior.se host=dynamic canreinvite=no disallow=all allow=alaw dtmfmode=rfc2833 46xxsettings.txt SET HTTPSRVR 213.88.138.183 SET HTTPPORT 80 SET DNSSRVR 10.242.10.10 SET DOMAIN inputinterior.se SET SIPDOMAIN inputinterior.se SIP_MODE 0 SET RTCPCONT 1 SET ENABLE_G711A 1 SET SEND_DTMF_TYPE 2 SET SIP_CONTROLLER_LIST sip.inputinterior.se:5060;transport=udp SET ENABLE_CONTACTS 0 Fixed IP on the phone, will look in DHCP later. Comments are most welcome. Med vänliga hälsningar MAGNUS BENNGRD Direktnr 031-799 89 75 Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: 7___ -- 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] ABCTI: first usable beta
Did a quick try, but I am said to say that I lack some setup info. In manager.conf enabled = yes webenabled = yes port = 5038 ... [abcti] secret = secret . read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan write = system,call,agent,user,config,command,reporting,originate I have an extension in sip.conf [0317998985] type=friend regexten=0317998985 secret=secret defaultuser=0317998985 callerid=Cecilia Benngard mailbox=0317998...@inputinterior.se host=dynamic canreinvite=yes nat=no disallow=all allow=alaw Did configure ABCTI with ip/user/pass/extension in asterisk cli: Connected to Asterisk SVN-trunk-r233358 currently running on sip2 (pid = 6317) Verbosity is at least 3 Core debug is at least 1 == Manager 'abcti' logged on from 90.230.92.67 But when I dial 0317998985, the phone rings but no reaction fron ABCTI so I do understand that I have missed something... :( but what? On Sun, 06 Dec 2009 17:22:05 +0100, Oliver Nittka wrote: Hallo, ABCTI (an open-source CTI client for Asterisk) has moved to beta stage. Find it on: http://abcti.sourceforge.net For the first time, we now have windows installers that actually work ;-) We would appreciate any feedback you can give. Regards, -- 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
[asterisk-users] spandsp version
Hi! What version of spandsp is recommended to use when u compile asterisk-trunk? Best regards MAGNUS BENNGRD ___ -- 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] Patch for app_dial.c: exit when just one ext is busy.
Hi! Would be a very nice feature for example the following scenario: Me has 2 phones, one ordinary SIP phone attached to the SIP server and one Cell phone. If someone calls my extension it will ring in both, but if I talk in for example the SIP phone I dont want it to ring on my cell phone. I sure hope it will be implementet. On Tue, 01 Dec 2009 21:35:11 +1100, Rob Hillis wrote: Leif Neland wrote: I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B) return busy when just one extension is busy. Forgive me for the question, but /why/ do you want this behaviour? Isn't the whole point of dialling multiple extensions so that a call has a greater chance of being answered? ___ -- 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] Asterisk - Segmentation fault
Gentlemen,
Forgive me if I am posting at the wrong place!
I was going to test the new chan_ooh323 driver so I did install:
debian: Linux sip2 2.6.26-2-686 #1 SMP
dahdi-linux-complete-2.2.0.2+2.2.0
Asterisk SVN-trunk-r231692
Did enable chan_ooh323, everything compiled without any problems.
Hardware setup:
Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975)
X-Lite can dial MeetMe (955) no problem but when
975 dials X-Lite, I get connectio hear X-Lite ringing but Asterisk dumps:
-- Registered SIP '0317998985' at 10.242.10.209 port 22796
Saved useragent X-Lite release 1103k stamp 53621 for peer 0317998985
-- Executing [...@inputinterior.se:1] Dial(OOH323/avaya-1,
SIP/0317998985) in new stack
== Using UDPTL CoS mark 5
-- Called 0317998985
-- SIP/0317998985-0001 is ringing
Segmentation fault
cat /var/log/messages
Dec 1 12:02:25 sip2 kernel: [13455.390240] asterisk[15013]:
segfault at 0 ip b7edde94 sp b6971170 error 6
in
libc-2.7.so[b7e68000+155000]
Can some guru give me a hint how I should go on?
sip2:/etc/asterisk# cat sip.conf
[general]
context=inputinterior.se
allowoverlap=yes
bindport=5060
bindaddr=10.242.10.122
srvlookup=yes
t38pt_udptl=yes
[0317998985]
type=friend
regexten=0317998985
secret=1234
defaultuser=0317998985
callerid=Cecilia Benngard
mailbox=0317998...@inputinterior.se
host=dynamic
canreinvite=no
nat=yes
disallow=all
allow=alaw
sip2:/etc/asterisk# cat extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[inputinterior.se]
exten = 955,1,Set(CHANNEL(language)=en)
exten = 955,2,MeetMe(955)
exten = 955,3,Hangup()
;
exten = 985,1,Dial(SIP/0317998985)
;
exten = _0X!,1,Dial(OOH323/0${EXTEN}/avaya)
sip2:/etc/asterisk# cat
ooh323.conf
[general]
context=inputinterior.se
bindaddr=10.242.10.122
port=5087
dtmfmode=rfc2833
disallow=all
allow=alaw
[avaya]
type=friend
context=inputinterior.se
ip=10.242.14.11
port=5087
dtmfmode=rfc2833
disallow=all
allow=alaw
Best regards
MAGNUS BENNGRD ___
-- 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] No application 'ReceiveFAX'
Hi! Have probably not understand how fax is working in Asterisk 1.6. I did install: ptlib-v1_12_0 h323plus-v1_19_7 dahdi-linux-complete-2.2.0.2+2.2.0 spandsp-0.0.5 asterisk-1.6.2 asterisk-addons-1.6.2 make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax But core show applications doesnt show me any fax applications and when I try to receive a fax: exten = 960,1,Answer() exten = 960,2,Wait(3) exten = 960,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) [Nov 30 09:29:18] WARNING[19893]: pbx.c:3677 pbx_extension_helper: No application 'ReceiveFAX' for extension (inputinterior.se, 960, 3) Can any guru guide me what I am doing wrong? Best regards, MAGNUS BENNGRD Direktnr 031-799 89 75 Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: 7___ -- 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] No application 'ReceiveFAX' - Solved
Did a recompile of everything, and then it started to work. Must have missed somthing when I did the first compile, or I did something in wrong order. DId a test with a fax machine attached to a POTS interface on an Avaya CM, H.323 trunk to Asterisk. Manage to send from the fax machine to the Asterisk server. Need to work on the T.38 and H.323 in the Asterisk, had to disable T.38 in the Avaya CM to be able to get the fax through. Can keep u posted if u are intrested in how it goes. On Mon, 30 Nov 2009 09:32:14 +0100, Magnus Benngård wrote: Hi! Have probably not understand how fax is working in Asterisk 1.6. I did install: ptlib-v1_12_0 h323plus-v1_19_7 dahdi-linux-complete-2.2.0.2+2.2.0 spandsp-0.0.5 asterisk-1.6.2 asterisk-addons-1.6.2 make menuselect in asterisk-1.6.2 source directory shows: [*] app_fax But core show applications doesnt show me any fax applications and when I try to receive a fax: exten = 960,1,Answer() exten = 960,2,Wait(3) exten = 960,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) [Nov 30 09:29:18] WARNING[19893]: pbx.c:3677 pbx_extension_helper: No application 'ReceiveFAX' for extension (inputinterior.se, 960, 3) Can any guru guide me what I am doing wrong? Best regards, MAGNUS BENNGRD Direktnr 031-799 89 75 Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: 7___ -- 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] Interconnect Asterisk with another PBX
I am doing what u wanna atm but instead of an Alcatlet with SIP support i have to struggle with an Avaya CM without SIP but with H.323. So far putting a trunk over Ethernet with SIP is the way I gonna go. I havent run in to any show-stopper so far with my CM H.323 - Asterisk integration. On Mon, 23 Nov 2009 11:17:22 -0500, Ryan Wagoner wrote: Either use SIP or PRIs to do the integration. FXO and FXS interfaces are a single port, where as a PRI will provide you with 23 channels. Use QSIG signaling over the PRI so Caller ID names will show between the systems. I just integrated a Toshiba CIX with Asterisk due to the cost for SIP licensing and the reliability of the Toshiba VOIP Phones. They were having hardware failures every few months. I went with Sangoma PRI cards using QSIG. Everything has been working great and I have rolled out 12 Snom 370 phones to work with the 150 Toshiba Digital phones. To the end users the experience is seamless as they can 4 digit dial any extension and the call will be routed to the correct system. This does take a bit of duplicate setup on the two systems, but was worth the hassle for the end result. Ryan On Mon, Nov 23, 2009 at 6:17 AM, Alex Balashov wrote: PRI is likely the simplest and most reliable. Xavier Mesquida wrote: Hi, I want to interconnect a Alcatel OmniPCX PBX with SIP support with an Asterisk PBX. My intention is Alcatel PBX manage all external calls and analog extensions and Asterisk manage all the SIP users (because I have to pay for every SIP license in Alcatel PBX and I can't edit configuration or password in that PBX) What's the best way to interconnect the 2 PBX? With SIP, with a FXO interface or FXS? How can I do that? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Prevent Dial if any extension is busy
On Sun, 22 Nov 2009 15:38:00 +0100, Leif Neland wrote: Magnus Benngård
skrev: Hi!
Part of extensions.conf:
exten = 985,1,Dial(SIP/0317998985H323/00702221...@avaya,20)
exten = 985,2,Goto(985-${DIALSTATUS},1)
exten = 985-BUSY,1,VoiceMail(0317998...@inputinterior.se,b [1])
exten = 985-BUSY,2,PlayBack(vm-goodbye)
exten = 985-BUSY,3,HangUp()
exten = 985-NOANSWER,1,VoiceMail(0317998...@inputinterior.se,u [2])
exten = 985-NOANSWER,2,PlayBack(vm-goodbye)
exten = 985-NOANSWER,3,HangUp()
0317998985 is a direct connected SIP phone
0702221448 is a celluar phone.
When dialing 985 both phones rings, perfect
If none answer within 20 seconds, VoiceMail(0317998...@inputinterior.se,u
[3]), perfect
But my problem comes when I speak on 0317998985 and someone calls on 985,
the call
get to my celluar phone and ofc the other way around.
Is there a way to check if any extension is busy and in that case jump to
VoiceMail(0317998...@inputinterior.se,b [4])?
If both
phones were directly connected sip, it could be done.
The problem is that you can't determine if the cellular is busy before
you call it.
If the cell was only called via asterisk, you could set a flag, when
asterisk called extension 985, and clear it, when hanging up, but I guess
the phone is used for call out via regular cell service, and also called
directly on its own number.
You don't own the cell-company, and can setup an API to get the status of
the cell, right? I didn't think so :-)
No i dont own the cell-company but they route the cell-call to my main
Avaya pbx and the Avaya route it back (with a new b-number) so I have
pretty much control over the cell-call.
Just have to route it to my Asterisk and set the flag there, will do some
reading and figure out how.
You could do this:
check if sip is busy, using ChanIsAvail
I am running Asterisk SVN-branch-1.6.2-r230384 so I thougt i can do
something like:
(For checking if I am talking on the SIP phone)
exten =
985,1,GotoIf($[${DEVICE_STATE(SIP/0317998985)}=BUSY]?11)
exten = 985,2,Dial(SIP/0317998985H323/00702221...@avaya,20)
exten = 985,3,Goto(985-${DIALSTATUS},21)
exten = 985,4,HangUp()
exten = 985-BUSY,11,VoiceMail(0317998...@inputinterior.se,b)
exten = 985-BUSY,12,PlayBack(vm-goodbye)
exten = 985-BUSY,13,HangUp()
exten = 985-NOANSWER,21,VoiceMail(0317998...@inputinterior.se,u)
exten = 985-NOANSWER,22,PlayBack(vm-goodbye)
exten = 985-NOANSWER,23,HangUp()
But there is something wrong with the first line, tried INUSE aswell.
When I place a call from 0317998985 and some1 call 985, the call goes to
the cell phone. :(
Can any1 se what I am doing wrong?
If so, go to voicemail.
Else, dial cell, timeout 20 sec
if busy go to voicemail
else dial sip, timeout 20 sec
if not answered. go to voicemail.
But this will give 20 seconds delay before sip rings, and 40 seconds
timeout for the caller before voicemail.
The other option is to modify the source, and add an option to
the
dial-command, to exit if any extension dialled is busy.
After all, this is open source :-)
Leif
Links:
--
[1] mailto:0317998...@inputinterior.se,b
[2] mailto:0317998...@inputinterior.se,u
[3] mailto:0317998...@inputinterior.se,u
[4] mailto:0317998...@inputinterior.se,b
___
-- 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] Mix of Swedish and English voice prompts
Hi! I did installed a Swedish voice prompts package, and added: language=se to [general] section in sip.conf. A SIP endpoint calling a conference get Swedish voice prompts but a call that comes through a H.323 trunk got English voice prompts. :( I did try to add: language=se to [general] section of h323.conf but no luck. Guess i have to add it in some more files but cant figure out what files. Any ideas? Best regards, MAGNUS BENNGRD Direktnr +46-31-799 89 75 Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: 7___ -- 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] Mix of Swedish and English voice prompts
Hi! That did the trick! Thx m8! exten = 959,1,Set(CHANNEL(language)=se) exten = 959,2,MeetMe(959) exten = 959,3,Hangup() On Fri, 20 Nov 2009 09:26:50 -0600, Tilghman Lesher wrote: On Friday 20 November 2009 08:21:05 Magnus Benngård wrote: Hi! I did installed a Swedish voice prompts package, and added: language=se to [general] section in sip.conf. A SIP endpoint calling a conference get Swedish voice prompts but a call that comes through a H.323 trunk got English voice prompts. :( I did try to add: language=se to [general] section of h323.conf but no luck. Guess i have to add it in some more files but cant figure out what files. From a quick look, it appears that chan_h323 does not support setting any language whatsoever. Probably the quickest workaround would be to set the language manually as the first step in your dialplan. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] VeriFone Omni VX-510 Credit Card Machine
Gentlemen, I am trying to find a solution for running a VX-510 over SIP. I know they have a BTB box that u can use for that purpose but it is, at least in Sweden, very expensive. What I would like to do is something like below. VX-510 -- SPA2102 -- Asterisk --H.323 trunk-- Avaya CM -- PSTN Asterisk version 1.6.2 PTlib version 1.12.0 H323Plus version 1.19.7 Running on Debian Lenny The VX.510 dials, get connection with other side but when the VX-510 tries to upgrade itself, the call get disconnected. :( I think I have messed a round with almost all parameters. Do u think it i possible or should i drop it? Any ideas that I can try? Med vänliga hälsningar MAGNUS BENNGRD Direktnr Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: 7___ -- 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] VeriFone Omni VX-510 Credit Card Machine
I know, we can attach something called btb-box (encrypt tcp/ip package) at the vx-510 and run the transactions over ethernet, I have tested it and ofc it works but... Our credit card processor charge us around 15 dollar per month for the btb-box. We need one btb-box per office, we have 20 offices atm and are growing. I will try to hassle around some more, prefer to spend that money on funnier things. Unless it's not possible with your credit card processor, I would recommend switching to the ethernet version of the vx-510--no hassle and faster processing. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax - FROM: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] ON BEHALF OF Magnus Benngård SENT: Sunday, November 15, 2009 7:29 AM TO: asterisk-users@lists.digium.com SUBJECT: [asterisk-users] VeriFone Omni VX-510 Credit Card Machine Gentlemen, I am trying to find a solution for running a VX-510 over SIP. I know they have a BTB box that u can use for that purpose but it is, at least in Sweden, very expensive. What I would like to do is something like below. VX-510 -- SPA2102 -- Asterisk --H.323 trunk-- Avaya CM -- PSTN Asterisk version 1.6.2 PTlib version 1.12.0 H323Plus version 1.19.7 Running on Debian Lenny The VX.510 dials, get connection with other side but when the VX-510 tries to upgrade itself, the call get disconnected. :( I think I have messed a round with almost all parameters. Do u think it i possible or should i drop it? Any ideas that I can try? Med vänliga hälsningar MAGNUS BENNGRD Direktnr Fältspatsgatan 2 421 30 Västra Frölunda Tel. 031-799 89 00 Fax 031-799 89 01 www.inputinterior.se [1] Links: -- [1] http://www.inputinterior.se inline: image001.jpg___ -- 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] IVR reports?
Hi all, I'm struggling with figuring out how to get management information with regard to where users are within a IVR system. Does anyone have any tips on reporting process available on where users are if call to IVR is disconnected or abandoned? Thanks Magnus ___ -- 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] IVR reports?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: 23 October 2009 22:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR reports? On Fri, 2009-10-23 at 22:24 +0100, Magnus Kelly wrote: Hi all, I'm struggling with figuring out how to get management information with regard to where users are within a IVR system. Does anyone have any tips on reporting process available on where users are if call to IVR is disconnected or abandoned? What we do is build a string with all the options a customer has pressed and append it to the userfield in CDR at the end of the call. That way we can follow exactly where the customer was throughout the call by parsing this string. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 [Magnus] Excellent prompt, does this mean it's difficult to get a real time view? In the sense of using ivr as a complex queuing system with hold advertorial and then agents at last filter. By chance do you have a snipe of your dial plan that collects the info? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
Hi, SNOM dosent show the number, it shows user realname. http://wiki.snom.com/wiki/index.php/Settings/user_realname // Magnus Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Cary Fitch Skickat: den 31 augusti 2009 09:06 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: Re: [asterisk-users] Inquiry:How to hide Caller Id A Google of that model showed a discontinued Telstra corded phone. But in any case SNOM and Grandstream phones Do show the number before you pick up the handset. I would suggest you use a Grandstream 286 voip adapter and a standard corded or wireless phone so that the caller doesn't have a display to see. Cary Fitch From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Monday, August 31, 2009 1:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inquiry:How to hide Caller Id Sorry for mis-typing in phone type . Please be informed that the current phone type our subscribers are using is TP6000 ones . On Mon, Aug 31, 2009 at 7:24 AM, Paul Hales pdha...@optusnet.com.aumailto:pdha...@optusnet.com.au wrote: I couldn't find any information on this brand of phone on the internet at all. PaulH hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Looking forward your reply Regards H.Motamedi On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.commailto:li...@venturevoip.com mailto:li...@venturevoip.commailto:li...@venturevoip.com wrote: On 31/08/09 5:24 PM, hadi motamedi wrote: Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . You're not really giving enough information. Who sees the number? Where do they see it? What type of phone? What is a subs? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.nethttp://www.astricon.net/ http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.nethttp://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.nethttp://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pause/Unpause agent based on devstate
Hi, I dont know if this is possible, but I want to pause a queue member if another member are busy in the phone. We have agents that has 2 phones and both are logged in to the same queue. I don't want the second phone to call if the first are in use. Any ideas? Magnus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream SX2000 attended tranfer
Hi all, could anyone share how to perform attended transfers with Asterisk and Grandstream SX2000's - we are able to perform blind transfers with no problem, but attended transfers fail - is it necessary to set two line identities on the phones to be able to do this? Appreciate all input, thanks - Magnus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/ShoreTel REFER support
Hello All, Here's the problem, we have happily set up several Asterisk servers to offer commercial service in the UK, our wholesale SIP termination partner (Magrathea - use SER/CiscoGW to provide us the service on a public IP address) - till now we have used Asterisk to connect clients on private IP's with Asterisk doing the required conversion for SIP/IAX between public and private IP's. The current issue is that we have recently agreed to support ShoreTel PBX's with their new SIP trunk feature, and in staging the first install we have found that certain features (blind transfer) require support for both SIP Refer and Refer Replace - which are not supported by the current VoIP provider SER config. (For some good reasons as they use public IP's) So the challenge is to quickly work out the possibility of either adding a SER setup in-between the ShoreTel PBX and the VoIP provider SER unit or preferably finding a way to use one of our current asterisk servers to provide support for this need. The intent of this setup is to both allow for NAT - E.g. use private IP's for the ShoreTel system and public Ip for the VoIP provider, as well as ensuring that the local Asterisk/SER server supports the required Refer and Refer replace commands to allow the ShoreTel PBX to be able to offer blind transfer support. ShoreTel uses the below call control steps during a transfer with the current architecture: . Blind transfer: A calls B. A puts B on hold. A sends a REFER to B transferring it to C. . Consult transfer: A calls B. A puts B on hold. A calls C. A puts C on hold. A sends REFER to C transferring it to B. ShoreTel architecture uses SIP REFER method for blind transfers and SIP REFER with Replaces header to do consult transfers. This means that since (For NAT reasons) our SIP.conf has two contexts - Sip trunk and ShoreTel trunk both have reinvite=no (also to maintain billing records) the SIP Refer functions are not working as planned or hoped.Or Refer is not supported? My problems are: a) My friend Google has little to offer in exactly which RFC's Asterisk supports (particularly as recently Google does not search correctly the list archives?) - Is the SIP Refer function supported? b) Very short timetable to deliver the working solution - 1 week-Particularly if we have to plunge into adding SER to the mix - Steep learning curve with SER? - as some (most?) of IpTels web site is down? Can any one offer guidance on whether my proposed solution will work and share any tips on problems I should be aware of? If any one is interested in taking this on as an Easter project for minor commercial reward - email me off list (magnus at mcomwifi dot net) If this is the wrong list for this type of thing - Apologies Thanks Magnus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK Caller ID - Asterisk 1.2.5 - TDM4 Card
Has any one found any issues (bug?) with ver 1.2.5 as CallerID generated on an outgoing FXS port (to the handset) fails when UK tones are used, with a message 'Didn't finish Caller-ID spill. Cancelling.' Any tips on getting this running ? Looked at Mantis, but only known bugs seem to relate to XP100 cards - not the TDM card. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail and MS Exchange
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Pajari Sent: Thursday, June 09, 2005 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this being developed for Asterisk and yet it would appear to be a critical component needed to migrate customers used to fully integrated Unified Messaging systems to Asterisk. (a) Has anyone cracked this nut (or started on it)? (b) Anyone interested if we post a bounty? From my perspective, not sure I would want Exchange (Which is difficult enough to manage) to be cluttered up with potentially large voicemail files, I would have thought that most Exchange clients are most likely to be Outlook based, who could use pst Imap (Or pop3 if asterisk could auto forward and then delete voice mail) to retrieve voicemail via email without having to worry about central Exchange issues. Might be my lack of knowledge but this would appear to be able to be written as a mapi outlook add in that could update Asterisk to purge voice mail at the same time the user deletes the local copy? Based on the assumption on Imap? I would support a bounty either way that offered this feature. Magnus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] static database config gui
Hi, Nice work! Is it easy to make it run without mod_perl? Is the source online? best regards, Magnus On 5/26/05, snacktime [EMAIL PROTECTED] wrote: I threw together a web gui for the static database configuration over the last couple of days. I built it using mod perl and the template toolkit. If enough people show an interest in this I'll put up a distribution, although it could take a few days. The interface is as generic as possible so you can throw pretty much any asterisk .conf file in and it works. The interface assumes you already know how to edit the config files. The database schema is the same as on the wiki. I'm working on making it a multi user interface. So that you can have multiple end users with their own copies of the config files all on the same server. The separation will be done through a naming convention that will be applied appropriately. A kind of asterisk virtual hosting. I have a demo setup at the following url: http://catalog1.paymentonline.com/voip/demo/index.html One note on the gui. The numbers on the very left are the order of the statements in the config file. For extensions, when you change the location of an extension priority the system will automatically renumber the order and the dialplan automatically. To insert a new priority in the middle of an extension, use a number with a fraction. When you add, delete, or update the system will automatically renumber everything. For example if you have the following extension: exten = 999,1,Answer exten = 999,2,Dial exten = 999,3,Hangup And you want to insert a new priority after 1, add the new priority as 1.5 which when added would give you something like this: exten = 999,1,Answer exten = 999,2,Ringing exten = 999,3,Dial exten = 999,4,Hangup Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rxfax(spandsp-0.0.2pre18) and HT488
Magnus wrote: Snip What do you think of http://www.soft-switch.org/foip.html now? Imho very good primer that explains situation well, raises question in terms of market opportunity for fax service at Asterisk zap pri location, does anyone have experience of linking Asterisk/Zap to RightFax style service? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why always getting max retries error during idle?
Hi Mike, Probably the same problem i had i while back. The ATA-box dont support message waiting indicatons from asterisk and therefore dont respond to the message, asterisk restries 5 times before giving up with a warning in the log. Iresolved it by removing the mailbox= in sip.conf for that ATA-box. //Magnus From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael StahlSent: Friday, May 13, 2005 5:09 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Why always getting "max retries" error during idle? My home asterisk seems to work- I can call from one internal phone to another. However, just leaving my system idle always generates an error message relating to a NOTIFY. See the log below. Any ideas? Thanks, Mike --MESSAGE FILE- to 172.31.254.106:5065May 13 11:01:28 VERBOSE[716]: Retransmitting #5 (no NAT):NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0Via: SIP/2.0/UDP 172.31.254.2:5060;branch=z9hG4bK6e7b6440From: "asterisk" sip:[EMAIL PROTECTED];tag=as116e0fc1To: sip:[EMAIL PROTECTED]:5065Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: Asterisk PBXEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 42 Messages-Waiting: noVoice-Message: 0/0 to 172.31.254.106:5065May 13 11:01:29 WARNING[716]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)May 13 11:01:32 VERBOSE[716]: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP NOTIFY retries exceeded.
Hello, I get warnings in my asterisk log: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call. I've used sip debugging to figure out the cause. It's my D-link DVG-1120S that don't understand message-summary events that asterisk sends out for MWI indication to the client. Is there any way to disable this in asterisk for this particular client? Tanks in advance, Magnus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 legacy multi PBX integration?
Hi, I have an asterisk server with a quad E1 card, Span 1 to PSTN, and calls flowing fine to and from the PSTN. I would now like to connect a legacy PBX to span 2 of the E1 card, I made a crossover, configured zapta.conf and zaptel.conf. Zttools shows both spans as up and active, but I am now stuck on how to a) Make calls out from legacy PBX to pstn and b) how to receive calls from the PSTN E1 bridged via span 1 onto span 2 and into the legacy pbx. c) let the two switches extensions calls each other. I have reviewed the WiKi and the documents cover the x-over cable the settings for the legacy pbx, but not the steps (config) needed in I guess extensions.conf - If someone could share experience and a sample configuration to achieve this, I would be grateful. Fyi from the 100 or so ddi's on the PSTN E1 I would like to share 50/50 between the switches. Do all 100 ddi's have to be present in both extensions.conf? Regards Magnus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone already pionered outing calling with user selcted background noise?
Hello, We are at the beginning of an asterisk project to be able to have callers call in on premium rate number, (Which funds outgoing call) be presented with ivr menu choice of background noises, then be presented with external dial tone, outgoing dialled digits collected and then dialled and once connected, background music invoked as well as connecting original caller and dialled person. For example, guy's late home, needs an excuse, wants to call home and pretend his flight is/was delayed, thus he would dial in, select airport background music with canned tannoy recording of flight delays and then enter his home telephone number. Or variations of theme. We think it can be done with Asterisk, simple IVR, E1 Zap channels, but thoughts on how would we mix canned selected audio file and outgoing call? Input from anyone's that's tried this welcome. Thanks Magnus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice controlled calling?
Hello all, rumours reach me of a way that the UK incumbent operator is planning to compete with VOIP by offering voice activated dialling, e.g. pick up the handset and through speech dial from your personnel directory, this leads me to wonder if this could be performed with Asterisk and Festival? I have looked in the WiKi and goggled, but can find no information on if this is possible, (particularly with SIP?) hence this question, has anyone achieved this? Intent would be to make is simple for non technical person - E.g. Grandma picks up the phone, does not have to worry about entering any digits and then makes call by voice control - for example call daughter etc. The key here is not to need any human interaction with the phone, other then picking up handset, the rest controlled by voice. Many thanks Magnus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice controlled calling? Pt 2
Hello again, many thanks for the feedback, very interested in Dean's comments that there could be work in progress else where in the world. Are the other features related to how the personnel directory could be viewable/recordable via web interface? This would seem worthwhile, also it would be good to be able to sync somehow with the gsm handsets that have device voice control as well as various PDA's etc. + through further goggling it seems that Orange offer a network version (wildfire) that would seem to show that it can be done using the gsm codec? Apologies for confusing use of Festival, perhaps that's why my research failed, but seems to me that if the incumbent PTT's/RBOCs get to monopolise this type of service it will give them a chance to slow down the move to VOIP, but it could also have impact on weaning people off mobiles (How many people lazy and thus use gsm phone based phone directory as can't be bothered to extract telephone numbers from directory?) Many thanks Magnus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP peer registration interval - SOLUTION
This is what I tryied on last Tuesday. It ran fine until yesterday (4 days) then asterisk stopped re-registering again. A sip reload fixed the problem and asterisk now re-registers happily again. I'm just unsure for how long ... Stefan Gofferje wrote: Stefan Gofferje schrieb: Hi folks, I'm registered with sipgate, a German SIP provider. Configs works fine so far. Trouble is, after a while, it seems, my registration is dropped by sipgate. How do I tell * the interval for * registering with a provider? I suppose, the re-registration interval is to long... I finally found a solution. THe SER of Sipgate seem to dislike being qualified, so setting qualify=no solved this problem. I also set defaultexpirey=60, which is respected by Sipgate's SER and makes re-registration after change of dynamic IP a bit faster and more reliable. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP peer registration interval
Hi, I'm having the same problem with the same Provider running asterisk on an open server that is in a hosted environment. The Problem is definately not the defaultexpirey. I can watch asterisk refreshing the registration every few minutes when using sip debugging. After a while (days) asterisk stops refreshing the registration. sip show registry shows that asterisk thinks it has a valid registration that expires after 120sec, BUT it doesn't do anything about it. After a while the SIP Provider assumes that asterisk is offline My solution is to restart asterisk once a day which is not satisdying but works fine so far. Im using stable 1.0.2. The sipgate is using SER as proxy... greetings, Magnus Robert Webb wrote: I'm registered with sipgate, a German SIP provider. Configs works fine so far. Trouble is, after a while, it seems, my registration is dropped by sipgate. How do I tell * the interval for * registering with a provider? I suppose, the re-registration interval is to long... defaultexpirey=120 :Default length of incoming/outoing registration ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with SIP Registration at PSTN Provider
Hi together, I have a asterisk running on a Debian testing system running flawlessly at least after starting the asterisk. The Server its running on has a fixed IP, no NAT, whatsoever and is reachable all the time. The Firewall has holes on port 5060 and for the RTP-range that asterisk is configured on. In my sip.conf I have a few register=lines and I can receive calls over those accounts. However after a few hours, days whatsoever asterisk is still thinking that those registrations are valid(CLI) but my provider (sipgate.de) sees the asterisk as offline and hence does not forward calls to my asterisk box. I tried playing with the *expirey values in the sip.conf but I have no clue how to test it properly cause it can work fine for days and then stop working ... Sipgate is running SER btw. Is that a known problem ? Has anyone a solution other than restarting asterisk every hour or so ? regards, Magnus Jungsbluth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] External Callforward (Vanity CLI)
Hello all, We have been asked if we can forward (for vanity reasons) one number to another number whilst retaining the original callers Caller ID. For example caller (ie 02027 xxx ) comes in on ISDN pri, and is then auto forwarded to 0208 xxx can the original caller id e.g. 02027 xxx be presented to remote site? We have tried a variety of options, but can not achieve this, checked the wiki, but we are arriving at the conclusion that this is not possible unless the carrier allows complete control on setting CLI, currently they only seem to allow the CLI to be set as one of the DDI number on the PRI. Yet this can be done when you divert on a GSM handset and if I remember correctly on my old office definity pabx. Have I missed something? Can asterisk send Qsig call divert information? Thanks for any and all thoughts - Magnus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: [Asterisk-Users] RE: Q: Can I over-ride the value of caller ID
On Sat, 29 Jan 2005 12:53:11 -0600
-Original Message-
From: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: Q: Can I over-ride the value of
${CALLERIDNAME} ?
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Folks,
Many thanks to Howard Lowndes who helped solve this problem; I ended
up using SetCallerID() instead of SetCIDName() as Howard suggested.
Although
SetCIDName() changed the value correctly, desk set CID displays displayed
either unavailable or out of area on incoming calls from my cell phone.
Here's what I ended up with:
...
exten = s/3125882300,1,SetCallerID(ROB CELL ${CALLERIDNUM})
exten = s/3125882300,2,Goto(100,1)
exten = 100,1,Macro(exten_vm,Zap/1)
...
Cheers,
Rob
Hello all, further to Rob's message, could I ask does zaptel.conf or
Zapata.conf need anything further (for caller ID) to allow the setting of
caller ID, as my below E1 pri debug shows that Asterisk seems to be
correctly sending the necessary Q931 instructions to the carrier (Colt) to
set the caller ID, yet cell phone still shows call as Withheld
Any thoughts? Or is this now an issue that now must be directed to the
carrier? (Note that actual tel number below has had digits blanked for this
post, but prior to blanking digit length correct)
Many thanks for all and any thoughts.
Magnus
pstn*CLI
-- Executing SetCallerID(SIP/amarlaptop-01d5, Magnus
0207---) in new stack
-- Executing Dial(SIP/amarlaptop-01d5, Zap/g1/077||r) in new
stack
-- Making new call for cr 32790
Protocol Discriminator: Q.931 (8) len=53
Call Ref: len= 2 (reference 22/0x16) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
[28 06 4d 61 67 6e 75 73]
Display (len= 6) [ Magnus ]
[6c 0d 21 80 30 32 30 37 31 38 39 33 37 34 33]
Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '0207---' ]
[70 0c a1 30 37 37 31 31 35 39 30 33 31 31]
Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '07711590311' ]
[a1]LI
Sending Complete (len= 1)
-- Called g1/077
Protocol Discriminator: Q.931 (8) len=10
Call Ref: len= 2 (reference 22/0x16) (Terminator)
Message type: CALL PROCEEDING (2)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
-- Processing IE 24 (cs0, Channel Identification)
Protocol Discriminator: Q.931 (8) len=5
Call Ref: len= 2 (reference 22/0x16) (Terminator)
Message type: ALERTING (1)
-- Zap/1-1 is ringing
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slackware zttool
New to list, going through steep learning curve with *, could someone assist with how to install zttool with slackware10, the normal make zttool fails, assume I am missing a dependency? (newt and newt-devel?) Trying to test Q.031 pri span with TE405p card, Google has little clue on Zttool and the * manual has a heading, but no entry? Thanks Magnus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercept HOLD of Snom phones
Yeah, thats what I figured, BUT, if you transfer an incoming call to another internal user, music on hold switches to INTERNAL, and if the 2nd agent does a another transfer, the incoming call gets INTERNAL music. I search for a way to define somewhere in extensions.conf a extension that is used when the call is put on hold, so I can decide by callerid. I tryied the snom Music on Hold Server Option and it seems to work: Define an extension like 1000,1,MusicOnHold(Something) and set [EMAIL PROTECTED] as Music on Hold Server in the snom phone. But I still see in the Asterisk CLI when pressing hold(verbose) -playing Music On Hold (default) -playing Music On Hold (Something) So it triggers twice somehow, but anyway, doesn't seem to cause trouble Nick Barnes wrote: Magnus: I would like to decide using the callerid which music on hold is tobe played: That is: play free music to calls from the outside but play copyrighted music if I put an internal call on hold (i.e. a co-worker). Is this possible ? Yes, and it's easier than intercepting the hold request. Add the following lines to your musiconhold.conf: INTERNAL = mp3:/var/lib/asterisk/mohmp3/internal EXTERNAL = mp3:/var/lib/asterisk/mohmp3/external and put your music into the appropriate directories. In the dial plan, for internal calls insert the line: exten = whatever,whatever,SetMusicOnHold(INTERNAL) and for external calls, insert the line: exten = whatever,whatever,SetMusicOnHold(EXTERNAL) in the appropriate places. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intercept HOLD of Snom phones
Hi, I'm running the 1.0 release of Asterisk an it is working nicely with our snom 105 phones. Hold puts the caller on hold, attended / unattended Transfer works directly with the snom buttons ... I have one question though: what does the snom exactly do to tell the * to put the call on hold (can I intercept this somewhere)? I would like to decide using the callerid which music on hold is tobe played: That is: play free music to calls from the outside but play copyrighted music if I put an internal call on hold (i.e. a co-worker). Is this possible ? regards, Magnus Jungsbluth ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
