[asterisk-users] Two lines for outgoing calls
Dear All,
I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel
2.6.18.
I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm
using below context for dialing out.
[outbound-local]
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9X,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9ZXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9ZXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _90900X,1,Dial(Zap/g1/${EXTEN:1},30m,tTr)
When Zap/1 is busy and I try to call, it will use Zap/2 which is fine
but there is something wrong cause I hear one ring and then a weird
sound like a noise or something and then hangup. I have to reload zaptel
modules and then everything works fine for a while.
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/200-08221590,
Zap/g1/150|30|tTr) in new stack
-- Called g1/150
-- Zap/2-1 answered SIP/200-08221590
-- Hungup 'Zap/2-1'
I even thought that second fxo module is broken so I changed it. No
results.
Any ideas?
Thank you in advance,
Dominik
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[asterisk-users] Two lines for outgoing calls
Dear All,
I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel
2.6.18.
I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm
using below context for dialing out.
[outbound-local]
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9X,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9ZXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9ZXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _90900X,1,Dial(Zap/g1/${EXTEN:1},30m,tTr)
When Zap/1 is busy and I try to call, it will use Zap/2 which is fine
but there is something wrong cause I hear one ring and then a weird
sound like a noise or something and then hangup. I have to reload zaptel
modules and then everything works fine for a while.
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/200-08221590,
Zap/g1/150|30|tTr) in new stack
-- Called g1/150
-- Zap/2-1 answered SIP/200-08221590
-- Hungup 'Zap/2-1'
I even thought that second fxo module is broken so I changed it. No
results.
Any ideas?
Thank you in advance,
Dominik
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Re: [asterisk-users] Two lines for outgoing calls
On Wed, 2007-12-26 at 11:36 -0500, Steve Totaro wrote:
Dominik Zalewski wrote:
Dear All,
I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel
2.6.18.
I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm
using below context for dialing out.
[outbound-local]
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9X,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9ZXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _9ZXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
exten = _90900X,1,Dial(Zap/g1/${EXTEN:1},30m,tTr)
When Zap/1 is busy and I try to call, it will use Zap/2 which is fine
but there is something wrong cause I hear one ring and then a weird
sound like a noise or something and then hangup. I have to reload zaptel
modules and then everything works fine for a while.
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/200-08221590,
Zap/g1/150|30|tTr) in new stack
-- Called g1/150
-- Zap/2-1 answered SIP/200-08221590
-- Hungup 'Zap/2-1'
I even thought that second fxo module is broken so I changed it. No
results.
Any ideas?
Thank you in advance,
Dominik
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Not sure why you would have intermittent problems like that. Is this a
Digium card?
Yes it's TDM400P from Digium.
I once had a four port FXO Digium card and thought that I had a bad FXO
module but it turned out that it was actually the slot on the card
itself. The issue I was experiencing was large amounts of static so
that you could not hear anything.
I changed the slot on the card but no results.
Maybe you can find something helpful in /var/log/messages that
corresponds with the issue.
Nothing.
Anyways, I would contact the manufacturer and see what they can do for you.
Thanks,
Steve Totaro
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[asterisk-users] Dialout Macro and transfer call in progress
Dear All,
I can not transfer call in progress. What's wrong with my macro? I think tT
flags is enough right?
[macro-stdexten]
exten = s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key
exten = s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key
exten = s,3,GotoIf($[${temp} = ]?5); If not existing, goto
priority 5
exten = s,4,Dial(Local/[EMAIL PROTECTED]/n); Unconditional Forward
exten = s,5,GoToIf($[${DNDStatus} = ]?7) ; If not existing ring the
interface
exten = s,6,Voicemail(u${ARG1}); If CFU failed, send to
voicemail w/ unavail announce
exten = s,7,Dial(${ARG2},20,tTrR) ; Ring the interface
exten = s,8,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Set(temp=${DB(CFNA/${ARG1})}) ; Get CFNA key
exten = s-NOANSWER,2,GotoIf($[${temp} = ]?4) ; If not existing, goto
voicemail
exten = s-NOANSWER,3,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on No Answer
exten = s-NOANSWER,4,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,5,Goto(s,5) ; If they press #, return to
ring the interface
exten = s-BUSY,1,Set(temp=${DB(CFB/${ARG1})}) ; Get CFB key
exten = s-BUSY,2,GotoIf($[${temp} = ]?4) ; If not existing, goto
voicemail
exten = s-BUSY,3,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on Busy
exten = s-BUSY,4,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
exten = s-BUSY,5,Goto(s,5) ; If they press #, return to
ring the interface
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer
exten = a,1,VoicemailMain([EMAIL PROTECTED]) ; If they press *, send the
user into VoicemailMai
[from-internal]
exten = 200,1,Macro(stdexten,200,SIP/user200)
...
Thanks in advance,
Dominik___
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Re: [asterisk-users] Call transfer in asterisk
On Monday 02 July 2007 01:45:44 pm satish patel wrote: dear all I am new in asterisk and i have now setup asterik for 40 phone now i want to configure call transfer between phone so how it is possible and what configuration part in asterisk will perfomed for this task give me suggestion for my solution Regards Satish Patel - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://www.voip-info.org/wiki/index.php?page=PBX+CallTransfer -- Dominik Zalewski | System Administrator OpenCraft t- +2 02 336 0003 w- http://www.open-craft.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer outgoing call - macro
Dear All,
I have a problem with call transfer. When I dial a number and then I want to
transfer current call to an extension, I'm getting disconnected. Transfering
incoming call works fine. I'm using macro for dialing.
extensions.conf:
[from-internal]
ignorepat = 9
exten = 200,1,Macro(stdexten,200,SIP/dzalewski)
[macro-stdexten]
exten = s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key
exten = s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key
exten = s,3,GotoIf($[${temp} = ]?5); If not existing, goto
priority 5
exten = s,4,Dial(Local/[EMAIL PROTECTED]/n); Unconditional Forward
exten = s,5,GoToIf($[${DNDStatus} = ]?7) ; If not existing ring the
interface
exten = s,6,Voicemail(u${ARG1}); If CFU failed, send to
voicemail w/ unavail announce
exten = s,7,Dial(${ARG2},20,tTrR) ; Ring the interface
exten = s,8,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Set(temp=${DB(CFNA/${ARG1})}) ; Get CFNA key
exten = s-NOANSWER,2,GotoIf($[${temp} = ]?4) ; If not existing, goto
voicemail
exten = s-NOANSWER,3,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on No Answer
exten = s-NOANSWER,4,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,5,Goto(s,5) ; If they press #, return to
ring the interface
exten = s-BUSY,1,Set(temp=${DB(CFB/${ARG1})}) ; Get CFB key
exten = s-BUSY,2,GotoIf($[${temp} = ]?4) ; If not existing, goto
voicemail
exten = s-BUSY,3,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on Busy
exten = s-BUSY,4,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
exten = s-BUSY,5,Goto(s,5) ; If they press #, return to
ring the interface
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer
exten = a,1,VoicemailMain([EMAIL PROTECTED]) ; If they press *, send the
user into VoicemailMai
features.conf:
[featuremap]
blindxfer = # ; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *2 ; One Touch Record
atxfer = *2; Attended transfer
Thanks in advance,
Dominik___
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Re: [asterisk-users] VPN on Asterisk
On Monday 18 June 2007 03:09:40 pm Biju wrote: Somebody sugested that we can do this with open VPN . 1st Asterisk PBX - install OpenVPN and configure it to run as a server 2nd Asterisk PBX - install OpenVPN and configure it as a client http://openvpn.net/install.html http://openvpn.net/howto.html#examples What OS is running there? But somehow. I couldn't do that. I will be greatful if somebodycan provide more details, Thanks Regards, Biju.V.P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, June 17, 2007 8:25 AM To: Commercial and Business-Oriented Asterisk Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VPN on Asterisk Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. For this first i need to setup VPN on my server .. Am i right? Well if anyone has experience in the whole setup how to make it run, a guide would be much appreciated with some pointer to equipment that are wel suited for the setup. Thanks in advance. Danny ___ --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 -- Dominik Zalewski | System Administrator OpenCraft t- +2 02 336 0003 w- http://www.open-craft.com ___ --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] VPN on Asterisk
On Sunday 17 June 2007 08:25:23 am [EMAIL PROTECTED] wrote: Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. For this first i need to setup VPN on my server .. Am i right? Well if anyone has experience in the whole setup how to make it run, a guide would be much appreciated with some pointer to equipment that are wel suited for the setup. Thanks in advance. Danny ___ --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 I'm in Middle East also and don't have problems with SIP and IAX2:) Try OpenVPN. It's easy to setup and has many features. -- Dominik Zalewski | System Administrator OpenCraft t- +2 02 336 0003 w- http://www.open-craft.com ___ --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] WiFi SIP phones
On Wednesday 23 May 2007 01:01:53 pm Chris Bagnall wrote: Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since then. What models are currently out there people would recommend I look at? Thanks in advance. Regards, Chris I'm using D-Link DPH-541 (g729) with asterisk 1.2.x without any problems. Battery life is poor but the range is acceptable. I tried also Zyxel WiFi SIP phone but I wasn't able to register to my pbx. It restarts everytime. Probably a firmware problem or something. Dominik Zalewski | System Administrator OpenCraft t- +2 02 336 0003 w- http://www.open-craft.com ___ --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] Asterisk 1.2.17 and BRIstuff - SOLVED
Hi All, I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is that current version of BRIstuff is for Asterisk 1.2.14. BRIstuff 0.3.0-PRE-1y (* 1.2.14) If I'm misunderstanding how to apply patches for 1.2.17? Thank you in advance, Dominik I found it:) http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y-e.tar.gz Regards, Dominik ___ --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] Asterisk 1.2.17 and BRIstuff
Hi All, I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is that current version of BRIstuff is for Asterisk 1.2.14. BRIstuff 0.3.0-PRE-1y (* 1.2.14) If I'm misunderstanding how to apply patches for 1.2.17? Thank you in advance, Dominik ___ --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] MP3Player
Hi All, I'm having problem with MP3Player app. I want the caller to hear mp3 when he is waiting until I answer my phone. -- from extentions.conf -- exten = 200,1,Answer() exten = 200,2,MP3Player(/home/user200/mp3/hanna-hais.mp3) exten = 200,3,Dial(SIP/200|20|tTrR) exten = 200,4,Hangup() -- end -- here is debug from CLI: -- Executing Answer(SIP/200-08a64d98, ) in new stack -- Executing MP3Player(SIP/200-08a64d98, /home/user200/mp3/hanna-hais.mp3) in new stack Mar 15 11:25:32 NOTICE[4991]: app_mp3.c:121 timed_read: Poll timed out/errored out with 0 -- Executing Dial(SIP/200-08a64d98, SIP/200|20|tTrR) in new stack -- Called 200 -- SIP/200-08a6a2d8 is ringing Asterisk 1.2.16 and mpg123 installed. Any ideas? Thank you in advance, Dominik ___ --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] Asterisk PPPD with analog lines
Hi All, Is it possible to use asterisk as a internet link backup callback solution? I mean when my main DSL link is down at my server room I would like to dial to asterisk , then it will call back me and provide a connection to a LAN network. Regards, Dominik ___ --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] Call forwarding
Hi All,
I'm using asterisk 1.2.15 and call forwarding doesnt work for me.
from my extensions.conf:
; Unconditional Call Forward
exten = _*21*X.,1,NoCDR
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
exten = _*21*X.,3,Playback(vm-saved)
exten = _*21*X.,4,Hangup
exten = #21#,1,NoCDR
exten = #21#,2,DBdel(CFIM/${CALLERID(NUM)})
exten = #21#,3,Playback(auth-thankyou)
exten = #21#,4,Hangup
debug from asterisk CLI:
-- Executing NoCDR(SIP/dzalewski-081afaf0, ) in new stack
Feb 15 15:00:19 NOTICE[32307]: cdr.c:443 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' not posted
Feb 15 15:00:19 NOTICE[32307]: cdr.c:445 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' lacks end
-- Executing Set(SIP/dzalewski-081afaf0, DB(CFIM/200)=204) in new
stack
-- Executing Playback(SIP/dzalewski-081afaf0, vm-saved) in new stack
-- Playing 'vm-saved' (language 'en')
-- Executing Hangup(SIP/dzalewski-081afaf0, ) in new stack
== Spawn extension (from-internal, *21*204, 4) exited non-zero
on 'SIP/dzalewski-081afaf0'
Thank you in advance,
Dominik
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Re: [asterisk-users] Call forwarding
On Thursday 15 February 2007 03:22:58 pm Stefan Wintermeyer wrote:
Am 15.02.2007 um 14:06 schrieb Dominik Zalewski:
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
Just use ${CALLERID(num)} and not ${CALLERID(NUM)}.
Stefan
it didnt help :( Is there is other way to implement call forwarding?
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Re: [asterisk-users] Call forwarding
On Thursday 15 February 2007 04:00:52 pm Pavel Jezek wrote: you just post only call forward activation part of dialplan, but you must also make dialplan part, that reflect, how is set this callforward mark, ie. if callforward is set, dial that number, if not, dial peer... Do you have any example of this diaplan part? Thanks, Dominik ___ --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] Can Asterisk handle 7000 SIP users?
Hi All, One of my customer asked me if Asterisk can handle 7000 SIP users. They want anyone that have access to wireless hotspot to make voice calls to the office using software phone or SIP cordless phone. Does anybody did such a setup? What are hardware requirements for server and how much bandwidth I will need using comercial codec? Thank you in advance, Dominik ___ --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] Can not compile latest zaptel -1.2.13
I'm trying to compile latest zaptel-1.2.13 and I'm getting following errors: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c: In function ‘debugfs_open’: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c:171: error: ‘struct inode’ has no member named ‘i_private’ make[5]: *** [/usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.o] Error 1 make[4]: *** [/usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp] Error 2 make[3]: *** [_module_/usr/src/Asterisk-1.2.14/zaptel-1.2.13] Error 2 make[2]: *** [modules] Error 2 make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.18.2-34-obj/i386/default' make: *** [all] Error 2 Any ideas? Thank you in advance, Dominik ___ --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] Can not compile latest zaptel -1.2.13 - SOLVED
If you look a number of lines above the function debugfs_open, you'll see: /* * As part of the inode diet the private data member of struct inode * has changed in 2.6.19. However, Fedore Core 6 adopted this change * a bit earlier (2.6.18). If you use vanila kernel (or Debian Etch) * Change the following test from 2,6,18 to 2,6,19. */ #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,18) #define I_PRIVATE(inode)((inode)-u.generic_ip) #else #define I_PRIVATE(inode)((inode)-i_private) #endif Looks like we're gonna break the Fedoras here (Debian Etch is less of an issue, as it doesn't have DEBUGFS enabled by default). Change that line to: #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,19) Does this help? I changed that line and zaptel compiled fine. Thank you:) Dominik ___ --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] Asterisk 1.2.14 and bristuff 0.2.0-RC8s
Hi All, How to install bristuff on asterisk 1.2.14? install scripts are trying to download and compile those versions: asterisk-1.0.10 zaptel-1.0.10 libpri-1.0.9 and I'm running: asterisk-1.2.14 zaptel-1.2.12 libpri-1.2.4 I only need Pickup application from bristuff to be able to pickup channel independent calls e.g. when I have incoming call from PSTN and I would like to answer ringing ext. from any SIP headset. Thank you in advance, Dominik ___ --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] Enhanced PickupChan
Hi All,
I've installed Enhanced PickupChan on asterisk 1.2.14 using howto from
http://www.thorsten-knabe.de/linux/asterisk/pickup.jsp .
from extensions.conf:
exten = 0,1,Dial(SIP/eosoiris|20|tTrR)
exten = 200,1,Dial(SIP/dzalewski|20|tTrR)
exten = _7.,1,Pickup2(${EXTEN:1})
When I try to pickup ringing SIP channel from other IP headset I go
disconnected.
here is debug from asterisk CLI:
-- Executing Dial(SIP/eosoiris-081b5e40, SIP/dzalewski|15|tTrR) in new
stack
-- Called dzalewski
-- SIP/dzalewski-081bb380 is ringing
-- Executing PickUp2(SIP/kitchen-081c08c0, 200) in new stack
find_matching_channel: pattern='200' state=5
find_matching_channel: trying channel='SIP/kitchen-081c08c0' state=4
pattern='200'
find_matching_channel: trying channel='SIP/dzalewski-081bb380'
state=5 pattern='200'
find_matching_channel: trying channel='SIP/eosoiris-081b5e40' state=4
pattern='200'
find_matching_channel: trying channel='Zap/3-1' state=6 pattern='200'
find_matching_channel: trying channel='SIP/developers-0819bca0'
state=6 pattern='200'
== Auto fallthrough, channel 'SIP/kitchen-081c08c0' status is 'UNKNOWN'
Any ideas?
Thank you in advance,
Dominik
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[asterisk-users] Asterisk 1.2.14 bristuff app_pickup.so
Hi All, I'm using Asterisk 1.2.14 with Wildcard TDM400P. I need app_pickup.so application so I can pickup channel-independent calls from any IP Phone headset. How to compile and install only this application from bristufff package? Thank you in advance, Dominik ___ --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] Pickup() ringing extension and call waiting
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would
like to be able to pickup ringing extention from any SIP phone using Pickup()
application.
from my dial plan:
[incoming]
exten = s,1,Dial(SIP/somebody1|60|tTrR)
[internal]
include = outbound-local
include = parkedcalls
exten = 200,1,Dial(SIP/somebody1|20|tTrR)
exten = 201,1,Dial(SIP/somebody2|20|tTrR)
exten = 202,1,Dial(SIP/somebody3|20|tTrR)
exten = _8.,1,Pickup(${EXTEN:1})
[outbound-local]
ignorepat = 9
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9X,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9ZXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
When there is incoming call and extension 200 rings, I press 8200 to pickup a
call and I get disconnected.
here is debug from asterisk CLI:
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1, SIP/somebody1|60|tTrR) in new stack
-- Called somebody1
-- SIP/somebody1-081bea58 is ringing
-- Executing Pickup(SIP/somebody3-081b3cd8, 200) in new stack
== Spawn extension (internal, 8200, 1) exited non-zero
on 'SIP/somebody3-081b3cd8'
Also,
Call waiting seems to not work. While having a conversation I hear beep in
my phone but the M2 (second line) button doesn't blink so I can not pickup
second call and put first one on hold.
from my zapata.conf:
[channels]
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
callerid=asreceived
usecallerid=no
hidecallerid=no
threewaycalling=yes
transfer=yes
callwaiting=yes
cancallforward=yes
;;hanguponpolarityswitch
busydetect=yes
faxdetect=both
group=1
callgroup=1
pickupgroup=1
context=incoming
channel = 3
;;channel = 4 ;; no line yet
Any ideas?
Thank you in advance,
Dominik
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Re: [asterisk-users] Pickup() ringing extension and call waiting
On Monday 29 January 2007 03:20:16 pm Steve Davies wrote:
On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have
Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I
would like to be able to pickup ringing extention from any SIP phone
using Pickup() application.
from my dial plan:
[incoming]
exten = s,1,Dial(SIP/somebody1|60|tTrR)
[internal]
include = outbound-local
include = parkedcalls
exten = 200,1,Dial(SIP/somebody1|20|tTrR)
exten = 201,1,Dial(SIP/somebody2|20|tTrR)
exten = 202,1,Dial(SIP/somebody3|20|tTrR)
exten = _8.,1,Pickup(${EXTEN:1})
[outbound-local]
ignorepat = 9
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9X,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
exten = _9ZXX,1,Dial(Zap/g1/${EXTEN:1}|60|tT)
When there is incoming call and extension 200 rings, I press 8200 to
pickup a call and I get disconnected.
here is debug from asterisk CLI:
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1, SIP/somebody1|60|tTrR) in new stack
-- Called somebody1
-- SIP/somebody1-081bea58 is ringing
-- Executing Pickup(SIP/somebody3-081b3cd8, 200) in new stack
== Spawn extension (internal, 8200, 1) exited non-zero
on 'SIP/somebody3-081b3cd8'
Pickup works on a channel, not on an extension number, so in the above
example you effectively execute
Pickup(200)
but need to have mapped the 200 so that you do
Pickup(SIP/somebody1)
Regards,
Steve
What do you mean by mapping the 200 ?
In this example I can pickup any ringing extension:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
If phone with number 42 rings you can catch the call by dialing 742. You
don't need to use the context
exten = _7.,1,Pickup(${EXTEN:1}) works for all contexts.
Regarding call waiting, internally when I'm having a conversation and someone
calls me, then my second line button blinks and I can pickup a second call
putting first one on hold. Problem just with real call waiting from PSTN.
Thanks,
Dominik
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