[asterisk-users] Query
Hi, Can any body tell me (i) Does digium TE-120P card can be installed on Redhat linux 9i (2.4.20-8) kernel (ii) It is written in documentation that TE120P card be installed only above 2.6.xx . So, which is the best suited one for it( 2.6.15 or 2.6.18 os some other release) (iii) Redhat 9i (2.4.20-8) is installed on my system. I downloaded 2.6.18 kernel. compiled and installed it. After booting through the new one, when I give lsmiod command, it gives the following error lsmod: QM_MODULES: Function not implemented Unable to load iptables module I tried the following way of kernel trap 1. Download the latest version of module-init-tools. 2. ./configure --prefix=/ make make instal 3. Now translate your old /etc/modules.conf into /etc/modprobe.conf with the ./generate-modprobe.conf script that comes with module-init-tools: ./generate-modprobe.conf /etc/modprobe.conf It worked for once. But everyday morning same problem of lsmod comes. I could not find out the way to remove this error of lsmod. Can anybody tell me the way to sort it out. Thanx and regards sanchal ___ --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] Audio going one way for a few seconds during thecall
Two reccomendations: 1) Enable nat for the SIP channels of the phones in SIP.conf. Or 2) If all the remote phones are in the same location, an IPSEC tunnel between the remote router, and your Asterisk machine. Jason. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: Saturday, 23 June 2007 1:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Audio going one way for a few seconds during thecall Hi, This question was posted earlier, but there was no satisfactory answer to it. Afterwards I tried everything but to no avail. The problem of audio going one way during the call for a few seconds is still there. Its Asterisk 1.2.18 hosted Dell server with no NAT. Phones connect remotely through a hi-speed Internet connection, they are behind NAT on a D-Link router, UDP ports 5060, 10001-2 are forwarded to LAN,*, which means they are forwarded to all the IPs. How can I fix this problem. -- Zeeshan A Zakaria ___ --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] IAX client USB phone
Am Samstag, den 23.06.2007, 09:52 -0300 schrieb Ronaldo Z. Afonso: Hi all, Does anybody know any USB phone that I can use as an IAX Client? The USB phones I saw on the market just behave like an additional sound card, with some control buttons perhaps, and those will not work without a software (like twinkle, x-lite...). The devices that work without PC software usually come with a network plug instead of USB - the point is that they work without PC, so why use a USB connection and depend of a PC in that respect? In my experience (and I only had two cheapo Ebay USB-phones) sound works without problems _but_ the keys might work or not, seems to be a bit of luck involved. One of those two phones had the number keys working, but neither hangup nor dial would do anything, the other one would not work at all - except the sound, which was OK. BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan problem
I already noticed the hisax problem, so I removed the module from the modules directory so it cannot be loaded anymore. Are you referring to this driver in specific, or other misdn specific driver. BTW it seems that messages from the list have about 2 days delay, that is why I did not see the message until this morning (monday 26th??) Thx, Hans Feringa On Thu, Jun 21, 2007 at 09:50:03AM +0200, [EMAIL PROTECTED] wrote: Thx, However it appears to be something else. Still need to find out what it is. Loading during boot does not work. After unloading (rmmod) modules mISDN, zaptel, wctdm etc, then reloading them manually in any particular order it works. Have you also unloaded the low-level misdn driver? One usual suspect whose name keeps popping up: hisax. TDM400P seems to use a certain chipset that is also used by some ISDN cards. So if you look at its aliases stirngs: alias: pci:vE159d0001svB1D9sd*bc*sc*i* alias: pci:vE159d0001svB118sd*bc*sc*i* alias: pci:vE159d0001svB119sd*bc*sc*i* alias: pci:vE159d0001svA9FDsd*bc*sc*i* alias: pci:vE159d0001svA8FDsd*bc*sc*i* alias: pci:vE159d0001svA800sd*bc*sc*i* alias: pci:vE159d0001svA801sd*bc*sc*i* alias: pci:vE159d0001svA908sd*bc*sc*i* alias: pci:vE159d0001svA901sd*bc*sc*i* Read: PCI cards with vendor ID E159, product ID 1 and a bunch of more specific sub-vendor IDs. The hisax driver has: alias: pci:vE159d0001sv*sd*bc*sc*i* That is: it is a generic driver that will try to probe all PCI cards with vendor ID E159 and product ID 1. I don't have misdn drivers installed, but you can narrow your search a bit by: grep e159 /lib/modules/`uname -r`/modules.pciids -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 ___ --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 not able to hear calling party ring sound
Dear sir I have setup Avaya with mediant with asterisk [sip_phone]---[ * ]---[mediant]---E1-trunk--[Avaya]---[analog_phone] This is my configuration when i call from SIP phone i got ringing sound of phone but whn i call from analog_phone behind avaya i didn't get ring sound of ring but SIP phone speaker ring why i am not able to hear ring sound from analog phone Regards satish patel - Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. ___ --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] Xorcom Bri 4 Port USB
Hi, I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches. So I'm running zaptel-1.2.17.1 and asterisk-1.2.18. The problem I'm having is that for one I get no LEDs showing if the unit is in TE and NT mode (not a issue for me but may have some impact on things) I have no errors in any logs I can see but once zaptel and asterisk are started up I get a lots of warnings in asterisk such as the following Jun 25 18:18:07 WARNING[2833]: chan_zap.c:2662 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! == Primary D-Channel on span 2 down Jun 25 18:18:07 WARNING[2834]: chan_zap.c:2662 pri_find_dchan: No D-channels available! Using Primary channel 6 as D-channel anyway! == Primary D-Channel on span 3 down Jun 25 18:18:07 WARNING[2835]: chan_zap.c:2662 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! == Primary D-Channel on span 1 down It errors for all for ports and makes no difference if I have the ISDN cables connected or not. I want to run in ptp mode and currently use a digium B410P card on the connections that work fine so I know that the lines work and ptp is the correct mode. Following are my configs. Any pointers you can give would be greatly appreciated. We are running Fedora 7. Kernel Linux 2.6.21-1.3194.fc7 #1 SMP i686 i686 i386 GNU/Linux (Standard Kernel with install) Device has jumpers all set to TE mode. /etc/init.d/zaptel.conf # Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE span=1,1,1,ccs,ami # termtype: te bchan=1-2 dchan=3 # Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE span=2,2,1,ccs,ami # termtype: te bchan=4-5 dchan=6 # Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE span=3,3,1,ccs,ami # termtype: te bchan=7-8 dchan=9 # Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED #span=4,4,1,ccs,ami # termtype: te #bchan=10-11 #dchan=12 # Global data loadzone= au defaultzone = au /etc/asterisk/zapata.conf [channels] ; echocancel = yes ; transfer = yes ; threewaycalling = yes #include zapata-channels.conf /etc/asterisk/zapata-channels.conf ; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE callerid=asreceived group=0 context=from-pstn switchtype = euroisdn signalling = bri_cpe channel = 1-2 callerid= group= context=default ; Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE callerid=asreceived group=0 context=from-pstn switchtype = euroisdn signalling = bri_cpe channel = 4-5 callerid= group= context=default ; Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE callerid=asreceived group=0 context=from-pstn switchtype = euroisdn signalling = bri_cpe channel = 7-8 callerid= group= context=default ; Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED ;callerid=asreceived ;group=0 ;context=from-pstn ;switchtype = euroisdn ;signalling = bri_cpe ;channel = 10-11 ;callerid= ;group= ;context=default ___ --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] g729 problem
Hi iam using asterisk 1.2 version I have purchased g729 license from Digium when iam making calls, iam getting this error ? Jun 25 14:41:45 NOTICE[4424]: frame.c:183 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end any help ram ___ --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] Best wifi IP phone for asterisk
Siemens GigaSet SL75 On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the only telco's get documentation crap) Does anyone have a suggestion? Thanks, MD ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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] Forcing Dial application to skip if called server is unreachable
In fact, Dial() doesn't return instantly like it should, in the case it
is used with ENUM. Dial application using the ENUMLOOKUP function
doesn't skip to the next priority like it was expected, if destination
server doesn't answer to the INVITE messages sent by our server.
For example, in the following code, if the first Dial using ENUM fails
to reach the contact's server, instead of skipping to the next priority
Dialing Zap channel instead, Asterisk keeps sending INVITE messages to
the destination server published in ENUM until dial timeout expires
(120), and only then jumps to the next priority, Dialing Zap:
exten = _X.,1,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0)
exten = _X.,2,GotoIf($[${counter}${sipcount}]?3:6)
exten = _X.,3,Set(counter=$[${counter}+1])
exten = _X.,4,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter})})
exten = _X.,5,GotoIf($[${counter}${sipcount}]?3:6)
exten = _X.,6,Dial(Zap/g1/${EXTEN})
Is this an Asterisk BUG or is it there some way I can solve this problem?
Regards,
Ricardo.
Alex Balashov wrote:
On Wed, 20 Jun 2007, [EMAIL PROTECTED] wrote:
Is it possible to force the Dial function to skip to the next priority if it
doesn't find the server of the called contact within a few seconds?
I know I can use:
Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL])
where I can use some short timeout in the timeout option, but if I do so,
when some call is well succeeded, it will only ring for that time!
I think you basically have to pick one or the other. Either set a long
timeout (15-30 sec, e.g. Dial(SIP/whatever,20) or don't use this feature.
The good news is that if the destination SIP server is actually
unreachable, Dial() should return almost instantly, at which point it
should jump to the failure priority.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
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--
---
Ricardo Carvalho
ITEC / IRICUP / Reitoria UP
tel: +351220408108
sip:[EMAIL PROTECTED]
[EMAIL PROTECTED]
---
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Re: [asterisk-users] Xorcom Bri 4 Port USB
Hi
On Mon, Jun 25, 2007 at 06:38:37PM +1000, Nathan Dennis wrote:
Hi,
I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled
asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches.
So I'm running zaptel-1.2.17.1 and asterisk-1.2.18.
The problem I'm having is that for one I get no LEDs showing if the unit
is in TE and NT mode (not a issue for me but may have some impact on
things) I have no errors in any logs I can see but once zaptel and
asterisk are started up I get a lots of warnings in asterisk such as
the following
What is the output of:
modinfo xpp | grep version
if this is something of the sort of 'r3495' then you indeed have an
older version of the driver where BRI support has not been matuire
enough and specifically leds display was not as it is today. In current
version (e.g: the one in zaptel 1.2.18/1.4.3) you will always see an
orange LED for NT or green led for TE on the port.
Please get the version of bristuff from:
http://updates.xorcom.com/astribank/bristuff/
http://updates.xorcom.com/astribank/bristuff/bristuff-0.3.0-PRE-1y-g-xr1.tar.gz
At least until we see a new version of bristuff.
and also see:
http://updates.xorcom.com/astribank/bristuff/INSTALL.html
Also, for the sake of those who will see the messages in a search:
Jun 25 18:18:07 WARNING[2833]: chan_zap.c:2662 pri_find_dchan: No D-channels
available! Using Primary channel 3 as D-channel anyway!
== Primary D-Channel on span 2 down
Jun 25 18:18:07 WARNING[2834]: chan_zap.c:2662 pri_find_dchan: No D-channels
available! Using Primary channel 6 as D-channel anyway!
== Primary D-Channel on span 3 down
Jun 25 18:18:07 WARNING[2835]: chan_zap.c:2662 pri_find_dchan: No D-channels
available! Using Primary channel 9 as D-channel anyway!
== Primary D-Channel on span 1 down
This message comes from chan_zap when a span is down. If a span has
pri_{cpe,net} signalling or bri_{cpe,net} signalling (bristuff BRI ptp)
then you'll get those messages for spans that are down. If the
signalling is bri_{cpe,net}_ptmp they'll be debug messages.
It errors for all for ports and makes no difference if I have the
ISDN cables connected or not. I want to run in ptp mode and
currently use a digium B410P card on the connections that work fine
so I know that the lines work and ptp is the correct mode.
Following are my configs. Any pointers you can give would be greatly
appreciated.
We are running Fedora 7.
Kernel Linux 2.6.21-1.3194.fc7 #1 SMP i686 i686 i386 GNU/Linux (Standard
Kernel with install)
Device has jumpers all set to TE mode.
/etc/init.d/zaptel.conf
# Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
span=1,1,1,ccs,ami
# termtype: te
bchan=1-2
dchan=3
# Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
span=2,2,1,ccs,ami
# termtype: te
bchan=4-5
dchan=6
# Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
span=3,3,1,ccs,ami
# termtype: te
bchan=7-8
dchan=9
# Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED
#span=4,4,1,ccs,ami
# termtype: te
#bchan=10-11
#dchan=12
# Global data
loadzone= au
defaultzone = au
/etc/asterisk/zapata.conf
[channels]
; echocancel = yes
; transfer = yes
; threewaycalling = yes
#include zapata-channels.conf
/etc/asterisk/zapata-channels.conf
; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
callerid=
group=
context=default
; Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 4-5
callerid=
group=
context=default
; Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 7-8
callerid=
group=
context=default
; Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED
;callerid=asreceived
;group=0
;context=from-pstn
;switchtype = euroisdn
;signalling = bri_cpe
;channel = 10-11
;callerid=
;group=
;context=default
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Query
On Mon, Jun 25, 2007 at 12:10:04PM +0530, [EMAIL PROTECTED] wrote: Hi, Can any body tell me (i) Does digium TE-120P card can be installed on Redhat linux 9i (2.4.20-8) kernel Why do you keep starting a new thread and not bother following up to answers in existing threads? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outging select
Hello everybody.
I have a analog line in the office and a ISDN (with mISDN) line.
I want to call outside from the analog line, but when this is busy, I want
to call outside the second call from the ISDN line.
That my extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,tm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,tm)
exten = 102,2,Hangup
exten = 103,1,Dial(SIP/103,30,tm)
exten = 103,2,Hangup
exten = 104,1,Dial(SIP/104,30,tm)
exten = 104,2,Hangup
include = outgoing_RTB
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/103,30,tm)
exten = fax,1,Dial(IAX2/200)
[incoming]
exten = 943712666,1,Wait(2)
exten = 943712666,2,Answer()
exten = 943712666,3,Background(/home/lazkano/bienvenido)
exten = 943712666,4,Wait(1)
exten = 943712666,5,Background(/home/lazkano/extension)
exten = 943712666,6,Wait(4)
exten = 943712666,7,Dial(SIP/104|30|tm)
exten = 943712666,8,Hangup()
exten = 101,1,Dial(SIP/101|30|tm)
exten = 102,1,Dial(SIP/102|30|tm)
exten = 103,1,Dial(SIP/103|30|tm)
exten = 104,1,Dial(SIP/104|30|tm)
How can I do that?
Thanks for all.
Have a nice day.
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Re: [asterisk-users] ChanSkype
Hi Kyle, You need to set up a inbound route from DID=skype1 and tell him where to finish. Something like: exten = skype1,1,Set(FROM_DID=skype1) exten = skype1,n,Goto(ext-local,1000,1) Hope it helps. Best Regards, Hugo Picão Link Consulting - RedesSegurança Tel: 213 100 182 Av. Duque de Ávila, 23 1000-138 Lisboa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Vorster Sent: quinta-feira, 21 de Junho de 2007 14:07 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSkype Hello, I recently installed chanskype on my asterisk box and it works like a dream, can phone out. But no idea how to setup the incoming calls, every time I phone my skype name it just connects and disconnect the call right away. I get the following on asterisk -rvv Verbosity was 1 and is now 14 == Sent cmd 'GET CALL 175 TYPE' to fd 18 on Skype dev 'skype1' == Sent cmd 'GET CALL 175 PARTNER_HANDLE' to fd 18 on Skype dev 'skype1' == Sent cmd 'GET CALL 175 PSTN_NUMBER' to fd 18 on Skype dev 'skype1' == Sent cmd 'GET CALL 175 STATUS' to fd 18 on Skype dev 'skype1' == Sent cmd 'ALTER CALL 175 END HANGUP' to fd 18 on Skype dev 'skype1' == Unknown event 'ALTER CALL 175 END HANGUP' from Skype device 'skype1' == Sent cmd 'GET CALL 175 STATUS' to fd 18 on Skype dev 'skype1' Any one got some advice ? Kind Regards, Kyle Vorster ___ --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 ___ --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 ___ --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] Hi ability solution
Hi all, On one of our client, I must to install an asterisk over a hi ability cluster. I have no experience with clusters an linux neither asterisk. Someone has installed an asterisk in a hi-ability enbviroment? How do you install the cluster? Witch solution did you use? Witch is the best cluster solution to use with asterisk? Thanks in advance, Voipcrazy ___ --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] Best wifi IP phone for asterisk
MM == Marco Mouta [EMAIL PROTECTED] writes: MM Siemens GigaSet SL75 The SL75 is DECT, not Wifi. Apart from that, was it really necessary to quote 20 lines and add a ridiculous 15 line disclaimer telling me that I'm not allowed to read the message? /Benny ___ --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] Hi ability solution
voip crazy wrote: Hi all, On one of our client, I must to install an asterisk over a hi ability cluster. I have no experience with clusters an linux neither asterisk. Someone has installed an asterisk in a hi-ability enbviroment? How do you install the cluster? Witch solution did you use? Witch is the best cluster solution to use with asterisk? Thanks in advance, Voipcrazy Do you mean High Ability, or High Availability I think Rocks is pretty good but I just started playing with it. I think it is more of a High Ability thing. http://www.rocksclusters.org/wordpress/ Thanks, Steve Totaro ___ --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] g729 problem
Disable Voice Activity Detection ram wrote: Hi iam using asterisk 1.2 version I have purchased g729 license from Digium when iam making calls, iam getting this error ? Jun 25 14:41:45 NOTICE[4424]: frame.c:183 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end any help ram ___ --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] outging select
KO, thank you very much.
i will try it.
2007/6/25, Steve Totaro [EMAIL PROTECTED]:
You could combine your two contexts or use goto.
Instead of:
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
Do:
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Goto(outgoing_RDSI,_9,1)
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
Or
[outgoing_trunks]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,3,Hangup()
Thanks,
Steve Totaro
Josu Lazkano wrote:
Hello everybody.
I have a analog line in the office and a ISDN (with mISDN) line.
I want to call outside from the analog line, but when this is busy, I
want to call outside the second call from the ISDN line.
That my extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,tm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,tm)
exten = 102,2,Hangup
exten = 103,1,Dial(SIP/103,30,tm)
exten = 103,2,Hangup
exten = 104,1,Dial(SIP/104,30,tm)
exten = 104,2,Hangup
include = outgoing_RTB
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/103,30,tm)
exten = fax,1,Dial(IAX2/200)
[incoming]
exten = 943712666,1,Wait(2)
exten = 943712666,2,Answer()
exten = 943712666,3,Background(/home/lazkano/bienvenido)
exten = 943712666,4,Wait(1)
exten = 943712666,5,Background(/home/lazkano/extension)
exten = 943712666,6,Wait(4)
exten = 943712666,7,Dial(SIP/104|30|tm)
exten = 943712666,8,Hangup()
exten = 101,1,Dial(SIP/101|30|tm)
exten = 102,1,Dial(SIP/102|30|tm)
exten = 103,1,Dial(SIP/103|30|tm)
exten = 104,1,Dial(SIP/104|30|tm)
How can I do that?
Thanks for all.
Have a nice day.
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Re: [asterisk-users] Hi ability solution
I would say High Availability, sorry for my english. Any High availiability solution for asterisk? VoipCrazy 2007/6/25, Steve Totaro [EMAIL PROTECTED]: voip crazy wrote: Hi all, On one of our client, I must to install an asterisk over a hi ability cluster. I have no experience with clusters an linux neither asterisk. Someone has installed an asterisk in a hi-ability enbviroment? How do you install the cluster? Witch solution did you use? Witch is the best cluster solution to use with asterisk? Thanks in advance, Voipcrazy Do you mean High Ability, or High Availability I think Rocks is pretty good but I just started playing with it. I think it is more of a High Ability thing. http://www.rocksclusters.org/wordpress/ Thanks, Steve Totaro ___ --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] Ring/Off-hook in strange state 6
I don't think my cards are bad, but maybe there is a problem with the one. It has been two weeks since I put my ticket in with Digium...and still no word. I am starting to get frustrated. On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is 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 ___ --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] Rining 180 and 183
Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-[mediant 2000][Avaya] when i call from avaya side to --- asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of this type of problem regards Satish patel - Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. ___ --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] outging select
You could combine your two contexts or use goto.
Instead of:
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
Do:
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Goto(outgoing_RDSI,_9,1)
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
Or
[outgoing_trunks]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,3,Hangup()
Thanks,
Steve Totaro
Josu Lazkano wrote:
Hello everybody.
I have a analog line in the office and a ISDN (with mISDN) line.
I want to call outside from the analog line, but when this is busy, I
want to call outside the second call from the ISDN line.
That my extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten = 101,1,Dial(SIP/101,30,tm)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/102,30,tm)
exten = 102,2,Hangup
exten = 103,1,Dial(SIP/103,30,tm)
exten = 103,2,Hangup
exten = 104,1,Dial(SIP/104,30,tm)
exten = 104,2,Hangup
include = outgoing_RTB
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/103,30,tm)
exten = fax,1,Dial(IAX2/200)
[incoming]
exten = 943712666,1,Wait(2)
exten = 943712666,2,Answer()
exten = 943712666,3,Background(/home/lazkano/bienvenido)
exten = 943712666,4,Wait(1)
exten = 943712666,5,Background(/home/lazkano/extension)
exten = 943712666,6,Wait(4)
exten = 943712666,7,Dial(SIP/104|30|tm)
exten = 943712666,8,Hangup()
exten = 101,1,Dial(SIP/101|30|tm)
exten = 102,1,Dial(SIP/102|30|tm)
exten = 103,1,Dial(SIP/103|30|tm)
exten = 104,1,Dial(SIP/104|30|tm)
How can I do that?
Thanks for all.
Have a nice day.
___
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Re: [asterisk-users] g729 problem
On 6/25/07, Karl J. Vesterling [EMAIL PROTECTED] wrote: Disable Voice Activity Detection yes i have disabled at my eyebeam, still i see this error iam using 1.2.18 ram ___ --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] inband DTMF for g729
- Original Message - From: Darrick Hartman (lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, June 24, 2007 11:25 AM Subject: Re: [asterisk-users] inband DTMF for g729 Gang Chen wrote: On 6/22/07, Gary Chen [EMAIL PROTECTED] wrote: We are using Level 3. At this point, changing carrier is not an option. Gary, I use Level(3) with G729a and RFC2833. No problems, no requirement for inband G729. -- Kristian Kielhofner I can connect to Asterisk IVR using a SIP phone and send RFC2833 with g729. It works fine. But when test call from PSTN to Asterisk, if I set dtmf=auto with g729, I got warning saying something like * does not support inband for g729 and sutomaticlly switch to rfc2833. If I set dtmf=g729, there is no warning but I have the same problem. This tells me that Level3 does use inband for g729 or maybe I am doing something wrong . Gary Gary, I'll restate what Kristian just said above. You do NOT need inband for Level 3. Set dtmf=RFC2833. Do you have the correct g729 codec licenses installed? This may be more of a transcoding issue than anything else. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com We have not yet purchase the g729 codec licenses. I want to test it out first before we buy any license. I download g729 from Internet. I did set dtmfmode=rfc2833. It worked if I use an SIP phone connect to Asterisk using g729 and send dtmf tone using rfc2833. But not from PSTN through Level 3 . Gary ___ --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] Hi ability solution
There is a whole wiki page on the subject. Google is your friend. http://www.google.com/search?q=high+availability+asteriskstart=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official This is what I am currently playing with: http://linux-ha.org voip crazy wrote: I would say High Availability, sorry for my english. Any High availiability solution for asterisk? VoipCrazy 2007/6/25, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: voip crazy wrote: Hi all, On one of our client, I must to install an asterisk over a hi ability cluster. I have no experience with clusters an linux neither asterisk. Someone has installed an asterisk in a hi-ability enbviroment? How do you install the cluster? Witch solution did you use? Witch is the best cluster solution to use with asterisk? Thanks in advance, Voipcrazy Do you mean High Ability, or High Availability I think Rocks is pretty good but I just started playing with it. I think it is more of a High Ability thing. http://www.rocksclusters.org/wordpress/ Thanks, Steve Totaro ___ --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] Hi ability solution
Any High availiability solution for asterisk? VoipCrazy http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf ___ --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] Best wifi IP phone for asterisk
Benny Amorsen schrieb: MM Siemens GigaSet SL75 The SL75 is DECT, not Wifi. Apart from that, was it really necessary to quote 20 lines and add a ridiculous 15 line disclaimer telling me that I'm not allowed to read the message? There is a GigaSet SL75 WLAN. http://gigaset.siemens.com/shc/0,1935,hq_en_0_122755_rArNrNrNrN,00.html Hmm, I did not see any DECT SL75.. Marcus ___ --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] Nokia N95 + Dial Plan
Thanks Benny...
Let me give it a try...
Cheers,
Nitesh
Benny Amorsen wrote:
ND == Nitesh Divecha [EMAIL PROTECTED] writes:
ND Hello All, Recently I added some Nokia N95 customers and it worked
ND pretty good. Now the customers are complaining about the dialing
ND rules... They are used to dialing +12486543210 and +4479XX for
ND long distance calls.
ND Is there anyway to create a + sign dial plan which will allow
ND them to dial a number with + sign.
You have your standard dialplan, usually something like:
exten = _X.,1,...
You just put this in:
exten = _+.,1,Goto(00${EXTEN:1},1)
And poof, all numbers starting with + get it replaced with 00.
You may need to do more munching of course, and 00 may not be right
for you.
/Benny
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Re: [asterisk-users] asterisk-users Digest, Vol 35, Issue 89
Greetings! Due to high workload, I am currently checking and responding to e-mail twice daily at 12:00 PM EST and 9:00PM EST. If you require urgent assistance (please ensure it is urgent) that cannot wait until either 12:00 PM or 9:00 PM, please contact me via phone at: 305-338-3867. Thank you for understanding this move to more efficiency and effectiveness. It helps me accomplish more to serve you better. Sincerely, Harold Riley ___ --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] Rining 180 and 183
On 6/25/07, satish patel [EMAIL PROTECTED] wrote: I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya I'm assuming that you're talking SIP here... typically, if Asterisk receives a 180 response (without SDP), it will pass that on to the end device, which will generate the ringback tone itself. On the other hand, if Asterisk receives a 183 with SDP (or a 180 with SDP, which is less common), it will try to bridge the end device with the media specified in the SDP message. Your best bet would be to look at the SIP messaging, specifically looking to see if there is SDP specified. If SDP is specified in the 183 or 180 message, then I'd try to figure out why Asterisk can't connect to that IP address and port. (In my experience, nine times out of ten this is a NAT firewall problem.) -Jared ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.5
Turn off debug From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Friday, June 22, 2007 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: [asterisk-users] 1.4.5 I am seeing a peculiar message on my console screen on my new installation of Asterisk 1.4.5I would appreciate any comments. Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED] Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS ___ --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] Call Path Optimization
Yes, it does. --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 23, 2007, at 7:48 PM, Ronaldo Z. Afonso wrote: Hi all, I was reading an IAX RFC, or a kind of, and it mentioned something about Call Path Optimization. Does Asterisk provide such a feature? Thanks. Ronaldo. ___ --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] Best wifi IP phone for asterisk
I can't find reference to TFTP for provisioning - does this phone support it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcus Franke Sent: Monday, June 25, 2007 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best wifi IP phone for asterisk Benny Amorsen schrieb: MM Siemens GigaSet SL75 The SL75 is DECT, not Wifi. Apart from that, was it really necessary to quote 20 lines and add a ridiculous 15 line disclaimer telling me that I'm not allowed to read the message? There is a GigaSet SL75 WLAN. http://gigaset.siemens.com/shc/0,1935,hq_en_0_122755_rArNrNrNrN,00.html Hmm, I did not see any DECT SL75.. Marcus ___ --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] international numbers...
This is the required dial plan: 0+61|XXX. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall Sent: Friday, June 22, 2007 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] international numbers... Using trixbox (or a custom dialplan if needed) has anyone been able to convert a number dialled like +61242110 to something like 02422110 ie (remove the +61 and replace with 0) i just dont know how to set it up, there seems to be no dialplan wildcard i can use to match +. I was thinking of something like .61XX but that still seems wrong to me. it could match other numbers. anyone had to do this in the past ? thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 35, Issue 89
Dude you got to be freaking kidding me - are you really sending this email to everyone who posts on the Asterisk list? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 25 June 2007 10:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users]asterisk-users Digest, Vol 35, Issue 89 Greetings! Due to high workload, I am currently checking and responding to e-mail twice daily at 12:00 PM EST and 9:00PM EST. If you require urgent assistance (please ensure it is urgent) that cannot wait until either 12:00 PM or 9:00 PM, please contact me via phone at: 305-338-3867. Thank you for understanding this move to more efficiency and effectiveness. It helps me accomplish more to serve you better. Sincerely, Harold Riley ___ --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] two channels, each dropping into the same context, different behavior.
So, incoming calls on zap work just as I expect them - an intro is played, the
caller hits 1 for sale 2 for support or dials an extension. I'm using the
privacy option for all extensions. When calls come in from zap, they caller
is played the priv-recordintro recording, they say their name, and everything
happens normally from there on out. However, when the call comes in from sip
and id dropped into the same context, they never hear priv-recordintro, and
the callee gets the unknown caller message, but from there the call can
proceed as normal.
I'm not getting any error messages. Below is the output of asterisk - -c
-U asterisk, and the relevant output of show dialplan.
Note that the sip calls come in on extension 666.
Thanks much in advance.
Ryan
===
show dialplan:
[ Context 'in-from-7869101' created by 'pbx_config' ]
'1' =1. Dial(SIP/lesnet_peer/1218348|30|ptT) [pbx_config]
2. VoiceMail([EMAIL PROTECTED])
[pbx_config]
3. Playback(vm-goodbye) [pbx_config]
4. Hangup() [pbx_config]
'101' = 1. Dial(SIP/lesnet_peer/1218348|30|ptT) [pbx_config]
2. VoiceMail([EMAIL PROTECTED]) [pbx_config]
3. Playback(vm-goodbye) [pbx_config]
4. Hangup() [pbx_config]
'102' = 1. Dial(SIP/lesnet_peer/1218390|30|ptT) [pbx_config]
2. VoiceMail([EMAIL PROTECTED]) [pbx_config]
3. Playback(vm-goodbye) [pbx_config]
4. Hangup() [pbx_config]
'103' = 1. Dial(SIP/lesnet_peer/1218348|30|ptT) [pbx_config]
2. VoiceMail([EMAIL PROTECTED]) [pbx_config]
3. Playback(vm-goodbye) [pbx_config]
4. Hangup() [pbx_config]
'104' = 1. Dial(SIP/lesnet_peer/1218428|30|ptT) [pbx_config]
2. VoiceMail([EMAIL PROTECTED]) [pbx_config]
3. Playback(vm-goodbye) [pbx_config]
4. Hangup() [pbx_config]
'105' = 1. Dial(SIP/lesnet_peer/1218348|30|ptT) [pbx_config]
2. VoiceMail([EMAIL PROTECTED]) [pbx_config]
3. Playback(vm-goodbye) [pbx_config]
4. Hangup() [pbx_config]
'2' =1. Dial(SIP/lesnet_peer/1218348|30|ptT) [pbx_config]
2. VoiceMail([EMAIL PROTECTED])
[pbx_config]
3. Playback(vm-goodbye) [pbx_config]
4. Hangup() [pbx_config]
'666' = 1. Answer() [pbx_config]
2. Goto(s|2) [pbx_config]
's' =1. Answer() [pbx_config]
2. Set(CALLERID(all)=${CALLERID(all)})[pbx_config]
3. Background(sbb-greets) [pbx_config]
4. Goto(s|1) [pbx_config]
't' =1. Goto(s|1) [pbx_config]
===
incoming call on zap works fine:
-- Starting simple switch on 'Zap/4-1'
[Jun 25 10:09:36] NOTICE[31818]: chan_zap.c:6351 ss_thread: Got event 18 (Ring
Begin)...
[Jun 25 10:09:39] NOTICE[31818]: chan_zap.c:6351 ss_thread: Got event 2
(Ring/Answered)...
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(Zap/4-1, CALLERID(all)= )
in new stack
-- Executing [EMAIL PROTECTED]:3] BackGround(Zap/4-1, sbb-greets) in
new stack
-- Zap/4-1 Playing 'sbb-greets' (language 'en')
== CDR updated on Zap/4-1
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1,
SIP/lesnet_peer/12183486879|30|ptT) in new stack
-- Privacy-- callerid is empty
-- Zap/4-1 Playing 'priv-recordintro' (language 'en')
-- Zap/4-1 Playing 'beep' (language 'en')
-- x=0, open writing: priv-callerintros/NOCALLERID_2Zap=4-1 format: gsm,
0x81d9920
-- Recording automatically stopped after a silence of 2 seconds
-- Zap/4-1 Playing 'auth-thankyou' (language 'en')
-- Zap/4-1 Playing 'vm-dialout' (language 'en')
-- Called lesnet_peer/12183486879
-- SIP/lesnet_peer-081d61a0 is ringing
-- Call on
[asterisk-users] callback and bridge problem
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts).
i've been practicing with callback for a while, but i'm at a dead end.
I hope somebody can help me to move on.
i have troubles getting two calls bridged together. Scenario is the
following:
- asterisk calls my cell via a SIP provider called neophone
- my cell rings, i pick up, and i find myself in:
[internal]
; callback is directed here
exten = s,1,WaitExten,50
include = voicemail-context
include = internal_extensions-context
include = dialout_prefix-context
because my call file looks like this:
Channel: SIP/[EMAIL PROTECTED]
Context: internal
Extension: s
Priority: 1
where 0620222 is my cell.
- after picking up, i dial 9520630111 where 952 is the dialing
prefix, 0630... is another cell. 952 is a prefix for another
registered account at the same provider (one account is allowed to place
one call at a time).
After this as you can see, the second number (..) is dialed.
However when i pick up the phone, the call hangs up.
This also happens when i use another prefix (another provider, even
PSTN) for the second call too.
The relevant part from asterisk console is at the end of this e-mail, i
don't really understand the warning messages.
- configs:
In sip.conf, the configuration for the two SIP accounts are:
register = 0621380:[EMAIL PROTECTED]
register = 0621381:[EMAIL PROTECTED]
[neophonex]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621380
authname=0621380
fromuser=0621380
secret=password
callerid=0621380
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no
[neophonex-out]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621381
authname=0621381
fromuser=0621381
secret=password
callerid=0621381
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no
extension.conf:
exten = _952.,1,Playback(kapcsolas,noanswer)
exten = _952.,n,Set(CALLERID(name)=0621380)
exten = _952.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
I have tried every possible setting i know about, but still, when i call
outside, via 'turning around' in asterisk, both cells hung up when
answering the call. I have tried calling a regular landline phone
number but still hanging up.
Both accounts are valid, registered and have enough credit to dial
outside its voice network.
The only way the call does not hung up is when i dial extensions within
asterisk.
The asterisk log:
-- Called [EMAIL PROTECTED]
-- Call on SIP/neophonex-out-081a9cc0 left from hold
-- SIP/neophonex-out-081a9cc0 is making progress passing it to
SIP/neophonex-081ab240
[Jun 25 16:57:07] WARNING[18232]: chan_sip.c:11839
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '[EMAIL PROTECTED]'. Giving up.
-- Call on SIP/neophonex-out-081a9cc0 left from hold
-- SIP/neophonex-out-081a9cc0 answered SIP/neophonex-081ab240
-- Native bridging SIP/neophonex-081ab240 and
SIP/neophonex-out-081a9cc0
[Jun 25 16:57:10] WARNING[18232]: chan_sip.c:11839
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '[EMAIL PROTECTED]'. Giving up.
== Spawn extension (internal, 9520630111, 3) exited non-zero on
'SIP/neophonex-081ab240'
[Jun 25 16:57:10] NOTICE[18440]: pbx_spool.c:351 attempt_thread: Call
completed to SIP/[EMAIL PROTECTED]
Please help me to figure out why the calls are hung up.
Thanks
Adam
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[asterisk-users] Set a global queue policy
Hi list: Is there any way or an idea of how to made a global queue policy. I need to have a Global Policy or a common policy to many queues. What i need is: I have 20 agents they are members of 5 queues, i have a last recent strategy for all the queues, the problem is that the strategy is for each queue. So one agent recive a call from all the queues, then another, and so on... So we have agent that don't work for more than 20 minutes. And anothers that have long brakes. So what i need is that 5 queues share the strategy, so the agent who get the call the agent that have more idle time in all five queues. So any idea ? -- Alvaro I. Parres Peredo Director de IT Grupo Xmarts SA de CV Tel: +52 (33) 35 63 6261 Ext. 112 01 800 087 2260 Cel: +52 (33) 33 68 1087 [EMAIL PROTECTED] ___ --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] Hi ability solution
Senad Jordanovic wrote: Any High availiability solution for asterisk? VoipCrazy http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf Is this free? Benefits over opensource packages? Thanks, Steve Totaro ___ --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] Rining 180 and 183
Thanks for reply dear
See i am going to explain my setup here
[asterisk]-[Mediant 2000]E1--[Avaya media g/w]
1) This is my setup i am useing asterisk 1.2.14 and this setup working fine but
one issuse is when i call from asterisk to avaya phone i got ringback tone in
my sip phone and i can talk to party or ppls
2) when i call from Avaya analog phone to asterisk IP phone or SIP phone i can
not hear ringback tone in analog phone so how to asterisk genrate tone for
avaya what is the configuration problem in my asterisk or any problem related
mediant 2000 device
i can't figure out where is the problem in avaya or in mediant or asterisk
i give you my configuration of sip
sip.conf
[115]
type=friend
context=mysip
username=115
host=dynamic
callerid=Video Phone 111
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p
subscribecontext=internal
[EMAIL PROTECTED]
[521]
type=user
host=71.5.250.52
dtmfmode=info
secret=12345
nat=yes
context=from-trunk
[mediant]
type=friend
disallow=all
allow=alaw
context=mysip
host=dynamic
dtmfmode=info
username=mediant
secret=12345
extention.conf
[mysip]
exten = 222,1,Dial(SIP/222)
exten = 333,1,Dial(SIP/333)
exten = 555,1,Dial(SIP/555)
exten = 100,1,Dial(SIP/100)
exten = 112,1,Dial(SIP/112)
exten = 115,1,Dial(SIP/115)
exten = 611,1,Echo()
exten = _79.,1,Dial(SIP/mediant/${EXTEN:2})
exten = _79.,2,Congestion
Jared Smith [EMAIL PROTECTED] wrote: On 6/25/07, satish patel wrote:
I have confusion how to asterisk genrate tone and what
ringing code use default 180 or 183 i have setup asterisk with mediant 2000
with avaya
I'm assuming that you're talking SIP here... typically, if Asterisk
receives a 180 response (without SDP), it will pass that on to the end
device, which will generate the ringback tone itself. On the other
hand, if Asterisk receives a 183 with SDP (or a 180 with SDP, which is
less common), it will try to bridge the end device with the media
specified in the SDP message.
Your best bet would be to look at the SIP messaging, specifically
looking to see if there is SDP specified. If SDP is specified in the
183 or 180 message, then I'd try to figure out why Asterisk can't
connect to that IP address and port. (In my experience, nine times
out of ten this is a NAT firewall problem.)
-Jared
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-
Get your own web address.
Have a HUGE year through Yahoo! Small Business.___
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[asterisk-users] Threading troubles 1.4.5 IAX2- SIP (FreeBSD specific??)
Hi there, I've asked this question to the BSD group too, but I'd like to know whether anybody else had similar experiences on Linux 2.6.20 etc.?? FreeBSD 6.2 Asterisk 1.4.5 (and 1.4.3 from ports) Sip phone - PBX(*) -IAX2-VROUTER(*)- SIP-Voip provider (SPA901 SPA922 phones) We've see a situation where the IAX2 appears to loose/drop the voice data to be sent to the SIP side of things. This happens semi intermittently, but we can reliably regenerate it at 40 alaw calls (even on a dedicated 1G network) and also with G729 (but a tad more calls). It appears to happen using both trunking and non-trunking modes. This happened with DONT_OPTIMIZE setting on or off, but with it ON it doesn;t dump core. At least when it was dumping core, it appeared to have been in the pthread_cancel calls. We've recompiled the PBX asterisk with no threading, and the milliwat/etc. tests to the vrouter from the SIP phones ran clean (other than when we pushed the bandwidth limits grin) This morning it was consistently the agent (on the SIP Phones) who could hear the remote side complaining that the remote side can't hear them anymore. After we've recompiled the VROUTER Asterisk with non-threading, the calls stayed stable. What appears to happen is that somewhere in the threading the IAX voice data is discarded or something on the way to the SIP side. Anybody else anything like this? Any other work around for this issue/problem?? -- Hendrik Visage ___ --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] international numbers...
exten = +61242110,1,Goto(0${EXTEN:3},1))
Gary Mensenares wrote:
This is the required dial plan:
0+61|XXX.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Kevin
Withnall
*Sent:* Friday, June 22, 2007 5:11 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] international numbers...
Using trixbox (or a custom dialplan if needed) has anyone been able to
convert a number dialled like
+61242110 to something like 02422110 ie (remove the +61 and
replace with 0)
i just dont know how to set it up, there seems to be no dialplan
wildcard i can use to match +.
I was thinking of something like .61XX but that still seems
wrong to me. it could match other numbers.
anyone had to do this in the past ?
thanks.
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Re: [asterisk-users] Best wifi IP phone for asterisk
MF == Marcus Franke [EMAIL PROTECTED] writes: MF There is a GigaSet SL75 WLAN. MF http://gigaset.siemens.com/shc/0,1935,hq_en_0_122755_rArNrNrNrN,00.html MF Hmm, I did not see any DECT SL75.. You are indeed correct, and I apologise. I was thinking of the SL37; how I messed them up I don't know. The SL37 is nice, but obviously needs a SIP DECT base to be any use. I have no experience with the SL75 WLAN. /Benny ___ --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] Hi ability solution
Steve Totaro wrote: Senad Jordanovic wrote: Any High availiability solution for asterisk? VoipCrazy http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf Is this free? Benefits over opensource packages? Thanks, Steve Totaro Steve... (and anyone else)I made a mistake replying to asterisk-users list thinking it is asterisk-biz list. Anything else please contact me of this list. Thanks Senad ___ --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] IAX client USB phone
On 09:52, Sat 23 Jun 07, Ronaldo Z. Afonso wrote: Hi all, Does anybody know any USB phone that I can use as an IAX Client? Thanks. For what I know, an USB phone is just an USB sound device with a phonelike piece of plastic to hold the mic and speaker. So you can use it with every softphone that works with asterisk. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Res: Record CDR in a Oracle database
What does the cdr table you created in oracle look like ? Tim. On 20 Jun 2007, at 13:37, Everton Goularth wrote: Hi All, Thank's for your hint Tim Panton I could connect my asterisk machine to my oracle machine. I used unixODBC-2.2.11.tar.gz, oracle-instantclient-basic-10.2.0.3-1.i386.rpm, oracle-instantclient-sqlplus-10.2.0.3-1.i386.rpm and the drive from www.oracle.com (odbc-oracle-3.1.0-linux-x86-glibc.tar) to configure my asterisk machine. I can connect to my oracle machine with isql and in the asterisk CLI I can see that the odbc is connected to oracle with the following command: asterisk*CLI odbc show Name: oracle DSN: oracle Connected: yes And I can see with netstat in my oracle machine that the my asterisk machine is connected in the DB oracle. But when I finish a call I received the following error: cdr_odbc: Connected to oracle cdr_odbc: Error in Query -1 cdr_odbc: Query FAILED Call not logged! cdr_odbc: Reconnecting to dsn oracle cdr_odbc: Connected to oracle cdr_odbc: Trying Query again! cdr_odbc: Error in Query -2 cdr_odbc: Query FAILED Call not logged! Does anyone have any ideia? Did anyone have this error?? or know how can I do this? Thank's in advanced... Everton Goularth GoVoIP - Uberlandia - MG Brasil ___ Yahoo! Mail - Sempre a melhor opção para você! Experimente já e veja as novidades. http://br.yahoo.com/mailbeta/tudonovo/ ___ --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 Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] two channels, each dropping into the same context, different behavior.
Ryan Goldberg wrote: So, incoming calls on zap work just as I expect them - an intro is played, the Ah, ignore all that- it had to do with caller id being empty vs unknown or something of that nature - at any rate some problem I can solve myself. I jumped the gun by posting. Ryan ___ --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 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer
Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not work. This error would occur: Spawn extension (companyname-default, 304, 1) exited non-zero on 'SIP/302-0824f618' In the case above, phone extension 304 and 302 were talking, and 304 pressed 'hold'. 302 gets dropped, as indicated above. I enabled sip debuggint (fully below) and notice that I get: SIP/2.0 488 Not Acceptable Here Warning: 399 SDP Not Acceptable Lots of info about the 488 implies some codec issue between the endpoints, so I changed my sip.conf [general] to only permit ulaw, as well as the same in the phone's config (SIPxxx.conf). Didn't help. Since it's cleaner this way, this is how I currently have left it. Strangely, transfers work if they come from a ZAP channel TO a queue or directly to voicemail (via an extension) but will NOT work if anything is being sent directly TO a SIP client. I also tested with a Grandstream Budgetone phone, I have the exact same issue, so it doesnt appear to be a firmware issue with the Cisco's (which are on the latest, 8.6) Here are all of the headers starting from when someone presses hold = = --- SIP read from 192.168.96.91:5060 --- - --- (0 headers 0 lines) Nat keepalive --- ivan*CLI --- SIP read from 192.168.96.18:50422 --- INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.96.18:5060;branch=z9hG4bK50f5380a From: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;tag=001200348d021a566e942586-7ba72702 To: User Name 1 sip:[EMAIL PROTECTED];tag=as2a70a5c3 Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 Date: Mon, 25 Jun 2007 17:09:59 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7960G/8.0 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Supported: replaces,join,norefersub Content-Length: 278 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 26847 2 IN IP4 192.168.96.18 s=SIP Call t=0 0 m=audio 26612 RTP/AVP 0 8 18 101 c=IN IP4 192.168.96.18 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly - --- (16 headers 13 lines) --- Sending to 192.168.96.18 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.96.18:26612 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.96.18:26612 Audio is at 192.168.96.5 port 12846 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 192.168.96.18:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.96.18:5060;branch=z9hG4bK50f5380a;received=192.168.96.18 From: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;tag=001200348d021a566e942586-7ba72702 To: User Name 1 sip:[EMAIL PROTECTED];tag=as2a70a5c3 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 240 v=0 o=root 1431 1433 IN IP4 192.168.96.16 s=session c=IN IP4 192.168.96.16 t=0 0 m=audio 27002 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.96.16, port 5060 Audio is at 192.168.96.5 port 16816 Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.96.16:5060: INVITE sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.96.5:5060;branch=z9hG4bK7f41a807;rport From: sip:[EMAIL PROTECTED];user=phone;tag=as2c9302cc To: User Name 1 sip:[EMAIL PROTECTED];tag=001200347d27001a7e20b127-2129053d Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported:
[asterisk-users] Outbound proxy setting with outbound proxy port in sip.conf
Hi, I'm going to forward SIP request to special outbound SIP proxy with none SIP port. I have this in my sip.conf [sip_proxy-out] type=peer ; we only want to call out, not be called username=408 host=192.168.0.95 outboundproxy=192.168.0.74 port=9097 I want a To: [EMAIL PROTECTED] by proxy 192.168.0.74:9097 but it turns out the To also has the port To: [EMAIL PROTECTED]:9097 How to give the peer an outboundproxy only port? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer
On Mon, 2007-06-25 at 12:51 -0500, falz wrote: Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not work. This error would occur: I had this problem when I first upgraded from 1.2 to 1.4 on all my IP Phones. What I did to fix it was add canreinvite=no to all phones and this solved the problem. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does outboundproxyport still work in 1.4.4
Hi, I specific outboundproxyport=9097 in version 1.4.4, but it doesn't work. It still connects sip port 5060. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer
On 6/25/07, Carlos Chavez [EMAIL PROTECTED] wrote: On Mon, 2007-06-25 at 12:51 -0500, falz wrote: However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not work. This error would occur: I had this problem when I first upgraded from 1.2 to 1.4 on all my IP Phones. What I did to fix it was add canreinvite=no to all phones and this solved the problem. Interesting. Just after posting to this list, I downgraded back to 1.2. I'll re-upgrade again later and try this out in the next few days. I didn't have this defined anywhere, so it must default to yes in the each device's area in sip.conf. Thanks for the tip, I was way off track on what I was debugging (codecs and such) --falz ___ --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] BRI Layer 2 status
Hi, Is there a way to set the status of layer 2 in a BRI circuit to either permanent or call. I have searched the wiki and can't find any info on the subject but seem to recall a post a couple of years ago detailing the process. Thanks Fadge -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 21 June 2007 13:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk config files and #include On Thu, Jun 21, 2007 at 04:07:03PM +0530, Deepak Bhat wrote: Im sure its not a circular include. Like you said its mostly realted to the number of nested includes but the exact meaning is not clear to me. I repeat: To trace this, enable debugging and debug logging. There is a debug comment for each included file. enable 'debug' for some log file in logger.conf , and then run: logger reload reload -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 ___ --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] Rining 180 and 183
I think what Jared recommended, looking at the sip messaging, will help
you here. He means to type sip debug in the asterisk CLI and look for
hints that SDP is being specified in the conversation. If it IS being
specified, then check into NAT/firewall issues, as he recommended also.
Mojo
satish patel wrote:
Thanks for reply dear
See i am going to explain my setup here
[asterisk]-[Mediant 2000]E1--[Avaya media g/w]
1) This is my setup i am useing asterisk 1.2.14 and this setup working
fine but one issuse is when i call from asterisk to avaya phone i got
ringback tone in my sip phone and i can talk to party or ppls
2) when i call from Avaya analog phone to asterisk IP phone or SIP phone
i can not hear ringback tone in analog phone so how to asterisk genrate
tone for avaya what is the configuration problem in my asterisk or any
problem related mediant 2000 device
i can't figure out where is the problem in avaya or in mediant or asterisk
i give you my configuration of sip
sip.conf
[115]
type=friend
context=mysip
username=115
host=dynamic
callerid=Video Phone 111
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p
subscribecontext=internal
[EMAIL PROTECTED]
[521]
type=user
host=71.5.250.52
dtmfmode=info
secret=12345
nat=yes
context=from-trunk
[mediant]
type=friend
disallow=all
allow=alaw
context=mysip
host=dynamic
dtmfmode=info
username=mediant
secret=12345
extention.conf
[mysip]
exten = 222,1,Dial(SIP/222)
exten = 333,1,Dial(SIP/333)
exten = 555,1,Dial(SIP/555)
exten = 100,1,Dial(SIP/100)
exten = 112,1,Dial(SIP/112)
exten = 115,1,Dial(SIP/115)
exten = 611,1,Echo()
exten = _79.,1,Dial(SIP/mediant/${EXTEN:2})
exten = _79.,2,Congestion
*/Jared Smith [EMAIL PROTECTED]/* wrote:
On 6/25/07, satish patel wrote:
I have confusion how to asterisk genrate tone and what
ringing code use default 180 or 183 i have setup asterisk with
mediant 2000
with avaya
I'm assuming that you're talking SIP here... typically, if Asterisk
receives a 180 response (without SDP), it will pass that on to the end
device, which will generate the ringback tone itself. On the other
hand, if Asterisk receives a 183 with SDP (or a 180 with SDP, which is
less common), it will try to bridge the end device with the media
specified in the SDP message.
Your best bet would be to look at the SIP messaging, specifically
looking to see if there is SDP specified. If SDP is specified in the
183 or 180 message, then I'd try to figure out why Asterisk can't
connect to that IP address and port. (In my experience, nine times
out of ten this is a NAT firewall problem.)
-Jared
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Get your own web address.
http://us.rd.yahoo.com/evt=49678/*http://smallbusiness.yahoo.com/domains/?p=BESTDEAL
Have a HUGE year through Yahoo! Small Business.
http://us.rd.yahoo.com/evt=49678/*http://smallbusiness.yahoo.com/domains/?p=BESTDEAL
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[asterisk-users] Transfer Call to Cell Phone
Hello All, I apologize if this question has already been answered but how do you transfer a call to a cell phone or another land line outside the PBX? Setup I have two pots lines into my current Asterisk Box. I have an outsides sales guy who wants to work off his cell phone or transfer his calls from his extension and the main sales extensions. How can I do this right? -- Otis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help. Help. Help. How to make outbound proxy and host URI with different port?
Looks like outboundproxyport doesn't support in 1.4.4 If you set the port, then it conflit with the one in To URI with host. I saw the code for outboundproxyport from the source, but is it a bug? ___ --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] two channels, each dropping into the same context, different behavior.
vant output of show dialplan. Note that the sip calls come in on extension 666. it's cursed, Thanks much in advance. Ryan ___ --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] BRI Layer 2 status
Hi Welcome to the Asterisk users mailing list. When writing a message to the list, don't just reply to an existing message. When replying to a message from the list, please don't quote irrelevant content. And now to answer your question: On Mon, Jun 25, 2007 at 08:16:09PM +0100, asterisk wrote: Hi, Is there a way to set the status of layer 2 in a BRI circuit to either permanent or call. I have searched the wiki and can't find any info on the subject but seem to recall a post a couple of years ago detailing the process. What Asterisk channel do you use? mISDN ? Zaptel? CAPI? Anything else? What version of Asterisk? It may also help to mention the versions of relevant program. E.g: the specific driver, the linux distribution you use and your kernel version. For instance, With Zaptel (chan_zap) you would use 'pri show span N' where N is the number of the span. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] four ringing and hangup with error
Dear All I have this setup [asterisk][mediant2000]---E1 Trunk--[Avaya] When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error *CLI Jun 26 01:26:08 NOTICE[]: chan_local.c:523 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel Jun 26 01:26:08 NOTICE[]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause = 0) My sip.conf [41] type=friend context=mysip username=41 host=dynamic callerid=SIP Phone 41 canreinvite=no dtfmode=rfc2833 disallow=all allow=alaw allow=ulaw [42] type=friend context=mysip username=42 host=dynamic callerid=SIP Phone 42 canreinvite=no dtfmode=rfc2833 disallow=all allow=alaw allow=ulaw [43] type=friend context=mysip username=43 host=dynamic callerid=SIP Phone 43 canreinvite=no dtfmode=rfc2833 disallow=all allow=alaw allow=ulaw MY extension.conf exten = 42,1,Answer exten = 42,2,Dial(SIP/42) exten = 42,3,Hangup exten = 43,1,Answer exten = 43,2,Dial(SIP/43) exten = 43,3,Hangup Regards Satish Patel - Need a vacation? Get great deals to amazing places on Yahoo! Travel. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic DUNDi weight support in * - HELP!
Hi all On the Asterisk website in the blog its announced that in a next release Asterisk would support dynamic DUNDi weitht values. I've installed Asterisk 1.4.4 (via aptitude install) but this doesn't seem to work. Has somebody some experience with this or know whether this feature is already implemented or not? Thanks! Andre ___ --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] Problems with ChanIsAvail always return status 0
Hi list:
I'm having the next problem, it appear that the application ChanIsAvail
is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
I add my dialplan and the output to the cli.
THanks.
In the example i'm dialing from extension SIP/112
My DialPlan Secction:
[macro-callonlyiffree]
exten = s,1,ChanIsAvail(${ARG1}|s)
exten = s,n,NoOp(${AVAILCHAN})
exten = s,n,NoOp(${AVAILORIGCHAN})
exten = s,n,NoOp(${AVAILSTATUS})
exten = s,n,GoToIf($[${AVAILSTATUS} 1]?autoanswer:fail)
exten = s,n,NoOp()
exten = s,n(autoanswer),Dial(${ARG1}||)
exten = s,102(fail),Hangup
[pruebas]
exten = *99,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]||r)
[inpuerta]
exten = _1XX,1,Macro(callonlyiffree,SIP/${EXTEN})
The Log:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/112-08236be8,
Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]||r)
in new stack
-- Called [EMAIL PROTECTED]
-- Called [EMAIL PROTECTED]
-- Executing [EMAIL PROTECTED]:1] Macro(Local/[EMAIL PROTECTED],2,
callonlyiffree|SIP/111) in new stack
-- Executing [EMAIL PROTECTED]:1] ChanIsAvail(
Local/[EMAIL PROTECTED],2, SIP/111|s) in new stack
-- Executing [EMAIL PROTECTED]:1] Macro(Local/[EMAIL PROTECTED],2,
callonlyiffree|SIP/112) in new stack
-- Executing [EMAIL PROTECTED]:1] ChanIsAvail(
Local/[EMAIL PROTECTED],2, SIP/112|s) in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(Local/[EMAIL PROTECTED],2,
SIP/111-081f7d18) in new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(Local/[EMAIL PROTECTED],2,
SIP/111) in new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(Local/[EMAIL PROTECTED],2,
0) in new stack
-- Executing [EMAIL PROTECTED]:5] GotoIf(Local/[EMAIL PROTECTED],2,
1?autoanswer:fail) in new stack
-- Goto (macro-callonlyiffree,s,7)
-- Executing [EMAIL PROTECTED]:7] Dial(Local/[EMAIL PROTECTED],2,
SIP/111||) in new stack
-- Called 111
-- Executing [EMAIL PROTECTED]:2] NoOp(Local/[EMAIL PROTECTED],2,
SIP/112-0822a4b8) in new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(Local/[EMAIL PROTECTED],2,
SIP/112) in new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(Local/[EMAIL PROTECTED],2,
0) in new stack
-- Executing [EMAIL PROTECTED]:5] GotoIf(Local/[EMAIL PROTECTED],2,
1?autoanswer:fail) in new stack
-- Goto (macro-callonlyiffree,s,7)
-- Executing [EMAIL PROTECTED]:7] Dial(Local/[EMAIL PROTECTED],2,
SIP/112||) in new stack
-- Called 112
-- SIP/111-08342ec0 is ringing
-- Local/[EMAIL PROTECTED],1 is ringing
-- SIP/112-08346e28 is ringing
-- Local/[EMAIL PROTECTED],1 is ringing
--
Alvaro I. Parres Peredo
Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
01 800 087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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Re: [asterisk-users] asterisk-users Digest, Vol 35, Issue 89
Dude you got to be freaking kidding me - are you really sending this email to everyone who posts on the Asterisk list? No, most likely he has an autoreply/vacation/out-of-office message enabled. I would expect us to get more of them. Just be thankful he is on digest mode! John Faubion ___ --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] Ring the second line when 1st line is busy
Hi,
I ma using Asterisk 1.2.18 FreePBX 2.2.1. I have assigned every
users in office with Polycom with 2 extensions as below
555
8555
I have configured Follow-me to ring when the users doesn't picks the phone on
line 1(555) after 10 seconds then ring the line 2(8555). But this is not a
standard telephony which I have been advised to change like below.
If someone calls Ext 555 its busy(means on a phone with someone), then only
ring the second line ie 8555 even if that is busy send on voicemail.
If the first line 555 is free no one picks up then let it go on the
voicemail not second line, bcos now no one has picked the phone nor busy.
I could find anyway to do in FreePBX, so was wondering how about doing this.
Thanx for any input.
exten = 555,1,Macro(exten-vm,555,555)
exten = 555,n,Hangup
exten = 555,hint,SIP/555
exten = ${VM_PREFIX}555,1,Macro(vm,555,DIRECTDIAL)
exten = ${VM_PREFIX}555,n,Hangup
--
Deepak
-
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Re: [asterisk-users] Best wifi IP phone for asterisk
On Mon, 25 Jun 2007, Marcus Franke wrote: Benny Amorsen schrieb: MM Siemens GigaSet SL75 The SL75 is DECT, not Wifi. Apart from that, was it really necessary to quote 20 lines and add a ridiculous 15 line disclaimer telling me that I'm not allowed to read the message? There is a GigaSet SL75 WLAN. http://gigaset.siemens.com/shc/0,1935,hq_en_0_122755_rArNrNrNrN,00.html Is this strictly a European phone? I can't find anyone who is selling them in the US... at least not a company I've ever heard of or dealt with before. Tried Amazon.com, voipsupply.com, Tech Data, and 3 pages of Google search results, etc. -- Nick ___ --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] AstPligg
Hello list, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes, I know, it's not much; but at least it's there and can be used immediately. You can find it at http://oinko.net/astpligg I'm looking forward to your comments (and stories) to make it a useful tool for the * community! l. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Call to Cell Phone
I have two pots lines into my current Asterisk Box. I have an outsides sales guy who wants to work off his cell phone or transfer his calls from his extension and the main sales extensions. How can I do this right? Do it right? You really haven't provided enough information to make the right decision. Do you have more than two lines? Surely you have more than two lines. You mention his extension and the main sales extensions. I can't imagine a sales department with only two lines. Well I can, but they don't sell much! 8) If you have other lines available, such as through an ITSP, T1/E1, or etc, then you only need to map his extension to an outside line. This could be done either through a follow-me, call forwarding, fixed routing, or etc. As an example, we have several agents (we're a real estate brokerage office), that only come into the office occasionally. Since most of them use their cell phones for nearly all of their business, I have fixed routing to send calls to them. I will soon have an IVR for them to be able to change that fixed routing on their own. We also have some agents that have a regular desk here in the office. For them, the use call forward unanswered at the phone to route the calls to their cell phones when they are out of the office. The owner uses follow-me to route her calls to the office phone, her home phone and her cell phone. Another way to do it would be to install a SIP/IAX/TDM to TDMA/GSM gateway. Make sure the provider is the same as the salesman's cell phone provider and your mobile to mobile minutes can be free. If you have more than a couple salesmen, this route will likely entail a multi-port gateway but the idea is still the same. As far as the right way, that depends on way to many factors tat you haven't addressed. John Faubion ___ --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] AstPligg
Great! Another one. With such a catchy name too! On Tue, 2007-06-26 at 01:42 +0200, lenz wrote: Hello list, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes, I know, it's not much; but at least it's there and can be used immediately. You can find it at http://oinko.net/astpligg I'm looking forward to your comments (and stories) to make it a useful tool for the * community! l. ___ --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] Best wifi IP phone for asterisk
i believe www.voipango.de sell them to US On 6/26/07, Nick Seraphin [EMAIL PROTECTED] wrote: On Mon, 25 Jun 2007, Marcus Franke wrote: Benny Amorsen schrieb: MM Siemens GigaSet SL75 The SL75 is DECT, not Wifi. Apart from that, was it really necessary to quote 20 lines and add a ridiculous 15 line disclaimer telling me that I'm not allowed to read the message? There is a GigaSet SL75 WLAN. http://gigaset.siemens.com/shc/0,1935,hq_en_0_122755_rArNrNrNrN,00.html Is this strictly a European phone? I can't find anyone who is selling them in the US... at least not a company I've ever heard of or dealt with before. Tried Amazon.com, voipsupply.com, Tech Data, and 3 pages of Google search results, etc. -- Nick ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 35, Issue 91
Greetings! Due to high workload, I am currently checking and responding to e-mail twice daily at 12:00 PM EST and 9:00PM EST. If you require urgent assistance (please ensure it is urgent) that cannot wait until either 12:00 PM or 9:00 PM, please contact me via phone at: 305-338-3867. Thank you for understanding this move to more efficiency and effectiveness. It helps me accomplish more to serve you better. Sincerely, Harold Riley ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Records s as dst
I am using VoiceOne http://voiceone.it/ as my management interface.
I am not 100% sure when it started, but my CDR is now full of s as
the DST instead of the actual dialed number.
As I understand it - it is because it is being recorded in the CDR
while in a macro (as below).
Is there any work around so that I can record the actual dialed number?
[macro-dialout]
exten = s,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
exten = s,n,NoOp(CID_NAME : ${CID_NAME})
exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER})
exten = s,n,NoOp(CID_CLIR : ${CID_CLIR})
exten = s,n,NoOp(TRUNK : ${TRUNK})
exten = s,n,Set(CALLERID(name)=${CID_NAME})
exten = s,n,Set(CALLERID(num)=${CID_NUMBER})
exten =
s,n,Set(PRESENTATION=${IF($[${CID_CLIR}=1]?prohib_not_screened:allowed_not_screened)})
exten = s,n,SetCallerPres(${PRESENTATION})
exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1)
exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW) ;Ring the interface
exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1) ;Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-BUSY,1,Playtones(busy)
exten = s-CONGESTION,1,Playtones(congestion)
exten = _s-.,1,Goto(s-CONGESTION,1) ;Treat anything else as no answer
--
Regards,
Troy Kelly
Director
Purple Oranges Pty Ltd
http://purpleoranges.com/
--
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Re: [asterisk-users] Problems with ChanIsAvail always return status 0
On 6/25/07, Alvaro Parres [EMAIL PROTECTED] wrote: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. This isn't really a problem with ChanIsAvail... it's more of a misunderstanding of what's going on. In your case, it appears that your SIP device will accept multiple calls at the same time from Asterisk. So even if your phone is on a call, Asterisk will come along, try to make another call to it, and the phone says Hey, go ahead! I don't mind! You've got quite a few options to solve your problem. While none of them are exactly perfect, it's good to have lots of options: o Try using the 's' option to ChanIsAvail(). (You might have to turn on call limits in sip.conf to get this to work correctly. Last time I played with this, it seems that the limitonpeers setting had to be set to yes as well.) o Use the GROUP() dialplan function to assign calls to call groups, and then use the GROUP_COUNT() function to check to see if that phone is already on any calls. o Turn off call waiting on your IP phone, so that it'll only accept one call at a time o Simply get call limits in sip.conf working correctly. (This is probably the hardest to do, unfortunately.) Hopefully, one of those options will help you out. (I've placed them in the order I'd try... but your mileage may vary.) -Jared ___ --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] Threading troubles 1.4.5 IAX2- SIP (FreeBSD specific??)
On 6/25/07, Hendrik Visage [EMAIL PROTECTED] wrote: We've see a situation where the IAX2 appears to loose/drop the voice data to be sent to the SIP side of things. This happens semi intermittently, but we can reliably regenerate it at 40 alaw calls (even on a dedicated 1G network) and also with G729 (but a tad more calls). It appears to happen using both trunking and non-trunking modes. I'm making a wild guess here, but I'd say that if you're using trunking, then you're probably getting close to exceeding the MTU size or possibly the MAX_TRUNKDATA size as defined in chan_iax2.c. If it's happening without IAX2 trunking turned on, then I have no idea what's happening... you'd have to look at the IAX2 and SIP packets when the problem is happening, and try to figure out what's causing the issue. -Jared ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DUNDi weight support in * - HELP!
On 6/25/07, Andre Wangler [EMAIL PROTECTED] wrote: On the Asterisk website in the blog its announced that in a next release Asterisk would support dynamic DUNDi weitht values. I've installed Asterisk 1.4.4 (via aptitude install) but this doesn't seem to work. Has somebody some experience with this or know whether this feature is already implemented or not? I answered your question on this list two days ago... maybe you didn't see my reply. If not, here's the summary: the feature will not be available until 1.6.0 is released, unless you want to check out the trunk code by using Subversion. When new features are added to Asterisk, they're only added to the next *major* release. Currently, the 1.4.x versions only get bug fixes, not new features. Once 1.6.0 is released, the 1.6.x releases will only get bug fixes, and any new features will go into 1.8. -Jared ___ --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] Spy a specific Channel
Hello Friends,
I have successfully being able to initiate a automatic Call with AMI that
leads me to a Extension XXX.
In my extension.conf I have: exten = XXX,1,ChanSpy(SIP/).
The problem that I have is to listen to a Specific channel that's using SIP.
I tried out this:
exten = XXX,1,Read(SPYNUM,extension)
exten = XXX,n,ChanSpy(SIP/${SPYNUM},q)
It asks for a specific extension when I dial XXX, but want I'm trying to do
is to find the way to bring the SPYNUM variable from AMI, I've tried to pass
Throw the originate command the variable SPYNUM, but with No luck, does any
one already done this???
Best Regards, Carlos
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Re: [asterisk-users] Transfer Call to Cell Phone
John thanks for the input. forget about my right way ok! by the way selling does not depend on the amount of lines you have and we are very productive trust me I have seen a million dollar corp work off four lines so your statement is quite vague... Otis John Faubion wrote: I have two pots lines into my current Asterisk Box. I have an outsides sales guy who wants to work off his cell phone or transfer his calls from his extension and the main sales extensions. How can I do this right? Do it right? You really haven't provided enough information to make the right decision. Do you have more than two lines? Surely you have more than two lines. You mention his extension and the main sales extensions. I can't imagine a sales department with only two lines. Well I can, but they don't sell much! 8) If you have other lines available, such as through an ITSP, T1/E1, or etc, then you only need to map his extension to an outside line. This could be done either through a follow-me, call forwarding, fixed routing, or etc. As an example, we have several agents (we're a real estate brokerage office), that only come into the office occasionally. Since most of them use their cell phones for nearly all of their business, I have fixed routing to send calls to them. I will soon have an IVR for them to be able to change that fixed routing on their own. We also have some agents that have a regular desk here in the office. For them, the use call forward unanswered at the phone to route the calls to their cell phones when they are out of the office. The owner uses follow-me to route her calls to the office phone, her home phone and her cell phone. Another way to do it would be to install a SIP/IAX/TDM to TDMA/GSM gateway. Make sure the provider is the same as the salesman's cell phone provider and your mobile to mobile minutes can be free. If you have more than a couple salesmen, this route will likely entail a multi-port gateway but the idea is still the same. As far as the right way, that depends on way to many factors tat you haven't addressed. John Faubion ___ --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] Modification of Caller ID based on context
Hi, I have been looking for an example of accomplishing this, but I've been unable to locate something similar to what I'm trying to do. Here's the scenario: Users caller ID is set to their internal extension (200-250). This is set in sip.conf for each user. Each user has a local DID as well (hosted through Vitelity, for example (555)111-). The problem is that this extension was being passed to the outside world. I currently have a SetCallerID command changing the CallerID to our main office number, but some users want their DID sent, not the general number. The problem is that if their caller ID is set to their DID, when users hit redial on their phones internally they dial out and back in. I corrected this by putting each DID in extensions.conf under their three digit extension, but that seems a bit like a kludge obviously. I'm looking for a method of sending the internal three digit extension only when a user is dialing another user internally, otherwise it will send their DID. Is their a method to do this in the dial plan? Anyone have an example of how to accomplish this? Thanks in advance. -Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best wifi IP phone for asterisk
Yes. I have so On 6/25/07, Nick Seraphin [EMAIL PROTECTED] wrote: Is this strictly a European phone? I can't find anyone who is selling them in the US... at least not a company I've ever heard of or dealt with before. Tried Amazon.com, voipsupply.com, Tech Data, and 3 pages of Google search results, etc. -- Nick When I contacted Siemens they suggested I search Google to find a supplier in the US, as there are none offiicial. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR changes in 1.4.5 are confusing
6I am using asterisk 1.4.5 and storing cdr in mysql . Here's example of one cdr generated 2007-06-26 00:44:28 682345xxx 6823456xxx s macro-dialout-trunk SIP/343684-09544f20 SIP/provider-0938de98 Dial SIP/provider/1386734|300| 28 5 60 1 0 ANSWERED I have maximum entries like this one in my cdr . Here outbound extension is shown as 's' and creates problem but still i can manage to get outgoing number in SIP//provider/number . However in between at many many places there are cdr's generated in old way like this one ( its same as generated by previous versions ). 2007-06-26 00:38:13 68234 68234 190437 from-internal SIP/343684-0954bf88 SIP/prov-09544f20 ResetCDR w 29 17 60 1 0 ANSWERED Here it again shows properly the outgoing number instead of 's' . In both cases 68234xxx is my caller id . I am using freepbx along with asterisk . However this is very much confusing .. why is asterisk generating 2 types of cdr ? Is it a bug ? ___ --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 Throughput
I've tried timing faxes two ways: From a fax machine on a station port of an AltiGen PC/PBX served by an MCI PRI calling back into the same PRI and reaching a RightFax server on a station port behind the AltiGen. From the same fax machine on the same station port of the AltiGen PC/PBX served by the same MCI PRI calling a number on an XO PRI connected to an Asterisk system (Digium TE410P), dialing out on another channel on the same PRI back into the MCI PRI and reaching the RightFax server on the station port behind the AltiGen. extensions.conf includes: exten = 6122353002,1,dial(zap/g1/6122590773) Sending a one-page fax with moderate density (no graphics) takes almost five minutes longer going through the Asterisk server. Any suggestions? --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax ___ --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] call transfer problem
Dear ALL I have asterisk with sip and it is integrated with avaya through mediant [*]-[mediant 2000]-E1--[Avaya] Now i want to call transfer feature in asterisk means transfer call from one phone 2 another phone how could it possible with asterisk Regrads Satish - Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains.___ --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] Inexpensive Layer 3 Switch?
Any recommendations on an economical layer 3 switch? Preferably something that you have hands on experience with connecting to IP phones with attached PCs? Specifically I need the ability to set the VLAN in the phone to tag voice packets and to set a native VLAN on a per port basis on the switch to put the untagged packets from the attached PC into a separate VLAN. POE is not a requirement but if you have suggestions for an economical layer 3 switch with POE I’d be glad to hear them…so far I’m looking at the SFE2000 from Linksys. thanks No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.7/868 - Release Date: 6/25/2007 12:20 PM ___ --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] Inexpensive Layer 3 Switch?
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote: Any recommendations on an economical layer 3 switch? Preferably something that you have hands on experience with connecting to IP phones with attached PCs? Specifically I need the ability to set the VLAN in the phone to tag voice packets and to set a native VLAN on a per port basis on the switch to put the untagged packets from the attached PC into a separate VLAN. POE is not a requirement but if you have suggestions for an economical layer 3 switch with POE I'd be glad to hear them…so far I'm looking at the SFE2000 from Linksys. I'm using a SFE2000 with PoE with Asterisk. Besides * and my management box (MySQL, ARI, Queuemetrics, etc.), I have the 10 desk phones that I need PoE for plugged into it, a mix of GXP2000, Linksys SPA941 and a couple of Aastra 480i and 9133i units. One of the things that sold me on it was that it can do 185W across all ports; you're not stuck giving 7W or 15W to each port (which was a problem with many models I looked at and limits you to only 12 of 24 ports getting power). I'm told the Grandsteams pull about 4W, and since that's the phone with the widest deployment, I expect to be able to drive 24 per switch eventually. So far I've had no problems with it, though I'm not using it's layer 3 functionality. I trunk two VLANs to a Baystack layer 3 switch, which was pretty simple to set up and has worked properly ever since. My setup doesn't have both tagged and untagged packets coming into the same port, so I can't speak to that. The configuration certainly seems to support it, and I suspect that the default admit all / no ingress filter combined with the fact that every port has to have a PVID assigned means that it would work pretty much out of the box after you configure your VLAN numbers. The UI interface needs some improvement. It's not quite sure if it's a linksys or a cisco right now. You can make all the configuration via a Web GUI as you would with a typical Linksys SOHO router, but if you don't go to Admin - File Management - Copy Files and choose to copy running-config to startup-config (using drop-down boxes, naturally), it loses all your changes on reboot. :) If you have other questions, feel free to contact me off-list. -- j. ___ --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] Nagios asterisk monitoring
Brandon,
I wanted to show my support this module as well. I would appreciate
information on how to obtain the finished product and or help for beta
testing.
Kenny
On Wed, 2007-04-11 at 14:11 -0500, Brandon Kruse wrote:
I wrote a very extensive plugin for cacti to monitor asterisk.
It uses the manager interface to poll and get statistics for 1.4 and 1.2.
Let me know if you interested, ill post it, or email me directly.
-bkruse
voip crazy wrote:
Dear list,
I am trying to configure the nagios plugin called check_sip. I just
read the README file included with the plugin. I follow the readme
steps to configure the plugin, without success. In the nagios web
interface I can see (No output!) In the status information column. If
I run the chech_sip plugin from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u sip:[EMAIL PROTECTED]
SIP 200 OK: 0.00 second response time
I do not know why If I run the plugin from the consle it works ok, but
if I run it from Nagios web interface it does not run.
Anyone are using this plugin?
Could you helpme to solve that?
Any clue will be appreciated.
Thanks for your time.
VoipCrazy
Here goes my nagios check_sip plugin configuration.
define command{
command_namecheck_sip
command_line$USER1$/check_sip -u $ARG1$ -H $HOSTADDRESS$ -w 5
}
define service{
use generic-service
host_name -PBX
service_description SIP test
check_command check_sip!sip:[EMAIL PROTECTED]
contact_groups admins
max_check_attempts 4
normal_check_interval 5
retry_check_interval1
notification_interval 240
check_period24x7
notification_period 24x7
notification_optionsc,r
}
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Re: [asterisk-users] CDR Records s as dst
This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if
it was intended to be that way .
On 25/06/07, Troy - Purple Oranges [EMAIL PROTECTED] wrote:
I am using VoiceOne http://voiceone.it/ as my management interface.
I am not 100% sure when it started, but my CDR is now full of s as
the DST instead of the actual dialed number.
As I understand it - it is because it is being recorded in the CDR
while in a macro (as below).
Is there any work around so that I can record the actual dialed number?
[macro-dialout]
exten = s,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
exten = s,n,NoOp(CID_NAME : ${CID_NAME})
exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER})
exten = s,n,NoOp(CID_CLIR : ${CID_CLIR})
exten = s,n,NoOp(TRUNK : ${TRUNK})
exten = s,n,Set(CALLERID(name)=${CID_NAME})
exten = s,n,Set(CALLERID(num)=${CID_NUMBER})
exten =
s,n,Set(PRESENTATION=${IF($[${CID_CLIR}=1]?prohib_not_screened:allowed_not_screened)})
exten = s,n,SetCallerPres(${PRESENTATION})
exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1)
exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW) ;Ring the interface
exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1) ;Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-BUSY,1,Playtones(busy)
exten = s-CONGESTION,1,Playtones(congestion)
exten = _s-.,1,Goto(s-CONGESTION,1) ;Treat anything else as no answer
--
Regards,
Troy Kelly
Director
Purple Oranges Pty Ltd
http://purpleoranges.com/
--
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Fax (07) 3105 5987
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