[asterisk-users] CDRs on call forward
In some circumstances I am transferring incoming calls to an external number (cell phone). Whenever this happens at the end of the call I get a single CDR representing the outgoing leg. There is no CDR for the incoming leg and no trace of incoming caller id in the CDR for outgoing leg. Is this expected behaviour? Is there a way to generate two CDRs one for the incoming and for the outgoing leg of forwarded calls? thanks John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDRs on call forward
Hi r u forwarding call using Originate action.. Which version of asterisk u used. On Thu, Sep 24, 2009 at 12:44 PM, John Fawcett john...@erba.tv wrote: In some circumstances I am transferring incoming calls to an external number (cell phone). Whenever this happens at the end of the call I get a single CDR representing the outgoing leg. There is no CDR for the incoming leg and no trace of incoming caller id in the CDR for outgoing leg. Is this expected behaviour? Is there a way to generate two CDRs one for the incoming and for the outgoing leg of forwarded calls? thanks John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Chandrakant Solanki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium transcoding card
Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent generic board out there that I could be investigating? It would be such a shame to waste a PCI slot that could have a voice-card in it. Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Xorcom FXS USB Bank is not loading firmware
Not sure, how can I check, but older astribanks work pretty fine on that system. Loic On Wed, 2009-09-23 at 15:00 +0300, Tzafrir Cohen wrote: On Wed, Sep 23, 2009 at 08:37:19AM +0200, Loic Didelot wrote: Hi Tzafrir, I just compiled the tarball, but now there seem to be some problems with the script lszaptel. Can't call method is_twinstar on unblessed reference at /usr/local/share/perl/5.8.8/Zaptel/Hardware/USB.pm line 108. Is usbfs mounted under /pruc/bus/usb ? -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error When Using Postgresql Schema With Realtime Sip
I have investigated further and found that it is a bug in ODBC, not Asterisk. The SQLColumns function, which asterisk uses to describe the table, does not return any columns when the table name includes the schema specification. You can show this by using isql to do help table which returns info about all the columns, and then help public.table which returns nothing. As chan_sip seems to be the only application that tests the structure of the table before writing to it this is why REGISTER fails. When I have time I will chase up ODBC and see if the issue is tracked there. Do you still want me to raise it as an issue on bugtracker? The problem manifests itself in res_odbc.c inside the ast_odbc_find_table function, around line 176 in my copy of the code. Tilghman Lesher wrote: Yep, I never bothered to include support for specifying either the catalog or the schema, since I've never had reason to use either one. Please report this issue on the bugtracker (https://issues.asterisk.org) and I'll get a patch up straightaway, but I'll need your testing to ensure the patch works. ++ But I won't be able to test for awhile. Stephen. As a test/work-around/option you could try setting the search_path for the user connecting to the database. This has worked for me with RT and LedgerSMB. Steve Hindmarch BT Design ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Xorcom FXS USB Bank is not loading firmware
On Thu, Sep 24, 2009 at 10:54:17AM +0200, Loic Didelot wrote: Not sure, how can I check, but older astribanks work pretty fine on that system. ls /proc/bus/usb What is the output of: lsusb -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/WiFi handsets?
I think that if I could go back and do this project over, I would have chosen DECT as well. We have intermittent problems with the wifi AP's also. Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com mgra...@mstvp.com wrote: I had a good experience with that Polycom/Spectralink phone. Very rugged as you say. The experience did highlight the weaknesses in consumer Wifi AP, which reinforced my commitment to continue using DECT around my office. Michael Original Message Subject: Re: [asterisk-users] SIP/WiFi handsets? From: Jason Baker jba...@glastender.com Date: Wed, September 23, 2009 10:02 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Ken, I did lots of research on this for my VoIP deployment here where I work. We have a huge manufacturing floor and all the supervisors have wifi phones. We evetually settled on the Polycom Spectralink 8002. A nice rugged little phone with great sound quality and some good features. We use a managed switch to create seamless wifi coverage over all of our AP's. Provisioning the phone is pretty easy, but no web browser if you were planning on using the phone to travel with, some hotels require login for internet access. I also tried a clamshell wifi SIP phone by D-Link. This phone actually works really well, but we had some minor issues with it so we went with all Spectralink phones. But the D-Link phone would be good choice if you plan to take your wifi phone on the road. I also tested the Linksys WIP330 which I thought was a terrible phone. Very difficult to use. Good luck. http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html http://www.dlink.com/products/?pid=485 http://www.voipsupply.com/linksys-wip330-na Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Ken D'Ambrosio wrote: Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent price:quality ratio, of possible. Keep seeing handsets for Vonage, etc., in Best Buy and the like, but I imagine it's locked to Vonage, and can't be re-appropriated. Thanks! -Kenhr___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting home intercom to Asterisk?
Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting home intercom to Asterisk?
AIPHONE makes all that stuff, I would not try to reinvent that. Vincent wrote: Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/WiFi handsets?
Just out of curiosity, what managed switch you used on this project? Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP - Jason Baker jba...@glastender.com escreveu: I think that if I could go back and do this project over, I would have chosen DECT as well. We have intermittent problems with the wifi AP's also. Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com mgra...@mstvp.com wrote: I had a good experience with that Polycom/Spectralink phone. Very rugged as you say. The experience did highlight the weaknesses in consumer Wifi AP, which reinforced my commitment to continue using DECT around my office. Michael Original Message Subject: Re: [asterisk-users] SIP/WiFi handsets? From: Jason Baker jba...@glastender.com Date: Wed, September 23, 2009 10:02 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Ken, I did lots of research on this for my VoIP deployment here where I work. We have a huge manufacturing floor and all the supervisors have wifi phones. We evetually settled on the Polycom Spectralink 8002. A nice rugged little phone with great sound quality and some good features. We use a managed switch to create seamless wifi coverage over all of our AP's. Provisioning the phone is pretty easy, but no web browser if you were planning on using the phone to travel with, some hotels require login for internet access. I also tried a clamshell wifi SIP phone by D-Link. This phone actually works really well, but we had some minor issues with it so we went with all Spectralink phones. But the D-Link phone would be good choice if you plan to take your wifi phone on the road. I also tested the Linksys WIP330 which I thought was a terrible phone. Very difficult to use. Good luck. http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html http://www.dlink.com/products/?pid=485 http://www.voipsupply.com/linksys-wip330-na Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Ken D'Ambrosio wrote: Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent price:quality ratio, of possible. Keep seeing handsets for Vonage, etc., in Best Buy and the like, but I imagine it's locked to Vonage, and can't be re-appropriated. Thanks! -Kenhr___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 Transfer issue
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 101
) in a queue..When a caller arrives in queue , it lands on first 100 , 100
then does a blind transfer to 101 .. so that the caller can converse with
101 .. strangely enough the queue_log shows :
1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123
1253814093|1253814090.12|55365|SIP/100|CONNECT|3|1253814090.13|2
1253814104|1253814090.12|55365|SIP/100|TRANSFER|101|from-internal|3|11|1
The third leg of the call that is the transferred part is not at all
reflecting in the queue log.I;ve tried the same with lot many calls .I also
tried with asterisk 1.6.0 version but same problem persists.. my dial plan
is ttached below along with sip.conf.
Extensions.conf
[incoming]
exten = _X.,1,Queue(55365,tT,,,90)
exten = _X.,2,Hangup
[from-internal]
exten = _X.,1,Answer
exten = _X.,2,Dial(SIP/{EXTEN},20,tT)
queues.conf
[general]
persistentmembers = yes
autofill = yes
Canreinvite=yes ; (tried with NO also)
monitor-type = MixMonitor
[55365]
fullname = Frontdesk
strategy = roundrobin
context=from-internal
ringinuse=no
setinterfacevar=yes
setqueueentryvar=yes
timeout = 10
wrapuptime =
autofill = yes
autopause = no
maxlen =
joinempty = no
leavewhenempty = no
reportholdtime = no
musicclass =
call-limit = 20
member = SIP/100
member = SIP/101
member = SIP/102
Please help , I m in a total mess .Thanks Sriram
___
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[asterisk-users] Asterisk 1.6 Transfer issue[Edited]
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 101
) in a queue..When a caller arrives in queue , it lands on first 100 , 100
then does a blind transfer to 101 .. so that the caller can converse with
101 .. strangely enough the queue_log shows :
1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123
1253814093|1253814090.12|55365|SIP/100|CONNECT|3|1253814090.13|2
1253814104|1253814090.12|55365|SIP/100|TRANSFER|101|from-internal|3|11|1
The third leg of the call that is the CALLERCOMPLETED part (Caller's talk
time with 101) is not at all reflecting in the queue log.I;ve tried the same
with lot many calls .I also tried with asterisk 1.6.0 version but same
problem persists.. my dial plan is ttached below along with sip.conf.
Extensions.conf
[incoming]
exten = _X.,1,Queue(55365,tT,,,90)
exten = _X.,2,Hangup
[from-internal]
exten = _X.,1,Answer
exten = _X.,2,Dial(SIP/{EXTEN},20,tT)
queues.conf
[general]
persistentmembers = yes
autofill = yes
Canreinvite=yes ; (tried with NO also)
monitor-type = MixMonitor
[55365]
fullname = Frontdesk
strategy = roundrobin
context=from-internal
ringinuse=no
setinterfacevar=yes
setqueueentryvar=yes
timeout = 10
wrapuptime =
autofill = yes
autopause = no
maxlen =
joinempty = no
leavewhenempty = no
reportholdtime = no
musicclass =
call-limit = 20
member = SIP/100
member = SIP/101
member = SIP/102
Please help , I m in a total mess .Thanks Sriram
___
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Re: [asterisk-users] Connecting home intercom to Asterisk?
On Thu, Sep 24, 2009 at 02:47:18PM +0200, Vincent wrote: Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. The social security number may be considered sensitive information. Some people may not want to have it displayed around. Anyway, typing a 9(?) digit number with no typos may be a problem to some of the patients. Why would they bother? Typing the number doesn't sound like the nicest interface to me. Choosing from a list of names sounds more probable. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/WiFi handsets?
DECT rocks - I understand the reasons for wanting to use wifi but sometimes when it's raining it makes more sense to drive a motorcar instead of ride a motorcycle :-) Cheers, Dean From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Baker Sent: Thursday, September 24, 2009 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP/WiFi handsets? I think that if I could go back and do this project over, I would have chosen DECT as well. We have intermittent problems with the wifi AP's also. Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com http://www.glastender.com/ mgra...@mstvp.com wrote: I had a good experience with that Polycom/Spectralink phone. Very rugged as you say. The experience did highlight the weaknesses in consumer Wifi AP, which reinforced my commitment to continue using DECT around my office. Michael Original Message Subject: Re: [asterisk-users] SIP/WiFi handsets? From: Jason Baker jba...@glastender.com mailto:jba...@glastender.com Date: Wed, September 23, 2009 10:02 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Ken, I did lots of research on this for my VoIP deployment here where I work. We have a huge manufacturing floor and all the supervisors have wifi phones. We evetually settled on the Polycom Spectralink 8002. A nice rugged little phone with great sound quality and some good features. We use a managed switch to create seamless wifi coverage over all of our AP's. Provisioning the phone is pretty easy, but no web browser if you were planning on using the phone to travel with, some hotels require login for internet access. I also tried a clamshell wifi SIP phone by D-Link. This phone actually works really well, but we had some minor issues with it so we went with all Spectralink phones. But the D-Link phone would be good choice if you plan to take your wifi phone on the road. I also tested the Linksys WIP330 which I thought was a terrible phone. Very difficult to use. Good luck. http://www.polycom.com/products/voice/wireless_solutions/wifi_communicat ions/handsets/spectralink_8002_wireless.html http://www.dlink.com/products/?pid=485 http://www.voipsupply.com/linksys-wip330-na Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Ken D'Ambrosio wrote: Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent price:quality ratio, of possible. Keep seeing handsets for Vonage, etc., in Best Buy and the like, but I imagine it's locked to Vonage, and can't be re-appropriated. Thanks! -Kenhr___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan
just forget about the dial(a,G()) approach ... you already posted that
it doesn't work ...
either call sendfax on the 1st step
to send fax to the channel that called in to asterisk or
use that call to trigger sending a fax with originate/system
Martin
On Wed, Sep 23, 2009 at 7:45 PM, sean darcy seandar...@gmail.com wrote:
Martin wrote:
well maybe it doesn't work as it should ... anyways like the other
poster said that's not the way you use it ...
either call the sendfax app directly or use Originate / call file
spooling...
BTW there should be an Originate app executable from dialplan ...
But since there's none you can do
exten = _X.,n,System(echo -e Channel: SIP/num...@gateway\\ncontext:
send\\nExtension: s\\nPriority: 1\\n
/var/spool/asterisk/outgoing/call-${UNIQUEID})
and at send,s,1 call sendfax
Martin
On Wed, Sep 23, 2009 at 1:44 AM, sean darcy seandar...@gmail.com wrote:
Martin wrote:
from RTFM
G(context^exten^pri) - If the call is answered, transfer the calling party
to
the specified priority and the called party to the
specified priority+1.
Optionally, an extension, or extension and context may be
specified.
Otherwise, the current extension is used. You cannot use
any additional
action post answer options in conjunction with this option.
your priority+1 is Hangup ...
is that it ?
Martin
On Tue, Sep 22, 2009 at 7:32 PM, sean darcy seandar...@gmail.com wrote:
Using Digium fax I've tried a simple dialplan:
'8447' = 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyy) [pbx_config]
3. Dial(DAHDI/g0/1bbbccc,,G(send)) [pbx_config]
[send] 4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif)
[pbx_config]
5. HangUp()
But I doesn't work. It executes hangup:
DAHDI/g0/1bbbccc,,G(send)) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/1bbbccc
-- DAHDI/1-1 is proceeding passing it to SIP/173-b55f7448
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
-- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
-- DAHDI/1-1 answered SIP/173-b55f7448
-- Executing [8...@outbound-fax:4] SendFAX(SIP/173-b55f7448,
/var/spool/asterisk/fax/20090922_1301.tif) in new stack
-- Channel 'SIP/173-b55f7448' sending fax
'/var/spool/asterisk/fax/20090922_1301.tif'
-- Channel 'SIP/173-b55f7448' fax session '16' started
-- Executing [8...@outbound-fax:5] Hangup(DAHDI/1-1, ) in new stack
== Spawn extension (outbound-fax, 8447, 5) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
-- Channel 'SIP/173-b55f7448' fax session '16', [ 000.003512 ],
STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX
So why does it hangup before completing the fax?
Does anyone have a SendFax dialplan that works for an analog channel?
Thanks for any help.
sean
Well, I had RTFM :) And I've tried this, without success:
'8447' = 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyy) [pbx_config]
3. Dial(DAHDI/g0/1bbbccc,,G(send)) [pbx_config]
[send] 4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif)
[pbx_config]
5. Wait() [pbx_config]
6. HangUp() [pbx_config]
The dialplan didn't wait. Also tried without the HangUp(), but the
dialplan just fell through. What should priority 5 (priority + 1) be?
Does anyone use SendFax for analog faxing?
sean
OK, I set up context [send-test]
dialplan show send-test
[ Context 'send-test' created by 'pbx_config' ]
's' = 1.
SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config]
newharborpbx*CLI
-= 1 extension (1 priority) in 1 context. =-
Then I tried:
3. Dial(DAHDI/g0/abbbccc,,G(send)) [pbx_config]
[send] 4. GoTo(really-send) [pbx_config]
[wait] 5. Wait(999) [pbx_config]
6. HangUp() [pbx_config]
[really-send] 7. System(env echo -e
Channel:${CHANNEL}\\nContext:send-test\\nExtension: s\\nPriority: 1\\n
/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
8. Wait(99) [pbx_config]
-- Executing [8...@outbound-fax:3] Dial(Console/dsp,
DAHDI/g0/abbbccc,,G(send)) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/abbbccc
-- DAHDI/1-1 is proceeding passing it to Console/dsp
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is making progress passing it to Console/dsp
-- DAHDI/1-1 is making progress passing it to Console/dsp
-- DAHDI/1-1 answered Console/dsp
-- Executing [8...@outbound-fax:4] Goto(Console/dsp,
really-send) in new stack
-- Goto
Re: [asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan
if you're trying to send the same fax to both parties, then do
exten = s,1,System()
exten = s,2,Sendfax()
step1 will spool the call to dial a number and send a fax
step2 will transmit the fax to the incoming call
Martin
On Wed, Sep 23, 2009 at 7:45 PM, sean darcy seandar...@gmail.com wrote:
Martin wrote:
well maybe it doesn't work as it should ... anyways like the other
poster said that's not the way you use it ...
either call the sendfax app directly or use Originate / call file
spooling...
BTW there should be an Originate app executable from dialplan ...
But since there's none you can do
exten = _X.,n,System(echo -e Channel: SIP/num...@gateway\\ncontext:
send\\nExtension: s\\nPriority: 1\\n
/var/spool/asterisk/outgoing/call-${UNIQUEID})
and at send,s,1 call sendfax
Martin
On Wed, Sep 23, 2009 at 1:44 AM, sean darcy seandar...@gmail.com wrote:
Martin wrote:
from RTFM
G(context^exten^pri) - If the call is answered, transfer the calling party
to
the specified priority and the called party to the
specified priority+1.
Optionally, an extension, or extension and context may be
specified.
Otherwise, the current extension is used. You cannot use
any additional
action post answer options in conjunction with this option.
your priority+1 is Hangup ...
is that it ?
Martin
On Tue, Sep 22, 2009 at 7:32 PM, sean darcy seandar...@gmail.com wrote:
Using Digium fax I've tried a simple dialplan:
'8447' = 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyy) [pbx_config]
3. Dial(DAHDI/g0/1bbbccc,,G(send)) [pbx_config]
[send] 4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif)
[pbx_config]
5. HangUp()
But I doesn't work. It executes hangup:
DAHDI/g0/1bbbccc,,G(send)) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/1bbbccc
-- DAHDI/1-1 is proceeding passing it to SIP/173-b55f7448
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
-- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
-- DAHDI/1-1 answered SIP/173-b55f7448
-- Executing [8...@outbound-fax:4] SendFAX(SIP/173-b55f7448,
/var/spool/asterisk/fax/20090922_1301.tif) in new stack
-- Channel 'SIP/173-b55f7448' sending fax
'/var/spool/asterisk/fax/20090922_1301.tif'
-- Channel 'SIP/173-b55f7448' fax session '16' started
-- Executing [8...@outbound-fax:5] Hangup(DAHDI/1-1, ) in new stack
== Spawn extension (outbound-fax, 8447, 5) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
-- Channel 'SIP/173-b55f7448' fax session '16', [ 000.003512 ],
STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX
So why does it hangup before completing the fax?
Does anyone have a SendFax dialplan that works for an analog channel?
Thanks for any help.
sean
Well, I had RTFM :) And I've tried this, without success:
'8447' = 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyy) [pbx_config]
3. Dial(DAHDI/g0/1bbbccc,,G(send)) [pbx_config]
[send] 4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif)
[pbx_config]
5. Wait() [pbx_config]
6. HangUp() [pbx_config]
The dialplan didn't wait. Also tried without the HangUp(), but the
dialplan just fell through. What should priority 5 (priority + 1) be?
Does anyone use SendFax for analog faxing?
sean
OK, I set up context [send-test]
dialplan show send-test
[ Context 'send-test' created by 'pbx_config' ]
's' = 1.
SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config]
newharborpbx*CLI
-= 1 extension (1 priority) in 1 context. =-
Then I tried:
3. Dial(DAHDI/g0/abbbccc,,G(send)) [pbx_config]
[send] 4. GoTo(really-send) [pbx_config]
[wait] 5. Wait(999) [pbx_config]
6. HangUp() [pbx_config]
[really-send] 7. System(env echo -e
Channel:${CHANNEL}\\nContext:send-test\\nExtension: s\\nPriority: 1\\n
/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
8. Wait(99) [pbx_config]
-- Executing [8...@outbound-fax:3] Dial(Console/dsp,
DAHDI/g0/abbbccc,,G(send)) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/abbbccc
-- DAHDI/1-1 is proceeding passing it to Console/dsp
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is making progress passing it to Console/dsp
-- DAHDI/1-1 is making progress passing it to Console/dsp
-- DAHDI/1-1 answered Console/dsp
-- Executing [8...@outbound-fax:4] Goto(Console/dsp,
really-send) in new stack
-- Goto (outbound-fax,8447,7)
-- Executing
Re: [asterisk-users] Asterisk 1.6 Transfer issue[Edited]
Sriram escribió:
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100
101 ) in a queue..When a caller arrives in queue , it lands on
first 100 , 100 then does a blind transfer to 101 .. so that the
caller can converse with 101 .. strangely enough the queue_log shows :
1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123
1253814093|1253814090.12|55365|SIP/100|CONNECT|3|1253814090.13|2
1253814104|1253814090.12|55365|SIP/100|TRANSFER|101|from-internal|3|11|1
Hi,
The queue_log content you are showing is the expected behavior. Once the
call is transfered, is out of the domain of the Queue() application (it
ends handling of the call and passes the control to the context and
extension), so you get no more queue_log events. That is, the TRANSFER
event counts as the last event on that queue for that call. The only
chance to have queue_log events for the transferred call is to transfer
to another queue, for example:
;Transfer context with queue
[from-internal]
exten = _X.,1,Answer
exten = _X.,2,*Queue*(transfer-queue,tT,,,90)
That way you will see the transferred call as an incoming call to
transfer-queue. Now how to make the queue ring the desired extension is
another issue. If using queues to transfer to any other available (not a
specific one) agent suits your needs, it works like a charm. Obviously
if someone transfers to the queue with no available members, it would be
queued again in the transfer-queue until someone is ready to take the
call. You can use attended transfer and blind transfer, it works the
same way.
The third leg of the call that is the CALLERCOMPLETED part (Caller's
talk time with 101) is not at all reflecting in the queue log...I;ve
tried the same with lot many calls ...I also tried with asterisk 1.6.0
version but same problem persists.. my dial plan is ttached below
along with sip.conf.
Extensions.conf
[incoming]
exten = _X.,1,*Queue*(55365,tT,,,90)
exten = _X.,2,Hangup
[from-internal]
exten = _X.,1,Answer
exten = _X.,2,Dial(SIP/{EXTEN},20,tT)
queues.conf
[general]
persistentmembers = yes
autofill = yes
Canreinvite=yes ; (tried with NO also)
monitor-type = MixMonitor
[55365]
fullname = Frontdesk
strategy = roundrobin
context=from-internal
ringinuse=no
setinterfacevar=yes
setqueueentryvar=yes
timeout = 10
wrapuptime =
autofill = yes
autopause = no
maxlen =
joinempty = no
leavewhenempty = no
reportholdtime = no
musicclass =
call-limit = 20
member = SIP/100
member = SIP/101
member = SIP/102
Please help , I m in a total mess ...Thanks Sriram
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] SIP/WiFi handsets?
Dean Collins wrote: Earlier in the thread someone made a comment about using gsm since everyone had gsm handsets already. Can you explain in detail please ? (what hardware specifically, and how does this actually work ?) My ignorant assumption is something like the end user has a cell phone that actually works with 2 carriers - yours and the real carrier. DECT rocks – I understand the reasons for wanting to use wifi but sometimes when it’s raining it makes more sense to drive a motorcar instead of ride a motorcycle J Cheers, Dean *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jason Baker *Sent:* Thursday, September 24, 2009 8:22 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SIP/WiFi handsets? I think that if I could go back and do this project over, I would have chosen DECT as well. We have intermittent problems with the wifi AP's also. *Jason Baker */IT Coordinator/ *Glastender, Inc.* 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com http://www.glastender.com/ mgra...@mstvp.com mailto:mgra...@mstvp.com wrote: I had a good experience with that Polycom/Spectralink phone. Very rugged as you say. The experience did highlight the weaknesses in consumer Wifi AP, which reinforced my commitment to continue using DECT around my office. Michael Original Message Subject: Re: [asterisk-users] SIP/WiFi handsets? From: Jason Baker jba...@glastender.com mailto:jba...@glastender.com Date: Wed, September 23, 2009 10:02 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Ken, I did lots of research on this for my VoIP deployment here where I work. We have a huge manufacturing floor and all the supervisors have wifi phones. We evetually settled on the Polycom Spectralink 8002. A nice rugged little phone with great sound quality and some good features. We use a managed switch to create seamless wifi coverage over all of our AP's. Provisioning the phone is pretty easy, but no web browser if you were planning on using the phone to travel with, some hotels require login for internet access. I also tried a clamshell wifi SIP phone by D-Link. This phone actually works really well, but we had some minor issues with it so we went with all Spectralink phones. But the D-Link phone would be good choice if you plan to take your wifi phone on the road. I also tested the Linksys WIP330 which I thought was a terrible phone. Very difficult to use. Good luck. http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html http://www.dlink.com/products/?pid=485 http://www.voipsupply.com/linksys-wip330-na Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com http://www.glastender.com Ken D'Ambrosio wrote: Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent price:quality ratio, of possible. Keep seeing handsets for Vonage, etc., in Best Buy and the like, but I imagine it's locked to Vonage, and can't be re-appropriated. Thanks! -Kenhr___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Error When Using Postgresql Schema With Realtime Sip
On Thursday 24 September 2009 05:06:02 stephen.hindma...@bt.com wrote: I have investigated further and found that it is a bug in ODBC, not Asterisk. The SQLColumns function, which asterisk uses to describe the table, does not return any columns when the table name includes the schema specification. You can show this by using isql to do help table which returns info about all the columns, and then help public.table which returns nothing. As chan_sip seems to be the only application that tests the structure of the table before writing to it this is why REGISTER fails. When I have time I will chase up ODBC and see if the issue is tracked there. Do you still want me to raise it as an issue on bugtracker? Yes, I want you to raise this on the bugtracker, and no, this is not a bug in ODBC, but a deficiency in my code. Since you tracked this down to the code in res_odbc.c, I might as well tell you that the first two NULL sets of arguments (NULL, 0) are for specifying the catalog and schema, respectively, of the database table, and it is because I never bothered parsing the schema out of the tablename that this does not work. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting home intercom to Asterisk?
On Thu, Sep 24, 2009 at 8:47 AM, Vincent vincent.delpo...@bigfoot.comwrote: I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Valcom makes a SIP door phone, but they're fairly expensive so I've not tried one. -- Heath Roberts htrobe...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium transcoding card
haven't heard of Digium miniPCI transcoding card ... but who knows maybe they're working on it ... Martin On Thu, Sep 24, 2009 at 3:42 AM, Steve Davies davies...@gmail.com wrote: Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent generic board out there that I could be investigating? It would be such a shame to waste a PCI slot that could have a voice-card in it. Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan
Martin wrote:
if you're trying to send the same fax to both parties, then do
exten = s,1,System()
exten = s,2,Sendfax()
step1 will spool the call to dial a number and send a fax
step2 will transmit the fax to the incoming call
Martin
On Wed, Sep 23, 2009 at 7:45 PM, sean darcy seandar...@gmail.com wrote:
Martin wrote:
well maybe it doesn't work as it should ... anyways like the other
poster said that's not the way you use it ...
either call the sendfax app directly or use Originate / call file
spooling...
BTW there should be an Originate app executable from dialplan ...
But since there's none you can do
exten = _X.,n,System(echo -e Channel: SIP/num...@gateway\\ncontext:
send\\nExtension: s\\nPriority: 1\\n
/var/spool/asterisk/outgoing/call-${UNIQUEID})
and at send,s,1 call sendfax
Martin
On Wed, Sep 23, 2009 at 1:44 AM, sean darcy seandar...@gmail.com wrote:
Martin wrote:
from RTFM
G(context^exten^pri) - If the call is answered, transfer the calling
party to
the specified priority and the called party to the
specified priority+1.
Optionally, an extension, or extension and context may be
specified.
Otherwise, the current extension is used. You cannot use
any additional
action post answer options in conjunction with this option.
your priority+1 is Hangup ...
is that it ?
Martin
On Tue, Sep 22, 2009 at 7:32 PM, sean darcy seandar...@gmail.com wrote:
Using Digium fax I've tried a simple dialplan:
'8447' = 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyy) [pbx_config]
3. Dial(DAHDI/g0/1bbbccc,,G(send))[pbx_config]
[send]4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif)
[pbx_config]
5. HangUp()
But I doesn't work. It executes hangup:
DAHDI/g0/1bbbccc,,G(send)) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/1bbbccc
-- DAHDI/1-1 is proceeding passing it to SIP/173-b55f7448
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
-- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
-- DAHDI/1-1 answered SIP/173-b55f7448
-- Executing [8...@outbound-fax:4] SendFAX(SIP/173-b55f7448,
/var/spool/asterisk/fax/20090922_1301.tif) in new stack
-- Channel 'SIP/173-b55f7448' sending fax
'/var/spool/asterisk/fax/20090922_1301.tif'
-- Channel 'SIP/173-b55f7448' fax session '16' started
-- Executing [8...@outbound-fax:5] Hangup(DAHDI/1-1, ) in new
stack
== Spawn extension (outbound-fax, 8447, 5) exited non-zero on
'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
-- Channel 'SIP/173-b55f7448' fax session '16', [ 000.003512 ],
STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX
So why does it hangup before completing the fax?
Does anyone have a SendFax dialplan that works for an analog channel?
Thanks for any help.
sean
Well, I had RTFM :) And I've tried this, without success:
'8447' = 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyy) [pbx_config]
3. Dial(DAHDI/g0/1bbbccc,,G(send))[pbx_config]
[send]4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif)
[pbx_config]
5. Wait() [pbx_config]
6. HangUp()[pbx_config]
The dialplan didn't wait. Also tried without the HangUp(), but the
dialplan just fell through. What should priority 5 (priority + 1) be?
Does anyone use SendFax for analog faxing?
sean
OK, I set up context [send-test]
dialplan show send-test
[ Context 'send-test' created by 'pbx_config' ]
's' =1.
SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config]
newharborpbx*CLI
-= 1 extension (1 priority) in 1 context. =-
Then I tried:
3. Dial(DAHDI/g0/abbbccc,,G(send)) [pbx_config]
[send] 4. GoTo(really-send) [pbx_config]
[wait] 5. Wait(999) [pbx_config]
6. HangUp() [pbx_config]
[really-send] 7. System(env echo -e
Channel:${CHANNEL}\\nContext:send-test\\nExtension: s\\nPriority: 1\\n
/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
8. Wait(99) [pbx_config]
-- Executing [8...@outbound-fax:3] Dial(Console/dsp,
DAHDI/g0/abbbccc,,G(send)) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/abbbccc
-- DAHDI/1-1 is proceeding passing it to Console/dsp
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is making progress passing it to Console/dsp
-- DAHDI/1-1 is making progress passing it to Console/dsp
-- DAHDI/1-1 answered Console/dsp
-- Executing [8...@outbound-fax:4] Goto(Console/dsp,
really-send) in new stack
-- Goto
[asterisk-users] rtp.conf dtmftimeout
What unit is dtmftimeout measured in? The sample configuration is provided below. Does it mean to say that the sample configuration file's dtmftimeout=3000 equates 1/8000th of a second? ; The amount of time a DTMF digit with no 'end' marker should be ; allowed to continue (in 'samples', 1/8000 of a second) ; ;dtmftimeout=3000 -- Brian Camp IT Freedom direct 512.351.4959 brian.c...@itfreedom.com helpdesk 512.419.0070 : fax 512.419.0080 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting home intercom to Asterisk?
On Thu, 24 Sep 2009, Vincent wrote: Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. What? And deprive the patient the pleasure of being sneered at by a snooty nosed doctors receptionist? Or don't you have them where you live? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting home intercom to Asterisk?
Yes - I have a similar access control using VoIP Pantel (Aleen) and Viking Units w- a C1000 module -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent Sent: Thursday, September 24, 2009 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Connecting home intercom to Asterisk? Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom push application for microbrowser
Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get Push message cannot be displayed back from the Polycom phone, and all I am sending is the Polycom example : PolycomIPPhone Data priority=critical h1 Fire Drill at 2pm /h1 Please exit and congregate at your appropriate location outside /Data /PolycomIPPhone Using curl to send it to the phone (192.168.1.54/push) on the LAN as a first test. (all urlencoded, yes) Did anyone ever succeed in doing this here? I'd appreciate any tips. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and VoIP Users Friday Meeting
Greetings, We'll be getting together as usual at 12 Noon Eastern US Time for a chat with David Duffet, a well-known member of the Asterisk community and hopefully one or more of his co-authors of the new book Asterisk 1.4 Professionals Guide. In fact, I've been offered two ebook version to give away during tomorrow's meeting, so you might want to participate. Connect with us by grabbing the SIP or PSTN numbers at http://VUC.me and getting on IRC #voip-users-conference I'm happy to be able to say that several regulars of the conference will be at Astricon and we hope to meet any of you who plan to go. We'll probably do a live version from the hotel Friday, after Astricon is over: http://Astricon.net for more on that. Hope to hear you tomorrow. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
Mike wrote: Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get Push message cannot be displayed back from the Polycom phone, and all I am sending is the Polycom example : PolycomIPPhone Data priority=”critical” h1 Fire Drill at 2pm /h1 Please exit and congregate at your appropriate location outside /Data /PolycomIPPhone Using curl to send it to the phone (192.168.1.54/push) on the LAN as a first test. (all urlencoded, yes) Did anyone ever succeed in doing this here? I'd appreciate any tips. Mike I've never done it (or heard of it until now), it looks pretty cool. Is the apps.push.messageType field set in sip.cfg? Did you set the apps.push.username and apps.push.password fields and is curl sending that username/password to the phone? Just stabs in the dark. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
This is also a stab-in-the-dark as my 501 doesn't have a microbrowser; Have you tried communicating with the phone via telnet to debug the problem? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, September 24, 2009 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom push application for microbrowser Mike wrote: Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get Push message cannot be displayed back from the Polycom phone, and all I am sending is the Polycom example : PolycomIPPhone Data priority=critical h1 Fire Drill at 2pm /h1 Please exit and congregate at your appropriate location outside /Data /PolycomIPPhone Using curl to send it to the phone (192.168.1.54/push) on the LAN as a first test. (all urlencoded, yes) Did anyone ever succeed in doing this here? I'd appreciate any tips. Mike I've never done it (or heard of it until now), it looks pretty cool. Is the apps.push.messageType field set in sip.cfg? Did you set the apps.push.username and apps.push.password fields and is curl sending that username/password to the phone? Just stabs in the dark. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/WiFi handsets?
I am using the 3Com Unified Gigabit Wired and Wireless PoE Switch. See the link below. http://www.3com.com/products/en_US/detail.jsp?tab=featurespathtype=purchasesku=3CRUS2475 Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Vincius Fontes wrote: Just out of curiosity, what managed switch you used on this project? Vincius Fontes www.asteriskforum.com.br - Informaes e discusso sobre Asterisk e telefonia IP - "Jason Baker" jba...@glastender.com escreveu: I think that if I could go back and do this project over, I would have chosen DECT as well. We have intermittent problems with the wifi AP's also. Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com mgra...@mstvp.com wrote: I had a good experience with that Polycom/Spectralink phone. Very rugged as you say. The experience did highlight the weaknesses in consumer Wifi AP, which reinforced my commitment to continue using DECT around my office. Michael Original Message Subject: Re: [asterisk-users] SIP/WiFi handsets? From: Jason Baker jba...@glastender.com Date: Wed, September 23, 2009 10:02 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Ken, I did lots of research on this for my VoIP deployment here where I work. We have a huge manufacturing floor and all the supervisors have wifi phones. We evetually settled on the Polycom Spectralink 8002. A nice rugged little phone with great sound quality and some good features. We use a managed switch to create seamless wifi coverage over all of our AP's. Provisioning the phone is pretty easy, but no web browser if you were planning on using the phone to travel with, some hotels require login for internet access. I also tried a clamshell wifi SIP phone by D-Link. This phone actually works really well, but we had some minor issues with it so we went with all Spectralink phones. But the D-Link phone would be good choice if you plan to take your wifi phone on the road. I also tested the Linksys WIP330 which I thought was a terrible phone. Very difficult to use. Good luck. http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html http://www.dlink.com/products/?pid=485 http://www.voipsupply.com/linksys-wip330-na Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Ken D'Ambrosio wrote: Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent price:quality ratio, of possible. Keep seeing handsets for Vonage, etc., in Best Buy and the like, but I imagine it's locked to Vonage, and can't be re-appropriated. Thanks! -Kenhr___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
In case it's important to you, microbrowser support was added to the 501 and 430 back in SIP 2.1.0. Though how you could use a microbrowser on a 430 for much I don't know. -Dave Danny Nicholas wrote: This is also a stab-in-the-dark as my 501 doesn't have a microbrowser; Have you tried communicating with the phone via telnet to debug the problem? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, September 24, 2009 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom push application for microbrowser Mike wrote: Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get Push message cannot be displayed back from the Polycom phone, and all I am sending is the Polycom example : PolycomIPPhone Data priority=critical h1 Fire Drill at 2pm /h1 Please exit and congregate at your appropriate location outside /Data /PolycomIPPhone Using curl to send it to the phone (192.168.1.54/push) on the LAN as a first test. (all urlencoded, yes) Did anyone ever succeed in doing this here? I'd appreciate any tips. Mike I've never done it (or heard of it until now), it looks pretty cool. Is the apps.push.messageType field set in sip.cfg? Did you set the apps.push.username and apps.push.password fields and is curl sending that username/password to the phone? Just stabs in the dark. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDRs on call forward
Chandrakant Solanki wrote: Hi r u forwarding call using Originate action.. Which version of asterisk u used. Hi asterisk 1.6.2.0 I'm using freepbx, but I looked into the generated files: if I read it correctly it ends up using Dial cmd. thanks, John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
Hi, Take a look at this: http://food4wine.ning.com/forum/topics/submit-an-application-for Way down the page Dave VG submitted some scripts that hold the answers. We also did a Polycom App conference at the VUC, but I can't find the link right now. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
On Thu, 24 Sep 2009, randulo wrote: Take a look at this: http://food4wine.ning.com/forum/topics/submit-an-application-for Grrr. Have to have a Ning ID and you have to be invited. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
Double Grrr... I have a NING ID, but no invite. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, September 24, 2009 2:50 PM To: randulo2...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom push application for microbrowser On Thu, 24 Sep 2009, randulo wrote: Take a look at this: http://food4wine.ning.com/forum/topics/submit-an-application-for Grrr. Have to have a Ning ID and you have to be invited. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan
sean darcy wrote:
Martin wrote:
if you're trying to send the same fax to both parties, then do
exten = s,1,System()
exten = s,2,Sendfax()
step1 will spool the call to dial a number and send a fax
step2 will transmit the fax to the incoming call
Martin
On Wed, Sep 23, 2009 at 7:45 PM, sean darcy seandar...@gmail.com wrote:
Martin wrote:
well maybe it doesn't work as it should ... anyways like the other
poster said that's not the way you use it ...
either call the sendfax app directly or use Originate / call file
spooling...
BTW there should be an Originate app executable from dialplan ...
But since there's none you can do
exten = _X.,n,System(echo -e Channel: SIP/num...@gateway\\ncontext:
send\\nExtension: s\\nPriority: 1\\n
/var/spool/asterisk/outgoing/call-${UNIQUEID})
and at send,s,1 call sendfax
Martin
On Wed, Sep 23, 2009 at 1:44 AM, sean darcy seandar...@gmail.com wrote:
Martin wrote:
from RTFM
G(context^exten^pri) - If the call is answered, transfer the calling
party to
the specified priority and the called party to the
specified priority+1.
Optionally, an extension, or extension and context may be
specified.
Otherwise, the current extension is used. You cannot use
any additional
action post answer options in conjunction with this option.
your priority+1 is Hangup ...
is that it ?
Martin
On Tue, Sep 22, 2009 at 7:32 PM, sean darcy seandar...@gmail.com wrote:
Using Digium fax I've tried a simple dialplan:
'8447' = 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyy) [pbx_config]
3. Dial(DAHDI/g0/1bbbccc,,G(send))[pbx_config]
[send]4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif)
[pbx_config]
5. HangUp()
But I doesn't work. It executes hangup:
DAHDI/g0/1bbbccc,,G(send)) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/1bbbccc
-- DAHDI/1-1 is proceeding passing it to SIP/173-b55f7448
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
-- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
-- DAHDI/1-1 answered SIP/173-b55f7448
-- Executing [8...@outbound-fax:4] SendFAX(SIP/173-b55f7448,
/var/spool/asterisk/fax/20090922_1301.tif) in new stack
-- Channel 'SIP/173-b55f7448' sending fax
'/var/spool/asterisk/fax/20090922_1301.tif'
-- Channel 'SIP/173-b55f7448' fax session '16' started
-- Executing [8...@outbound-fax:5] Hangup(DAHDI/1-1, ) in new
stack
== Spawn extension (outbound-fax, 8447, 5) exited non-zero on
'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
-- Channel 'SIP/173-b55f7448' fax session '16', [ 000.003512 ],
STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX
So why does it hangup before completing the fax?
Does anyone have a SendFax dialplan that works for an analog channel?
Thanks for any help.
sean
Well, I had RTFM :) And I've tried this, without success:
'8447' = 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyy) [pbx_config]
3. Dial(DAHDI/g0/1bbbccc,,G(send))[pbx_config]
[send]4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif)
[pbx_config]
5. Wait() [pbx_config]
6. HangUp()[pbx_config]
The dialplan didn't wait. Also tried without the HangUp(), but the
dialplan just fell through. What should priority 5 (priority + 1) be?
Does anyone use SendFax for analog faxing?
sean
OK, I set up context [send-test]
dialplan show send-test
[ Context 'send-test' created by 'pbx_config' ]
's' =1.
SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config]
newharborpbx*CLI
-= 1 extension (1 priority) in 1 context. =-
Then I tried:
3. Dial(DAHDI/g0/abbbccc,,G(send)) [pbx_config]
[send] 4. GoTo(really-send) [pbx_config]
[wait] 5. Wait(999) [pbx_config]
6. HangUp() [pbx_config]
[really-send] 7. System(env echo -e
Channel:${CHANNEL}\\nContext:send-test\\nExtension: s\\nPriority: 1\\n
/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
8. Wait(99) [pbx_config]
-- Executing [8...@outbound-fax:3] Dial(Console/dsp,
DAHDI/g0/abbbccc,,G(send)) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/abbbccc
-- DAHDI/1-1 is proceeding passing it to Console/dsp
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is making progress passing it to Console/dsp
-- DAHDI/1-1 is making progress passing it to Console/dsp
-- DAHDI/1-1 answered Console/dsp
-- Executing [8...@outbound-fax:4] Goto(Console/dsp,
really-send) in new stack
--
Re: [asterisk-users] Digium transcoding card
On Thu, 24 Sep 2009 09:42:25 +0100, Steve Davies wrote: Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent generic board out there that I could be investigating? It would be such a shame to waste a PCI slot that could have a voice-card in it. Thanks, Steve Looking ahead, but not that far, I'd like to see that card extended to transcode between wideband codecs (G.722, G.722.1, G.722.1C, AMR-WB SILK) in addition to G.729a and G.723.1. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium transcoding card
On Thu, Sep 24, 2009 at 05:32:24PM -0500, Michael Graves wrote: On Thu, 24 Sep 2009 09:42:25 +0100, Steve Davies wrote: Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent generic board out there that I could be investigating? It would be such a shame to waste a PCI slot that could have a voice-card in it. Thanks, Steve Looking ahead, but not that far, I'd like to see that card extended to transcode between wideband codecs (G.722, G.722.1, G.722.1C, AMR-WB SILK) in addition to G.729a and G.723.1. I would like to see something to plug into a tdm410 it could take up one of the ports of the 4 port card (not sure if it exists already) - that would be cool Michael -- I want to thank you for the importance that you've shown for education and literacy. - George W. Bush 04/13/2005 Washington, DC signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting home intercom to Asterisk?
On Thu, 2009-09-24 at 16:20 +0300, Tzafrir Cohen wrote: On Thu, Sep 24, 2009 at 02:47:18PM +0200, Vincent wrote: Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. The social security number may be considered sensitive information. Some people may not want to have it displayed around. Anyway, typing a 9(?) digit number with no typos may be a problem to some of the patients. Why would they bother? Typing the number doesn't sound like the nicest interface to me. Choosing from a list of names sounds more probable. Also handy for people with wounds to their eyes/fingers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
Hi, yes I did, I did have errors at first but that hurdle has been cleared. Thanks for the try :-) Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, September 24, 2009 14:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom push application for microbrowser Mike wrote: Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get Push message cannot be displayed back from the Polycom phone, and all I am sending is the Polycom example : PolycomIPPhone Data priority=critical h1 Fire Drill at 2pm /h1 Please exit and congregate at your appropriate location outside /Data /PolycomIPPhone Using curl to send it to the phone (192.168.1.54/push) on the LAN as a first test. (all urlencoded, yes) Did anyone ever succeed in doing this here? I'd appreciate any tips. Mike I've never done it (or heard of it until now), it looks pretty cool. Is the apps.push.messageType field set in sip.cfg? Did you set the apps.push.username and apps.push.password fields and is curl sending that username/password to the phone? Just stabs in the dark. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
I've tried turning logging way up for the relevant portions of the sip application, but no telnet. Not sure how I would go about this to get more info that what I already have. The phone is giving me a response, it's just that the response is push message cannot be displayed Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, September 24, 2009 14:20 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom push application for microbrowser This is also a stab-in-the-dark as my 501 doesn't have a microbrowser; Have you tried communicating with the phone via telnet to debug the problem? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, September 24, 2009 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom push application for microbrowser Mike wrote: Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get Push message cannot be displayed back from the Polycom phone, and all I am sending is the Polycom example : PolycomIPPhone Data priority=critical h1 Fire Drill at 2pm /h1 Please exit and congregate at your appropriate location outside /Data /PolycomIPPhone Using curl to send it to the phone (192.168.1.54/push) on the LAN as a first test. (all urlencoded, yes) Did anyone ever succeed in doing this here? I'd appreciate any tips. Mike I've never done it (or heard of it until now), it looks pretty cool. Is the apps.push.messageType field set in sip.cfg? Did you set the apps.push.username and apps.push.password fields and is curl sending that username/password to the phone? Just stabs in the dark. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/WiFi handsets?
On Thu, 2009-09-24 at 09:56 -0400, jon pounder wrote: Dean Collins wrote: Earlier in the thread someone made a comment about using gsm since everyone had gsm handsets already. Can you explain in detail please ? (what hardware specifically, and how does this actually work ?) My ignorant assumption is something like the end user has a cell phone that actually works with 2 carriers - yours and the real carrier. Your assumption is correct. We set up our own wireless network on several locations / ships, using nano-BTS systems. Sites were interconnected via VPN's and satelite links. And made a roaming agreement with other GSM-providers. On location (Withing the reach of our own transmitters) you see our name as gsm-provider and if you move away several kilometers, yoo switch automagically to a national provider. Word of caution, probably only viable for large organisations/companies ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting home intercom to Asterisk?
Sorry but AIPHONE is a terrible choice for this. On Thu, Sep 24, 2009 at 8:53 AM, Chris Mason (Lists) li...@masonc.com wrote: AIPHONE makes all that stuff, I would not try to reinvent that. Vincent wrote: Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting home intercom to Asterisk?
I have done something similar using the following: 1. An Adit 600 with FXS card. 2. A door box from Viking http://vikingelectronics.com/products/view_product.php?pid=428 3. An inline dialer from viking: http://vikingelectronics.com/products/view_product.php?pid=137 4. A relay activated using an FXS port ring signal from Mike Sandman: http://www.sandman.com/wizard.html#UniversalRingRelay First an explanation of the problem. Big office building they want during off hours when there is no receptionist and the door is locked that a visitor can ring an office without actually knowing their extension/phone number. The Viking door box is actually a regular phone packaged in a door box that supports Auto Answer and uses power from the FXS port, no additional power is needed. When the call button is pressed it just goes off hook. The dialer plugs in inline with the door box and provides touch tone, needs a 9VDC battery The relay will make contact between C and NO and break contact between C and NC while there is ring on the line (FXS port). In zapata.conf I have the FXS port that the door box is connected to configured as hot line which takes it as soon as it goes off hook to an IVR. The IVR asks the caller to spell the name of the person/office the are looking for or just dial their suite number or extension number. The spelling invokes app_dir, anything else just dials that person. Dynamic features is set so that if *5 is pressed it will dial the FXS port that has the relay connected to it so that the door unlocks. Hope this helps. On Thu, Sep 24, 2009 at 8:47 AM, Vincent vincent.delpo...@bigfoot.com wrote: Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then display a pop-up on the doctor's PC screen that such and such patient just walked in the waiting room. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
