[asterisk-users] Auto Answer or AgentLogin stay on the line?
Hi, I am in process developing Multi-Tenant system for Call Centers. I am considering what are the best option for Agent to Login and and wait for the calls from the Queue. Option 1: AgentLogin (staying on the line with music on hold and bridging the call when a customer enters the queue) Option 2: After AgentLogin then enable Auto-Answer feature on the SIP Phones. I am considering going for Option 1 because it will work on any sip phones but it might be impossible to use AgentLogin if calls are distributed to many queues on multiple servers and leastrecent would not work? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Answer
2015-03-23 11:08 GMT-06:00 ricky gutierrez xserverli...@gmail.com: Hi , I'm having some problems with functions enable auto answer in some Grandstream GXP 1405 , I have enabled this feature in the snom 821 phone and work gr8 , in the gandstream not work, I enable the function on the phone Allow Auto Answer by Call-Info: yes Dialplan: exten = 501,1,SIPAddHeader(Call-Info: answer-after=2) exten = 501,n,Page(SIP/140SIP/110,d) exten = 501,n,Hangup() not work for me, it ring but does the function of auto answer Any idea? I found the problem, my mistake, annex the solution for someone else to help exten = 501,1,SIPAddHeader(Call-Info: answer-after=0) exten = 501,n,Dial(SIP/140SIP/137SIP/112SIP/113SIP/122SIP/120SIP/131SIP/132SIP/116SIP/136SIP/111SIP/125SIP /124) -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto Answer
Hi , I'm having some problems with functions enable auto answer in some Grandstream GXP 1405 , I have enabled this feature in the snom 821 phone and work gr8 , in the gandstream not work, I enable the function on the phone Allow Auto Answer by Call-Info: yes Dialplan: exten = 501,1,SIPAddHeader(Call-Info: answer-after=2) exten = 501,n,Page(SIP/140SIP/110,d) exten = 501,n,Hangup() not work for me, it ring but does the function of auto answer Any idea? -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto-answer call
Ø when i use the Dial the sip/105 still ringing This should help you out http://wiki.snom.com/FAQ/How_to_make_Asterisk_send_INVITEs_to_trigger_the_phone_for_Intercom Dan Journo Kesher Communications (UK) www.keshercommunications.comhttp://www.keshercommunications.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto-answer call
hi when i try to this with page() exten = 506,1,SIPAddHeader(Call-Info:__\; answer-after=0) exten = 506,n,page(SIP/105) CLIAccepting call from '0661xx' to '506' on channel 1/13, span 1 -- Executing [506@default:1] SIPAddHeader(DAHDI/13-1, Call-Info:__; answer-after=0) in new stack -- Executing [506@default:2] Page(DAHDI/13-1, SIP/105) in new stack -- Called 105 -- DAHDI/13-1 Playing 'beep' (language 'en') -- SIP/105-00c7 is ringing -- SIP/105-00c7 is ringing -- SIP/105-00c7 is ringing -- Created MeetMe conference 1023 for conference '1894843837d' -- SIP/105-00c7 is ringing -- Span 1: Channel 1/13 got hangup, cause -1 -- Hungup 'DAHDI/pseudo-358137724' == Spawn extension (default, 506, 2) exited non-zero on 'DAHDI/13-1' -- Hungup 'DAHDI/13-1' and the call hungup when i use the Dial the sip/105 still ringing thanks and regards 2014-02-05 Larry Moore lmo...@omninet.net.au: On 6/02/2014 2:21 AM, Salaheddine Elharit wrote: thanks for your response , i test this solution but the issue still the same any other solution thanks and regards 2014-02-04 Steve Edwards asterisk@sedwards.com mailto:asterisk@sedwards.com: On Tue, 4 Feb 2014, Salaheddine Elharit wrote: i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:__\; answer-after=0) I'm just a 1.2 Luddite... I have this for a Sipura: exten = _!.,n,sipaddheader(Call-Info:\__;answer-after=0) Maybe the quotes or the space after the semi-colon? Maybe wireshark would yield a clue? -- Thanks in advance, Here is a list of headers used for various vendors, I can't remember which one is for Polycom. SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(Alert-Info: Info=Alert-Autoanswer); SIPAddHeader(Call-Info:\;Answer-After=0); SIPAddHeader(P-Auto-Answer: normal); SIPAddHeader(Answer-Mode: Auto); Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto-answer call
thanks for your response , i test this solution but the issue still the same any other solution thanks and regards 2014-02-04 Steve Edwards asterisk@sedwards.com: On Tue, 4 Feb 2014, Salaheddine Elharit wrote: i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:\; answer-after=0) I'm just a 1.2 Luddite... I have this for a Sipura: exten = _!.,n,sipaddheader(Call-Info:\;answer-after=0) Maybe the quotes or the space after the semi-colon? Maybe wireshark would yield a clue? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto-answer call
Please don't top post. On Wed, 5 Feb 2014, Salaheddine Elharit wrote: i test this solution but the issue still the same How does what you see in wireshark compare to what the snom expects? Can you enable debug/verbose syslogging on the phone to see if it complains about anything? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto-answer call
On 6/02/2014 2:21 AM, Salaheddine Elharit wrote: thanks for your response , i test this solution but the issue still the same any other solution thanks and regards 2014-02-04 Steve Edwards asterisk@sedwards.com mailto:asterisk@sedwards.com: On Tue, 4 Feb 2014, Salaheddine Elharit wrote: i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:__\; answer-after=0) I'm just a 1.2 Luddite... I have this for a Sipura: exten = _!.,n,sipaddheader(Call-Info:\__;answer-after=0) Maybe the quotes or the space after the semi-colon? Maybe wireshark would yield a clue? -- Thanks in advance, Here is a list of headers used for various vendors, I can't remember which one is for Polycom. SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(Alert-Info: Info=Alert-Autoanswer); SIPAddHeader(Call-Info:\;Answer-After=0); SIPAddHeader(P-Auto-Answer: normal); SIPAddHeader(Answer-Mode: Auto); Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto-answer call
hello list, i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:\; answer-after=0) exten = 506,n,Dial(SIP/105) when i call the 506 the SIP/105 still ringing, i have snom 320 and i have set the Auto Answer Indication: on i test with Dial and page() but the issue still the same any help please -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto-answer call
On Tue, 4 Feb 2014, Salaheddine Elharit wrote: i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:\; answer-after=0) I'm just a 1.2 Luddite... I have this for a Sipura: exten = _!.,n,sipaddheader(Call-Info:\;answer-after=0) Maybe the quotes or the space after the semi-colon? Maybe wireshark would yield a clue? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto answer
Hello; Is it possible to configure in the sip.conf for the Phone to be auto answer? Regards Bilal-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto answer
Hello, You could use Answer-After for that. But, afaik there is no definitive description in the RFCs on how it is used. You would have to enable such features on the telephones too. And I would expect that different phone manufacturers would probably use different mechanisms to enable such an option. Furthermore, considering the security issues this would create i wouldn' t recommend taking such a path. On Wed, Jul 17, 2013 at 12:04 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; Is it possible to configure in the sip.conf for the Phone to be auto answer? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards. Yasin SULUHAN Contact Information Mobile: +90 535 656 35 55 http://planetvoip.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto answer
On Wednesday 17 July 2013, bilal ghayyad wrote: Hello; Is it possible to configure in the sip.conf for the Phone to be auto answer? That is a phone feature, not an Asterisk feature (in fact, not all phones even support it. A GPO 746 plugged into a Grandstream HandyTone 286, for instance -- well, OK, that's a bit of an extreme case). You would have to put it in the phone's configuration file (which is usually sent to the phone by TFTP or HTTP, after it obtains an IP address by DHCP). The config file usually is named after the MAC address of the phone, with a manufacturer-dependent extension. You really need to refer to the manual for the phones you are using. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto answer
But this not in the sip.conf, this in the extensions.conf, right? Regards Bilal From: Yasin Suluhan ysulu...@gmail.com To: bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 17, 2013 12:21 PM Subject: Re: [asterisk-users] auto answer Hello, You could use Answer-After for that. But, afaik there is no definitive description in the RFCs on how it is used. You would have to enable such features on the telephones too. And I would expect that different phone manufacturers would probably use different mechanisms to enable such an option. Furthermore, considering the security issues this would create i wouldn' t recommend taking such a path. On Wed, Jul 17, 2013 at 12:04 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; Is it possible to configure in the sip.conf for the Phone to be auto answer? RegardsBilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards. Yasin SULUHAN Contact Information Mobile: +90 535 656 35 55-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto answer
On Wednesday 17 July 2013, bilal ghayyad wrote: But this not in the sip.conf, this in the extensions.conf, right? Regards Bilal No. This would be set up in the phone's own configuration file, which in turn depends on the make and model of phone (and its location depends on your site setup). -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto answer
So it is not at asterisk configuration? Regards Bilal From: A J Stiles asterisk_l...@earthshod.co.uk To: bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 17, 2013 12:57 PM Subject: Re: [asterisk-users] auto answer On Wednesday 17 July 2013, bilal ghayyad wrote: But this not in the sip.conf, this in the extensions.conf, right? Regards Bilal No. This would be set up in the phone's own configuration file, which in turn depends on the make and model of phone (and its location depends on your site setup). -- AJS Answers come *after* questions.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto answer
yes its not asterisk configuration, its phone feature and phone configuration. On Wed, Jul 17, 2013 at 3:27 PM, bilal ghayyad bilmar...@yahoo.com wrote: So it is not at asterisk configuration? Regards Bilal -- *From:* A J Stiles asterisk_l...@earthshod.co.uk *To:* bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Wednesday, July 17, 2013 12:57 PM *Subject:* Re: [asterisk-users] auto answer On Wednesday 17 July 2013, bilal ghayyad wrote: But this not in the sip.conf, this in the extensions.conf, right? Regards Bilal No. This would be set up in the phone's own configuration file, which in turn depends on the make and model of phone (and its location depends on your site setup). -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto answer
Dialplan auto answer
; intercom
exten=_*,1,SIPAddHeader(Call-Info:
sip:xxx.xxx.xxx.xxx\;answer-after=0) ;xxx.xxx.xxx.xxx is the address of
your asterisk box
exten=_*,n,Dial(SIP/${EXTEN:1})
As long as your phones are compatible, this MIGHT work.
Worked for me. Sadly, I cannot recall which phones we were using. Long
time ago.
Hope it helps,
Pat
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan
N
Sent: Wednesday, July 17, 2013 9:02 AM
To: bilal ghayyad; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] auto answer
yes its not asterisk configuration, its phone feature and phone
configuration.
On Wed, Jul 17, 2013 at 3:27 PM, bilal ghayyad bilmar...@yahoo.com wrote:
So it is not at asterisk configuration?
Regards
Bilal
_
From: A J Stiles asterisk_l...@earthshod.co.uk
To: bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Wednesday, July 17, 2013 12:57 PM
Subject: Re: [asterisk-users] auto answer
On Wednesday 17 July 2013, bilal ghayyad wrote:
But this not in the sip.conf, this in the extensions.conf, right?
Regards
Bilal
No. This would be set up in the phone's own configuration file, which in
turn
depends on the make and model of phone (and its location depends on your
site
setup).
--
AJS
Answers come *after* questions.
--
_
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Re: [asterisk-users] auto answer
Please don't top post. On Wed, 17 Jul 2013, bilal ghayyad wrote: So it is not at asterisk configuration? 1) The phone has to be configured to allow it. 2) Asterisk has to set the appropriate SIP header for your specific model phone prior to 'dialing' the phone for each call. I.e. the added SIP header for a Cisco is different than for a Polycom. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto answer
If am not wrong even without doing any setting in asterisk side, if the phone has Auto Answer it works.. ! Correct me if am wrong. On Wed, Jul 17, 2013 at 9:14 PM, Steve Edwards asterisk@sedwards.comwrote: Please don't top post. On Wed, 17 Jul 2013, bilal ghayyad wrote: So it is not at asterisk configuration? 1) The phone has to be configured to allow it. 2) Asterisk has to set the appropriate SIP header for your specific model phone prior to 'dialing' the phone for each call. I.e. the added SIP header for a Cisco is different than for a Polycom. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto answer
Please don't top post. On Thu, 18 Jul 2013, Gopalakrishnan N wrote: If am not wrong even without doing any setting in asterisk side, if the phone has Auto Answer it works.. ! Correct me if am wrong. My experience is limited to Sipura, Polycom, and Cisco endpoints. All required configuration on the phone to enable the feature AND the addition of an 'endpoint specific' SIP header to ask the phone to auto-answer in the dialplan. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto answer Asterisk ; Unable to create channel of type
Hello, I'm trying to get s extensions to autoanswer to Centos computer speakers, the computer is a Dell Optiplex 170L embeded sound card. I'm running Centos 6.2 i386 with Asterisk 1.8.10 Does anybody know how to fix error below? -- Executing [s@default:1] Dial(SIP/publicip-0001, console/sda1,20,A(trek)) in new stack [Apr 19 14:25:25] WARNING[2966]: chan_oss.c:377 find_desc: could not find sda1 [Apr 19 14:25:25] WARNING[2966]: chan_oss.c:850 oss_request: oss_request ty console data 0x0xb6a448e8 sda1 [Apr 19 14:25:25] NOTICE[2966]: chan_oss.c:852 oss_request: Device sda1 not found [Apr 19 14:25:25] WARNING[2966]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'console' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@default:2] Hangup(SIP/publicip-0001, ) in new stack == Spawn extension (default, s, 2) exited non-zero on 'SIP/publicip-0001' #lspci -v 00:1f.5 Multimedia audio controller: Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller (rev 02) Subsystem: Dell Device 017a Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at ee00 [size=256] I/O ports at edc0 [size=64] Memory at feb7fa00 (32-bit, non-prefetchable) [size=512] Memory at feb7f900 (32-bit, non-prefetchable) [size=256] Capabilities: [50] Power Management version 2 Kernel driver in use: Intel ICH Kernel modules: snd-intel8x0 Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto Answer in manager
Hi All, I am doing auto answering call from manager but it seems not working any idea ? following commands i am passing to my manager. My phone only ringing not answering we have asterisk 1.8 Action: Originate Channel: SIP/7527 Context: all-page Priority: 1 Variable: SIPAddHeader Value: Alert-Info: Ring Answer CallerID: System Page Action: Logoff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto Answer (Paging)
I'm trying to duplicate a behavior we had with our old avaya system, and I've come across Auto Answer (Ring Answer). However, its not quite the same yet. Right now, when I dial **5053, it will add the SIP header for Ring Answer and it will call 5053. The phone auto pickups just fine. However, we need that call to be muted. If you were to call into a meeting, we wouldn't want them to hear that meeting, but instead the people in the meeting could hear the hellohello, and then that's it. Is it possible to have a auto-muted auto-pickup call? Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto answer when already on a call
I have a customer that has Aastra phones on an Asterisk 1.2.13 system. The big boss want to be able to interrupt someones phone even if they are in the middle of a call. What he wants is basically that when he dials the busy extension that he gets on the speaker so he can say something to that person. I tried to use the example from the paging section of the Wiki and if there is no other call on the phone then I can get directly on the speaker. But if that phone already has another call then it gives me a busy tone. The phone can handle at least 3 calls (Aastra 9133i) and if I make a regular call I do get into the second line. Is there a way to make the phone auto answer on speaker and interrupting the first call? -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chàvez Prats Director de Tecnologìa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto answer an IAXY how
Can an IAXY be setup to auto answer? If so how? I mean any call coming into it automatically connect it to the phone and send voice traffic. Thanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto answer an IAXY how
I see on http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels that an option 'a' is available meaning 'request autoanswer'. Never tested this before, so please do. Another possibility might be setting immediate=yes in iax.conf for the iaxy? just a guess. Moj Jerry Geis wrote: Can an IAXY be setup to auto answer? If so how? I mean any call coming into it automatically connect it to the phone and send voice traffic. Thanks, Jerry ___ --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 !DSPAM:500,44a5778f215925167217508! -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto answer an IAXY how
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 It would not be the iaxy... it would be the phone that is attached to it... there are plenty of phones/answering machines /other FXS signalling devices that can do auto answer... the iaxy is not capable of doing that... Sean Jerry Geis wrote: Can an IAXY be setup to auto answer? If so how? I mean any call coming into it automatically connect it to the phone and send voice traffic. Thanks, Jerry ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEpXvo1Kolm8VQlAURAt5BAJ91hIBpkCABT5buMVqiau5K61pL2ACfYLwG WCp55L0L4OHM64pASfWJCgg= =frDI -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto answer an IAXY how
I see on http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels that an option 'a' is available meaning 'request autoanswer'. Never tested this before, so please do. Another possibility might be setting immediate=yes in iax.conf for the iaxy? just a guess. Moj I tried both of those just now and it did not work. I am trying to use the IAXY to connect to an analog intercom system. I can put a normal analog line on the intercom system, pick up the phone (off hook), and select my zone and talk. I want to do this with an IAXY. So when I call into the IAXY it comes off hook and I would be connect to the intercom. Thanks for any other suggestions. Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto answer an IAXY how
Ah, the problem is that you are connecting FXO to FXO. The IAXy provides dialtone and o does your Intercom system. You can try to use an FXO to FXS converter or simply replace it with an FXO adapter. I would also check the documentation on your intercom device. There may be a way to switch the port type around. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Friday, June 30, 2006 3:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Auto answer an IAXY how I see on http://www.voip- info.org/wiki/index.php?page=Asterisk+IAX+channels that an option 'a' is available meaning 'request autoanswer'. Never tested this before, so please do. Another possibility might be setting immediate=yes in iax.conf for the iaxy? just a guess. Moj I tried both of those just now and it did not work. I am trying to use the IAXY to connect to an analog intercom system. I can put a normal analog line on the intercom system, pick up the phone (off hook), and select my zone and talk. I want to do this with an IAXY. So when I call into the IAXY it comes off hook and I would be connect to the intercom. Thanks for any other suggestions. Jerry ___ --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
[Asterisk-Users] Auto Answer Fax
Can anyone point me to a good howto or example on how to get * to recognize inbound faxes and adjust accordingly? Ideally I would like it to grab the fax and email it to me, but I dont know if that is really possible yet or not. Thanks Neri ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Answer Fax
You can use SpanDSP. http://www.voip-info.org/tiki-index.php?page=spandsp -- Tom On 9/29/05, Rene Nelson [EMAIL PROTECTED] wrote: Can anyone point me to a good howto or example on how to get * to recognize inbound faxes and adjust accordingly? Ideally I would like it to grab the fax and email it to me, but I dont know if that is really possible yet or not. Thanks Neri ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Answer Fax
I have been just working on the same thing today. You can start by taking a look at this application for inbound IP faxes: http://www.voip-info.org/tiki-index.php?page=NVFaxDetect I can currently detect the fax but can't seem to capture it. - Scott On Thursday 29 September 2005 4:52 pm, Rene Nelson wrote: Can anyone point me to a good howto or example on how to get * to recognize inbound faxes and adjust accordingly? Ideally I would like it to grab the fax and email it to me, but I dont know if that is really possible yet or not. Thanks Neri ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto Answer BEEP
I've just received a couple of the Grandstream GXP-2000 enterprise phones for evaluation. When a line on the phone is configured for auto answer, it connects silently. Has anyone been successful in havein a beep sound played to alert the user that he has an autoanswer call? Thanks Bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
