Re: [asterisk-users] Conference feature
Would this be of any help to you? http://lists.digium.com/pipermail/asterisk-users/2011-June/263339.html [SATISH] Mumbai, India. On Mon, Jun 27, 2011 at 7:14 AM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: I am referring to 3-way conference Att, Rafael Saraiva 2011/6/26 Flavio Miranda flaviormira...@hotmail.com Very simple.. Just edit the meetme.conf in /etc/asterisk like this : [rooms] conf = 888 And then, in /etc/asterisk/ extensions.conf , put something like that: [conference] exten = 888,1,Set(CHANNEL(language)=pt_BR)if you have pt_BR audio exten = 888,n,MeetMe(888,pdM) exten = 888,n,Playback(vm-goodbye) exten = 888,n,Hangup When an user call 888 he will be in a conference room. I hope it help! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 26 Jun 2011 22:25:00 -0300 From: rafaels...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Conference feature Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Asterisk 1.8.4 - Google iCal not working
When i reload asterisk, calendar show calendars does not show this. What I am missing? I really need to get this to work! You are missing that you should take out passwords from config files. Hope your gmail account didn't get hacked. -- _ -- 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] Agi script for working hours PBX
Hi ,
can you any buddy provide agi script in perl or php for,
only working hours incomming calls forward to his cellno., and after working
hours should be play one playback msg then forward voicemail to his
extension.
working hours(sun - thu, 9:00 to 19:00)
ex: dialplan
exten = 4578901,2,AGI(agi://127.0.0.1:4577/call_log)
exten =
4578901,3,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = 4578901,4,Dial(SIP/5001,30,tTo)
exten = 4578901,5,Set(CALLERID(all)=044578901)
exten = 4578901,6,Dial(Zap/g0/0554721368,,tTo)
exten = 4578901,7,Hangup
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] Agi script for working hours PBX
On 06/27/2011 05:30 AM, mahesh katta wrote: can you any buddy provide agi script in perl or php for, only working hours incomming calls forward to his cellno., and after working hours should be play one playback msg then forward voicemail to his extension. working hours(sun - thu, 9:00 to 19:00) Have you considered using the GotoIfTime() dial plan application? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] Agi script for working hours PBX
Wasn't that helpful?
http://lists.digium.com/pipermail/asterisk-users/2011-June/264082.html
Use GotoIfTime in agi of your choice with condition part being calculated
dynamically as per your requirement.
But I really don't see any usefulness of AGI if your working hours are fixed
i.e. Mon - Thu, 9:00 to 19:00.
[SATISH]
Mumbai, India.
On Mon, Jun 27, 2011 at 3:00 PM, mahesh katta maheshka...@flexydial.comwrote:
Hi ,
can you any buddy provide agi script in perl or php for,
only working hours incomming calls forward to his cellno., and after
working hours should be play one playback msg then forward voicemail to his
extension.
working hours(sun - thu, 9:00 to 19:00)
ex: dialplan
exten = 4578901,2,AGI(agi://127.0.0.1:4577/call_log)
exten =
4578901,3,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = 4578901,4,Dial(SIP/5001,30,tTo)
exten = 4578901,5,Set(CALLERID(all)=044578901)
exten = 4578901,6,Dial(Zap/g0/0554721368,,tTo)
exten = 4578901,7,Hangup
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
(E) Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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[asterisk-users] ReceiveFax to G.711
Hello, While using FFA, how can we program a specific extensions.conf script to accept the fax in G.711, while for other calls (from a different source), to allow T.38 with or without fallback? The current ReceiveFax(filename,f) is not working for us with one of our providers. Thanks, Michael -- _ -- 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] not find files in asterisk 1.8
hi list, I have a problem with Asterisk 1.8 I installed the software via the yum repositories of asterisk.org but if I go to the /etc/asterisk/ I do not find any files in it? possible? thanks in advance p -- _ -- 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] not find files in asterisk 1.8
Have you installed sample configuration files package? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paolo De Michele Sent: Monday, June 27, 2011 4:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] not find files in asterisk 1.8 hi list, I have a problem with Asterisk 1.8 I installed the software via the yum repositories of asterisk.org but if I go to the /etc/asterisk/ I do not find any files in it? possible? thanks in advance p -- _ -- 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] not find files in asterisk 1.8
the file already installed: Updating: asterisk-sounds-core-en-gsm asterisk18 asterisk18-addons asterisk18-addons-bluetooth asterisk18-addons-core asterisk18-addons-mysql asterisk18-addons-ooh323 asterisk18-core asterisk18-dahdi asterisk18-doc asterisk18-voicemail missing something? On 06/27/2011 01:28 PM, Faisal Hanif wrote: Have you installed sample configuration files package? *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Paolo De Michele *Sent:* Monday, June 27, 2011 4:26 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] not find files in asterisk 1.8 hi list, I have a problem with Asterisk 1.8 I installed the software via the yum repositories of asterisk.org but if I go to the /etc/asterisk/ I do not find any files in it? possible? thanks in advance p -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk changing SIP INFO dtmf duration
Hello List, We are facing a problem in broadcasting DTMF from MeetMe. Our client is sending DTMF with duration=160 (in SIP INFO) to asterisk but asterisk is changing this header to different values like 162, 175 etc while broadcasting to all the participants. Is it possible to restrict asterisk from changing this header value or this is a common behavior of all the PBXs. Regards, Rajib Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- _ -- 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] ReceiveFax to G.711
On 06/27/2011 06:11 AM, Michael wrote: Hello, While using FFA, how can we program a specific extensions.conf script to accept the fax in G.711, while for other calls (from a different source), to allow T.38 with or without fallback? The current ReceiveFax(filename,f) is not working for us with one of our providers. You can't control this in the dialplan. You control this by disabling T.38 support in sip.conf for that provider (setting 't38pt_updtl' to 'no'). However, don't expect very high reliability transferring FAX over G.711 over the Internet. Alternatively, if your provider claims to support T.38, then ensure you are running an updated version of Asterisk, and if you still can't make FAX work over T.38 with them, post debug logs here and we can try to help you figure why it's not working. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] Cisco IP Phones 7942 and Skinny/SIP in asterisk
Thanks a lot. OK, from where you got these files? I am trying to know the source so I can get from it any missing file that the phone is needed. Regards Bilal - bilal ghayyad wrote: Dears; The Cisco 7942 worked in SIP and did not work in skinny firmware (in skinny, it register but no voice can be heared). My phones are Cisco 7940s, so it may be a different layout then expected: cat dialplan.xml DIALTEMPLATE TEMPLATE MATCH=0 Timeout=2 User=Phone/ !-- Local operator-- TEMPLATE MATCH=8. Timeout=0 User=Phone/ !-- Unpark call-- TEMPLATE MATCH=9,.11 Timeout=0 User=Phone/ !-- Service numbers -- TEMPLATE MATCH=9,1.. Timeout=0 User=Phone/ !-- Long Distance -- TEMPLATE MATCH=9,... Timeout=0 User=Phone/ !-- Local numbers -- TEMPLATE MATCH=5... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=44 Timeout=0 User=Phone/ !-- Paging-- TEMPLATE MATCH=700 Timeout=0 User=Phone/ !-- Parking a call-- TEMPLATE MATCH=45 Timeout=0 User=Phone/ !-- Previous Page-- TEMPLATE MATCH=4... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=3... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=\*... Timeout=0 User=Phone/ !-- 3 digit Speed Dials -- TEMPLATE MATCH=\*.. Timeout=2 User=Phone/ !-- 2 digit Speed Dials -- TEMPLATE MATCH=\*. Timeout=1 User=Phone/ !-- 1 digit Speed Dials -- TEMPLATE MATCH=..\* Timeout=0 User=Phone/ !-- 2 digit Access Codes -- TEMPLATE MATCH=* Timeout=15/ !-- Anything else -- /DIALTEMPLATE Doug -- _ -- 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] Festival VOICE MAIL TO FEMAIL
Hi all How can I change the festival application voice in asterisk from mailvoice to femailvoice. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.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] Cisco IP Phones 7942 and Skinny/SIP in asterisk
bilal ghayyad wrote: from where you got these files? I found the link on http://www.voip-info.org I searched for Cisco 7940 http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] not find files in asterisk 1.8
If you're installing from source you need to do make samples On Mon, 2011-06-27 at 13:38 +0200, Paolo De Michele wrote: the file already installed: Updating: asterisk-sounds-core-en-gsm asterisk18 asterisk18-addons asterisk18-addons-bluetooth asterisk18-addons-core asterisk18-addons-mysql asterisk18-addons-ooh323 asterisk18-core asterisk18-dahdi asterisk18-doc asterisk18-voicemail missing something? On 06/27/2011 01:28 PM, Faisal Hanif wrote: Have you installed sample configuration files package? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paolo De Michele Sent: Monday, June 27, 2011 4:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] not find files in asterisk 1.8 hi list, I have a problem with Asterisk 1.8 I installed the software via the yum repositories of asterisk.org but if I go to the /etc/asterisk/ I do not find any files in it? possible? thanks in advance p -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] [ resolved ] not find files in asterisk 1.8
I solved by installing asterisk18-configs regards, p On 06/27/2011 01:38 PM, Paolo De Michele wrote: the file already installed: Updating: asterisk-sounds-core-en-gsm asterisk18 asterisk18-addons asterisk18-addons-bluetooth asterisk18-addons-core asterisk18-addons-mysql asterisk18-addons-ooh323 asterisk18-core asterisk18-dahdi asterisk18-doc asterisk18-voicemail missing something? On 06/27/2011 01:28 PM, Faisal Hanif wrote: Have you installed sample configuration files package? *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Paolo De Michele *Sent:* Monday, June 27, 2011 4:26 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] not find files in asterisk 1.8 hi list, I have a problem with Asterisk 1.8 I installed the software via the yum repositories of asterisk.org but if I go to the /etc/asterisk/ I do not find any files in it? possible? thanks in advance p -- _ -- 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] not find files in asterisk 1.8
Sorry, read your problem properly this time yum install asterisk18-configs On Mon, 2011-06-27 at 13:53 +0100, Ishfaq Malik wrote: If you're installing from source you need to do make samples On Mon, 2011-06-27 at 13:38 +0200, Paolo De Michele wrote: the file already installed: Updating: asterisk-sounds-core-en-gsm asterisk18 asterisk18-addons asterisk18-addons-bluetooth asterisk18-addons-core asterisk18-addons-mysql asterisk18-addons-ooh323 asterisk18-core asterisk18-dahdi asterisk18-doc asterisk18-voicemail missing something? On 06/27/2011 01:28 PM, Faisal Hanif wrote: Have you installed sample configuration files package? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paolo De Michele Sent: Monday, June 27, 2011 4:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] not find files in asterisk 1.8 hi list, I have a problem with Asterisk 1.8 I installed the software via the yum repositories of asterisk.org but if I go to the /etc/asterisk/ I do not find any files in it? possible? thanks in advance p -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Google Voice receiving call problem
On Thu, Jun 23, 2011 at 3:12 PM, Tim Panton t...@westhawk.co.uk wrote: You should probably not mention the voipusersconfere...@gmail.com address this for week's VUC as at the moment the gateway ignores any calls to it. If/when it comes back to life, we can realistically expect wideband through to zipdx. This said, I see that http://Bluejeans.com/vuc works with Gtalk so we'll see if anyone shows up there today or Tues-Wed. :r -- _ -- 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] ReceiveFax to G.711
Hi Kevin, Controlling it through the sip.conf peers is sufficient for us for this case (because this particular provider doesn't support T.38 at all), but I think it would be a good idea to add the option to enable/disable T.38 from the dialplan. If I recall correctly, that's how callweaver worked at the time. Also, we just checked it, and since for that provider, we have other codecs in higher priorities (like GSM, for example) than G.711, G.711 was not chosen as the only codec, so the fax transmission failed. We can not prioritize G.711 over the other codecs in the sip.conf, for the obvious reasons, so for this, we need to do it in the dialplan. How can we do it? Thanks. On Mon, Jun 27, 2011 at 2:41 PM, Kevin P. Fleming kpflem...@digium.comwrote: You can't control this in the dialplan. You control this by disabling T.38 support in sip.conf for that provider (setting 't38pt_updtl' to 'no'). However, don't expect very high reliability transferring FAX over G.711 over the Internet. Alternatively, if your provider claims to support T.38, then ensure you are running an updated version of Asterisk, and if you still can't make FAX work over T.38 with them, post debug logs here and we can try to help you figure why it's not working. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] ReceiveFax to G.711
On Mon, Jun 27, 2011 at 9:06 AM, Michael voip.quest...@gmail.com wrote: Hi Kevin, Controlling it through the sip.conf peers is sufficient for us for this case (because this particular provider doesn't support T.38 at all), but I think it would be a good idea to add the option to enable/disable T.38 from the dialplan. If I recall correctly, that's how callweaver worked at the time. Also, we just checked it, and since for that provider, we have other codecs in higher priorities (like GSM, for example) than G.711, G.711 was not chosen as the only codec, so the fax transmission failed. We can not prioritize G.711 over the other codecs in the sip.conf, for the obvious reasons, so for this, we need to do it in the dialplan. How can we do it? Obvious solution is to move this particular number off this particular provider to a provider where you don't require prioritizing GSM and where that provider properly supports T.38. Outside of that idea, you would have to do some edge SIP routing before the call hit asterisk, to deflect this particular call to an asterisk configured differently. You could do that with OpenSIPS for example. Doesn't that sound like a lot more work than just porting the number to a different provider? Or if you insist on keeping the provider, get a second account with them for just this number, and you can apply different rules in sip.conf for just that account. -- _ -- 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] Asterisk 1.8.4 - Google iCal not working
That's not the password. I switched it to that in the config file for realism. I always give some honey out to those who have a sugar tooth. Any ideas on the fix? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell Sent: Monday, June 27, 2011 3:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8.4 - Google iCal not working When i reload asterisk, calendar show calendars does not show this. What I am missing? I really need to get this to work! You are missing that you should take out passwords from config files. Hope your gmail account didn't get hacked. -- _ -- 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] Asterisk 1.8.4 - Google iCal not working
That's not the password. I switched it to that in the config file for realism. I always give some honey out to those who have a sugar tooth. Any ideas on the fix? On Mon, Jun 27, 2011 at 3:24 AM, Matt Darnell mattdarn...@gmail.com wrote: When i reload asterisk, calendar show calendars does not show this. What I am missing? I really need to get this to work! You are missing that you should take out passwords from config files. Hope your gmail account didn't get hacked. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] interface to gstreamer
Hey - Is there an asterisk interface to gstreamer? I have not found anything... Perhaps using a web cam and v4l to get some video going on a PC with asterisk running. Call come into extensions.conf and take the video and send it to xvimagesink and audio to alsa with outgoing video coming from v4l. Is anything like that out there? Jerry -- _ -- 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] FW: Asterisk 1.8.4 - Google iCal not working
Emailed wrong address -Original Message- From: ERIC HERRON [mailto:e...@lanline.com] Sent: Monday, June 27, 2011 10:58 AM To: 'asterisk-users@lists.digium.com' Subject: FW: [asterisk-users] Asterisk 1.8.4 - Google iCal not working Forgot link.. That's not the password. I switched it to that in the config file for realism. I always give some honey out to those who have a sugar tooth. Any ideas on the fix? have you tried this http://highsecurity.blogspot.com/2010/12/asterisk-18-calendaring-with-text-t o.html -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell Sent: Monday, June 27, 2011 3:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8.4 - Google iCal not working When i reload asterisk, calendar show calendars does not show this. What I am missing? I really need to get this to work! You are missing that you should take out passwords from config files. Hope your gmail account didn't get hacked. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rtptimeout on 1.8.4
Hi Since switching from 1.6.x to 1.8.4 I have noticed the following 1. When you do a 'core show channel channel name' the resulting information only shows data for Frames In , Frames out is always 0. 2. The rtptimeout option in the sip.conf no longer seems to work. I have this set to 60 seconds but have had channels which have not timeout when the rtp stops. If I subsequently do a channel request hangup then the CLI reports that there was a rtptimeout but will be hugely over the set amount. For instance on requesting the hangup on one such channel today it reported rtp timeout for 18000 seconds. Anybody got any ideas how to get 1.8.4 rtptimeout correctly? Regards Jon Jon Farmer Tel 07795 118140 -- _ -- 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] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI
We just finished an upgrade of our Asterisk system to an HA
environment on a pair of servers using Linux-HA. As part of the
upgrade, we also moved to Asterisk version 1.8.4.3
Most things are working quite nicely on the new system. However, I’m
having trouble getting a paging feature to work. In Asterisk 1.4, we
simply used the Page() application like this:
3400,n,Page(SIP/3011SIP/3021SIP/3110SIP/3120SIP/3121SIP/3122SIP/3124SIP/3125SIP/3126SIP/3127SIP/3221SIP/3222SIP/3223SIP/3250SIP/3261SIP/3262SIP/3310SIP/3311SIP/3324SIP/3329SIP/3331SIP/3332SIP/3350SIP/3455SIP/3457)
However, the Page() application seems to rely on the Meetme()
application which also relies on the DAHDI channel driver for mixing
of the audio streams. I have tried using the DAHDI channel driver on
this system, but that seems to make the Music On Hold application use
the DAHDI timing module instead of the pthread module. With the DAHDI
timing module, Music On Hold does not playback Shoutcast streams which
is also a requirement for this system.
As an alternate solution, we have tried implementing a workaround
which simply uses a set of .call files to dial each phone. Those
phones then auto-answer the call and are placed into a conference
bridge on mute using the ConfBridge application. At this point, the
initiating caller speaks the announcement and the phones automatically
hangup after about 10 second. This worked perfectly in our small scale
tests. However, when we ramped this up to the 25 phones that are
required and tested it this morning, somehow this caused the Asterisk
service to restart. I suspect that processing the 25 call files and
placing them into the conference all at the same time somehow made the
system crash and it immediately started up again.
Here's the relevant dialplan:
exten = 3400,1,Answer
exten = 3400,n,playback(beep)
exten = 3400,n,system(cp /etc/asterisk/testPage/*.call
/var/spool/asterisk/outgoing_staging/)
exten = 3400,n,system(mv /var/spool/asterisk/outgoing_staging/*.call
/var/spool/asterisk/outgoing/)
exten = 3400,n,ConfBridge(testPage,1)
exten = 3400,n,hangup
[testPage]
exten = s,1,Answer
exten = s,n,playback(beep)
exten = s,n,Set(TIMEOUT(absolute)=10)
exten = s,n,ConfBridge(testPage,m)
exten = s,n,hangup
exten = _,1,SIPAddHeader(Alert-Info: Auto Answer)
exten = _,n,Dial(SIP/${EXTEN})
exten = _,n,Hangup()
and here's a sample call file:
channel: Local/3011@testPage
callerid: Page
context: testPage
extension: s
priority: 1
archive: no
waittime: 120
Does anyone have insight into how we could accomplish this paging
feature or of anything that we may have missed?
I suspect we could get this all to work with the original Page()
application if there was a way to force MusicOnHold to use the pthread
timing module instead of the Dahdi timing module. Is that configurable
somewhere?
Thanks for your help,
Bob
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[asterisk-users] Question regarding progressinband
Hello, I have question regarding the changes that are made in the sip protocol in Asterisk - the option progressinband. When this option is set to yes in asterisk version 1.4.21.1 - the call flow is: sip.conf: progressinband=yes DeviceAsterisk ---INVITE SDP- -100 Trying -183 Session Prgoress-- After version 1.4.2X+ (tested with 1.4.36/1.4.41.1) the call flow changes to: DeviceAsterisk ---INVITE SDP- -100 Trying -180 Ringing-- -183 Session Prgoress-- From the information that I was able to found in the last version it will always send 180 Ringing and after that the 183 Session Progress. Is there any way to configure the system to behave as in version 1.4.21.1? The show stopper for me is the new behaver. I known that - there is no problem to send 180 Ringing before sending the 183 Session Progress and if there is a problem it is in the other device, not in the Asterisk. Also is the new behaver (1.4.2X+) presented in all new versions of Asterisk 1.6+/1.8+? Regards, Anatoliy -- _ -- 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] ReceiveFax to G.711
On 06/27/2011 08:06 AM, Michael wrote: Controlling it through the sip.conf peers is sufficient for us for this case (because this particular provider doesn't support T.38 at all), but I think it would be a good idea to add the option to enable/disable T.38 from the dialplan. If I recall correctly, that's how callweaver worked at the time. In Asterisk trunk (soon to become Asterisk 1.10), there is an 'F' option to ReceiveFAX and SendFAX that forces audio mode FAX even if the channel is T.38 capable. That would do what you want. I posted a patch some time ago for Asterisk 1.8 to add the same ability, but it probably doesn't apply any more... the it's only a few lines though, it should be fairly easy to replicate in the Asterisk 1.8 version of res_fax.c Also, we just checked it, and since for that provider, we have other codecs in higher priorities (like GSM, for example) than G.711, G.711 was not chosen as the only codec, so the fax transmission failed. We can not prioritize G.711 over the other codecs in the sip.conf, for the obvious reasons, so for this, we need to do it in the dialplan. How can we do it? How are you going to determine that you need to force G.711 to be used? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] ReceiveFax to G.711
You could force g711 inbound by using Set(SIP_CODEC=ulaw) -Original Message- From: Kevin P. Fleming kpflem...@digium.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 27 Jun 2011 14:08:00 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ReceiveFax to G.711 On 06/27/2011 08:06 AM, Michael wrote: Controlling it through the sip.conf peers is sufficient for us for this case (because this particular provider doesn't support T.38 at all), but I think it would be a good idea to add the option to enable/disable T.38 from the dialplan. If I recall correctly, that's how callweaver worked at the time. In Asterisk trunk (soon to become Asterisk 1.10), there is an 'F' option to ReceiveFAX and SendFAX that forces audio mode FAX even if the channel is T.38 capable. That would do what you want. I posted a patch some time ago for Asterisk 1.8 to add the same ability, but it probably doesn't apply any more... the it's only a few lines though, it should be fairly easy to replicate in the Asterisk 1.8 version of res_fax.c Also, we just checked it, and since for that provider, we have other codecs in higher priorities (like GSM, for example) than G.711, G.711 was not chosen as the only codec, so the fax transmission failed. We can not prioritize G.711 over the other codecs in the sip.conf, for the obvious reasons, so for this, we need to do it in the dialplan. How can we do it? How are you going to determine that you need to force G.711 to be used? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax with Asterisk and T38Modem
Hi, I'm fighting with HylaFAX, Asterisk and T38Modem since some time, to get fax2mail and mail2fax working, my SIP-provider only supports T38 and thus using G711 with IAXModem is not an option. I have got running mail2fax with HylaFAX+ 5.5.0, Asterisk 1.4.20 and T38Modem 2.0.0 successfull, but I'm not able to upgrade to anything newer than Asterisk 1.4.24. I was able to break down this problem to some changes in the chan_sip.c, where a improved support for T38 on initial INVITE has been implemented (see: https://reviewboard.asterisk.org/r/208/diff/#index_header). I'm not sure what the problem is, but it seems that the T38 handshake does not work correct any more. Goal is, to get running this with the latest Asterisk, right now 1.8.4.3, Is there s.b. who can give me a hint how to solve the problem? For fax2mail I use Asterisk 1.8.4.3, but it is not working too, after a fax call, the connection with T38Modem is established, HylaFAX reports ANSWER: FAX CONNECTION DEVICE '/dev/ttyT380', but a fax is never received. In the tcpdump I see the correct handshaking between Asterisk and T38Modem, but finaly T38Modem switches to recvonly and Asterisk answers with SIP/2.0 488 Not acceptable here and the call ends shortly after that. Probably there is a problem in T38Modem, so I filed a bug at sourceforge for that: https://sourceforge.net/tracker/?func=detailaid=3337581group_id=152230atid=783657 But may be, somebody has a solution for this too. Thanks a lot, -- Chau y hasta luego, Thorolf -- _ -- 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] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI
Call file are not suitable for you as asterisk process these files in serial
mode (single threaded) and in case of large number of files processing of
last file can be that much delayed that some portion of message may be
already played or the 1st phone may be hanged.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Pierce
Sent: Monday, June 27, 2011 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast
with DAHDI
We just finished an upgrade of our Asterisk system to an HA environment on a
pair of servers using Linux-HA. As part of the upgrade, we also moved to
Asterisk version 1.8.4.3
Most things are working quite nicely on the new system. However, I'm having
trouble getting a paging feature to work. In Asterisk 1.4, we simply used
the Page() application like this:
3400,n,Page(SIP/3011SIP/3021SIP/3110SIP/3120SIP/3121SIP/3122SIP/3124S
IP/3125SIP/3126SIP/3127SIP/3221SIP/3222SIP/3223SIP/3250SIP/3261SIP/3
262SIP/3310SIP/3311SIP/3324SIP/3329SIP/3331SIP/3332SIP/3350SIP/3455
SIP/3457)
However, the Page() application seems to rely on the Meetme() application
which also relies on the DAHDI channel driver for mixing of the audio
streams. I have tried using the DAHDI channel driver on this system, but
that seems to make the Music On Hold application use the DAHDI timing module
instead of the pthread module. With the DAHDI timing module, Music On Hold
does not playback Shoutcast streams which is also a requirement for this
system.
As an alternate solution, we have tried implementing a workaround which
simply uses a set of .call files to dial each phone. Those phones then
auto-answer the call and are placed into a conference bridge on mute using
the ConfBridge application. At this point, the initiating caller speaks the
announcement and the phones automatically hangup after about 10 second. This
worked perfectly in our small scale tests. However, when we ramped this up
to the 25 phones that are required and tested it this morning, somehow this
caused the Asterisk service to restart. I suspect that processing the 25
call files and placing them into the conference all at the same time somehow
made the system crash and it immediately started up again.
Here's the relevant dialplan:
exten = 3400,1,Answer
exten = 3400,n,playback(beep)
exten = 3400,n,system(cp /etc/asterisk/testPage/*.call
/var/spool/asterisk/outgoing_staging/)
exten = 3400,n,system(mv /var/spool/asterisk/outgoing_staging/*.call
/var/spool/asterisk/outgoing/)
exten = 3400,n,ConfBridge(testPage,1)
exten = 3400,n,hangup
[testPage]
exten = s,1,Answer
exten = s,n,playback(beep)
exten = s,n,Set(TIMEOUT(absolute)=10)
exten = s,n,ConfBridge(testPage,m)
exten = s,n,hangup
exten = _,1,SIPAddHeader(Alert-Info: Auto Answer) exten =
_,n,Dial(SIP/${EXTEN}) exten = _,n,Hangup()
and here's a sample call file:
channel: Local/3011@testPage
callerid: Page
context: testPage
extension: s
priority: 1
archive: no
waittime: 120
Does anyone have insight into how we could accomplish this paging feature or
of anything that we may have missed?
I suspect we could get this all to work with the original Page() application
if there was a way to force MusicOnHold to use the pthread timing module
instead of the Dahdi timing module. Is that configurable somewhere?
Thanks for your help,
Bob
--
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