Re: [asterisk-users] Using SIP stack within Asterisk to reboot phones - Possible?
Dnia Mon, 27 Dec 2010 14:30:37 -0500 Bruce B bruceb...@gmail.com napisał(a): That is all I see and the phone is not restarted. I have the same problem with SPA508G and I think that there a bug in authorization - when I set Auth Resync-Reboot: No, I'm able to reset the phone. When I set Auth Resync-Reboot: Yes, I can see, that the phone is receiving notify in syslog and try to do auth: Dec 28 09:02:03 *.*.*.* [1]SipNotifyCmd 8 Dec 28 09:02:03 *.*.*.* SIP: Challenge NOTIFY On asterisk I can see: 0.00 asterisk - phone SIP Request: NOTIFY sip:t...@phone:5060 0.012148 phone - asterisk SIP Status: 401 Unauthorized As far i know, asterisk should continue digest authorization process at this point. Greets, -- Damian Ryszka aka Rychu rychu(at)sileman.net.pl -- _ -- 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] Using SIP stack within Asterisk to reboot phones - Possible?
2010/12/27 Bruce B bruceb...@gmail.com Hi Everyone, I use Asterisk for regularPBX use it's made for. But I want to take it a bit further and use it at cmmand level to be able to send SIP notifies to restart a phone Why would you need to restart a phone that is not registered yet ? For me, the first thing a SIP phone will try to do is to register itself, and then somehow restart. or take advantage of a phone's UPnP capabilities. Is Asterisk capable of that? If so, what is a simple SIP reboot message like and how can I invoke it from a Asterisk CLI? If Asterisk is not the best tool for this purpose what is a very simple to implement SIP stack out there that can do this? 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 -- _ -- 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] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone
Tim: Can you specify what problems you had with the openvox cards i.e. related to your comment other than a bad run of analog cards we experienced last year On Mon, Dec 27, 2010 at 10:20 AM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Anyone who has experience using Digium analog card clones from any of the following: 1. Zycoo 2. CTVON 3. Chinaroby 4. Etross 5. Immediate IT (IIT) 6. Realtone and can give review which one is good quality with easy configuration and error free running. Also since some of these manufacture only analog cards, does anyone have any experience using these in a single system with digital cards from other manufacturers like Openvox? I've used Openvox cards in production heavily for FXO/FXS applications and a few installations using their single port T1/E1 card. Other than a bad run of analog cards we experienced last year, they've been solid. Also, I've used Zycoo analog boards which work very well, not a problem anywhere. One other manufacturer you might want to look at is 'Phonic EQ'. [1] I just received a T1/E1 board (wanted something very inexpensive as this is for my test rig in the lab) that seems to work well but I have not used it long enough to know if there are any problems. --Tim [1] http://www.phoniceq.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 -- _ -- 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] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact
Am 28.12.10 07:26, schrieb Bruce B: Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a number and save the contact or save it without a number. Seems to me that Outcall is not harvesting the CLID number as it should or maybe it's not passing it to outlook so that the old contact which already exists for that number to be pulled. I am wondering if anyone else has experienced this or if you guys think OutCall is really not reliable and I should look for an alternative. Please let me know if there is a solid alternative out there. Thanks Hello, I dont know outcall but i can show you an alternative which really works nice for my needs. http://sourceforge.net/projects/siptapi/ this small program use the windows TAPI interface and dials out via sip. But it only works for outgoing calls in the freeware version. For inbound you have to buy it. best regards Stefan -- _ -- 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 -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- 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 1.4.38 - unknown signalling bri_cpe
Le 27/12/2010 20:09, Kevin P. Fleming a écrit : On 12/27/2010 12:37 PM, Administrator TOOTAI wrote: [...] d...@myphoneserver:/usr/src$ strings /usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony w/PRI DAHDI Telephony Driver w/PRI Asterisk 1.4 has never had BRI support in chan_dahdi. Ouch :-( Are Digiums HB8 cards working with mISDN and Asterisk 1.4? -- Daniel -- _ -- 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 1.4.38 - unknown signalling bri_cpe
On 12/28/2010 05:17 AM, Administrator TOOTAI wrote: Le 27/12/2010 20:09, Kevin P. Fleming a écrit : On 12/27/2010 12:37 PM, Administrator TOOTAI wrote: [...] d...@myphoneserver:/usr/src$ strings /usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony w/PRI DAHDI Telephony Driver w/PRI Asterisk 1.4 has never had BRI support in chan_dahdi. Ouch :-( Are Digiums HB8 cards working with mISDN and Asterisk 1.4? Not to my knowledge; Digium has not produced an mISDN driver for the HX series cards, and I doubt anyone else has. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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
[asterisk-users] How to reload queue on the fly?
Asterisk: 1.6.2.15 On the production server I've modify the /etc/asterisk/queues.conf file. Now in CLI I wan't to reload queue configuration gracefully. I did: virtual-pbx*CLI queue reload members office virtual-pbx*CLI But `queue show office` tells me that nothing has changed. I tried to reload all -- `queue reload all': virtual-pbx*CLI queue reload all [Dec 28 15:10:01] NOTICE[25405]: app_queue.c:5668 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. And still my configuration is not applied. Current queue for `office': virtual-pbx*CLI queue show 1telecom_office 1telecom_office has 1 calls (max unlimited) in 'linear' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/121 (Ringing) has taken no calls yet SIP/120 (Not in use) has taken no calls yet SIP/123 (Not in use) has taken no calls yet Callers: 1. SIP/Softswitch-000161ae (wait: 0:01, prio: 0) While modified configuration is: [office] strategy = linear timeout = 10 member = SIP/100 member = SIP/101 member = SIP/121 member = SIP/123 member = SIP/120 setinterfacevar=yes monitor-format = wav monitor-type = MixMonitor joinempty = yes What's may be wrong? -- С уважением, Денис Давыдов -- _ -- 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] How to reload queue on the fly?
Try: module reload app_queue.so 2010/12/28 Денис Давыдов dyna...@gmail.com Asterisk: 1.6.2.15 On the production server I've modify the /etc/asterisk/queues.conf file. Now in CLI I wan't to reload queue configuration gracefully. I did: virtual-pbx*CLI queue reload members office virtual-pbx*CLI But `queue show office` tells me that nothing has changed. I tried to reload all -- `queue reload all': virtual-pbx*CLI queue reload all [Dec 28 15:10:01] NOTICE[25405]: app_queue.c:5668 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. And still my configuration is not applied. Current queue for `office': virtual-pbx*CLI queue show 1telecom_office 1telecom_office has 1 calls (max unlimited) in 'linear' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/121 (Ringing) has taken no calls yet SIP/120 (Not in use) has taken no calls yet SIP/123 (Not in use) has taken no calls yet Callers: 1. SIP/Softswitch-000161ae (wait: 0:01, prio: 0) While modified configuration is: [office] strategy = linear timeout = 10 member = SIP/100 member = SIP/101 member = SIP/121 member = SIP/123 member = SIP/120 setinterfacevar=yes monitor-format = wav monitor-type = MixMonitor joinempty = yes What's may be wrong? -- С уважением, Денис Давыдов -- _ -- 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 -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.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] Asterisk 1.4.38 - unknown signalling bri_cpe
Le 28/12/2010 13:10, Kevin P. Fleming a écrit : On 12/28/2010 05:17 AM, Administrator TOOTAI wrote: Le 27/12/2010 20:09, Kevin P. Fleming a écrit : On 12/27/2010 12:37 PM, Administrator TOOTAI wrote: [...] d...@myphoneserver:/usr/src$ strings /usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony w/PRI DAHDI Telephony Driver w/PRI Asterisk 1.4 has never had BRI support in chan_dahdi. Ouch :-( Are Digiums HB8 cards working with mISDN and Asterisk 1.4? Not to my knowledge; Digium has not produced an mISDN driver for the HX series cards, and I doubt anyone else has. You should modify your ADL_quickstart document on Digium store to precise the Asterisk version compatible with those cards (perhaps also in datasheet or somewhere else). At this time you have Asterisk-X.X-current.tar.gz but as 1.4, 1.6.2 and 1.8 are existing in current, I will not be the only one making this mistake. -- Daniel -- _ -- 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] How to reload queue on the fly?
The same result. Colleague did remotely (in his words): `queue reload all office' - and it works for me. This is very strange why my variant didn't work :( On 12/28/2010 03:54 PM, Rodrigo Lang wrote: Try: module reload app_queue.so 2010/12/28 Денис Давыдов dyna...@gmail.com mailto:dyna...@gmail.com Asterisk: 1.6.2.15 On the production server I've modify the /etc/asterisk/queues.conf file. Now in CLI I wan't to reload queue configuration gracefully. I did: virtual-pbx*CLI queue reload members office virtual-pbx*CLI But `queue show office` tells me that nothing has changed. I tried to reload all -- `queue reload all': virtual-pbx*CLI queue reload all [Dec 28 15:10:01] NOTICE[25405]: app_queue.c:5668 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. And still my configuration is not applied. Current queue for `office': virtual-pbx*CLI queue show 1telecom_office 1telecom_office has 1 calls (max unlimited) in 'linear' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/121 (Ringing) has taken no calls yet SIP/120 (Not in use) has taken no calls yet SIP/123 (Not in use) has taken no calls yet Callers: 1. SIP/Softswitch-000161ae (wait: 0:01, prio: 0) While modified configuration is: [office] strategy = linear timeout = 10 member = SIP/100 member = SIP/101 member = SIP/121 member = SIP/123 member = SIP/120 setinterfacevar=yes monitor-format = wav monitor-type = MixMonitor joinempty = yes What's may be wrong? -- С уважением, Денис Давыдов -- _ -- 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 -- Rodrigo Lang Opening your mind - Just another Open Source site http://openingyourmind.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 -- С уважением, Денис Давыдов -- _ -- 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] Queue announce parameter problem
Hi list,
I´m using Asterisk 1.6.2.14.
I have some queues configured in my solution, using dynamics members.
All my members are IAX2 clients.
Each member will be at least in two queues at same time.
So i´m trying to use the announce parameter in queues.conf to inform the
member
from where this call arrived to him.
But it is not working. The queue in general is working fine, but the
audio for annouce is not played,
and i can´t see any error message about that, like missing file, codec
problem, etc ...
Here is my queues.conf:
[general]
persistentmembers=yes
autofill=yes
monitor-type=MixMonitor
shared_lastcall=no
[queue_template](!)
musicclass=default
strategy=random
joinempty=yes
leavewhenempty=no
ringinuse=no
monitor-format=gsm
timeout=20
retry=1
maxlen=5
[client_SP](queue_template)
announce=queue_client_sp
[client_RJ](queue_template)
announce=queue_client_rj
[client_PR](queue_template)
announce=queue_client_sp
[client_SC](queue_template)
announce=queue_client_sp
[client_RG](queue_template)
announce=queue_client_sp
In extensions.conf :
[entry-queue]
exten =
s,1,Set(MONITOR_FILENAME=/var/log/asterisk/calls/in/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${queue}_${CALLERID(num)})
exten = s,n,Queue(${queue}65)
Any idea why Asterisk dosnt´play the announce ??
--
Abraços ...
Eduardo Lobo Blanco
Spacecom Ltda.
edua...@spacecom.com.br
(41) 3270-6000 *03
(41) 9101-4450
--
_
-- 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] How to reload queue on the fly?
Try modify the queues.conf to this: [office] strategy = linear timeout = 10 setinterfacevar=yes monitor-format = wav monitor-type = MixMonitor joinempty = yes member = SIP/100 member = SIP/101 member = SIP/121 member = SIP/123 member = SIP/120 At, Rodrigo Lang. 2010/12/28 Давыдов Денис dyna...@gmail.com The same result. Colleague did remotely (in his words): `queue reload all office' - and it works for me. This is very strange why my variant didn't work :( On 12/28/2010 03:54 PM, Rodrigo Lang wrote: Try: module reload app_queue.so 2010/12/28 Денис Давыдов dyna...@gmail.com Asterisk: 1.6.2.15 On the production server I've modify the /etc/asterisk/queues.conf file. Now in CLI I wan't to reload queue configuration gracefully. I did: virtual-pbx*CLI queue reload members office virtual-pbx*CLI But `queue show office` tells me that nothing has changed. I tried to reload all -- `queue reload all': virtual-pbx*CLI queue reload all [Dec 28 15:10:01] NOTICE[25405]: app_queue.c:5668 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. And still my configuration is not applied. Current queue for `office': virtual-pbx*CLI queue show 1telecom_office 1telecom_office has 1 calls (max unlimited) in 'linear' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/121 (Ringing) has taken no calls yet SIP/120 (Not in use) has taken no calls yet SIP/123 (Not in use) has taken no calls yet Callers: 1. SIP/Softswitch-000161ae (wait: 0:01, prio: 0) While modified configuration is: [office] strategy = linear timeout = 10 member = SIP/100 member = SIP/101 member = SIP/121 member = SIP/123 member = SIP/120 setinterfacevar=yes monitor-format = wav monitor-type = MixMonitor joinempty = yes What's may be wrong? -- С уважением, Денис Давыдов -- _ -- 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 -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.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 -- С уважением, Денис Давыдов -- _ -- 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 -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.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] OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact
Thanks for feedback. I am looking mainly for pop-up of Outlook and don't need outgoing call at all but it would be nice to have. Regards, On Tue, Dec 28, 2010 at 4:01 AM, Stefan Schmidt s...@sil.at wrote: Am 28.12.10 07:26, schrieb Bruce B: Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a number and save the contact or save it without a number. Seems to me that Outcall is not harvesting the CLID number as it should or maybe it's not passing it to outlook so that the old contact which already exists for that number to be pulled. I am wondering if anyone else has experienced this or if you guys think OutCall is really not reliable and I should look for an alternative. Please let me know if there is a solid alternative out there. Thanks Hello, I dont know outcall but i can show you an alternative which really works nice for my needs. http://sourceforge.net/projects/siptapi/ this small program use the windows TAPI interface and dials out via sip. But it only works for outgoing calls in the freeware version. For inbound you have to buy it. best regards Stefan -- _ -- 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 -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- 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 1.4.38 - unknown signalling bri_cpe
On 12/28/2010 07:19 AM, Administrator TOOTAI wrote: Le 28/12/2010 13:10, Kevin P. Fleming a écrit : On 12/28/2010 05:17 AM, Administrator TOOTAI wrote: Le 27/12/2010 20:09, Kevin P. Fleming a écrit : On 12/27/2010 12:37 PM, Administrator TOOTAI wrote: [...] d...@myphoneserver:/usr/src$ strings /usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony w/PRI DAHDI Telephony Driver w/PRI Asterisk 1.4 has never had BRI support in chan_dahdi. Ouch :-( Are Digiums HB8 cards working with mISDN and Asterisk 1.4? Not to my knowledge; Digium has not produced an mISDN driver for the HX series cards, and I doubt anyone else has. You should modify your ADL_quickstart document on Digium store to precise the Asterisk version compatible with those cards (perhaps also in datasheet or somewhere else). At this time you have Asterisk-X.X-current.tar.gz but as 1.4, 1.6.2 and 1.8 are existing in current, I will not be the only one making this mistake. The Hx8 card manual (on the Digium website) clearly states on page 46 that the minimum required version of Asterisk for use with these cards is Asterisk 1.6.0.1. It is possible to use Asterisk 1.4.x with these cards, but only with analog (FXO and FXS) modules, not BRI modules. This is repeated on page 49, where the manual says to download the latest release version of Asterisk 1.6 or later. The ADL_quickstart document you are referring to is generic and does not take into account the specific requirements of particular card models and/or configurations; users need to refer to the manuals for the cards and modules they plan to use to get a complete understanding of the required software components and their versions. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] Panasonic KX-TGP500 w/Asterisk
Has anybody have any experience using the KX-TGP500 Cordless DEC 6.0 System with asterisk ? Hi, They sent us a few for free, so I guess I owe them a review. The Panasonic 500 and 550 worked very well with *sterisk 1.4. The full 550 phone seemed a little small, but all the advertised features worked. Audio quality was very good with a nice long range. The cordless phones are light weight with a belt clip, You just register the base set or phone, then connect the cordless phones. It pulls a config from central server, firmware options are thorough for business needs. One cordless phone (out of 5) developed a bad key. I really have to press hard to get the 1 key to work. The key went bad after 5 months. Overall, I really like the phones. Adrian Serafini -- _ -- 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] One way crappy audio in iax call - Asterisk 1.6.2.15
Le 24/12/2010 16:47, Steve Davies a écrit : On 24 December 2010 14:40, Administrator TOOTAIad...@tootai.net wrote: Hi, We had 2 asterisk 1.4 connected together in iax, all was fine. One of them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38 When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But calling from 1.6.2 to 1.4 give a bad audio to calling party (words are cutted, you can't understand the words). On callee party it's still good. We replace 1.6.2 version with 1.4.38 and everything is going back to normal, good audio on both side does'nt matter who call. I already opened another thread about problem with iax and Asterisk 1.6.2 (rsa auth not working anymore). Are there some known problems with iax and 1.6 version of Asterisk? Thanks for any hint Not 100% sure, but I think there was a fix for IAX audio in 1.6.2.16-rc1 - Perhaps try that? Done and it effectively seems to solve the problem. Thanks. -- Daniel -- _ -- 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 1.4.38 - unknown signalling bri_cpe
HI We are trying to setup Trixbox for Multiple tenant so please guide us. We are using TrixBox 2.6 CE Thanks Amardeep -- _ -- 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] Panasonic KX-TGP500 w/Asterisk
Well, my distributor states the 550 has been discontinued, I'm going to try out the 500 and see how well it works, as it appears to be the only Sip based cordless phone that looks to be half way decent.. (not a big fan of Sip/2.4/5.8ghz Wifi Phones) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Serafini Sent: Tuesday, December 28, 2010 11:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Panasonic KX-TGP500 w/Asterisk Has anybody have any experience using the KX-TGP500 Cordless DEC 6.0 System with asterisk ? Hi, They sent us a few for free, so I guess I owe them a review. The Panasonic 500 and 550 worked very well with *sterisk 1.4. The full 550 phone seemed a little small, but all the advertised features worked. Audio quality was very good with a nice long range. The cordless phones are light weight with a belt clip, You just register the base set or phone, then connect the cordless phones. It pulls a config from central server, firmware options are thorough for business needs. One cordless phone (out of 5) developed a bad key. I really have to press hard to get the 1 key to work. The key went bad after 5 months. Overall, I really like the phones. Adrian Serafini -- _ -- 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 1.4.38 - unknown signalling bri_cpe
On 12/28/2010 10:49 AM, Administrator TOOTAI wrote: Le 28/12/2010 16:53, Kevin P. Fleming a écrit : [...] The Hx8 card manual (on the Digium website) clearly states on page 46 that the minimum required version of Asterisk for use with these cards is Asterisk 1.6.0.1. Hmmh, reading *before buying* 45 pages from a manual to know exactly the working environment seems to me a little bit extreme ;-) And people which are buying worldwide from a reseller doesnt perhaps have the reaction to go to Digium website. If you have a suggestion for a better place for this information to be made available, please let us know. Our resellers are supposed to know these things as well, which is the best method of ensuring that customers in each country around the world are kept informed. In the meantime I'll suggest that the web site team update the Hx8 product page on the website to include a statement about Asterisk 1.4.x not supporting BRI modules on these cards. Please note that we don't blame you. We installed those cards on a 1.6.2.15 Asterisk version but faced 2 problems with IAX; solutions where to switch back to 1.4 or to replace the IAX trunk with a SIP one. But seems that 1.6.16-rc1 resolved the main problem (crappy audio) Glad to hear it; hopefully that will take care of your issues. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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
[asterisk-users] Replacing digital pri card
I need to replace our current 1 port pri card with a quad port card. I'm currently using the newest AsteriskNOW distro. Are there any issues I should expect to run into? I'm hoping the transition will be smooth, however I havent had to do this in the past. -- _ -- 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] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?
On Tue, Dec 28, 2010 at 11:33 AM, Bruce B bruceb...@gmail.com wrote: I appreciate your feedback and let me know what info I can post here that may help resolve the issue (such as output from dmesg or lspci?). Hi Bruce, The following would be useful for starters: 1. cat /etc/wanpipe/*.conf 2. ifconfig -a (from a working and non-working situation) 3. lspci -v and lsusb -v (from a working and non-working situation) 4. wanrouter hwprobe verbose (from a working and non-working situation) 5. /var/log/messages (near the date the problem happened) Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.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] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?
Thanks for the input. I can not replicate the situation as it happens randomely or maybe over the weekend. However I have sent you all the requested command and logs in a separate e-mail for your analyzes. The only thing that stood out at me was the output of lsusb -v at the very end where it timed out. Since all lines didn't work I am to assume that both module went down but per my diagnoses with hwprobe I could see one unit connected and the other was not when the problem happened. Simply connecting/disconnecting that unit or connecting it to another port solved the problem and it showed up in hwprobe This is an Acer Aspire Revo mini PC. I am wondering if the U100s draw too much power? The only other USB connected device is the thumb size wireless connector for the keyboard. Acer computer: http://reviews.cnet.com/desktops/acer-aspire-revo-ar1600/4505-3118_7-33777218.html Looking forward to your analysis. Regards, Bruce On Tue, Dec 28, 2010 at 3:58 PM, Moises Silva moises.si...@gmail.comwrote: On Tue, Dec 28, 2010 at 11:33 AM, Bruce B bruceb...@gmail.com wrote: I appreciate your feedback and let me know what info I can post here that may help resolve the issue (such as output from dmesg or lspci?). Hi Bruce, The following would be useful for starters: 1. cat /etc/wanpipe/*.conf 2. ifconfig -a (from a working and non-working situation) 3. lspci -v and lsusb -v (from a working and non-working situation) 4. wanrouter hwprobe verbose (from a working and non-working situation) 5. /var/log/messages (near the date the problem happened) Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.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 -- _ -- 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
